3 2014-05-03 Sebastian Dröge <slomo@coaxion.net>
8 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
11 Automatic update of common submodule
12 From bcb1518 to 211fa5f
14 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
19 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
21 * tests/check/gst/sessionmedia.c:
22 tests: fix memory leak in sessionmedia unit test
24 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
26 * gst/rtsp-server/rtsp-client.c:
27 client: emit a signal before sending a message
28 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
30 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
32 * gst/rtsp-server/rtsp-client.c:
33 client: pass context to send_message
34 Pass the current context to send_message, we will need it later.
36 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
38 * gst/rtsp-server/rtsp-client.c:
39 client: fix typo in comment
41 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
43 * gst/rtsp-server/rtsp-media.c:
44 media: Do not stop thread twice if default_prepare() fails
46 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
48 * gst/rtsp-server/rtsp-client.c:
49 client: set the watch to flushing before going to NULL
50 First set the watch to flushing so that we unblock any current and
51 future attempt to send data on the watch, Then set the pipeline to
53 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
55 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
57 * gst/rtsp-server/rtsp-session-pool.c:
58 * tests/check/gst/sessionpool.c:
59 rtsp-session-pool: Fixes annotation
60 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
61 in the sessionpool test.
62 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
64 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
66 * gst/rtsp-server/rtsp-media.c:
67 * gst/rtsp-server/rtsp-media.h:
68 media: make media_prepare virtual
69 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
71 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
73 * gst/rtsp-server/rtsp-media.c:
74 * tests/check/gst/media.c:
75 media: stop the thread in more error cases
77 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
79 * gst/rtsp-server/rtsp-media.c:
80 * tests/check/gst/media.c:
81 media: allow NULL as the thread
82 Use the default context whan passing a NULL thread.
84 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
86 * gst/rtsp-server/rtsp-client.c:
87 rtsp-client: indent cleanup
88 Coverity was moaning about unreachable code, and I think it was just
89 confused by { being before the label. We'll see if it pops up again.
92 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
94 * gst/rtsp-server/rtsp-client.c:
95 * gst/rtsp-server/rtsp-media.c:
96 client: Add drop-backlog property
97 When we have too many messages queued for a client (currently hardcoded
98 to 100) we overflow and drop the messages. Add a drop-backlog property
99 to control this behaviour. Setting this property to FALSE will retry
100 to send the messages to the client by waiting for more room in the
102 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
104 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
106 * gst/rtsp-server/rtsp-client.c:
107 client: support for POST before GET when setting up a tunnel
109 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
111 * gst/rtsp-server/rtsp-client.c:
112 client: remove watch of the second client after http tunnel setup
113 The second client will be freed after the HTTP tunnel has been set up.
114 Make sure it's RTSP watch is never dispatched again.
115 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
117 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
119 * gst/rtsp-server/rtsp-media.c:
120 * tests/check/gst/media.c:
121 media: Make media_prepare() fail if port allocation fails
122 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
124 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
126 * tests/check/gst/media.c:
127 media test: cleanup the thread pool in tests
129 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
131 * gst/rtsp-server/rtsp-media.c:
132 * tests/check/gst/media.c:
133 rtsp-media: Unblock blocked streams in unprepare
134 The streams will be blocked when a live media is prepared.
135 The streams should be unblocked in gst_rtsp_media_unprepare.
136 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
138 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
140 * gst/rtsp-server/rtsp-media.c:
141 media: release the state lock when going to NULL
142 Set our state to UNPREPARING and release the state-lock before
143 setting the pipeline to the NULL state. This way, any pad-added
144 callback will be able to take the state-lock and check that we are now
145 unpreparing instead of deadlocking.
146 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
148 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
150 * gst/rtsp-server/rtsp-media.c:
151 media: protect status with lock
152 Make sure we only update the status with the lock.
154 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
156 * gst/rtsp-server/rtsp-client.c:
157 * gst/rtsp-server/rtsp-sdp.c:
158 rtsp: update for MIKEY API changes
160 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
162 * gst/rtsp-server/rtsp-client.c:
163 client: parse the mikey response from the client
164 Parse the mikey response from the client and update the policy for
167 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
169 * gst/rtsp-server/rtsp-stream.c:
170 * gst/rtsp-server/rtsp-stream.h:
171 stream: add method to set crypto info
172 Make a method to configure the crypto information of a stream.
173 Set udpsrc in READY instead of PAUSED so that we can configure caps
176 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
178 * gst/rtsp-server/rtsp-client.c:
179 client: cleanup error paths
181 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
183 * gst/rtsp-server/rtsp-media.c:
186 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
188 * examples/test-video.c:
189 test: enable SRTP only on RTSPS
190 We only want to enable SRTP when doing rtsp over TLS so that we can
191 exchange the keys in a secure way.
193 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
195 * examples/test-video.c:
196 test: print an error on failure
198 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
201 * examples/test-video.c:
202 * gst/rtsp-server/rtsp-sdp.c:
203 * gst/rtsp-server/rtsp-stream.c:
204 * tests/check/Makefile.am:
205 stream: add SRTP support
206 Install srtp encoder and decoder elements in rtpbin
209 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
211 * tests/check/Makefile.am:
212 * tests/check/gst/sessionpool.c:
213 tests: Add unit tests for sessionpool
214 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
216 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
218 * tests/check/gst/threadpool.c:
219 tests: Improve code coverage of rtsp-threadpool tests
220 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
222 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
224 * tests/check/gst/sessionmedia.c:
225 tests: Improve code coverage for rtsp-session-media
226 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
228 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
230 gobject-introspection: Add annotations to support language bindings
231 In addition a few cosmetic changes:
232 * Adjust the order of arguments
233 * Fix typo: occured -> occurred
234 * Fix indentation after Return:-clauses
235 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
237 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
239 * gst/rtsp-server/rtsp-stream.c:
240 rtsp-stream: Don't mix IPv4 and IPv6 addresses
241 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
243 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
245 * gst/rtsp-server/rtsp-stream.c:
246 stream: take caps after the session manager
247 Take the caps for the SDP after they leave the rtpbin so that we can
248 also get the properties added by rtpbin elements.
250 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
252 * gst/rtsp-server/rtsp-stream.c:
253 stream: release lock while pushing out packets
254 Keep a cache of the transports and use this to iterate the transport
255 while pushing packets. This allows us to release the lock early.
256 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
258 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
260 * gst/rtsp-server/rtsp-client.c:
261 * gst/rtsp-server/rtsp-client.h:
262 rtsp-client: vmethod for modifying tunnel GET response
263 Add a vmethod tunnel_http_response where the response to the HTTP GET
264 for tunneled connections can be modified.
265 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
267 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
269 * gst/rtsp-server/rtsp-sdp.c:
270 sdp: make 1 media line per profile
271 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
272 line in the SDP for each profile. The client is then supposed to pick
273 one of the profiles in the SETUP request. Because the m= lines have the
274 same pt, the client also knows that only 1 option is possible.
276 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
278 * gst/rtsp-server/rtsp-media-factory.c:
279 * gst/rtsp-server/rtsp-media-factory.h:
280 * gst/rtsp-server/rtsp-media.c:
281 factory: add profile property and pass to media and streams
283 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
285 * examples/test-multicast.c:
286 * gst/rtsp-server/rtsp-sdp.c:
287 sdp: pass multicast connection for multicast-only stream
288 Pass the multicast address of the stream in the connection info in the
289 SDP so that clients try a multicast connection first.
290 Only allow multicast connections in the test-multicast example. Also
291 increase the TTL a little.
293 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
296 .gitignore: Ignore gcov intermediate files
297 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
299 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
301 * gst/rtsp-server/rtsp-stream.c:
302 stream: release some locks in error cases
304 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
306 docs: Enable and fix gtk-doc warnings
307 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
308 * addresspool/mediafactory: Add missing annotation colon
309 * stream: Annotate return value
310 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
312 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
315 Automatic update of common submodule
316 From fe1672e to bcb1518
318 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
321 Automatic update of common submodule
322 From 1a07da9 to fe1672e
324 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
326 * examples/Makefile.am:
327 examples: use LDADD for libs instead of LDFLAGS
329 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
332 configure: make sure releases are in .doap file
334 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
336 * examples/test-cgroups.c:
337 examples: test-cgroups: don't put code with side effects into g_assert()
338 The g_assert() might get compiled out with the right
339 compiler/preprocessor flags.
341 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
343 * examples/.gitignore:
344 examples: add cgroup test binary to .gitignore
346 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
348 * examples/test-cgroups.c:
349 examples: fix cgroup test build
350 Fixes build failure caused by compiler warning:
351 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
353 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
356 .gitignore: ignore temp files created in the course of 'make check'
358 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
360 * gst/rtsp-server/rtsp-media.c:
361 rtsp-media: don't loose frames handling new PLAY request
362 If client supplied a range check if the range specifies the start point.
363 If not, then do an accurate seek to the current position. If a start
364 point was specified do do a key unit seek to make sure the streaming
365 starts with decodeable frames.
366 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
368 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
370 * gst/rtsp-server/rtsp-media.c:
371 Revert "media: only flush when setting a new start position"
372 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
373 We need to do the flush in all cases, demuxer block currently for
376 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
378 * gst/rtsp-server/rtsp-media.c:
379 media: only flush when setting a new start position
380 Only flush the pipeline when we change the start position with
382 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
384 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
386 * gst/rtsp-server/rtsp-stream.c:
387 stream: set ttl-mc before adding the socket
388 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
389 never be set on socket.
390 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
392 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
394 * gst/rtsp-server/rtsp-media.c:
395 media: stop thread if media is already prepared
396 in gst_rtsp_media_prepare() the thread is not used if media is already
397 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
399 https://bugzilla.gnome.org/show_bug.cgi?id=724182
401 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
404 build: Ship gst-rtsp-server.doap file
406 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
408 * tests/check/gst/rtspserver.c:
409 tests: Fix another compiler warning with gcc
411 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
413 * gst/rtsp-server/rtsp-client.c:
414 * gst/rtsp-server/rtsp-mount-points.c:
415 * gst/rtsp-server/rtsp-stream.c:
416 * tests/check/gst/client.c:
417 rtsp-server: Fix lots of compiler warnings with clang
419 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
422 * gst-rtsp-server.doap:
424 configure: Synchronise with the configure scripts of the other modules
426 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
429 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
431 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
433 * gst/rtsp-server/rtsp-media.c:
434 * gst/rtsp-server/rtsp-stream.c:
435 Revert "rtsp-server: support build against last stable release"
436 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
437 Let us require 1.2.3 now, which is going to be released in a few
440 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
442 * gst/rtsp-server/rtsp-session-media.c:
443 * gst/rtsp-server/rtsp-stream-transport.c:
444 session: improve RTP-Info
445 Ignore streams that can't generate RTP-Info instead of failing.
446 Don't return the empty string when all streams are unconfigured but
447 return NULL so that we don't generate and empty RTP-Info header.
448 Improve docs a little.
450 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
452 * gst/rtsp-server/rtsp-session-media.c:
453 Don't free rtpinfo GString when it is NULL
454 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
456 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
458 * gst/rtsp-server/rtsp-media.c:
459 media: only set keyframe flag when modifying start
460 Only set the keyframe flag when we modify the start position. The
461 keyframe flag should probably be ignored when no change is requested but
462 until we can claim this is all documented properly and all demuxer
463 implement this, avoid setting the flag.
464 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
466 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
468 * gst/rtsp-server/rtsp-thread-pool.c:
469 thread-pool: Unref source after mainloop has quit to avoid races in GLib
470 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
472 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
474 * gst/rtsp-server/rtsp-stream.c:
475 stream: handle NULL seqnum and rtptime arguments
477 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
479 * gst/rtsp-server/rtsp-thread-pool.c:
480 * tests/check/gst/threadpool.c:
481 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
482 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
484 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
486 * gst/rtsp-server/rtsp-stream.c:
487 stream: add fallback for missing stats property
488 Use a fallback when the payloader does not have a stats property
489 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
491 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
494 Automatic update of common submodule
495 From f7bc1c3 to 1a07da9
497 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
499 * gst/rtsp-server/rtsp-stream.c:
500 stream: don't leak stats structure
501 Don't leak the stats structure and deal with NULL stats.
503 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
505 * gst/rtsp-server/rtsp-stream.c:
506 stream: Get rtpinfo properties atomically from payloader
507 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
509 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
511 * gst/rtsp-server/rtsp-media.c:
512 media: refactor state change functions and signals
513 Make functions to set the target state and the pipeline state and emit
514 the signals from those functions.
516 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
518 * gst/rtsp-server/rtsp-media.c:
519 * gst/rtsp-server/rtsp-media.h:
520 media: add signal to notify of pending state changes
522 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
524 * gst/rtsp-server/rtsp-media.c:
525 * gst/rtsp-server/rtsp-stream.c:
526 rtsp-server: support build against last stable release
527 Until 1.2.3 is out with the new get_type function and we
530 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
532 * gst/rtsp-server/rtsp-stream.c:
533 stream: fix compilation
535 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
537 * gst/rtsp-server/rtsp-media.c:
538 * gst/rtsp-server/rtsp-media.h:
539 * gst/rtsp-server/rtsp-stream.c:
540 * gst/rtsp-server/rtsp-stream.h:
541 stream: add property to configure profiles
543 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
545 * gst/rtsp-server/rtsp-client.c:
546 client: let stream check supported transport
547 Delegate the check if a transport is allowed to the stream.
548 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
550 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
552 * gst/rtsp-server/rtsp-stream.c:
553 * gst/rtsp-server/rtsp-stream.h:
554 stream: add method to check supported transport
555 Add a method to check if a transport is supported
557 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
560 configure.ac: Only check for gstreamer-check, not check
561 We include check in gstreamer-check since quite some time now.
563 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
565 * gst/rtsp-server/rtsp-session-media.c:
566 * gst/rtsp-server/rtsp-stream-transport.c:
567 * gst/rtsp-server/rtsp-stream.c:
568 * gst/rtsp-server/rtsp-stream.h:
569 stream: return clock-rate from get_rtpinfo
570 And use it to correct the rtptime to the requested start-time.
571 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
573 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
575 * gst/rtsp-server/rtsp-session-media.c:
576 * gst/rtsp-server/rtsp-stream-transport.c:
577 * gst/rtsp-server/rtsp-stream-transport.h:
578 session-media: calculate start-time
580 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
582 * gst/rtsp-server/rtsp-stream-transport.c:
583 * gst/rtsp-server/rtsp-stream.c:
584 * gst/rtsp-server/rtsp-stream.h:
585 stream: also return the running-time
586 Return the running-time in the rtpinfo as well.
588 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
590 * gst/rtsp-server/rtsp-client.c:
591 * gst/rtsp-server/rtsp-session-media.c:
592 * gst/rtsp-server/rtsp-session-media.h:
593 * gst/rtsp-server/rtsp-stream-transport.c:
594 * gst/rtsp-server/rtsp-stream-transport.h:
595 session-media: let the session-media make the RTPInfo
596 Add method to create the RTPInfo for a stream-transport.
597 Add method to create the RTPInfo for all stream-transports in a
599 Use the session-media RTPInfo code in client. This allows us to refactor
600 another method to link the TCP callbacks.
602 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
604 mount-points: sort sequence before g_sequence_lookup
605 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
606 sort sequence if dirty, otherwise lookup will fail.
607 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
609 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
612 configure: rename package from gst-rtsp to gst-rtsp-server
613 To match git module name and avoid confusion with the
614 rtsp lib in gst-plugins-base and rtsp plugin in -good.
616 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
619 configure: bump core/base/good requirement to 1.2.0
620 Bump to released stable version and make implicit
621 requirements explicit.
623 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
628 Fix broken gettext setup which is not used anyway
630 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
633 Automatic update of common submodule
634 From dbedaa0 to d48bed3
636 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
638 * gst/rtsp-server/rtsp-client.c:
639 * gst/rtsp-server/rtsp-media.c:
640 * gst/rtsp-server/rtsp-media.h:
641 media: add setup_sdp vmethod
642 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
643 gst_rtsp_media_setup_sdp.
644 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
646 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
648 * gst/rtsp-server/rtsp-stream.c:
649 rtsp-stream: Check return value of sscanf
650 streamid is only valid if sscanf matched something.
652 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
654 * gst/rtsp-server/rtsp-client.c:
655 rtsp-client: Fix iteration
656 Wouldn't even enter the code block otherwise (i++ was used as the check
657 and not the postfix).
659 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
661 * gst/rtsp-server/rtsp-client.c:
662 * gst/rtsp-server/rtsp-client.h:
663 client: add vmethod to configure media and streams
664 Implement a vmethod that can be used to configure the media and the
665 streams based on the current context. Handle the blocksize handling in
667 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
669 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
672 Make git ignore more unit test binaries
674 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
676 * gst/rtsp-server/rtsp-address-pool.h:
677 * gst/rtsp-server/rtsp-auth.h:
678 * gst/rtsp-server/rtsp-client.h:
679 * gst/rtsp-server/rtsp-context.h:
680 * gst/rtsp-server/rtsp-media-factory-uri.h:
681 * gst/rtsp-server/rtsp-media-factory.h:
682 * gst/rtsp-server/rtsp-media.h:
683 * gst/rtsp-server/rtsp-mount-points.h:
684 * gst/rtsp-server/rtsp-server.h:
685 * gst/rtsp-server/rtsp-session-media.h:
686 * gst/rtsp-server/rtsp-session-pool.h:
687 * gst/rtsp-server/rtsp-session.h:
688 * gst/rtsp-server/rtsp-stream-transport.h:
689 * gst/rtsp-server/rtsp-stream.h:
690 * gst/rtsp-server/rtsp-thread-pool.h:
691 * gst/rtsp-server/rtsp-token.h:
692 rtsp-server: add padding to many public structures
693 Not mini objects though, since they are not subclassable
694 anyway, nor kept on the stack or inlined in a structure.
696 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
698 media: add new create_rtpbin vmethod
699 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
700 https://bugzilla.gnome.org/show_bug.cgi?id=719734
702 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
704 * tests/check/gst/media.c:
705 tests: fix memory leak, free test's thread pool
706 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
708 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
710 * gst/rtsp-server/rtsp-stream-transport.c:
711 stream-transport: free url in finalize
713 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
715 * gst/rtsp-server/rtsp-media.c:
716 media: also do state change in suspended state
718 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
720 * gst/rtsp-server/rtsp-client.c:
721 * gst/rtsp-server/rtsp-media.c:
722 media: also handle prepare and range in suspended state
723 When we are suspended, we are already prepared.
724 We can get the range in the suspended state.
726 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
728 * tests/check/Makefile.am:
729 * tests/check/gst/sessionmedia.c:
730 check: add test for uri in setup
731 Added unit tests for the new functionality in GstRTSPStreamTransport.
732 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
734 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
736 * gst/rtsp-server/rtsp-client.c:
737 client: store setup uri and use in PLAY response
738 Store the uri used when doing the setup and use that in the PLAY
740 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
742 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
744 * gst/rtsp-server/rtsp-stream-transport.c:
745 * gst/rtsp-server/rtsp-stream-transport.h:
746 stream-transport: add method to get/set url
748 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
750 * gst/rtsp-server/rtsp-client.c:
751 client: suspend after SDP and unsuspend before PLAYING
752 Based on patches by Ognyan Tonchev <ognyan@axis.com>
753 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
755 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
757 * gst/rtsp-server/rtsp-media-factory.c:
758 * gst/rtsp-server/rtsp-media-factory.h:
759 * gst/rtsp-server/rtsp-media.c:
760 * gst/rtsp-server/rtsp-media.h:
761 * gst/rtsp-server/rtsp-session-media.c:
762 * gst/rtsp-server/rtsp-session.c:
763 * tests/check/gst/media.c:
764 * tests/check/gst/mediafactory.c:
765 media: add suspend modes
766 Add support for different suspend modes. The stream is suspended right after
767 producing the SDP and after PAUSE. Different suspend modes are available that
768 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
769 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
770 state and RESET will bring the pipeline to the NULL state.
771 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
772 this means that the pipeline needs to be prerolled again.
773 Base on patches by Ognyan Tonchev <ognyan@axis.com>
774 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
776 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
778 * gst/rtsp-server/rtsp-media.c:
779 media: start live streams in blocked state
780 Start live streams in the blocked state and make them preroll using the
781 messages. This ensure that no data is played by the sink until we explicitly
782 unblock the stream right before going to PLAYING.
783 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
785 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
787 * gst/rtsp-server/rtsp-media.c:
788 media: refactor starting and waiting for preroll
789 Based on patches from Ognyan Tonchev <ognyan@axis.com>
790 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
792 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
794 * gst/rtsp-server/rtsp-stream.c:
795 * gst/rtsp-server/rtsp-stream.h:
796 stream: add API to block streams
797 Add an API to block on the streams and make it post a message.
798 Based on patch by Ognyan Tonchev <ognyan@axis.com>
799 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
801 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
803 * docs/libs/Makefile.am:
804 docs: Specify the override file
805 Even if it's empty (for now) it avoids make distcheck complaining
807 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
809 * gst/rtsp-server/rtsp-media.c:
810 media: move default implementations to where they are used
812 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
814 * gst/rtsp-server/rtsp-media.c:
815 media: take the right lock in gst_rtsp_media_set_pipeline_state()
816 We need to take the state_lock when calling this method.
818 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
820 * gst/rtsp-server/rtsp-media.c:
821 media: handle add-added on non-bins too
822 Handle dynamic payloaders that are not bins, as used in the unit-test.
824 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
826 * gst/rtsp-server/rtsp-media-factory.c:
827 * gst/rtsp-server/rtsp-media-factory.h:
828 * gst/rtsp-server/rtsp-media.c:
829 rtsp-media/-factory: Fix request pad name comments
830 These must be escaped for gtk-doc to parse the comments without warnings.
832 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
834 rtsp-media: remove transports if media is in error status
835 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
836 trying to change to GST_STATE_NULL and media is in error status, we
837 remove all transports.
838 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
840 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
842 * gst/rtsp-server/rtsp-media.c:
843 rtsp-media: use element metadata to find payloader
844 Use the element metadata to find the payloader instead of checking
846 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
848 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
850 rtsp-stream: add getter for payload type
851 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
852 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
853 element and create the stream with this one instead of the dynpay%d
855 https://bugzilla.gnome.org/show_bug.cgi?id=712396
857 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
859 * gst/rtsp-server/rtsp-client.c:
860 * gst/rtsp-server/rtsp-context.h:
861 * gst/rtsp-server/rtsp-media.c:
862 * gst/rtsp-server/rtsp-mount-points.c:
863 * gst/rtsp-server/rtsp-server.c:
864 * gst/rtsp-server/rtsp-token.c:
865 rtsp-*: Refer to NULL as a constant in comments
867 https://bugzilla.gnome.org/show_bug.cgi?id=714988
869 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
871 rtsp-*: Fix type name typos in comments
872 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
873 * rtsp-auth: Refer to part of constant name as text
874 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
875 * rtsp-session-media: Fix GstRTSPSessionMedia typo
876 * rtsp-stream: Fix typo when refering to GstBin
877 https://bugzilla.gnome.org/show_bug.cgi?id=714988
879 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
882 * docs/libs/gst-rtsp-server-docs.sgml:
883 * docs/libs/gst-rtsp-server-sections.txt:
884 docs: Improve documentation
885 * Include annotation-glossary to quiet gtk-doc
886 * Rename remaining ClientState -> Context
887 * Rename object hierarchy file
888 * Remove stale chapter references
889 * Add missing function and object references
890 * Include missing GstRTSPAddressPoolResult
891 https://bugzilla.gnome.org/show_bug.cgi?id=714988
893 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
895 * gst/rtsp-server/rtsp-client.c:
896 * gst/rtsp-server/rtsp-server.c:
897 * gst/rtsp-server/rtsp-session-pool.c:
898 * gst/rtsp-server/rtsp-session.c:
899 * gst/rtsp-server/rtsp-stream.c:
900 rtsp-server: sprinkle some allow-none annotations for g-i
902 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
904 * gst/rtsp-server/rtsp-stream.c:
905 * gst/rtsp-server/rtsp-stream.h:
906 stream: add method to filter transports
907 Add a method to safely iterate and collect the stream transports
908 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
910 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
912 * gst/rtsp-server/rtsp-client.c:
913 * gst/rtsp-server/rtsp-server.c:
914 * gst/rtsp-server/rtsp-session-pool.c:
915 * gst/rtsp-server/rtsp-session.c:
916 rtsp: allow NULL func in filters
917 Passing a null function make the filters return a list of
920 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
922 * gst/rtsp-server/rtsp-address-pool.c:
923 * tests/check/gst/addresspool.c:
924 address-pool: fix address increment
925 Use a guint instead of guint8 to increment the address. It's still not
926 completely correct because a guint might not be able to hold the complete
927 address range, but that's an enhacement for later.
928 Add unit test to test improved behaviour.
929 https://bugzilla.gnome.org/show_bug.cgi?id=708237
931 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
933 * gst/rtsp-server/rtsp-client.c:
934 * tests/check/gst/client.c:
935 client: allow absolute path in requests
936 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
938 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
940 * gst/rtsp-server/rtsp-client.c:
941 * gst/rtsp-server/rtsp-client.h:
942 client: make make_path_from_uri a vmethod
944 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
946 * docs/libs/gst-rtsp-server-sections.txt:
947 * gst/rtsp-server/rtsp-stream.c:
948 * gst/rtsp-server/rtsp-stream.h:
949 * tests/check/Makefile.am:
950 * tests/check/gst/stream.c:
951 stream: Add functions to get rtp and rtcp sockets
952 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
954 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
956 * gst/rtsp-server/rtsp-context.c:
957 * gst/rtsp-server/rtsp-context.h:
958 context: defing a GType for the context
959 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
961 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
963 * gst/rtsp-server/Makefile.am:
964 * gst/rtsp-server/rtsp-auth.c:
965 * gst/rtsp-server/rtsp-context.c:
966 * gst/rtsp-server/rtsp-media.c:
967 * gst/rtsp-server/rtsp-mount-points.c:
968 * gst/rtsp-server/rtsp-server.h:
969 * gst/rtsp-server/rtsp-session-media.c:
970 * gst/rtsp-server/rtsp-session.c:
971 * gst/rtsp-server/rtsp-stream.c:
972 Fixed several GIR warnings
974 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
976 * gst/rtsp-server/rtsp-auth.c:
979 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
981 * tests/check/Makefile.am:
982 * tests/check/gst/token.c:
983 tests: Add unit tests for token
984 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
986 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
988 * gst/rtsp-server/rtsp-token.c:
989 token: Validate args for gst_rtsp_token_is_allowed
990 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
992 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
994 * gst/rtsp-server/rtsp-token.c:
995 token: Fix bug when creating empty token
996 We always want to have a valid GstStructure in the token.
997 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
999 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
1001 * gst/rtsp-server/rtsp-thread-pool.c:
1002 thread-pool: avoid race in shutdown
1003 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
1004 don't actually stop the mainloop ever. Solve this race by adding an idle source
1005 to the mainloop that calls the _quit. This way we immediately exit the mainloop
1006 if quit was called before we started it.
1008 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1010 * tests/check/Makefile.am:
1011 * tests/check/gst/permissions.c:
1012 tests: Add unit tests for permissions
1013 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
1015 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1017 * tests/check/gst/mediafactory.c:
1018 tests: Test mediafactory permissions
1019 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
1021 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1023 * gst/rtsp-server/rtsp-permissions.c:
1024 permissions: Fix refcounting when adding/removing roles
1025 Previously a role that was removed was unreffed twice, and when
1026 replacing an existing role the replaced role was freed while still being
1027 referenced. Both bugs are now fixed.
1028 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
1030 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1032 * tests/check/gst/media.c:
1033 * tests/check/gst/mediafactory.c:
1034 * tests/check/gst/rtspserver.c:
1035 tests: Check gst_rtsp_url_parse return value
1036 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
1038 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
1041 Automatic update of common submodule
1042 From 865aa20 to dbedaa0
1044 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
1046 * gst/rtsp-server/rtsp-server.c:
1047 rtsp-server: Fix socket leak
1048 https://bugzilla.gnome.org/show_bug.cgi?id=710088
1050 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
1052 * gst/rtsp-server/rtsp-session-pool.c:
1053 rtsp-session-pool: Make sure session IDs are properly URI-escaped
1054 https://bugzilla.gnome.org/show_bug.cgi?id=643812
1056 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
1058 * examples/.gitignore:
1059 * examples/test-video.c:
1060 examples: fix compilation when WITH_AUTH is defined
1061 https://bugzilla.gnome.org/show_bug.cgi?id=710228
1063 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
1066 gitignore: Add new test binary
1068 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
1070 * tests/check/Makefile.am:
1071 * tests/check/gst/threadpool.c:
1072 thread-pool: Add unit test for the thread pools
1073 https://bugzilla.gnome.org/show_bug.cgi?id=710228
1075 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
1077 * gst/rtsp-server/rtsp-thread-pool.c:
1078 thread-pool: Fix thread leak when reusing threads
1079 https://bugzilla.gnome.org/show_bug.cgi?id=709730
1081 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
1083 * gst/rtsp-server/rtsp-server.c:
1084 * tests/check/gst/rtspserver.c:
1085 tests: fixed racy behavior in rtspserver tests
1086 https://bugzilla.gnome.org/show_bug.cgi?id=710078
1088 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1090 * tests/check/gst/addresspool.c:
1091 tests: Improve address pool unit tests
1092 Add a range with mixed IPV4 and IPV6 addresses to pool.
1093 Get an IPV4 address from an IPV6-only pool.
1094 Get an IPV6 address from an IPV4-only pool.
1095 Reserve a IPV6 address from an IPV4-only pool.
1096 Check for unicast addresses in multicast-only pool.
1097 Check for unicast addresses in uni-/multicast-mixed pool.
1098 https://bugzilla.gnome.org/show_bug.cgi?id=710128
1100 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1102 * gst/rtsp-server/rtsp-client.c:
1103 client: append query string in PAUSE/PLAY/TEARDOWN as well
1105 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
1107 * gst/rtsp-server/rtsp-client.c:
1108 client: Add query to control path
1109 If the SETUP url contains a query it must be appended to the control
1110 path so that it matches any already created stream in the media. The
1111 query will also be appended to the session media path.
1113 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1115 * gst/rtsp-server/rtsp-media.c:
1116 rtsp-media: remove old line
1118 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
1120 * gst/rtsp-server/rtsp-stream.c:
1121 stream: Correct control comparison
1122 https://bugzilla.gnome.org/show_bug.cgi?id=709176
1124 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1126 * gst/rtsp-server/rtsp-media.c:
1127 media: Check dynamically if the pipeline supports seeking
1128 We should not depend on whether or not the pipeline state change
1129 returned NO_PREROLL or not. A media could dynamically change its
1130 element and switch from seekable to non seekable so it's best to test
1131 the seekable nature of the pipeline dynamically when we try to do a seek.
1133 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1135 * gst/rtsp-server/rtsp-media.c:
1136 media: Return FALSE if seeking is not supported
1138 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1140 * gst/rtsp-server/rtsp-media.c:
1141 rtsp-media: don't seek accurate by default
1142 Accurate seeking is perhaps a little overkill in the most common situation and
1143 causes some formats (mp3) over slow media to seek extremely slowly.
1145 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
1147 * tests/check/gst/rtspserver.c:
1148 tests: fix unit test
1149 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
1151 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
1153 * gst/rtsp-server/rtsp-client.c:
1154 client: Reply 400 if media cannot be constructed
1155 Reply 400 Bad Request instead of 503 Service Unavailable if media
1156 cannot be constructed in SETUP.
1157 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
1159 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
1161 * gst/rtsp-server/rtsp-client.c:
1162 client: Send setup reply once only
1163 If find_media() failed in handle_setup_request() two replies was sent.
1164 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
1166 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
1169 Automatic update of common submodule
1170 From 6b03ba7 to 865aa20
1172 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
1174 * gst/rtsp-server/rtsp-server.c:
1175 server: Emit client-connected signal earlier
1176 Emit client-connected before the client ref is given to a GSource,
1177 otherwise client-connected can be emitted after the client object has
1180 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
1182 * gst/rtsp-server/rtsp-address-pool.c:
1183 * gst/rtsp-server/rtsp-address-pool.h:
1184 * gst/rtsp-server/rtsp-stream.c:
1185 * tests/check/gst/addresspool.c:
1186 addresspool: return reason of failure
1187 Let gst_rtsp_address_pool_reserve_address() return the reason why
1188 the address could not be reserved.
1189 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
1191 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
1194 autogen.sh: Sync behaviour with other GStreamer modules
1195 Allows building from outside of tree amongst other things
1197 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
1200 Automatic update of common submodule
1201 From b613661 to 6b03ba7
1203 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
1206 Automatic update of common submodule
1207 From 74a6857 to b613661
1209 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
1212 Automatic update of common submodule
1213 From 01a7a46 to 74a6857
1215 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
1217 * gst/rtsp-server/rtsp-client.c:
1218 client: Do not read beyond end of path string
1219 If the setup was done without a control url, make sure we don't try to read the
1220 non-existing control string and crash.
1222 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1224 * gst/rtsp-server/rtsp-client.c:
1225 client: Fix RTPInfo header
1226 Refactor the method to make the content_base.
1227 Use the content-base and the control url to construct the RTPInfo
1230 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1232 * gst/rtsp-server/rtsp-client.c:
1233 client: map url to path only in describe
1234 Only map the request url to a path in the DESCRIBE method. The SDP then
1235 contains the base and control urls that should be used to SETUP/PAUSE/
1236 PLAY/TEARDOWN the media.
1238 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1240 * gst/rtsp-server/rtsp-client.c:
1241 Revert "client: map URL to path in requests"
1242 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
1243 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
1244 contains the base and control urls which are used in the SETUP, PLAY,
1245 PAUSE and TEARDOWN requests.
1247 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1249 * gst/rtsp-server/rtsp-client.c:
1250 client: map URL to path in requests
1252 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1254 * gst/rtsp-server/rtsp-client.c:
1255 * gst/rtsp-server/rtsp-mount-points.c:
1256 * gst/rtsp-server/rtsp-mount-points.h:
1257 mount-points: make vmethod to make path from uri
1258 Make a vmethod to transform an url into a path. The path is then used to lookup
1259 the factory. This makes it possible to also use other bits of the url, such as
1260 the query parameters, to locate the factory.
1262 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
1264 * gst/rtsp-server/rtsp-thread-pool.c:
1265 * gst/rtsp-server/rtsp-thread-pool.h:
1266 thread-pool: Add cleanup to wait for the threadpool to finish
1267 Also fix race condition if two threads are asking for the first
1268 thread from the thread pool at once. This would case two internal
1269 GThreadPools to be created.
1270 https://bugzilla.gnome.org/show_bug.cgi?id=707753
1272 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
1274 * gst/rtsp-server/rtsp-client.c:
1275 * tests/check/gst/client.c:
1276 client: free threadpool
1277 https://bugzilla.gnome.org/show_bug.cgi?id=707638
1279 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
1281 * tests/check/gst/mountpoints.c:
1282 mountpoints tests: unref matched factories
1283 https://bugzilla.gnome.org/show_bug.cgi?id=707638
1285 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
1287 * tests/check/gst/media.c:
1288 media tests: unref thread pool and caps
1289 https://bugzilla.gnome.org/show_bug.cgi?id=707638
1291 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
1293 * gst/rtsp-server/rtsp-auth.c:
1294 * gst/rtsp-server/rtsp-media-factory.c:
1295 * gst/rtsp-server/rtsp-media.c:
1296 auth, media, media-factory: unref permissions
1297 https://bugzilla.gnome.org/show_bug.cgi?id=707638
1299 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1301 * examples/Makefile.am:
1302 Makefile: add rule for appsrc example
1304 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1306 * examples/test-appsrc.c:
1307 tests: add appsrc example
1308 Add an example on how to use appsrc to feed the server pipeline with data.
1310 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
1312 * gst/rtsp-server/rtsp-client.c:
1313 rtsp-client: remove query part from content-base string
1314 Make sure that after the control url has been resolved, it's
1315 not a part of the query-string.
1316 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
1318 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1320 * gst/rtsp-server/rtsp-client.c:
1321 client: don't check url in response
1322 There is no url or method in the response to check
1324 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1326 * gst/rtsp-server/rtsp-client.c:
1327 * gst/rtsp-server/rtsp-client.h:
1328 Add handle-response signal for when we receive a GET_PARAMETER response
1330 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1332 * gst/rtsp-server/rtsp-server.c:
1333 Fix gst_rtsp_server_client_filter, using wrong variable type
1335 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
1337 * gst/rtsp-server/rtsp-media-factory-uri.c:
1338 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
1339 For AAC we need to check for framed=true instead of parsed=true.
1340 https://bugzilla.gnome.org/show_bug.cgi?id=701384
1342 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1344 * gst/rtsp-server/rtsp-stream.c:
1345 stream: optimize pipeline for protocols
1346 When TCP is not an allowed protocol for the stream, avoid creating the
1347 appsrc/appsink/queue and tee elements.
1349 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1351 * gst/rtsp-server/rtsp-media.c:
1352 media: set protocols on streams
1354 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1356 * gst/rtsp-server/rtsp-client.c:
1357 client: use protocols supported by stream
1359 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1361 * gst/rtsp-server/rtsp-media-factory.c:
1362 * gst/rtsp-server/rtsp-media.c:
1363 * gst/rtsp-server/rtsp-stream.c:
1364 media-factory: allow all protocols
1366 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1368 * gst/rtsp-server/rtsp-media.c:
1369 media: configure protocols in new streams
1371 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1373 * gst/rtsp-server/rtsp-stream.c:
1374 * gst/rtsp-server/rtsp-stream.h:
1375 stream: add protocols property
1377 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1379 * gst/rtsp-server/rtsp-media.c:
1380 rtsp-media: send state in "new-state" signal
1381 https://bugzilla.gnome.org/show_bug.cgi?id=705110
1383 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
1386 build: add subdir-objects to AM_INIT_AUTOMAKE
1387 Fixes warnings with automake 1.14
1388 https://bugzilla.gnome.org/show_bug.cgi?id=705350
1390 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1392 * docs/libs/gst-rtsp-server-sections.txt:
1393 * gst/rtsp-server/rtsp-client.c:
1394 * gst/rtsp-server/rtsp-server.c:
1395 * gst/rtsp-server/rtsp-server.h:
1396 server: add method to iterate clients of server
1398 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1400 * gst/rtsp-server/rtsp-media.c:
1401 * gst/rtsp-server/rtsp-media.h:
1402 Add vmethod for rtsp-media subclass to access rtpbin
1404 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1406 * gst/rtsp-server/rtsp-client.h:
1407 small documentation fix
1409 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1411 * gst/rtsp-server/rtsp-client.c:
1412 Do not take range header if range is invalid
1414 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1416 * docs/libs/gst-rtsp-server-sections.txt:
1417 * gst/rtsp-server/rtsp-media.c:
1418 media: add docs for new method
1420 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1422 * gst/rtsp-server/rtsp-media.c:
1423 * gst/rtsp-server/rtsp-media.h:
1424 Add API to rtsp-media set the pipeline's state
1426 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1428 * gst/rtsp-server/rtsp-media.c:
1429 Update current position/duration when gst_rtsp_media_get_range_string is called
1431 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1433 * examples/test-cgroups.c:
1434 tests: add some more docs
1436 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1438 * examples/test-cgroups.c:
1439 * gst/rtsp-server/Makefile.am:
1440 * gst/rtsp-server/rtsp-auth.c:
1441 * gst/rtsp-server/rtsp-auth.h:
1442 * gst/rtsp-server/rtsp-client.c:
1443 * gst/rtsp-server/rtsp-client.h:
1444 * gst/rtsp-server/rtsp-context.c:
1445 * gst/rtsp-server/rtsp-context.h:
1446 * gst/rtsp-server/rtsp-params.c:
1447 * gst/rtsp-server/rtsp-params.h:
1448 * gst/rtsp-server/rtsp-server.c:
1449 * gst/rtsp-server/rtsp-thread-pool.c:
1450 * gst/rtsp-server/rtsp-thread-pool.h:
1451 * tests/check/gst/client.c:
1452 ClientState -> Context
1453 Rename the clientstate to context and put the code in a separate file.
1455 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1457 * examples/test-auth.c:
1458 * gst/rtsp-server/rtsp-auth.c:
1459 * gst/rtsp-server/rtsp-auth.h:
1460 auth: add support for default token
1461 The default token is used when the user is not authenticated and can be used to
1462 give minimal permissions.
1464 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1466 * examples/test-auth.c:
1467 * gst/rtsp-server/rtsp-auth.c:
1468 auth: use defines when possible
1470 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1472 * gst/rtsp-server/rtsp-address-pool.c:
1473 address-pool: improve docs
1475 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1477 * gst/rtsp-server/rtsp-permissions.c:
1478 permissions: add the role to the copy
1480 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
1482 * gst/rtsp-server/rtsp-permissions.c:
1483 permissions: Also copy the roles
1485 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
1487 * gst/rtsp-server/rtsp-permissions.c:
1488 permissions: Make it build
1490 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1492 * gst/rtsp-server/rtsp-address-pool.h:
1495 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1497 * docs/libs/gst-rtsp-server-sections.txt:
1498 * gst/rtsp-server/rtsp-auth.c:
1499 * gst/rtsp-server/rtsp-auth.h:
1500 * gst/rtsp-server/rtsp-media.c:
1501 * gst/rtsp-server/rtsp-session-media.c:
1502 * gst/rtsp-server/rtsp-stream-transport.c:
1503 * gst/rtsp-server/rtsp-stream-transport.h:
1504 * gst/rtsp-server/rtsp-stream.c:
1505 * tests/check/gst/client.c:
1508 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1510 * docs/libs/gst-rtsp-server-sections.txt:
1511 * gst/rtsp-server/rtsp-address-pool.c:
1512 * gst/rtsp-server/rtsp-address-pool.h:
1513 * tests/check/gst/addresspool.c:
1514 * tests/check/gst/rtspserver.c:
1515 address-pool: cleanups
1516 Remove redundant method, improve docs.
1518 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1520 * docs/libs/gst-rtsp-server-sections.txt:
1521 * gst/rtsp-server/rtsp-auth.h:
1522 * gst/rtsp-server/rtsp-permissions.c:
1523 * gst/rtsp-server/rtsp-permissions.h:
1524 * gst/rtsp-server/rtsp-token.c:
1527 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1529 * gst/rtsp-server/rtsp-permissions.c:
1530 permissions: implement _remove_role
1532 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1534 * gst/rtsp-server/rtsp-permissions.c:
1535 permissions: update docs
1537 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1539 * tests/check/gst/client.c:
1540 tests: simplify tests
1541 Client settings are now disabled by default so we don't need an auth
1542 module to disable them.
1544 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1546 * gst/rtsp-server/rtsp-auth.c:
1547 auth: add default authorizations
1548 When no auth module is specified, use our table of defaults to look up the
1549 default value of the check instead of always allowing everything. This was
1550 we can disallow client settings by default.
1552 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1555 README: update readme
1557 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1559 * gst/rtsp-server/rtsp-thread-pool.c:
1560 * gst/rtsp-server/rtsp-thread-pool.h:
1561 thread-pool: add more docs
1563 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1565 * gst/rtsp-server/rtsp-thread-pool.c:
1566 * gst/rtsp-server/rtsp-thread-pool.h:
1567 thread-pool: fix race in thread reuse
1568 If we try to reuse a thread right after we made it stop, we end up using a
1569 stopped thread. Catch this case and only reuse threads that are not stopping.
1571 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1573 * gst/rtsp-server/rtsp-server.c:
1574 server: add small debug
1576 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1578 * tests/check/gst/client.c:
1580 Add some permissions to media so we can use the auth and enable
1583 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1585 * gst/rtsp-server/rtsp-client.c:
1586 client: support pushed context in handle_request
1587 If we already have a pushed state, reuse it and add our own things. This makes
1588 it easier to write tests.
1590 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1592 * gst/rtsp-server/rtsp-auth.c:
1593 auth: don't auth on methods
1594 Don't authorize on methods anymore but on the resources that we
1595 try to access, this is more flexible.
1596 Move the authorization checks to where they are needed and let the
1597 check return the response on error.
1599 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1601 * gst/rtsp-server/rtsp-mount-points.c:
1602 mount-points: add some debug
1604 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1606 * tests/check/gst/client.c:
1607 tests: almost fix test
1609 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1611 * gst/rtsp-server/rtsp-auth.c:
1612 * gst/rtsp-server/rtsp-auth.h:
1613 * gst/rtsp-server/rtsp-client.c:
1614 * gst/rtsp-server/rtsp-client.h:
1615 * gst/rtsp-server/rtsp-server.c:
1616 * gst/rtsp-server/rtsp-server.h:
1617 auth: let the auth module check client_settings
1618 Let the auth module decide if client settings are allowed for the
1621 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1623 * gst/rtsp-server/rtsp-token.c:
1624 * gst/rtsp-server/rtsp-token.h:
1625 token: add method to check boolean permission
1627 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1629 * examples/test-auth.c:
1630 * examples/test-cgroups.c:
1631 * gst/rtsp-server/rtsp-token.c:
1632 * gst/rtsp-server/rtsp-token.h:
1633 token: simplify token constructor
1634 Use variable arguments to make easier API.
1636 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1638 * examples/test-auth.c:
1639 * examples/test-cgroups.c:
1640 * gst/rtsp-server/rtsp-media-factory.c:
1641 * gst/rtsp-server/rtsp-media-factory.h:
1642 media-factory: add convenience API for factory
1644 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1646 * examples/test-auth.c:
1647 * examples/test-cgroups.c:
1648 * gst/rtsp-server/rtsp-permissions.c:
1649 * gst/rtsp-server/rtsp-permissions.h:
1650 permissions: simplify API a little
1651 Avoid passing GstStructure in the add_role method, use varargs instead
1652 to construct the structure behind the scenes. We can then also use the
1653 structure name as the role and simplify some more logic.
1655 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1657 * gst/rtsp-server/rtsp-auth.c:
1660 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1662 * gst/rtsp-server/rtsp-auth.c:
1663 * gst/rtsp-server/rtsp-auth.h:
1664 * gst/rtsp-server/rtsp-client.c:
1665 auth: handle unauthorized response
1666 Move handling of the unauthorized response to the auth module, it can add
1667 the appropriate headers to request authorization for the required method
1668 much better than the client.
1670 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1672 * gst/rtsp-server/rtsp-client.c:
1673 * gst/rtsp-server/rtsp-client.h:
1674 client: allow for sending any message, not only requests
1675 Change the _send_request() method to _send_message() so that we
1676 can both send requests and replies.
1678 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1680 * docs/libs/gst-rtsp-server-sections.txt:
1681 * gst/rtsp-server/rtsp-server.h:
1684 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1686 * examples/test-video.c:
1687 * gst/rtsp-server/rtsp-auth.c:
1688 * gst/rtsp-server/rtsp-auth.h:
1689 * gst/rtsp-server/rtsp-server.c:
1690 * gst/rtsp-server/rtsp-server.h:
1691 auth: move TLS handling to auth module
1692 Remove the TLS settings on the server and move it to the auth module because
1693 that is where security related bits go.
1695 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1697 * gst/rtsp-server/rtsp-client.c:
1698 * gst/rtsp-server/rtsp-client.h:
1699 client: add state push/pop
1701 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1703 * gst/rtsp-server/rtsp-client.c:
1704 * gst/rtsp-server/rtsp-client.h:
1705 client: add connection to state
1707 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1709 * gst/rtsp-server/rtsp-mount-points.c:
1710 mount-points: fix debug
1712 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1714 * tests/check/gst/media.c:
1715 tests: fix media test
1717 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1719 * gst/rtsp-server/rtsp-thread-pool.c:
1720 thread-pool: we don't require a state
1722 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1724 * gst/rtsp-server/rtsp-server.c:
1725 server: let context ref the server
1726 So that we don't risk losing the server object early anc crash.
1728 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1730 * tests/check/gst/client.c:
1731 tests: fix client test
1733 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1736 * docs/libs/gst-rtsp-server-docs.sgml:
1737 * docs/libs/gst-rtsp-server-sections.txt:
1738 * gst/rtsp-server/rtsp-address-pool.c:
1739 * gst/rtsp-server/rtsp-auth.c:
1740 * gst/rtsp-server/rtsp-client.c:
1741 * gst/rtsp-server/rtsp-client.h:
1742 * gst/rtsp-server/rtsp-media-factory-uri.c:
1743 * gst/rtsp-server/rtsp-media-factory.c:
1744 * gst/rtsp-server/rtsp-media-factory.h:
1745 * gst/rtsp-server/rtsp-media.c:
1746 * gst/rtsp-server/rtsp-mount-points.c:
1747 * gst/rtsp-server/rtsp-params.c:
1748 * gst/rtsp-server/rtsp-permissions.c:
1749 * gst/rtsp-server/rtsp-sdp.c:
1750 * gst/rtsp-server/rtsp-server.c:
1751 * gst/rtsp-server/rtsp-server.h:
1752 * gst/rtsp-server/rtsp-session-media.c:
1753 * gst/rtsp-server/rtsp-session-pool.c:
1754 * gst/rtsp-server/rtsp-session.c:
1755 * gst/rtsp-server/rtsp-stream-transport.c:
1756 * gst/rtsp-server/rtsp-stream.c:
1757 * gst/rtsp-server/rtsp-thread-pool.c:
1758 * gst/rtsp-server/rtsp-token.c:
1761 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1763 * gst/rtsp-server/rtsp-session-pool.c:
1764 * gst/rtsp-server/rtsp-session-pool.h:
1765 session-pool: make vmethod to create a session
1766 Make a vmethod to create a sessions so that subclasses can create
1767 custom session objects
1769 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1771 * gst/rtsp-server/rtsp-auth.c:
1772 * gst/rtsp-server/rtsp-media-factory.h:
1773 * gst/rtsp-server/rtsp-media.h:
1774 * gst/rtsp-server/rtsp-mount-points.h:
1775 * gst/rtsp-server/rtsp-session-pool.h:
1776 * gst/rtsp-server/rtsp-stream.h:
1779 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1781 * docs/libs/gst-rtsp-server-docs.sgml:
1782 * docs/libs/gst-rtsp-server-sections.txt:
1783 * gst/rtsp-server/rtsp-address-pool.c:
1784 * gst/rtsp-server/rtsp-address-pool.h:
1785 * gst/rtsp-server/rtsp-auth.c:
1786 * gst/rtsp-server/rtsp-client.h:
1787 * gst/rtsp-server/rtsp-media-factory.h:
1788 * gst/rtsp-server/rtsp-media.c:
1789 * gst/rtsp-server/rtsp-media.h:
1790 * gst/rtsp-server/rtsp-permissions.c:
1791 * gst/rtsp-server/rtsp-permissions.h:
1792 * gst/rtsp-server/rtsp-server.h:
1793 * gst/rtsp-server/rtsp-session-media.c:
1794 * gst/rtsp-server/rtsp-session-media.h:
1795 * gst/rtsp-server/rtsp-session-pool.h:
1796 * gst/rtsp-server/rtsp-session.h:
1797 * gst/rtsp-server/rtsp-stream-transport.h:
1798 * gst/rtsp-server/rtsp-stream.c:
1799 * gst/rtsp-server/rtsp-thread-pool.h:
1802 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1805 * examples/Makefile.am:
1806 configure: compile cgroup example conditionally
1807 Only compile the cgroup example when we have libcgroup
1809 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1812 * examples/Makefile.am:
1813 * examples/test-cgroups.c:
1814 examples: add cgroups example
1816 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1818 * tests/check/gst/rtspserver.c:
1819 tests: fix compilation
1821 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1823 * gst/rtsp-server/rtsp-thread-pool.c:
1824 thread-pool: fix vmethod invocation
1826 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1828 * gst/rtsp-server/rtsp-thread-pool.c:
1829 * gst/rtsp-server/rtsp-thread-pool.h:
1830 thread-pool: store thread type in thread
1832 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1834 * gst/rtsp-server/rtsp-client.c:
1835 client: pass thread from pool to media _prepare
1836 Get a thread from the configured threadpool and pass it to the prepare method of
1839 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1841 * gst/rtsp-server/rtsp-media.c:
1842 * gst/rtsp-server/rtsp-media.h:
1843 media: Accept a thread in _prepare
1844 Remove out own threadpool handling and use the provided thread and
1845 maincontext for the bus messages and the state changes.
1847 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1849 * gst/rtsp-server/rtsp-server.c:
1850 server: configure client thread pool
1852 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1854 * gst/rtsp-server/rtsp-client.c:
1855 * gst/rtsp-server/rtsp-client.h:
1856 client: add method to configure thread pool
1858 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1860 * gst/rtsp-server/rtsp-client.h:
1861 * gst/rtsp-server/rtsp-server.c:
1862 * gst/rtsp-server/rtsp-server.h:
1863 server: use thread pool
1864 Use the thread pool instead of doing our own thing.
1866 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1868 * gst/rtsp-server/Makefile.am:
1869 * gst/rtsp-server/rtsp-thread-pool.c:
1870 * gst/rtsp-server/rtsp-thread-pool.h:
1871 thread-pool: add object to manage threads
1872 Add an object to manage the client and media threads.
1874 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1876 * gst/rtsp-server/rtsp-auth.c:
1877 auth: debug authorization check
1879 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1881 * gst/rtsp-server/rtsp-media.c:
1882 media: start media pipeline in context
1883 Start the media pipeline in the provided context (or our default one
1884 when NULL). This makes sure that we run the bus thread in this context and that
1885 all media threads are children of this context.
1887 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1889 * gst/rtsp-server/rtsp-media-factory.c:
1890 factory: pass permissions to media by default
1892 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1894 * examples/test-auth.c:
1895 test: add permissions to auth test
1896 Ass some permissions to the media factory in the test.
1898 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1900 * gst/rtsp-server/rtsp-auth.c:
1901 * gst/rtsp-server/rtsp-auth.h:
1902 * gst/rtsp-server/rtsp-client.c:
1903 auth: simplify auth checks
1904 Remove client from methods, it's now in the state
1905 Perform the check specified by the string, use the information from the
1906 thread local context.
1908 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1910 * gst/rtsp-server/rtsp-client.c:
1911 * gst/rtsp-server/rtsp-client.h:
1912 client: add state to current thread
1913 Add the client to the ClientState object.
1914 Place the ClientState on the current thread.
1916 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1918 * gst/rtsp-server/rtsp-media-factory.c:
1919 * gst/rtsp-server/rtsp-media-factory.h:
1920 * gst/rtsp-server/rtsp-media.c:
1921 * gst/rtsp-server/rtsp-media.h:
1922 media: make it possible to set permissions
1923 Make it possible to set permissions on media and media factory objects
1925 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1927 * gst/rtsp-server/Makefile.am:
1928 * gst/rtsp-server/rtsp-permissions.c:
1929 * gst/rtsp-server/rtsp-permissions.h:
1930 permissions: add permissions object
1931 Add a mini object to store permissions based on a role.
1933 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1935 * examples/test-auth.c:
1936 * gst/rtsp-server/rtsp-auth.c:
1937 * gst/rtsp-server/rtsp-auth.h:
1938 * gst/rtsp-server/rtsp-client.c:
1939 auth: add auth checks
1940 Add an enum with auth checks and implement the checks in the auth object.
1941 Perform the checks from the client.
1943 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1945 * examples/test-auth.c:
1946 * gst/rtsp-server/rtsp-auth.c:
1947 * gst/rtsp-server/rtsp-auth.h:
1948 * gst/rtsp-server/rtsp-client.h:
1949 auth: use the token after authentication
1950 After we authenticated a user, keep the Token around in the state.
1952 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1954 * gst/rtsp-server/rtsp-client.c:
1955 * gst/rtsp-server/rtsp-media.c:
1956 * gst/rtsp-server/rtsp-media.h:
1957 * tests/check/gst/media.c:
1958 media: add optional context for bus messages
1959 Add an optional mainloop to _prepare that will handle the bus messages instead
1960 of always using the shared mainloop.
1962 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1964 * gst/rtsp-server/Makefile.am:
1965 * gst/rtsp-server/rtsp-token.c:
1966 * gst/rtsp-server/rtsp-token.h:
1967 token: add authorization token
1968 Add a simply miniobject that contains the authorizations. The object contains a
1969 GstStructure that hold all authorization fields. When a user is authenticated,
1970 the auth module will create a Token for the user. The token is then used to
1971 check what operations the user is allowed to do and various other configuration
1974 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1976 * examples/test-auth.c:
1977 * gst/rtsp-server/rtsp-auth.c:
1978 * gst/rtsp-server/rtsp-auth.h:
1979 * gst/rtsp-server/rtsp-client.c:
1980 * gst/rtsp-server/rtsp-client.h:
1981 * gst/rtsp-server/rtsp-media-factory.c:
1982 * gst/rtsp-server/rtsp-media-factory.h:
1983 * gst/rtsp-server/rtsp-media.c:
1984 * gst/rtsp-server/rtsp-media.h:
1985 auth: remove auth from media and factory
1986 Remove the auth object from media and factory. We want to have the RTSPClient
1987 authenticate and authorize resources, there is no need to place another auth
1988 manager on the media/factory.
1990 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1992 * examples/test-auth.c:
1993 * gst/rtsp-server/rtsp-auth.c:
1994 * gst/rtsp-server/rtsp-auth.h:
1995 * gst/rtsp-server/rtsp-client.h:
1996 auth: add support for multiple basic auth tokens
1997 Make it possible to add multiple basic authorisation tokens to one authorization
1998 object. Associate with each token an authorization group that will define what
1999 capabilities are allowed.
2001 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2003 * gst/rtsp-server/rtsp-client.c:
2004 client: error out on non-aggregate control
2005 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
2007 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2009 * gst/rtsp-server/rtsp-client.c:
2010 client: rework setup request a little
2011 Cache the media in DESCRIBE based on the longest matching path with the uri
2012 that we can find in the mount points.
2013 Rework the setup request a little to get the media from the session or from
2014 the longest matching path, this way we can derive the control string as
2015 everything after the path instead of hardcoding it.
2016 Find the stream based on the control string and only open a session when all
2019 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2021 * gst/rtsp-server/rtsp-media.c:
2022 * gst/rtsp-server/rtsp-media.h:
2023 media: add method to find a stream by control url
2025 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2027 * gst/rtsp-server/rtsp-stream.c:
2028 * gst/rtsp-server/rtsp-stream.h:
2029 stream: add method to check control url of stream
2031 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2033 * gst/rtsp-server/rtsp-client.c:
2034 * gst/rtsp-server/rtsp-session-media.c:
2035 * gst/rtsp-server/rtsp-session-media.h:
2036 * gst/rtsp-server/rtsp-session.c:
2037 * gst/rtsp-server/rtsp-session.h:
2038 session: use path matching for session media
2039 Use a path string instead of a uri to lookup session media in the sessions. Also
2040 use path matching to find the largest possible path that matches.
2042 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2044 * gst/rtsp-server/rtsp-client.c:
2045 * gst/rtsp-server/rtsp-mount-points.c:
2046 * gst/rtsp-server/rtsp-mount-points.h:
2047 * tests/check/gst/mountpoints.c:
2048 mount-points: remove useless vmethod
2049 Making lookups in the mount points should not be done with a URL, if there is a
2050 mapping to be done from URL to mount points, we'll need to do it somewhere
2053 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2055 * gst/rtsp-server/rtsp-mount-points.c:
2056 * gst/rtsp-server/rtsp-mount-points.h:
2057 * tests/check/gst/mountpoints.c:
2058 mount-points: improve mount point searching
2059 Use a GSequence to keep track of the mount points.
2060 Match a URL to the longest matching registered mount point. This should be the
2061 URL to perform aggreagate control and the remainder is the stream specific
2063 Add some unit tests for this.
2065 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
2067 * gst/rtsp-server/Makefile.am:
2068 rtsp-server: Allow building of static library
2070 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2072 * tests/check/gst/mediafactory.c:
2073 tests: fix compilation
2075 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2077 * gst/rtsp-server/rtsp-sdp.c:
2078 sdp: get control string from stream
2079 Use the control string as configured in the stream.
2081 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2083 * gst/rtsp-server/rtsp-stream.c:
2084 * gst/rtsp-server/rtsp-stream.h:
2085 stream: add methods and property to set control string
2087 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2089 * gst/rtsp-server/rtsp-client.c:
2091 Rename variables for clarity
2092 Keep media in state when we can
2094 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2096 * gst/rtsp-server/rtsp-client.c:
2097 * gst/rtsp-server/rtsp-stream.c:
2098 * gst/rtsp-server/rtsp-stream.h:
2099 stream: add more support for IPv6
2100 Rename _get_address to _get_multicast_address in GstRTSPStream to
2101 make it clear that this function only deals with multicast.
2102 Make it possible to have both an IPv4 and IPv6 multicast address on
2103 a stream. Give the client an IPv4 or IPv6 address depending on the
2104 address it used to connect to the server.
2105 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
2107 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2109 * gst/rtsp-server/rtsp-client.c:
2112 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2114 * gst/rtsp-server/rtsp-stream.c:
2115 stream: handle failed port allocation
2116 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
2117 can't allocate any family at all. Also keep track of what port families we
2119 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
2121 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2123 * gst/rtsp-server/rtsp-stream.c:
2124 stream: improve docs
2126 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2128 * gst/rtsp-server/rtsp-stream-transport.c:
2129 stream-transport: remove old if 0 block
2131 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
2133 * tests/check/gst/client.c:
2135 gst_rtsp_client_get_uri() has been removed
2136 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
2138 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2140 * gst/rtsp-server/rtsp-client.c:
2141 * gst/rtsp-server/rtsp-client.h:
2142 client: add method to filter managed sessions
2143 Add a method to filter the sessions managed by this client connection.
2144 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
2146 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2148 * gst/rtsp-server/rtsp-client.c:
2149 * gst/rtsp-server/rtsp-client.h:
2150 client: remove _get_uri() method
2151 Remove the get_uri() method on the client. A client has no uri, the uri
2152 property is an internal property to manage the last cached media for
2155 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2157 * gst/rtsp-server/rtsp-media-factory.h:
2158 media-factory: fix typo
2160 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
2162 * gst/rtsp-server/rtsp-media.c:
2163 rtsp-media: Do not leak the query in default_query_stop
2164 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
2166 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2168 * gst/rtsp-server/rtsp-media.c:
2169 media: don't unlock when conversion fails
2170 Don't unlock the state lock when conversion fails because it was not locked.
2172 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2174 * gst/rtsp-server/rtsp-media.c:
2175 * gst/rtsp-server/rtsp-media.h:
2176 Add query_position and query_stop vmethods to rtsp-media
2178 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2180 * gst/rtsp-server/rtsp-media.c:
2181 Fix typo in property install for rtsp-media's time-provider
2183 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2185 * gst/rtsp-server/rtsp-client.c:
2186 * gst/rtsp-server/rtsp-client.h:
2187 client: clean some variables
2188 Clean some variables and add some guards to _send_request()
2190 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2192 * gst/rtsp-server/rtsp-client.c:
2193 * gst/rtsp-server/rtsp-client.h:
2194 Add gst_rtsp_client_send_request API
2195 This makes it possible to send arbitrary messages to a client, such as
2196 SET_PARAMETER or GET_PARAMETER
2198 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2200 * gst/rtsp-server/rtsp-media.c:
2201 * gst/rtsp-server/rtsp-media.h:
2202 media: add _get_element() method
2203 Add method to get the element used when creating the media.
2204 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
2206 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2208 * gst/rtsp-server/rtsp-media.c:
2211 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
2213 * gst/rtsp-server/rtsp-stream.c:
2214 * gst/rtsp-server/rtsp-stream.h:
2215 stream: allow access to the rtp session
2216 https://bugzilla.gnome.org/show_bug.cgi?id=703004
2218 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
2220 * gst/rtsp-server/rtsp-stream.c:
2221 * gst/rtsp-server/rtsp-stream.h:
2222 dscp qos support in gst-rtsp-stream
2223 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
2225 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2227 * tests/check/gst/rtspserver.c:
2229 Actually do what the comment says. Also keep the old code around, not sure what
2230 should happen when you get a 454 from a TEARDOWN, does it close the connection?
2231 it currently doesn't.
2233 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2235 * gst/rtsp-server/rtsp-client.c:
2236 client: also watch newly created session
2237 When we newly created a session, start watching it immediately instead of
2238 on the next request.
2240 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
2242 * tests/check/gst/client.c:
2243 tests: add unit test for new-session
2244 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
2246 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2248 * gst/rtsp-server/rtsp-client.c:
2249 client: emit new-session when new session is created
2250 Only emit new-session when we created a new session for a client, not when a
2251 client picked up a previous session.
2252 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
2254 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
2256 * gst/rtsp-server/rtsp-client.c:
2257 client: handle asterisk as path in requests
2258 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
2260 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2262 * gst/rtsp-server/rtsp-media.c:
2263 media: handle segment query format mismatch
2264 It's possible that the segment query returns with a different format than what
2265 we asked for, handle this case also.
2267 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
2269 * gst/rtsp-server/rtsp-media.c:
2270 media: use segment stop in collect_media_stats
2271 Use segment stop instead of duration as range end point.
2272 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
2274 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
2276 * gst/rtsp-server/rtsp-media.c:
2277 * tests/check/gst/media.c:
2278 rtsp-media: Do not leak the element in take_pipeline
2279 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
2281 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
2283 * gst/rtsp-server/rtsp-client.c:
2284 * gst/rtsp-server/rtsp-client.h:
2285 rtsp-client: Make configure_client_transport virtual
2286 This patch makes configure_client_transport virtual. The functionality is
2287 needed to handle some weird clients sending multicast transport settings as url
2289 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
2291 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
2293 * gst/rtsp-server/rtsp-client.c:
2294 * gst/rtsp-server/rtsp-client.h:
2295 rtsp-client: Make param_set and param_get virtual
2296 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
2298 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
2300 * gst/rtsp-server/rtsp-client.c:
2301 * gst/rtsp-server/rtsp-media.c:
2302 * gst/rtsp-server/rtsp-media.h:
2303 media: convert_range replaces get_range_times
2304 get_range_times worked for handling UTC ranges for seeks, but we also
2305 need to convert back from NPT to the requested unit in
2306 get_range_string. convert_range is now used for both.
2307 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
2309 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2311 * gst/rtsp-server/rtsp-client.c:
2312 * gst/rtsp-server/rtsp-sdp.c:
2313 * gst/rtsp-server/rtsp-sdp.h:
2314 sdp: cleanup sdp info
2315 We don't need to pass the proto, we can more easily check a boolean.
2316 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
2318 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
2320 * gst/rtsp-server/rtsp-sdp.c:
2321 use 0.0.0.0 or :: for c= line instead of server address
2323 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
2325 * gst/rtsp-server/rtsp-client.c:
2326 use local address, not remote, in SDP
2327 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
2329 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2332 Automatic update of common submodule
2333 From 098c0d7 to 01a7a46
2335 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
2337 * gst/rtsp-server/rtsp-media.c:
2338 * gst/rtsp-server/rtsp-media.h:
2339 media: possibility to override range time conversion
2340 Make it possible to override the conversion from GstRTSPTimeRange to
2341 GstClockTimes, that is done before seeking on the media
2342 pipeline. Overriding can be useful for UTC ranges, where the default
2343 conversion gives nanoseconds since 1900.
2344 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
2346 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
2348 * gst/rtsp-server/rtsp-server.c:
2349 * gst/rtsp-server/rtsp-server.h:
2350 rtsp-server: Expose the use_client_settings API
2351 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
2353 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
2355 * gst/rtsp-server/rtsp-client.c:
2356 * gst/rtsp-server/rtsp-stream.c:
2357 * gst/rtsp-server/rtsp-stream.h:
2358 rtspstream: handle both ipv4 and ipv6 clients
2359 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
2361 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2363 * gst/rtsp-server/rtsp-sdp.c:
2364 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
2365 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
2366 We already have a way to place extra attributes in the SDP by using a string
2367 property with prefix x- or a- in the caps.
2369 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2371 * gst/rtsp-server/rtsp-sdp.c:
2372 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
2373 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
2374 We already have a way to place extra attributes in the SDP, just make a string
2375 property in the payloader with a- or x- prefix.
2377 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2379 * gst/rtsp-server/rtsp-sdp.c:
2380 rtsp: place a- and x- properties as attributes
2381 application/x-rtp has properties with a- and x- prefixes that should be
2382 placed as attributes in the SDP for the media instead of being added to the
2385 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2387 * examples/Makefile.am:
2388 * examples/test-video.c:
2389 example: add TLS example
2391 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2393 * gst/rtsp-server/rtsp-server.c:
2394 * gst/rtsp-server/rtsp-server.h:
2395 server: add support for TLS
2396 Add methods to set and get a TLS certificate.
2397 Add vmethod to configure a new connection. By default, configure the TLS
2398 certificate in a new connection if needed.
2400 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2402 * gst/rtsp-server/rtsp-server.c:
2403 * gst/rtsp-server/rtsp-server.h:
2404 server: remove accept_client vmethod
2405 This vmethod is not very useful so remove it.
2407 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2409 * gst/rtsp-server/rtsp-server.c:
2410 server: don't crash on NULL GError
2412 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
2414 * gst/rtsp-server/rtsp-session-pool.c:
2415 rtsp-session-pool: corrected session timeout detection
2416 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
2418 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2420 * gst/rtsp-server/rtsp-client.c:
2421 client: improve debug
2423 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2425 * gst/rtsp-server/rtsp-client.c:
2426 * gst/rtsp-server/rtsp-client.h:
2427 * gst/rtsp-server/rtsp-server.c:
2428 server: refactor connection setup
2429 Let the server accept the socket connection and construct a GstRTSPConnection
2430 from it. Remove the code from the client and let the client only deal with
2431 a fully configure GstRTSPConnection object.
2432 We will need this later when the server will configure the connection for
2435 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2437 * gst/rtsp-server/rtsp-stream.c:
2438 stream: keep the transport object alive
2439 Keep the transport object alive while we have it as qdata on the
2442 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
2444 * gst/rtsp-server/rtsp-client.c:
2445 * gst/rtsp-server/rtsp-server.c:
2446 rtsp-server: Do not crash on nmapping of server
2447 * generate error when gst_rtsp_connection_accept fails
2448 * do not stop accepting incoming connections because
2449 accepting a client fails
2450 https://bugzilla.gnome.org/show_bug.cgi?id=701072
2452 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
2454 * gst/rtsp-server/rtsp-client.c:
2455 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
2456 https://bugzilla.gnome.org/show_bug.cgi?id=700953
2458 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
2460 * gst/rtsp-server/rtsp-sdp.c:
2461 rtsp-sdp: Parse framerate caps field and set SDP attribute
2462 The SDP attribute and its format is described in RFC4566.
2463 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
2465 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
2467 * gst/rtsp-server/rtsp-sdp.c:
2468 rtsp-sdp: Parse width/height from caps and set SDP attribute
2469 The SDP attribute and its format is described in RFC6064.
2470 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
2472 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
2474 * gst/rtsp-server/rtsp-sdp.c:
2475 * tests/check/gst/client.c:
2476 rtsp-sdp: add bandwidth line
2477 https://bugzilla.gnome.org/show_bug.cgi?id=699220
2479 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2482 Automatic update of common submodule
2483 From 5edcd85 to 098c0d7
2485 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
2487 * tests/check/gst/media.c:
2488 tests: add dynamic payloader prepare/unprepare check
2490 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2492 * gst/rtsp-server/rtsp-media.c:
2493 media: release lock when removing fakesink
2495 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2497 * gst/rtsp-server/rtsp-stream.c:
2498 stream: set elements to NULL before removing
2499 When removing a stream, set the elements to NULL first. This avoids
2500 element-is-not-in-NULL-state errors when we dispose the elements.
2502 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
2505 Automatic update of common submodule
2506 From 3cb3d3c to 5edcd85
2508 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2510 * gst/rtsp-server/rtsp-media.c:
2511 * gst/rtsp-server/rtsp-media.h:
2512 media: listen to pad-removed signals
2513 Listen to the pad-removed signal and remove the stream associated with the
2515 Add signal to be notified of the removed pad.
2516 Remove the fakesink in unprepare()
2517 Fix signatures of the signal methods
2519 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2521 * examples/test-sdp.c:
2522 tests: add example of reusable pipelines
2524 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
2526 * gst/rtsp-server/rtsp-stream.c:
2527 * gst/rtsp-server/rtsp-stream.h:
2528 stream: add method to get the srcpad
2530 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
2532 * tests/check/gst/media.c:
2533 check: add media prepare/unprepare test
2534 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2536 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
2538 * gst/rtsp-server/rtsp-media.c:
2539 media: disconnect from signal handlers in unprepare()
2540 We connected to the pad-added and no-more-pads signals in prepare() so
2541 we need to disconnect from them in unprepare().
2542 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2544 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
2546 * gst/rtsp-server/rtsp-media.c:
2547 media: don't free streams array
2548 Don't free the streams array in the unprepare() method, they were not
2550 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2552 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
2554 * gst/rtsp-server/rtsp-media.c:
2555 media: don't unref the pipeline in unprepare
2556 Unprepare() should undo what prepare() does. Because the pipeline is
2557 not created in prepare(), we should not unref it in unprepare()
2559 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
2561 * gst/rtsp-server/rtsp-stream.c:
2562 stream: clear session and caps for reuse
2563 Set the session and caps to NULL after unref otherwise we might unref
2565 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2567 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
2569 * gst/rtsp-server/rtsp-client.c:
2570 client: send out teardown signal before tearing down
2571 The advantage is that in the signal handler you get direct access to
2572 information about what streams are about to get torn down (in the
2573 GstRTSPClientState).
2574 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
2576 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
2578 * gst/rtsp-server/rtsp-client.c:
2579 * gst/rtsp-server/rtsp-client.h:
2580 client: expose connection
2581 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
2583 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
2586 Automatic update of common submodule
2587 From aed87ae to 3cb3d3c
2589 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2591 * gst/rtsp-server/rtsp-media.c:
2592 * gst/rtsp-server/rtsp-media.h:
2593 * gst/rtsp-server/rtsp-session-media.c:
2594 * gst/rtsp-server/rtsp-session-media.h:
2595 media: add method to get the base_time of the pipeline
2596 Together with a shared clock, this base-time could eventually be sent to
2597 the client so that it can reconstruct the exact running-time of the clock
2600 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2602 * gst/rtsp-server/Makefile.am:
2603 * gst/rtsp-server/rtsp-media.c:
2604 * gst/rtsp-server/rtsp-media.h:
2605 * gst/rtsp-server/rtsp-sdp.c:
2606 media: add GstNetTimeProvider support
2607 Add a property to let the media provide a GstNetTimeProvider for its clock.
2608 Make methods to get the clock and nettimeprovider
2609 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
2610 provider and also the current time of the clock. This should make it possible
2611 for (GStreamer) clients to slave their clock to the server clock.
2613 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
2616 Automatic update of common submodule
2617 From 04c7a1e to aed87ae
2619 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2621 * gst/rtsp-server/rtsp-media.c:
2622 media: wait for buffering to complete
2623 Wait for buffering to complete before changing the state to the target state.
2625 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2627 * gst/rtsp-server/rtsp-media.c:
2628 media: small cleanup
2630 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
2632 * tests/check/gst/rtspserver.c:
2633 tests: remove extra unref in test_setup_non_existing_stream
2634 The unref is not needed anymore, teardown runs without it.
2635 https://bugzilla.gnome.org/show_bug.cgi?id=696542
2637 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
2639 * tests/check/gst/rtspserver.c:
2640 tests: GSocketService cleanup in test_bind_already_in_use
2641 Use g_socket_service_stop so the rtspserver test stops listening for
2642 incoming connections in test_bind_already_in_use.
2643 https://bugzilla.gnome.org/show_bug.cgi?id=696541
2645 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
2647 * gst/rtsp-server/rtsp-media-factory.c:
2648 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
2649 Instead use a GWeakRef which is safe to use
2650 This is a known GLib bug, see:
2651 https://bugzilla.gnome.org/show_bug.cgi?id=667145
2653 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
2655 * gst/rtsp-server/rtsp-client.c:
2656 * gst/rtsp-server/rtsp-media.c:
2657 * gst/rtsp-server/rtsp-media.h:
2658 * gst/rtsp-server/rtsp-sdp.c:
2659 * tests/check/gst/media.c:
2660 * tests/check/gst/rtspserver.c:
2661 rtsp-media/client: Reply to PLAY request with same type of Range
2662 Remember the type of Range from the PLAY request and use the same type for
2665 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
2667 * gst/rtsp-server/rtsp-client.c:
2668 * gst/rtsp-server/rtsp-client.h:
2669 * tests/check/gst/client.c:
2670 rtsp-client: expose uri
2672 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
2674 * tests/check/gst/mediafactory.c:
2675 tests: Hold ref while creating second media
2676 To test if the media aren't shared, make sure we keep the first one while creating a second
2677 otherwise the same memory address may be reused.
2679 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
2682 configure: remove out-of-date comment
2684 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
2687 .gitignore: ignore more build files
2689 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
2691 * tests/check/Makefile.am:
2692 tests: use right _LIBS variable for gst-plugins-base libs
2694 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2696 * tests/check/Makefile.am:
2697 check: add librtp to libs
2699 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
2701 * tests/check/gst/rtspserver.c:
2702 tests: Add test to check selecting a port the server will send from
2704 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
2706 * tests/check/gst/rtspserver.c:
2707 tests: Make sure packets are actually received
2709 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
2711 * gst/rtsp-server/rtsp-stream.c:
2712 stream: Select unicast address from pool if appropriate
2714 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
2716 * gst/rtsp-server/rtsp-stream.c:
2717 stream: Properties are always there in Gst 1.0
2719 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
2721 * tests/check/gst/addresspool.c:
2722 tests: Add tests for unicast addresses in pool
2724 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
2726 * gst/rtsp-server/rtsp-address-pool.c:
2727 * tests/check/gst/addresspool.c:
2728 address-pool: Verify that multicast addresses are used for multicast and vice-versa
2730 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
2732 * docs/libs/gst-rtsp-server-sections.txt:
2733 * gst/rtsp-server/rtsp-address-pool.c:
2734 * gst/rtsp-server/rtsp-address-pool.h:
2735 * gst/rtsp-server/rtsp-stream.c:
2736 * tests/check/gst/addresspool.c:
2737 address-pool: Add unicast addresses
2739 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
2742 * gst/rtsp-server/rtsp-server.c:
2743 * tests/check/gst/rtspserver.c:
2744 rtsp-server: Limit the number of threads per server instance
2745 If we exceed the maximum, just round robin the clients over the existing
2748 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
2750 * gst/rtsp-server/rtsp-server.c:
2751 rtsp-server: No need to store the GMainContext in the client context
2753 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
2755 * tests/check/gst/rtspserver.c:
2756 tests: Add test for client disconnection
2758 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
2760 * tests/check/gst/rtspserver.c:
2761 tests: Test client and session timeouts with multiple threads
2763 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
2765 * gst/rtsp-server/rtsp-address-pool.c:
2766 * gst/rtsp-server/rtsp-auth.c:
2767 * gst/rtsp-server/rtsp-client.c:
2768 * gst/rtsp-server/rtsp-media-factory-uri.c:
2769 * gst/rtsp-server/rtsp-media-factory.c:
2770 * gst/rtsp-server/rtsp-media.c:
2771 * gst/rtsp-server/rtsp-mount-points.c:
2772 * gst/rtsp-server/rtsp-server.c:
2773 * gst/rtsp-server/rtsp-session-media.c:
2774 * gst/rtsp-server/rtsp-session-pool.c:
2775 * gst/rtsp-server/rtsp-session.c:
2776 Document locking and its order
2778 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
2780 * tests/check/gst/rtspserver.c:
2781 tests: Test that slow DESCRIBE don't block other clients
2783 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
2785 * tests/check/gst/client.c:
2786 tests: Add tests for client-requested multicast address
2788 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
2790 * docs/libs/gst-rtsp-server-sections.txt:
2791 docs: Put the various functions in the right sections
2793 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
2795 * docs/libs/gst-rtsp-server-docs.sgml:
2796 * docs/libs/gst-rtsp-server-sections.txt:
2797 * gst/rtsp-server/rtsp-address-pool.c:
2798 * gst/rtsp-server/rtsp-address-pool.h:
2799 docs: Generate docs for GstRTSPAddressPool
2801 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
2803 * gst/rtsp-server/rtsp-client.c:
2804 * gst/rtsp-server/rtsp-stream.c:
2805 * gst/rtsp-server/rtsp-stream.h:
2806 client: Check client provided addresses against the address pool
2808 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
2810 * gst/rtsp-server/rtsp-address-pool.c:
2811 * gst/rtsp-server/rtsp-address-pool.h:
2812 * tests/check/gst/addresspool.c:
2813 address-pool: Add API to request a specific address from the pool
2814 Also add relevant unit tests.
2816 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
2818 * tests/check/gst/mediafactory.c:
2819 tests: Check the passing around of a RTSPAddressPool
2820 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
2821 way down to the stream.
2823 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
2825 * tests/check/gst/addresspool.c:
2826 tests: Add more tests for the address pool
2828 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
2830 * gst/rtsp-server/rtsp-address-pool.c:
2831 address-pool: Fix off by one error
2832 When splitting a port range, the port after a skip is not part of range.
2834 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
2837 Automatic update of common submodule
2838 From 2de221c to 04c7a1e
2840 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
2843 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
2844 AM_CONFIG_HEADER was removed in automake 1.13
2845 https://bugzilla.gnome.org/show_bug.cgi?id=693368
2847 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
2850 Automatic update of common submodule
2851 From a942293 to 2de221c
2853 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2855 * gst/rtsp-server/rtsp-client.c:
2856 client: make sure the watch exists while sending data
2857 Protect the send_func with a lock. This allows us to wait for sending
2858 to complete before changing the send_func and user_data. We add an
2859 extra ref to the watch to make sure that it remains valid during
2861 When closing the connection, set the send_func to NULL
2862 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
2864 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2866 * tests/check/Makefile.am:
2867 tests: use GST_*_1_0 environment variables everywhere
2868 The _1_0 suffixed environment variables override the
2869 non-suffixed ones, so if we're in an environment that
2870 sets the _1_0 suffixed ones, such as jhbuild, we need
2871 to set those to make sure ours actually always get
2874 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2877 Automatic update of common submodule
2878 From acb04d9 to a942293
2880 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2882 * gst/rtsp-server/rtsp-client.c:
2883 rtsp-client: set the client backlog
2884 Set the client backlog to a reasonable default
2886 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
2888 * gst/rtsp-server/rtsp-media.c:
2889 rtsp-media: Make the element a constructor parameter
2890 https://bugzilla.gnome.org/show_bug.cgi?id=689594
2892 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2894 * docs/libs/Makefile.am:
2895 docs: Link with gcov library when gcov is enabled
2896 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
2898 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2900 * gst/rtsp-server/rtsp-media.c:
2901 media: match prepare with unprepare
2902 Really unprepare when there were an equal amount of prepare calls.
2904 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2906 * gst/rtsp-server/rtsp-media.c:
2907 media: media has to be unprepared in finalize
2908 Because unprepare takes away the last ref on the media.
2910 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2912 * gst/rtsp-server/rtsp-client.c:
2913 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
2914 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
2915 We can't use the refcount to trigger unprepare because it is the unprepare call
2916 that removes the last refcount after all messages are consumed. What we should
2917 probably do is make a prepared refcount and only unprepare when the refcount
2920 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2922 * gst/rtsp-server/rtsp-media.c:
2923 media: let the source unref the last media ref
2924 the last ref to the media is held by the source so we don't need to add more ref
2925 and unrefs, we simply destroy the media when the source is gone.
2927 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2929 * gst/rtsp-server/rtsp-media.c:
2930 media: improve debug
2932 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2934 * gst/rtsp-server/rtsp-media.c:
2936 Make sure we are in the right state when collecting the position and duration.
2937 Only make ourselves PREPARED when we were previously PREPARING.
2939 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2941 * gst/rtsp-server/rtsp-media.c:
2942 media: use g_object_ref/unref for GObjects
2944 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
2946 * gst/rtsp-server/rtsp-client.c:
2947 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
2948 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
2949 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
2950 isn't being used anymore.
2952 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
2954 * gst/rtsp-server/rtsp-media.c:
2955 Fix compiler warning
2957 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
2959 * gst/rtsp-server/rtsp-media-factory-uri.c:
2960 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
2962 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2964 * gst/rtsp-server/rtsp-session-media.h:
2967 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2969 * gst/rtsp-server/rtsp-media.c:
2970 * tests/check/gst/media.c:
2971 media: avoid element leak
2973 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2975 * gst/rtsp-server/rtsp-media.c:
2976 media: require an element in media constructor
2978 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2980 * gst/rtsp-server/rtsp-client.c:
2981 Revert "client: TEARDOWN brings that state to Init again"
2982 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
2983 The object is already disposed, there is no point in setting the state.
2985 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2987 * gst/rtsp-server/rtsp-client.c:
2988 client: TEARDOWN brings that state to Init again
2990 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2992 * docs/libs/gst-rtsp-server-sections.txt:
2993 * examples/test-auth.c:
2994 * gst/rtsp-server/rtsp-auth.c:
2995 * gst/rtsp-server/rtsp-auth.h:
2996 * gst/rtsp-server/rtsp-client.c:
2997 * gst/rtsp-server/rtsp-client.h:
2998 * gst/rtsp-server/rtsp-media-factory-uri.c:
2999 * gst/rtsp-server/rtsp-media-factory-uri.h:
3000 * gst/rtsp-server/rtsp-media-factory.c:
3001 * gst/rtsp-server/rtsp-media-factory.h:
3002 * gst/rtsp-server/rtsp-media.c:
3003 * gst/rtsp-server/rtsp-media.h:
3004 * gst/rtsp-server/rtsp-mount-points.c:
3005 * gst/rtsp-server/rtsp-mount-points.h:
3006 * gst/rtsp-server/rtsp-sdp.c:
3007 * gst/rtsp-server/rtsp-server.c:
3008 * gst/rtsp-server/rtsp-server.h:
3009 * gst/rtsp-server/rtsp-session-media.c:
3010 * gst/rtsp-server/rtsp-session-media.h:
3011 * gst/rtsp-server/rtsp-session-pool.c:
3012 * gst/rtsp-server/rtsp-session-pool.h:
3013 * gst/rtsp-server/rtsp-session.c:
3014 * gst/rtsp-server/rtsp-session.h:
3015 * gst/rtsp-server/rtsp-stream-transport.c:
3016 * gst/rtsp-server/rtsp-stream-transport.h:
3017 * gst/rtsp-server/rtsp-stream.c:
3018 * gst/rtsp-server/rtsp-stream.h:
3019 * tests/check/gst/media.c:
3020 rtsp: make object details private
3021 Make all object details private
3022 Add methods to access private bits
3024 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3026 * tests/check/Makefile.am:
3027 * tests/check/gst/media.c:
3028 tests: add media tests
3030 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3032 * gst/rtsp-server/rtsp-media.c:
3033 media: check if prepared for some methods
3034 Check that the media object is prepared before doing seek and getting the
3035 current position etc.
3036 Add some g_return checks.
3038 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3040 * tests/check/Makefile.am:
3041 * tests/check/gst/mediafactory.c:
3042 tests: add mediafactory test
3044 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3046 * gst/rtsp-server/rtsp-stream.c:
3047 stream: improve debug
3049 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3051 * gst/rtsp-server/rtsp-media.c:
3052 * gst/rtsp-server/rtsp-media.h:
3053 media: unref pipeline in finalize to avoid leaking it
3055 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3057 * gst/rtsp-server/rtsp-media-factory-uri.c:
3058 * gst/rtsp-server/rtsp-media.c:
3059 rtsp: use gst_object_unref on GstObjects
3061 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3063 * gst/rtsp-server/rtsp-media-factory.c:
3064 media-factory: require an url
3066 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3068 * examples/test-uri.c:
3069 examples: fix include
3071 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3073 * gst/rtsp-server/rtsp-server.h:
3074 server: remove unused include
3076 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3078 * tests/check/Makefile.am:
3079 * tests/check/gst/mountpoints.c:
3080 tests: add test for mountpoints
3082 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3084 * gst/rtsp-server/rtsp-client.c:
3085 client: fix factory leak
3086 Keep the factory in the state object only for authorization checks and make
3087 sure we unref it on failure. Also don't keep invalid objects in the state
3090 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3092 * gst/rtsp-server/rtsp-mount-points.c:
3093 mounts: add g_return_if guards
3095 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3097 * tests/check/gst/client.c:
3098 tests: add more tests
3100 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3102 * gst/rtsp-server/rtsp-client.c:
3103 client: improve debug
3105 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3107 * gst/rtsp-server/rtsp-client.c:
3108 client: improve debug and fix leaks
3109 Cleanup the uri and session when there is a bad request.
3111 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3116 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3118 * tests/check/gst/client.c:
3119 test: add test for session in options request
3121 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3123 * gst/rtsp-server/rtsp-client.c:
3124 client: use 454 when session can't be found
3125 We should use 454 when a session can't be found because there was no session
3126 pool configured in the server. This is not a server configuration problem
3127 because the server on which the request is done might not be the same one that
3128 will keep the sessions for us and so it does not need to support sessions.
3130 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3132 * gst/rtsp-server/rtsp-client.c:
3133 client: only free connection when there is one
3134 It's possible that the client doesn't have a connection when we try to free it.
3136 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3138 * tests/check/Makefile.am:
3139 * tests/check/gst/client.c:
3140 tests: add unit test for the client object
3142 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3144 * gst/rtsp-server/rtsp-client.c:
3145 client: small cleanup
3147 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3149 * gst/rtsp-server/rtsp-client.h:
3150 client: remove unused include
3152 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3154 * gst/rtsp-server/rtsp-client.c:
3155 client: fix compilation
3157 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3159 * gst/rtsp-server/rtsp-client.c:
3160 client: call destroy without the lock
3162 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3164 * gst/rtsp-server/rtsp-client.c:
3165 * gst/rtsp-server/rtsp-client.h:
3166 client: make the client usable without a socket
3167 Make a method to let the client handle a message and a callback when the client
3168 wants us to send a response message back. This makes it possible to also use the
3169 client object without the sockets, which should make it easier to test.
3171 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3173 * gst/rtsp-server/rtsp-client.c:
3174 * gst/rtsp-server/rtsp-client.h:
3175 client: small cleanup
3177 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3179 * docs/libs/gst-rtsp-server-sections.txt:
3180 * gst/rtsp-server/rtsp-client.c:
3181 * gst/rtsp-server/rtsp-client.h:
3182 * gst/rtsp-server/rtsp-server.c:
3183 client: remove reference to server
3184 We don't need to keep a ref to the server
3186 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3188 * gst/rtsp-server/rtsp-client.c:
3189 * gst/rtsp-server/rtsp-client.h:
3191 Also add some g_return_if()
3193 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3195 * gst/rtsp-server/rtsp-client.c:
3196 client: log more errors
3198 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3200 * gst/rtsp-server/rtsp-client.c:
3201 client: fix compilation
3203 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3205 * gst/rtsp-server/rtsp-client.c:
3206 * gst/rtsp-server/rtsp-client.h:
3207 client: add generic close-after-send support
3208 Add a property to send_response() to close the connection after the response has
3209 been sent to the client.
3211 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3214 * docs/libs/gst-rtsp-server-docs.sgml:
3215 * docs/libs/gst-rtsp-server-sections.txt:
3216 * docs/libs/gst-rtsp-server.types:
3217 * examples/test-auth.c:
3218 * examples/test-launch.c:
3219 * examples/test-mp4.c:
3220 * examples/test-multicast.c:
3221 * examples/test-multicast2.c:
3222 * examples/test-ogg.c:
3223 * examples/test-readme.c:
3224 * examples/test-sdp.c:
3225 * examples/test-uri.c:
3226 * examples/test-video.c:
3227 * gst/rtsp-server/Makefile.am:
3228 * gst/rtsp-server/rtsp-auth.h:
3229 * gst/rtsp-server/rtsp-client.c:
3230 * gst/rtsp-server/rtsp-client.h:
3231 * gst/rtsp-server/rtsp-media-mapping.c:
3232 * gst/rtsp-server/rtsp-media-mapping.h:
3233 * gst/rtsp-server/rtsp-mount-points.c:
3234 * gst/rtsp-server/rtsp-mount-points.h:
3235 * gst/rtsp-server/rtsp-server.c:
3236 * gst/rtsp-server/rtsp-server.h:
3237 * gst/rtsp-server/rtsp-session-media.c:
3238 * gst/rtsp-server/rtsp-session-pool.c:
3239 * gst/rtsp-server/rtsp-session-pool.h:
3240 * tests/check/gst/rtspserver.c:
3241 MediaMapping -> MountPoints
3242 Describes better what the object manages.
3244 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3247 configure: bump required version of -base
3249 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3251 * gst/rtsp-server/rtsp-media.c:
3254 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3256 * gst/rtsp-server/rtsp-media.c:
3257 * gst/rtsp-server/rtsp-media.h:
3258 media: support more Range formats
3259 Use the new -base methods to convert the Range string into a seek start and stop
3262 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3264 * examples/test-launch.c:
3265 examples: fix whitespace
3267 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3269 * examples/test-auth.c:
3270 test-auth: add example of how to remove sessions
3271 Add an example of the session filter api.
3273 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3275 * examples/test-uri.c:
3276 test-uri: remove mapping example
3278 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3280 * examples/test-uri.c:
3281 test-uri: fix callback signature
3283 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3285 * gst/rtsp-server/rtsp-media-factory.c:
3286 factory: keep ref to factory while media active
3287 While the media from a factory is alive, keep a ref to the factory.
3288 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
3290 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3292 * gst/rtsp-server/rtsp-media-factory-uri.c:
3293 factory-uri: add some debug
3295 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3297 * gst/rtsp-server/rtsp-stream.c:
3298 stream: set udp sources to PLAYING
3299 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
3300 so that it doesn't cause our pipeline to produce ASYNC-DONE.
3302 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3304 * gst/rtsp-server/rtsp-media-factory-uri.c:
3305 factory-uri: take ref to factory
3306 Take a ref to the factory that we place in our list.
3308 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3310 * tests/Makefile.am:
3311 * tests/test-reuse.c:
3312 test: add test for server reuse
3313 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
3315 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
3317 * gst/rtsp-server/rtsp-server.c:
3318 server: start and stop multiple times
3319 Stop listening on the RTSP port when the GSource is removed, so clients
3320 can't connect and the server can be started again.
3321 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
3323 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3325 * gst/rtsp-server/rtsp-server.c:
3326 server: fix small leak
3328 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3330 * gst/rtsp-server/rtsp-media.c:
3331 media: unref source in finish_unprepare
3332 The source is created in prepare, unref it in finish_unprepare.
3333 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
3335 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
3337 * gst/rtsp-server/rtsp-client.c:
3338 * gst/rtsp-server/rtsp-media.c:
3339 rtsp-media: remove bus watch before finalizing
3340 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
3341 * An extra media ref is added for the bus watch. This extra ref is unreffed by
3342 the GDestroyNotify function.
3343 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
3344 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
3345 gst_rtsp_media_unprepare before unreffing the media.
3346 This way, the bus watch will be removed before the media is finalized.
3347 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
3349 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
3351 * gst/rtsp-server/rtsp-client.c:
3352 * gst/rtsp-server/rtsp-client.h:
3353 client: wait until the TEARDOWN response is sent to close the connection
3354 Responses can be sent async so we need to wait until the TEARDOWN response has
3355 been written before we close the connection to the client. This avoids the risk
3356 of writing/polling closed sockets.
3357 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
3359 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
3361 * gst/rtsp-server/rtsp-stream.c:
3362 rtsp-stream: plug socket leak
3363 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
3365 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
3368 Automatic update of common submodule
3369 From 6bb6951 to a72faea
3371 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
3373 * gst/rtsp-server/rtsp-media-factory-uri.c:
3374 rtsp-server: don't use deprecated API
3376 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
3378 * gst/rtsp-server/rtsp-client.c:
3379 rtsp-client: fix unused-but-set-variable compiler warning
3380 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
3382 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3385 * docs/libs/gst-rtsp-server-sections.txt:
3386 * gst/rtsp-server/rtsp-client.c:
3389 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3391 * examples/Makefile.am:
3392 * examples/test-multicast2.c:
3393 examples: add another multicast example
3394 Add an example for how to configure separate multicast ranges for each media
3397 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3399 * examples/test-multicast.c:
3402 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3404 * gst/rtsp-server/rtsp-client.c:
3405 * gst/rtsp-server/rtsp-media.c:
3406 * gst/rtsp-server/rtsp-session-media.c:
3407 * gst/rtsp-server/rtsp-session-media.h:
3408 * gst/rtsp-server/rtsp-stream-transport.c:
3409 * gst/rtsp-server/rtsp-stream-transport.h:
3410 stream: use the address managed by the stream
3411 Use the address managed by the stream for multicast. This allows us to have 1
3412 multicast address for each stream.
3413 Because the address is now managed by the stream we don't have to pass it around
3415 Set the address pool on the streams.
3417 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3419 * gst/rtsp-server/rtsp-client.c:
3420 * gst/rtsp-server/rtsp-media.c:
3421 * gst/rtsp-server/rtsp-stream.c:
3424 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3426 * gst/rtsp-server/rtsp-media.c:
3427 * gst/rtsp-server/rtsp-media.h:
3428 media: add signal for new streams
3429 This allows applications to listen for new streams and configure properties on
3430 them, like the address pool.
3432 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3434 * gst/rtsp-server/rtsp-media.c:
3435 media: configure address pool in new streams
3437 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3439 * gst/rtsp-server/rtsp-stream.c:
3440 * gst/rtsp-server/rtsp-stream.h:
3441 stream: add methods to deal with address pool
3442 Add methods to get and set the address pool for the stream
3443 Add method to allocate and get the multicast addresses for this stream.
3445 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3447 * docs/libs/gst-rtsp-server-sections.txt:
3448 * gst/rtsp-server/rtsp-media.c:
3449 * gst/rtsp-server/rtsp-media.h:
3450 media: remove MTU property
3451 It is a stream property
3453 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3455 * gst/rtsp-server/rtsp-client.c:
3456 client: set blocksize only on stream
3457 Set the blocksize only on the current stream.
3459 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3461 * gst/rtsp-server/rtsp-stream.c:
3462 stream: share src and sink sockets
3463 the allocated socket is in the used-socket property, not socket.
3465 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3467 * gst/rtsp-server/rtsp-address-pool.c:
3468 * gst/rtsp-server/rtsp-address-pool.h:
3469 * gst/rtsp-server/rtsp-client.c:
3470 * gst/rtsp-server/rtsp-session-media.c:
3471 * gst/rtsp-server/rtsp-session-media.h:
3472 * gst/rtsp-server/rtsp-stream-transport.c:
3473 * gst/rtsp-server/rtsp-stream-transport.h:
3474 * tests/check/gst/addresspool.c:
3475 rtsp: make address-pool return an address object
3476 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
3477 store more info in the structure and allows us to more easily return the address
3478 to the right pool when no longer needed.
3479 Pass the address to the StreamTransport so that we can return it to the pool
3480 when the stream transport is freed or changed.
3482 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3484 * examples/Makefile.am:
3485 * examples/test-multicast.c:
3486 examples: add multicast example
3487 Show how to set up the multicast address pool so that media can be
3488 server with multicast.
3490 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3492 * gst/rtsp-server/rtsp-client.c:
3493 * gst/rtsp-server/rtsp-media-factory.c:
3494 * gst/rtsp-server/rtsp-media-factory.h:
3495 * gst/rtsp-server/rtsp-media.c:
3496 * gst/rtsp-server/rtsp-media.h:
3497 rtsp: use AddressPool
3498 Remove the multicast_group property.
3499 Use the configured addresspool to allocate multicast addresses.
3501 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3503 * gst/rtsp-server/rtsp-address-pool.c:
3504 * gst/rtsp-server/rtsp-address-pool.h:
3505 address-pool: add clear method
3507 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3509 * gst/rtsp-server/rtsp-address-pool.c:
3510 address-pool: small cleanups
3512 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3514 * tests/check/Makefile.am:
3515 * tests/check/gst/addresspool.c:
3516 tests: add addresspool unit test
3518 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3520 * gst/rtsp-server/Makefile.am:
3521 * gst/rtsp-server/rtsp-address-pool.c:
3522 * gst/rtsp-server/rtsp-address-pool.h:
3523 address-pool: add object to manage multicast addresses
3524 Make an object that can manage a rage of multicast addresses and ports.
3526 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3528 * gst/rtsp-server/rtsp-server.c:
3529 server: set default max-threads property
3531 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3533 * gst/rtsp-server/rtsp-media.c:
3534 media: wait for concurrent _prepare
3535 If a prepare is busy, wait for the result.
3537 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3539 * gst/rtsp-server/rtsp-media.c:
3540 media: add lock around message handler
3541 We don't want to dispatch messages while we are still processing the result of
3544 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3546 * gst/rtsp-server/rtsp-media.c:
3547 * gst/rtsp-server/rtsp-media.h:
3548 media: add lock to protect state changes
3550 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3552 * gst/rtsp-server/rtsp-stream.c:
3553 * gst/rtsp-server/rtsp-stream.h:
3556 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3558 * gst/rtsp-server/rtsp-stream-transport.c:
3559 * gst/rtsp-server/rtsp-stream-transport.h:
3560 * gst/rtsp-server/rtsp-stream.c:
3561 stream-transport: add keep-alive method
3563 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3565 * gst/rtsp-server/rtsp-stream-transport.c:
3566 * gst/rtsp-server/rtsp-stream-transport.h:
3567 * gst/rtsp-server/rtsp-stream.c:
3568 stream-transport: add method to handle RTP/RTCP
3569 Call new methods instead of poking into the structures directly.
3571 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3573 * gst/rtsp-server/rtsp-session-media.c:
3574 * gst/rtsp-server/rtsp-session-media.h:
3575 session-media: add locking
3577 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3579 * gst/rtsp-server/rtsp-session.c:
3580 * gst/rtsp-server/rtsp-session.h:
3581 session: add locking
3583 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3585 * gst/rtsp-server/rtsp-server.c:
3586 server: free old socket
3588 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3590 * gst/rtsp-server/rtsp-media-mapping.c:
3591 * gst/rtsp-server/rtsp-media-mapping.h:
3592 mapping: add locking
3594 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3596 * gst/rtsp-server/rtsp-media-factory.c:
3597 media-factory: add locking
3599 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3601 * gst/rtsp-server/rtsp-auth.c:
3602 * gst/rtsp-server/rtsp-auth.h:
3605 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3607 * gst/rtsp-server/rtsp-server.c:
3608 * gst/rtsp-server/rtsp-server.h:
3609 server: add max-thread property
3611 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3613 * gst/rtsp-server/rtsp-server.c:
3614 * gst/rtsp-server/rtsp-server.h:
3615 server: use a threadpool for the mainloops
3617 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3619 * gst/rtsp-server/rtsp-client.c:
3620 * gst/rtsp-server/rtsp-client.h:
3621 client: rename method
3622 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
3623 don't really create the client from the socket, we use the socket for the
3626 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3628 * gst/rtsp-server/rtsp-client.c:
3629 * gst/rtsp-server/rtsp-client.h:
3630 * gst/rtsp-server/rtsp-server.c:
3631 server: rework maincontext handling in clients
3632 Make a separate method to attach a client to a MainContext.
3633 Let the server decide in what GMainContext the client will operate and give this
3634 context to the client in attach. Then the server can later decide to use a
3635 separate thread for each client or just use the mainthread.
3637 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3639 * gst/rtsp-server/rtsp-client.c:
3640 * gst/rtsp-server/rtsp-session.c:
3641 * gst/rtsp-server/rtsp-session.h:
3642 session: move session header code in session object
3644 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
3648 * examples/test-auth.c:
3649 * examples/test-launch.c:
3650 * examples/test-mp4.c:
3651 * examples/test-ogg.c:
3652 * examples/test-readme.c:
3653 * examples/test-sdp.c:
3654 * examples/test-uri.c:
3655 * examples/test-video.c:
3656 * gst/rtsp-server/rtsp-auth.c:
3657 * gst/rtsp-server/rtsp-auth.h:
3658 * gst/rtsp-server/rtsp-client.c:
3659 * gst/rtsp-server/rtsp-client.h:
3660 * gst/rtsp-server/rtsp-media-factory-uri.c:
3661 * gst/rtsp-server/rtsp-media-factory-uri.h:
3662 * gst/rtsp-server/rtsp-media-factory.c:
3663 * gst/rtsp-server/rtsp-media-factory.h:
3664 * gst/rtsp-server/rtsp-media-mapping.c:
3665 * gst/rtsp-server/rtsp-media-mapping.h:
3666 * gst/rtsp-server/rtsp-media.c:
3667 * gst/rtsp-server/rtsp-media.h:
3668 * gst/rtsp-server/rtsp-params.c:
3669 * gst/rtsp-server/rtsp-params.h:
3670 * gst/rtsp-server/rtsp-sdp.c:
3671 * gst/rtsp-server/rtsp-sdp.h:
3672 * gst/rtsp-server/rtsp-server.c:
3673 * gst/rtsp-server/rtsp-server.h:
3674 * gst/rtsp-server/rtsp-session-media.c:
3675 * gst/rtsp-server/rtsp-session-media.h:
3676 * gst/rtsp-server/rtsp-session-pool.c:
3677 * gst/rtsp-server/rtsp-session-pool.h:
3678 * gst/rtsp-server/rtsp-session.c:
3679 * gst/rtsp-server/rtsp-session.h:
3680 * gst/rtsp-server/rtsp-stream-transport.c:
3681 * gst/rtsp-server/rtsp-stream-transport.h:
3682 * gst/rtsp-server/rtsp-stream.c:
3683 * gst/rtsp-server/rtsp-stream.h:
3684 * tests/check/gst/rtspserver.c:
3685 * tests/test-cleanup.c:
3688 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
3690 * gst/rtsp-server/rtsp-media.c:
3691 * gst/rtsp-server/rtsp-session-media.c:
3692 * gst/rtsp-server/rtsp-session.c:
3693 rtsp-server: added annotations to indicate type of ownership transfer of return values
3694 https://bugzilla.gnome.org/show_bug.cgi?id=680777
3696 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
3699 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
3701 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
3704 * bindings/Makefile.am:
3705 * bindings/vala/Makefile.am:
3706 * bindings/vala/gst-rtsp-server-0.10.deps:
3707 * bindings/vala/gst-rtsp-server-0.10.vapi:
3708 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
3709 * bindings/vala/packages/gst-rtsp-server-0.10.files:
3710 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
3711 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
3712 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
3714 bindings: remove vala bindings
3715 They'll be reunited with the other GStreamer bindings
3716 https://bugzilla.gnome.org/show_bug.cgi?id=680777
3718 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3720 * gst/rtsp-server/rtsp-client.c:
3721 * gst/rtsp-server/rtsp-session-media.c:
3722 * gst/rtsp-server/rtsp-session-media.h:
3723 * gst/rtsp-server/rtsp-stream-transport.c:
3724 * gst/rtsp-server/rtsp-stream-transport.h:
3725 rtsp: only create transport when needed
3726 Only create the StreamTransport when configured.
3728 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3730 * gst/rtsp-server/rtsp-client.c:
3731 client: small cleanup
3733 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3735 * gst/rtsp-server/rtsp-client.c:
3736 * gst/rtsp-server/rtsp-client.h:
3737 * gst/rtsp-server/rtsp-stream-transport.c:
3738 * gst/rtsp-server/rtsp-stream-transport.h:
3739 rtsp: refactor configuration of transport
3740 Move the configuration of the transport to a place where it makes
3743 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3745 * gst/rtsp-server/rtsp-client.c:
3746 client: refactor transport parsing
3748 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3750 * gst/rtsp-server/rtsp-client.c:
3751 client: refuse to change the MTU on shared media
3752 If we change the MTU of chared media, it changes for all clients.
3753 We don't want to set the MTU to something large for clients that
3756 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3758 * examples/test-mp4.c:
3759 * gst/rtsp-server/rtsp-media.c:
3760 small fixes to docs and debug
3762 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3764 * gst/rtsp-server/rtsp-stream.c:
3765 stream: transports must already have been removed
3767 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3769 * gst/rtsp-server/rtsp-media.c:
3770 * gst/rtsp-server/rtsp-stream.c:
3771 * gst/rtsp-server/rtsp-stream.h:
3772 stream: improve join and leave of the pipeline
3774 Do the cleanup properly
3777 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3779 * gst/rtsp-server/rtsp-media.c:
3780 media: move unprepare below default implementation
3781 Makes it easier to find the default implementation
3783 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3785 * gst/rtsp-server/rtsp-media.c:
3786 media: signal unprepared when we actually finish
3788 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3790 * gst/rtsp-server/rtsp-media.c:
3791 media: no need to unlock, unprepare does that when needed
3793 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3795 * docs/libs/gst-rtsp-server-sections.txt:
3796 * gst/rtsp-server/rtsp-media-factory.h:
3797 * gst/rtsp-server/rtsp-media-mapping.c:
3798 * gst/rtsp-server/rtsp-media.h:
3799 * gst/rtsp-server/rtsp-params.c:
3800 * gst/rtsp-server/rtsp-server.c:
3801 * gst/rtsp-server/rtsp-session-pool.h:
3802 * gst/rtsp-server/rtsp-session.c:
3803 * gst/rtsp-server/rtsp-session.h:
3804 * gst/rtsp-server/rtsp-stream-transport.h:
3805 * gst/rtsp-server/rtsp-stream.h:
3808 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3810 * gst/rtsp-server/rtsp-client.c:
3811 * gst/rtsp-server/rtsp-media-mapping.h:
3812 * gst/rtsp-server/rtsp-media.c:
3813 * gst/rtsp-server/rtsp-media.h:
3814 * gst/rtsp-server/rtsp-server.h:
3815 * gst/rtsp-server/rtsp-stream.c:
3816 * gst/rtsp-server/rtsp-stream.h:
3817 rtsp: fix MTU setting
3818 Fix setting of the MTU. There is no need for a vmethod.
3820 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3825 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3828 configure: bump version number after refactoring
3830 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3832 * gst/rtsp-server/Makefile.am:
3833 * gst/rtsp-server/rtsp-client.c:
3834 * gst/rtsp-server/rtsp-client.h:
3835 * gst/rtsp-server/rtsp-media-factory-uri.c:
3836 * gst/rtsp-server/rtsp-media-factory.c:
3837 * gst/rtsp-server/rtsp-media-factory.h:
3838 * gst/rtsp-server/rtsp-media.c:
3839 * gst/rtsp-server/rtsp-media.h:
3840 * gst/rtsp-server/rtsp-sdp.c:
3841 * gst/rtsp-server/rtsp-session-media.c:
3842 * gst/rtsp-server/rtsp-session-media.h:
3843 * gst/rtsp-server/rtsp-session.c:
3844 * gst/rtsp-server/rtsp-session.h:
3845 * gst/rtsp-server/rtsp-stream-transport.c:
3846 * gst/rtsp-server/rtsp-stream-transport.h:
3847 * gst/rtsp-server/rtsp-stream.c:
3848 * gst/rtsp-server/rtsp-stream.h:
3849 rtsp: massive refactoring
3850 Make GObjects from the remaining simple structures.
3851 Remove GstRTSPSessionStream, it's not needed.
3852 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
3853 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
3854 a GstRTSPStream should be transported to a client.
3855 Rename GstRTSPMediaFactory::get_element -> create_element because that
3856 more accurately describes what it does.
3857 Make nice methods instead of poking in the structures.
3858 Move some methods inside the relevant object source code.
3859 Use GPtrArray to store objects instead of plain arrays, it is more
3860 natural and allows us to more easily clean up.
3861 Move the allocation of udp ports to the Stream object. The Stream object
3862 contains the elements needed to stream the media to a client.
3863 Improve the prepare and unprepare methods. Unprepare should now undo
3864 everything prepare did. Improve also async unprepare when doing EOS on
3865 shutdown. Make sure we always unprepare correctly.
3867 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
3869 * gst/rtsp-server/rtsp-client.c:
3870 rtsp-client: Unref server address clients connected to
3871 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
3873 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
3875 * gst/rtsp-server/rtsp-server.c:
3876 rtsp-server: don't ref server socket if it is NULL
3877 Fixes test_bind_already_in_use unit test again after commit 6a497440.
3878 https://bugzilla.gnome.org/show_bug.cgi?id=686644
3880 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
3882 * tests/check/Makefile.am:
3883 tests: Add libgio link dependency
3884 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
3886 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
3888 * gst/rtsp-server/rtsp-media-mapping.c:
3889 * gst/rtsp-server/rtsp-media-mapping.h:
3890 rtsp-media-mapping: rename find_media vfunc to find_factory
3891 The virtual method and class method should have the same name
3892 so it is correctly represented in GIR file
3893 https://bugzilla.gnome.org/show_bug.cgi?id=680777
3895 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
3897 * gst/rtsp-server/rtsp-auth.c:
3898 * gst/rtsp-server/rtsp-client.c:
3899 * gst/rtsp-server/rtsp-media-factory-uri.c:
3900 * gst/rtsp-server/rtsp-media-factory.c:
3901 * gst/rtsp-server/rtsp-media-mapping.c:
3902 * gst/rtsp-server/rtsp-media.c:
3903 * gst/rtsp-server/rtsp-server.c:
3904 * gst/rtsp-server/rtsp-session-pool.c:
3905 * gst/rtsp-server/rtsp-session.c:
3906 rtsp-server: fixed comments and GIR annotations
3907 https://bugzilla.gnome.org/show_bug.cgi?id=680777
3909 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
3911 * gst/rtsp-server/rtsp-media-mapping.c:
3912 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
3914 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
3916 * gst/rtsp-server/rtsp-server.c:
3917 rtsp-server: allow binding on port 0 (binds on a random port)
3919 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
3921 * gst/rtsp-server/rtsp-server.c:
3922 * gst/rtsp-server/rtsp-server.h:
3923 rtsp-server: add bound-port property
3924 bound-port can be used to retrieve the port number when the server is bound on
3925 port 0, which binds on a random port.
3927 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
3929 * gst/rtsp-server/rtsp-media-factory.c:
3930 * gst/rtsp-server/rtsp-media-factory.h:
3931 rtsp-media-factory: make ::get_element overridable by GI bindings
3932 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
3933 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
3934 as the invoker for ::get_element(), making it overridable by GI generated
3937 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
3939 * gst/rtsp-server/rtsp-media-factory-uri.c:
3940 rtsp-media-factory-uri: don't autoplug parsers in a loop
3941 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
3944 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
3946 * gst/rtsp-server/Makefile.am:
3947 Explicitly link against gio. Fix link error on mac.
3949 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
3951 * gst/rtsp-server/rtsp-session.c:
3952 session: add ttl to the transport header in SETUP
3953 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
3955 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
3957 * gst/rtsp-server/rtsp-client.c:
3958 * gst/rtsp-server/rtsp-client.h:
3959 * gst/rtsp-server/rtsp-media.c:
3960 client: Use client transport settings for multicast if allowed.
3961 This patch makes it possible for the client to send transport settings for
3962 multicast (destination && ttl). Client settings must be explicitly allowed or
3963 the server will use its own settings.
3964 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
3966 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
3969 Automatic update of common submodule
3970 From 6c0b52c to 6bb6951
3972 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
3974 * gst/rtsp-server/rtsp-client.c:
3975 rtsp-client: do not destroy the rtsp watch
3976 Don't destroy the client watch while dispatching. The rtsp watch is
3977 automatically destroyed after the rtsp watch function closed() has
3979 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
3981 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
3984 Automatic update of common submodule
3985 From 4f962f7 to 6c0b52c
3987 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
3989 * gst/rtsp-server/rtsp-media.c:
3990 media: fix check for seekability
3992 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3994 * gst/rtsp-server/rtsp-client.c:
3995 client: use more GIO
3996 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
3998 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4000 * gst/rtsp-server/rtsp-server.c:
4001 server: remove obsolete includes
4003 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4005 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
4006 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
4007 be available in "on_new_ssrc". The transports are added in
4008 gst_rtsp_media_set_state when going to PLAYING state. However,
4009 "on_new_ssrc" might be called before this happens.
4010 https://bugzilla.gnome.org/show_bug.cgi?id=683304
4012 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4014 * gst/rtsp-server/rtsp-client.c:
4015 * gst/rtsp-server/rtsp-client.h:
4016 rtsp-client: add signals for rtsp requests (fixes #683287)
4018 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4020 * gst/rtsp-server/rtsp-client.c:
4021 * gst/rtsp-server/rtsp-client.h:
4022 add new-session signal to rtsp-client (fixes #683058)
4024 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
4027 Automatic update of common submodule
4028 From 668acee to 4f962f7
4030 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
4032 * gst/rtsp-server/rtsp-server.c:
4033 * tests/check/gst/rtspserver.c:
4034 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
4035 Do not assume that *error is set in g_socket_address_enumerator_next.
4036 Added test_bind_already_in_use unit-test.
4037 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
4039 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
4042 Automatic update of common submodule
4043 From 94ccf4c to 668acee
4045 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
4047 * gst/rtsp-server/rtsp-client.c:
4048 * gst/rtsp-server/rtsp-client.h:
4049 rtsp-client: make create_sdp virtual method
4050 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
4052 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4055 Automatic update of common submodule
4056 From 98e386f to 94ccf4c
4058 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4060 * gst/rtsp-server/rtsp-client.c:
4063 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
4065 * gst/rtsp-server/rtsp-client.c:
4066 * gst/rtsp-server/rtsp-client.h:
4067 * gst/rtsp-server/rtsp-server.c:
4068 * gst/rtsp-server/rtsp-server.h:
4069 rtsp-server: use an existing socket to establish HTTP tunnel
4070 Make it possible to transfer a socket from an HTTP server to be used as
4071 an RTSP over HTTP tunnel.
4073 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
4075 * gst/rtsp-server/rtsp-client.c:
4076 * gst/rtsp-server/rtsp-media.c:
4077 * gst/rtsp-server/rtsp-media.h:
4078 rtsp: Handle the blocksize parameter
4079 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
4081 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
4083 * tests/check/Makefile.am:
4084 * tests/check/gst/rtspserver.c:
4085 Have unit test get header from source dir, not installed dir
4086 This makes compilation of unit tests work in a build directory other
4087 than the source directory.
4088 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
4090 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
4092 * gst/rtsp-server/rtsp-media.c:
4093 rtsp-media: update for gst_element_make_from_uri() changes
4095 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
4098 * tests/Makefile.am:
4099 * tests/check/Makefile.am:
4100 * tests/check/gst/rtspserver.c:
4102 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
4104 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
4106 * gst/rtsp-server/rtsp-media.c:
4107 rtsp-media: don't collect media stats when going to NULL
4108 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
4110 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4112 * gst/rtsp-server/rtsp-client.c:
4113 client: don't leak transports
4115 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
4117 * gst/rtsp-server/rtsp-client.c:
4118 rtsp-client: free transport on no_stream in SETUP handler
4120 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
4122 * gst/rtsp-server/rtsp-client.c:
4123 rtsp-client: changed session media iteration
4124 In client_unlink_session: now don't iterate in session->medias
4125 list where items are removed by gst_rtsp_session_release_media.
4126 Instead, repeatedly remove the first item.
4128 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
4130 * gst/rtsp-server/rtsp-client.c:
4131 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
4132 GstRTSPSessionMedia is not a GObject type. When the
4133 GstRTSPSession is freed, it will free the media.
4135 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
4137 * gst/rtsp-server/rtsp-media-factory.c:
4138 factory: plug pad leak in collect_streams
4139 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
4140 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
4141 will take one reference, and the other reference will otherwise
4144 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
4147 configure: suppress some warnings when debug is disabled
4148 Warnings about unused variables should be suppressed if core has the
4149 debug system disabled.
4150 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
4152 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4154 * docs/libs/Makefile.am:
4155 docs: fix build in uninstalled setup
4156 Include gst-plugins-base libs properly.
4158 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
4160 * docs/libs/gst-rtsp-server.types:
4161 docs: include headers defining rtsp-server object types
4162 Fixes compiler warnings during docs build.
4163 https://bugzilla.gnome.org/show_bug.cgi?id=676824
4165 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
4168 configure: Add warning flags for compiler when configuring
4169 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
4171 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4174 Automatic update of common submodule
4175 From 03a0e57 to 98e386f
4177 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4180 Automatic update of common submodule
4181 From 1fab359 to 03a0e57
4183 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
4185 * gst/rtsp-server/rtsp-client.c:
4186 client: fix GSocketAddress leak in gst_rtsp_client_accept
4187 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
4189 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4192 Automatic update of common submodule
4193 From f1b5a96 to 1fab359
4195 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4198 Automatic update of common submodule
4199 From 92b7266 to f1b5a96
4201 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4204 Automatic update of common submodule
4205 From ec1c4a8 to 92b7266
4207 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4210 Automatic update of common submodule
4211 From 3429ba6 to ec1c4a8
4213 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
4215 * gst/rtsp-server/rtsp-auth.c:
4216 * gst/rtsp-server/rtsp-client.c:
4217 * gst/rtsp-server/rtsp-media-factory-uri.c:
4218 * gst/rtsp-server/rtsp-server.c:
4219 rtsp: fix compiler warnings
4220 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
4222 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4225 Automatic update of common submodule
4226 From dc70203 to 3429ba6
4228 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4230 * gst/rtsp-server/rtsp-client.c:
4231 * gst/rtsp-server/rtsp-media-factory.c:
4232 * gst/rtsp-server/rtsp-media-factory.h:
4233 * gst/rtsp-server/rtsp-media.c:
4234 * gst/rtsp-server/rtsp-media.h:
4235 * gst/rtsp-server/rtsp-server.c:
4236 * gst/rtsp-server/rtsp-server.h:
4237 * gst/rtsp-server/rtsp-session-pool.c:
4238 * gst/rtsp-server/rtsp-session-pool.h:
4239 rtsp-server: port to new thread API
4241 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4244 Automatic update of common submodule
4245 From 6db25be to dc70203
4247 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4249 * gst/rtsp-server/rtsp-auth.c:
4250 * gst/rtsp-server/rtsp-auth.h:
4251 * gst/rtsp-server/rtsp-client.c:
4252 rtsp-server: Fix compilation and compiler warnings
4254 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4258 * gst/rtsp-server/Makefile.am:
4259 configure: Modernize autotools setup a bit
4260 Also we now only create tar.bz2 and tar.xz tarballs.
4262 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4265 Automatic update of common submodule
4266 From 464fe15 to 6db25be
4268 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4271 Automatic update of common submodule
4272 From 7fda524 to 464fe15
4274 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4277 * docs/libs/Makefile.am:
4278 * docs/version.entities.in:
4280 * gst/rtsp-server/Makefile.am:
4281 * pkgconfig/Makefile.am:
4282 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4283 * pkgconfig/gstreamer-rtsp-server.pc.in:
4284 * tests/Makefile.am:
4285 rtsp-server: Update versioning
4287 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4289 Merge remote-tracking branch 'origin/0.10'
4291 gst/rtsp-server/rtsp-session-pool.c
4293 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4295 * gst/rtsp-server/rtsp-session-pool.c:
4296 rtsp-server: Don't use deprecated GLib API
4298 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4300 Replace master with 0.11
4302 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4304 Merge branch 'master' into 0.11
4306 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4308 Merge branch 'master' into 0.11
4310 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
4313 A couple minor typo fixes
4315 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4317 * gst/rtsp-server/rtsp-media.c:
4318 media: fix state of the appqueue
4320 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4322 * gst/rtsp-server/rtsp-media-factory-uri.c:
4323 factory: use videoconvert
4325 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4327 * gst/rtsp-server/rtsp-media-factory-uri.c:
4328 factory: change to new style caps
4330 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4332 * gst/rtsp-server/rtsp-client.c:
4333 * gst/rtsp-server/rtsp-client.h:
4334 * gst/rtsp-server/rtsp-media-factory-uri.c:
4335 * gst/rtsp-server/rtsp-media.c:
4336 * gst/rtsp-server/rtsp-server.c:
4337 * gst/rtsp-server/rtsp-server.h:
4338 * gst/rtsp-server/rtsp-session-pool.c:
4339 rtsp-server: port to GIO
4342 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4345 configure: fix build
4347 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4350 docs: fix for gst_rtsp_server_set_port() -> _set_service()
4351 https://bugzilla.gnome.org/show_bug.cgi?id=666548
4353 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4356 * examples/Makefile.am:
4357 First rule of gst-rtsp-server club: don't talk about gst-phonon
4359 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4362 * pkgconfig/Makefile.am:
4363 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
4364 * pkgconfig/gst-rtsp-server.pc.in:
4365 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4366 * pkgconfig/gstreamer-rtsp-server.pc.in:
4367 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
4368 For consistency with all other modules.
4370 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4372 * gst/rtsp-server/rtsp-client.c:
4373 rtsp-client: update for new map API
4375 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4378 * bindings/Makefile.am:
4379 * bindings/python/Makefile.am:
4380 * bindings/python/arg-types.py:
4381 * bindings/python/codegen/Makefile.am:
4382 * bindings/python/codegen/__init__.py:
4383 * bindings/python/codegen/argtypes.py:
4384 * bindings/python/codegen/code-coverage.py:
4385 * bindings/python/codegen/codegen.py:
4386 * bindings/python/codegen/definitions.py:
4387 * bindings/python/codegen/defsparser.py:
4388 * bindings/python/codegen/docextract.py:
4389 * bindings/python/codegen/docgen.py:
4390 * bindings/python/codegen/fileprefix.override:
4391 * bindings/python/codegen/fileprefixmodule.c:
4392 * bindings/python/codegen/h2def.py:
4393 * bindings/python/codegen/mergedefs.py:
4394 * bindings/python/codegen/mkskel.py:
4395 * bindings/python/codegen/override.py:
4396 * bindings/python/codegen/reversewrapper.py:
4397 * bindings/python/codegen/scmexpr.py:
4398 * bindings/python/rtspserver-types.defs:
4399 * bindings/python/rtspserver.defs:
4400 * bindings/python/rtspserver.override:
4401 * bindings/python/rtspservermodule.c:
4402 * bindings/python/test.py:
4404 python: remove pygst-based python bindings
4405 pygi is the future, apparently.
4407 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
4410 Automatic update of common submodule
4411 From c463bc0 to 7fda524
4413 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4416 Automatic update of common submodule
4417 From 2a59016 to c463bc0
4419 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4422 Automatic update of common submodule
4423 From 0807187 to 2a59016
4425 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4428 Automatic update of common submodule
4429 From 11f0cd5 to 0807187
4431 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4433 * examples/test-auth.c:
4434 example: update for new caps
4436 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4438 * examples/test-video.c:
4439 * gst/rtsp-server/rtsp-client.c:
4440 * gst/rtsp-server/rtsp-media-factory-uri.c:
4441 * gst/rtsp-server/rtsp-media.c:
4442 * gst/rtsp-server/rtsp-media.h:
4443 * gst/rtsp-server/rtsp-session.c:
4444 * gst/rtsp-server/rtsp-session.h:
4445 rtsp-server: port some more to 0.11
4447 Remove bufferlist stuff
4449 Add queue before appsink now that preroll-queue-len is gone.
4450 Update for request pad changes.
4452 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4454 Merge branch 'master' into 0.11
4456 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
4458 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
4459 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
4460 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
4462 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
4464 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
4465 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
4466 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
4468 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4470 Merge branch 'master' into 0.11
4472 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4474 * gst/rtsp-server/rtsp-media.c:
4475 * gst/rtsp-server/rtsp-media.h:
4476 media: add a seekable boolean
4477 Maintain the seekable state with a new variable instead of reusing the
4480 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
4482 * gst/rtsp-server/rtsp-media.c:
4483 Disallow seek in live media
4485 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4487 Merge branch 'master' into 0.11
4489 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
4491 * gst/rtsp-server/rtsp-server.c:
4492 #ifdef statements for windows socket creation were missing
4494 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
4497 Automatic update of common submodule
4498 From a39eb83 to 11f0cd5
4500 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
4503 Automatic update of common submodule
4504 From 605cd9a to a39eb83
4506 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4508 Merge branch 'master' into 0.11
4510 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4512 * gst/rtsp-server/rtsp-client.c:
4513 client: use method to access property
4515 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4517 * gst/rtsp-server/rtsp-media-factory.c:
4518 * gst/rtsp-server/rtsp-media-factory.h:
4519 media-factory: add protocols property
4520 Add a property to configure the allowed protocols in the media created from the
4523 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4525 * gst/rtsp-server/rtsp-media-factory.c:
4526 * gst/rtsp-server/rtsp-media-factory.h:
4527 media-factory: add media-configure signal
4528 Add signal to allow the application to configure the media after it was created
4531 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4533 * gst/rtsp-server/rtsp-client.c:
4534 client: use method to access property
4536 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4538 * gst/rtsp-server/rtsp-media-factory.c:
4539 * gst/rtsp-server/rtsp-media-factory.h:
4540 media-factory: add protocols property
4541 Add a property to configure the allowed protocols in the media created from the
4544 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4546 * gst/rtsp-server/rtsp-media-factory.c:
4547 * gst/rtsp-server/rtsp-media-factory.h:
4548 media-factory: add media-configure signal
4549 Add signal to allow the application to configure the media after it was created
4552 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4554 Merge branch 'master' into 0.11
4556 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4558 * gst/rtsp-server/rtsp-client.c:
4559 client: use media multicast group
4561 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4563 * gst/rtsp-server/rtsp-media-factory.h:
4564 * gst/rtsp-server/rtsp-server.h:
4565 * gst/rtsp-server/rtsp-session-pool.h:
4566 * gst/rtsp-server/rtsp-session.h:
4569 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
4571 * gst/rtsp-server/rtsp-client.c:
4572 * gst/rtsp-server/rtsp-sdp.h:
4573 sdp: copy and free the server ip address
4574 Copy and free the server ip address to make memory management easier later.
4576 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4578 * gst/rtsp-server/rtsp-media-factory.c:
4579 media-factory: configure multicast in media
4581 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4583 * gst/rtsp-server/rtsp-media.c:
4584 * gst/rtsp-server/rtsp-media.h:
4585 media: add property for multicast group
4586 Add a property to configure the multicast group in the media.
4587 Based on patches from Marc Leeman and Robert Krakora.
4589 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4591 * gst/rtsp-server/rtsp-media-factory.c:
4592 * gst/rtsp-server/rtsp-media-factory.h:
4593 media-factory: add property for multicast group
4594 Add a property to configure the multicast group in the media factory.
4595 Based on patches from Marc Leeman and Robert Krakora.
4597 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4599 * gst/rtsp-server/rtsp-client.c:
4600 client: do configuration of transport in one place
4601 Move the configuration of the transport destination address to where we also
4602 configure the other bits.
4604 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4606 * gst/rtsp-server/rtsp-client.c:
4607 client: use media multicast group
4609 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4611 * gst/rtsp-server/rtsp-media-factory.h:
4612 * gst/rtsp-server/rtsp-server.h:
4613 * gst/rtsp-server/rtsp-session-pool.h:
4614 * gst/rtsp-server/rtsp-session.h:
4617 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
4619 * gst/rtsp-server/rtsp-client.c:
4620 * gst/rtsp-server/rtsp-sdp.h:
4621 sdp: copy and free the server ip address
4622 Copy and free the server ip address to make memory management easier later.
4624 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4626 * gst/rtsp-server/rtsp-media-factory.c:
4627 media-factory: configure multicast in media
4629 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4631 * gst/rtsp-server/rtsp-media.c:
4632 * gst/rtsp-server/rtsp-media.h:
4633 media: add property for multicast group
4634 Add a property to configure the multicast group in the media.
4635 Based on patches from Marc Leeman and Robert Krakora.
4637 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4639 * gst/rtsp-server/rtsp-media-factory.c:
4640 * gst/rtsp-server/rtsp-media-factory.h:
4641 media-factory: add property for multicast group
4642 Add a property to configure the multicast group in the media factory.
4643 Based on patches from Marc Leeman and Robert Krakora.
4645 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4647 * gst/rtsp-server/rtsp-client.c:
4648 client: do configuration of transport in one place
4649 Move the configuration of the transport destination address to where we also
4650 configure the other bits.
4652 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4654 Merge branch 'master' into 0.11
4656 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
4658 * gst/rtsp-server/rtsp-client.c:
4659 client: destroy pipeline on client disconnect with no prior TEARDOWN.
4660 The problem occurs when the client abruptly closes the connection without
4661 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
4662 server is where the pipeline gets torn down. Since this handler is not called,
4663 the pipeline remains and is up and running. Subsequent clients get their own
4664 pipelines and if the do not issue TEARDOWNs then those pipelines will also
4665 remain up and running. This is a resource leak.
4667 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4669 Merge branch 'master' into 0.11
4671 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
4673 * gst/rtsp-server/rtsp-media-factory.c:
4674 * gst/rtsp-server/rtsp-media-factory.h:
4675 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
4676 For example, it can be used to retrieve source elements like appsrc, in a more
4677 convenient way than subclassing get_element.
4679 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4681 Merge branch 'master' into 0.11
4683 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
4685 * gst/rtsp-server/rtsp-server.c:
4686 rtsp-server: hold on to reference while using object
4688 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4690 * gst/rtsp-server/rtsp-media.c:
4693 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4696 configure: use unstable api
4698 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
4700 * gst/rtsp-server/rtsp-client.c:
4701 client: fix reference counting
4703 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
4705 * gst/rtsp-server/rtsp-client.c:
4706 * gst/rtsp-server/rtsp-media.c:
4707 fix compiler warnings about unused variables
4709 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
4711 * examples/test-launch.c:
4712 * examples/test-readme.c:
4713 * examples/test-uri.c:
4714 * examples/test-video.c:
4715 examples: tell rtsp uri when ready
4717 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
4720 Automatic update of common submodule
4721 From 69b981f to 605cd9a
4723 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4725 * gst/rtsp-server/rtsp-client.c:
4726 client: update for buffer API change
4728 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4730 * gst/rtsp-server/Makefile.am:
4731 Makefile.am: 0.10 => @GST_MAJORMINOR@
4733 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4735 * gst/rtsp-server/rtsp-media-factory-uri.c:
4736 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
4738 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4740 * gst/rtsp-server/.gitignore:
4741 .gitignore: 0.10 => 0.11
4743 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4745 * gst/rtsp-server/Makefile.am:
4746 Makefile.am: 0.10 => @GST_MAJORMINOR@
4748 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4750 Merge branch 'master' into 0.11
4752 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
4755 Automatic update of common submodule
4756 From 9e5bbd5 to 69b981f
4758 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
4761 Automatic update of common submodule
4762 From fd35073 to 9e5bbd5
4764 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
4767 Automatic update of common submodule
4768 From 46dfcea to fd35073
4770 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4772 * gst/rtsp-server/rtsp-media-factory-uri.c:
4773 * gst/rtsp-server/rtsp-media.c:
4774 media: port to new caps API
4776 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4778 Merge branch 'master' into 0.11
4780 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
4782 * bindings/vala/gst-rtsp-server-0.10.vapi:
4783 Updated Vala bindings.
4784 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
4786 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
4788 * gst/rtsp-server/rtsp-server.c:
4789 * gst/rtsp-server/rtsp-server.h:
4790 Add a signal for newly connected clients.
4791 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
4793 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
4795 * bindings/python/rtspserver.override:
4796 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
4798 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4800 * gst/rtsp-server/Makefile.am:
4801 * gst/rtsp-server/rtsp-client.c:
4802 * gst/rtsp-server/rtsp-funnel.c:
4803 * gst/rtsp-server/rtsp-funnel.h:
4804 * gst/rtsp-server/rtsp-media.c:
4805 rtsp-server: port to 0.11
4807 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4812 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4814 Merge branch 'master' into 0.11
4819 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4822 Automatic update of common submodule
4823 From c3cafe1 to 46dfcea
4825 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
4827 * bindings/python/Makefile.am:
4828 * bindings/python/rtspserver.defs:
4829 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
4831 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
4833 * bindings/python/arg-types.py:
4834 python bindings: add GstRTSPUrlParam
4835 Needed to implement MediaFactory virtual proxies
4837 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
4839 * bindings/python/arg-types.py:
4840 python bindings: fix returning GstRTSPUrl types
4842 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
4844 * bindings/python/arg-types.py:
4845 python bindings: add arg type for GstRTSPUrl
4847 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
4849 * bindings/python/rtspserver.defs:
4850 python bindings: fix the definition of MediaFactory.collect_stream
4852 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
4855 Automatic update of common submodule
4856 From 1ccbe09 to c3cafe1
4858 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4861 Automatic update of common submodule
4862 From 193b717 to 1ccbe09
4864 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
4867 Automatic update of common submodule
4868 From b77e2bf to 193b717
4870 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4873 build: Include lcov.mak to allow test coverage report generation
4875 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4878 Automatic update of common submodule
4879 From d8814b6 to b77e2bf
4881 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4884 Automatic update of common submodule
4885 From 6aaa286 to d8814b6
4887 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
4890 Automatic update of common submodule
4891 From 6aec6b9 to 6aaa286
4893 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
4896 autogen: wingo signed comment
4898 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
4900 * gst/rtsp-server/rtsp-session-pool.c:
4901 session: use full charset for RTSP session ID
4902 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
4903 session ID more difficult.
4904 https://bugzilla.gnome.org/show_bug.cgi?id=643812
4906 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4908 * gst/rtsp-server/Makefile.am:
4909 rtsp-server: Don't install the funnel header
4911 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
4914 Automatic update of common submodule
4915 From 1de7f6a to 6aec6b9
4917 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4920 configure: require core/base 0.10.31
4921 Needed at least for gst_plugin_feature_rank_compare_func().
4923 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
4926 Automatic update of common submodule
4927 From f94d739 to 1de7f6a
4929 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4931 * gst/rtsp-server/rtsp-media.c:
4932 media: remove more unused code
4934 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4936 * gst/rtsp-server/rtsp-media.c:
4937 * gst/rtsp-server/rtsp-media.h:
4938 media: remove duplicate filtering
4939 Remove the duplicate filtering code now that we have a released -good version.
4940 Give a warning instead.
4942 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4944 * gst/rtsp-server/rtsp-media-factory.c:
4945 * gst/rtsp-server/rtsp-media.c:
4946 media: fix default buffer size
4948 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4950 * gst/rtsp-server/rtsp-media-factory.c:
4951 * gst/rtsp-server/rtsp-media-factory.h:
4952 media-factory: add property to configure the buffer-size
4953 Add a property to configure the kernel UDP buffer size.
4955 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4957 * gst/rtsp-server/rtsp-media.c:
4958 * gst/rtsp-server/rtsp-media.h:
4959 media: add property to configure kernel buffer sizes
4960 Add a property to configure the kernel UDP buffer size.
4962 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4965 configure: set PYGOBJECT_REQ before using it
4966 https://bugzilla.gnome.org/show_bug.cgi?id=640641
4968 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4971 docs: recursive into sub-directories on 'make upload'
4973 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4975 * docs/libs/gst-rtsp-server-docs.sgml:
4976 * docs/version.entities.in:
4977 docs: mention full version these docs are for, not just major-minor
4979 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4984 === release 0.10.8 ===
4986 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4991 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4993 * gst/rtsp-server/rtsp-server.c:
4994 rtsp-server: clarify docs a little
4996 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4998 * gst/rtsp-server/rtsp-media.c:
4999 media: init debug category before starting thread
5001 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5003 * gst/rtsp-server/rtsp-auth.c:
5004 auth: add realm to make it more spec compliant
5006 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5008 * gst/rtsp-server/rtsp-server.c:
5009 * gst/rtsp-server/rtsp-server.h:
5012 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5014 * examples/test-video.c:
5015 example: improve example docs a little
5017 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5019 * gst/rtsp-server/rtsp-server.c:
5020 server: ensure the watch has a ref to the server
5022 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5024 * gst/rtsp-server/rtsp-server.c:
5025 server: simpify channel function
5027 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5029 * gst/rtsp-server/rtsp-server.c:
5030 * gst/rtsp-server/rtsp-server.h:
5031 server: simplify management of channel and source
5032 We don't need to keep around the channel and source objects. Let the mainloop
5033 and the source manage the source and channel respectively.
5035 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5041 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5044 * tests/Makefile.am:
5045 * tests/test-cleanup.c:
5046 tests: add tests directory and cleanup test
5048 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5050 * gst/rtsp-server/rtsp-media-factory-uri.c:
5051 * gst/rtsp-server/rtsp-media-factory.c:
5052 * gst/rtsp-server/rtsp-media-mapping.c:
5053 * gst/rtsp-server/rtsp-media.c:
5054 * gst/rtsp-server/rtsp-session-pool.c:
5055 * gst/rtsp-server/rtsp-session.c:
5056 server: improve debugging in various objects
5058 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5060 * gst/rtsp-server/rtsp-server.c:
5061 server: chain up to the parent finalize
5063 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
5065 * bindings/python/rtspserver-types.defs:
5066 * bindings/python/rtspserver.defs:
5067 * bindings/python/rtspserver.override:
5068 * bindings/python/test.py:
5069 gst-rtsp-server: update python bindings
5071 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5073 * gst/rtsp-server/rtsp-client.c:
5074 client: use the response from the clientstate
5075 Create the response object only once and store in the client state.
5076 Make all methods use the state response,
5078 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5080 * gst/rtsp-server/rtsp-server.c:
5081 server: use signal to keep track of clients
5082 Keep track of all the clients that the server creates and remove them when they
5083 fire the 'closed' signal.
5085 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5087 * gst/rtsp-server/rtsp-client.c:
5088 * gst/rtsp-server/rtsp-client.h:
5089 client: emit signal when closing
5091 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5093 * examples/.gitignore:
5094 * examples/Makefile.am:
5095 * examples/test-auth.c:
5096 * examples/test-video.c:
5097 * gst/rtsp-server/rtsp-auth.c:
5098 * gst/rtsp-server/rtsp-auth.h:
5099 * gst/rtsp-server/rtsp-client.c:
5100 * gst/rtsp-server/rtsp-media-factory.c:
5101 * gst/rtsp-server/rtsp-media.c:
5102 * gst/rtsp-server/rtsp-media.h:
5103 * gst/rtsp-server/rtsp-session-pool.h:
5104 * gst/rtsp-server/rtsp-session.h:
5105 media: enable per factory authorisations
5106 Allow for adding a GstRTSPAuth on the factory and media level and check
5107 permissions when accessing the factory.
5108 Add hints to the auth methods for future more fine grained authorisation.
5109 Add example application for per factory authentication.
5111 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5113 * gst/rtsp-server/rtsp-auth.c:
5114 * gst/rtsp-server/rtsp-auth.h:
5115 * gst/rtsp-server/rtsp-client.c:
5116 * gst/rtsp-server/rtsp-client.h:
5117 * gst/rtsp-server/rtsp-params.c:
5118 * gst/rtsp-server/rtsp-params.h:
5119 rtsp-server: Pass ClientState structure arround
5120 Pass the collected information for the ongoing request in a GstRTSPClientState
5121 structure that we can then pass around to simplify the method arguments. This
5122 will also be handy when we implement logging functionality.
5124 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5126 * gst/rtsp-server/rtsp-media-factory.c:
5127 * gst/rtsp-server/rtsp-media-factory.h:
5128 media-factory: add methods to configure authorisation
5130 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5132 * gst/rtsp-server/rtsp-client.c:
5133 client: unref auth in finalize
5135 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5137 * gst/rtsp-server/rtsp-server.c:
5138 server: unref auth in finalize
5140 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5142 * docs/libs/gst-rtsp-server-docs.sgml:
5143 * docs/libs/gst-rtsp-server-sections.txt:
5144 * docs/libs/gst-rtsp-server.types:
5147 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5149 * gst/rtsp-server/rtsp-server.c:
5150 * gst/rtsp-server/rtsp-server.h:
5151 server: separate create and accept
5152 Create separate create and accept methods so that subclasses can create custom
5154 Configure the server in the client object and prepare for keeping track of
5157 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5159 * gst/rtsp-server/rtsp-client.c:
5160 * gst/rtsp-server/rtsp-client.h:
5161 client: add support for setting the server.
5162 Add support for keeping a ref to the server that started this client
5165 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5167 * gst/rtsp-server/rtsp-auth.c:
5168 auth: fix memleak and add some docs
5169 Fix a memleak of the basic auth token.
5170 Add docs for the helper function
5172 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5174 * gst/rtsp-server/rtsp-auth.c:
5175 * gst/rtsp-server/rtsp-auth.h:
5176 * gst/rtsp-server/rtsp-client.c:
5177 client: delegate setup of auth to the manager
5178 Delegate the configuration of the authentication tokens to the manager object
5181 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5183 * examples/test-video.c:
5184 * gst/rtsp-server/Makefile.am:
5185 * gst/rtsp-server/rtsp-auth.c:
5186 * gst/rtsp-server/rtsp-auth.h:
5187 * gst/rtsp-server/rtsp-client.c:
5188 * gst/rtsp-server/rtsp-client.h:
5189 * gst/rtsp-server/rtsp-server.c:
5190 * gst/rtsp-server/rtsp-server.h:
5191 auth: add authentication object
5192 Add an object that can check the authorization of requests.
5193 Implement basic authentication.
5194 Add example authentication to test-video
5196 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5198 * gst/rtsp-server/rtsp-server.c:
5199 * gst/rtsp-server/rtsp-server.h:
5200 server: move includes back
5201 the includes are needed for sockaddr_in.
5203 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5205 * gst/rtsp-server/rtsp-client.c:
5206 * gst/rtsp-server/rtsp-client.h:
5207 * gst/rtsp-server/rtsp-server.c:
5208 * gst/rtsp-server/rtsp-server.h:
5209 rtsp: move network includes where they are needed
5211 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
5213 * gst/rtsp-server/rtsp-media.h:
5214 rtsp-media.h: Minor corrections in comments.
5217 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
5220 Automatic update of common submodule
5221 From e572c87 to f94d739
5223 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5227 * docs/libs/.gitignore:
5228 * examples/.gitignore:
5229 * gst/rtsp-server/.gitignore:
5232 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5234 * docs/libs/Makefile.am:
5235 docs: We don't build ps/pdf for API reference docs
5237 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5240 Automatic update of common submodule
5241 From ccbaa85 to e572c87
5243 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5246 Automatic update of common submodule
5247 From 46445ad to ccbaa85
5249 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5251 * gst/rtsp-server/Makefile.am:
5252 * gst/rtsp-server/fs-funnel.c:
5253 * gst/rtsp-server/fs-funnel.h:
5254 * gst/rtsp-server/rtsp-funnel.c:
5255 * gst/rtsp-server/rtsp-funnel.h:
5256 * gst/rtsp-server/rtsp-media.c:
5257 funnel: rename fsfunnel to rtspfunnel
5258 Rename the funnel to avoid conflicts with the farsight one.
5260 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5262 * gst/rtsp-server/Makefile.am:
5263 * gst/rtsp-server/fs-funnel.c:
5264 * gst/rtsp-server/fs-funnel.h:
5265 * gst/rtsp-server/rtsp-media.c:
5266 rtsp-media: add and use fsfunnel
5267 Add a copy of fsfunnel to the build because input-selector removed the (broken)
5268 select-all property that we need.
5270 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5272 * gst/rtsp-server/Makefile.am:
5273 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
5274 Use PKG_CONFIG_PATH specified at configure time (if any) as well
5275 for the g-ir-compiler, rather than just assuming the env var has
5278 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5285 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
5287 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5290 * gst/rtsp-server/Makefile.am:
5291 gobject-introspection: fix g-i build for uninstalled setup
5292 Requires gst-plugins-base git (> 0.10.31.2).
5294 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5296 * examples/test-uri.c:
5297 examples: add some more options and comments
5299 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5301 * gst/rtsp-server/rtsp-media-factory-uri.c:
5302 factory-uri: use right property type
5304 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5306 * gst/rtsp-server/rtsp-media-factory-uri.c:
5307 factory-uri: attempt to configure buffer-lists
5308 Attempt to configure buffer lists in the payloader for improved performance.
5310 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5312 * gst/rtsp-server/rtsp-media.c:
5313 media: attempt to configure bigger UDP buffers
5314 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
5315 send buffers with high bitrate streams.
5317 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
5319 * gst/rtsp-server/rtsp-client.c:
5320 client: use the socket length from getsockname
5321 Use the length returned by getsockname to perform the getnameinfo call because
5322 the size can depend on the socket type and platform.
5325 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5327 * docs/libs/gst-rtsp-server-docs.sgml:
5328 * docs/libs/gst-rtsp-server-sections.txt:
5329 docs: add uri factory to the docs
5331 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5333 * gst/rtsp-server/rtsp-client.c:
5334 * gst/rtsp-server/rtsp-media.h:
5337 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5339 * gst/rtsp-server/rtsp-client.c:
5340 * gst/rtsp-server/rtsp-media.c:
5341 * gst/rtsp-server/rtsp-media.h:
5342 * gst/rtsp-server/rtsp-session.c:
5343 * gst/rtsp-server/rtsp-session.h:
5344 rtsp-server: add support for buffer lists
5345 Add support for sending bufferlists received from appsink.
5348 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5350 * gst/rtsp-server/rtsp-client.c:
5351 * gst/rtsp-server/rtsp-media.c:
5352 * gst/rtsp-server/rtsp-media.h:
5353 * gst/rtsp-server/rtsp-sdp.c:
5354 media: make method to retrieve the play range
5355 Make a method to retrieve the playback range so that we can conditionally create
5356 a different range for the SDP and the PLAY requests.
5358 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5360 * gst/rtsp-server/rtsp-media.c:
5361 * gst/rtsp-server/rtsp-media.h:
5362 media: add signal to notify of state changes
5364 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5366 * gst/rtsp-server/rtsp-client.h:
5367 client: cleanup headers
5369 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5371 * gst/rtsp-server/rtsp-client.c:
5374 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5376 * gst/rtsp-server/rtsp-media-factory-uri.c:
5377 * gst/rtsp-server/rtsp-media-factory-uri.h:
5378 factory-uri: add support for gstpay
5379 Add an option to prefer gstpay over decoder + raw payloader.
5381 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5383 * gst/rtsp-server/rtsp-media-factory-uri.c:
5384 * gst/rtsp-server/rtsp-media-factory-uri.h:
5385 factory-uri: rework the autoplugger.
5386 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
5389 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5391 * gst/rtsp-server/rtsp-media-factory-uri.c:
5392 factory-uri: use better factory filter
5393 Make better payloader filter based on autoplug rank and RTP use case.
5395 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5398 Automatic update of common submodule
5399 From 169462a to 46445ad
5401 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5403 * gst/rtsp-server/rtsp-server.c:
5404 server: set SO_REUSEADDR before bind
5405 Set the SO_REUSEADDR _before_ bind() to make it actually work.
5407 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5409 * gst/rtsp-server/rtsp-media.c:
5410 * gst/rtsp-server/rtsp-media.h:
5411 media: emit prepared signal when prepared
5412 Make a 'prepared' signal and emit it when we successfully prepared the element.
5413 This signal can be used to configure the media object after it has been prepared
5416 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
5419 Automatic update of common submodule
5420 From 011bcc8 to 169462a
5422 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
5424 python an optional dependency
5425 * configure.ac: Move up valgrind and g-i checks. Make the python
5426 dependency optional, as it was before.
5428 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5430 Merge branch 'master' into 0.11
5435 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5437 * gst/rtsp-server/rtsp-media.c:
5438 media: update range when active clients changed
5439 When we changed the number of active clients, update the current range
5440 information because we want the second client connecting to a shared resource
5441 continue from where the stream currently.
5443 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5445 * gst/rtsp-server/rtsp-media-factory-uri.c:
5446 * gst/rtsp-server/rtsp-media-factory-uri.h:
5447 factory-uri: add colorspace and fix pt
5448 Rework the way we pass data to the autoplugger.
5449 When we have raw caps, plug a converter element to make pluggin to raw
5450 payloaders more successful.
5451 Make sure all dynamically plugged payloaders have a unique payload types.
5453 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5455 * examples/Makefile.am:
5456 * examples/test-uri.c:
5457 example: add example of the uri factory
5459 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5461 * gst/rtsp-server/Makefile.am:
5462 * gst/rtsp-server/rtsp-media-factory-uri.c:
5463 * gst/rtsp-server/rtsp-media-factory-uri.h:
5464 * gst/rtsp-server/rtsp-server.h:
5465 factory-uri: add a factory to stream any URI
5466 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
5469 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5471 * gst/rtsp-server/rtsp-media.c:
5472 * gst/rtsp-server/rtsp-media.h:
5473 media: ignore spurious ASYNC_DONE messages
5474 When we are dynamically adding pads, the addition of the udpsrc elements will
5475 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
5476 the real ASYNC_DONE when everything is prerolled.
5478 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5480 * gst/rtsp-server/rtsp-media-factory.c:
5481 * gst/rtsp-server/rtsp-media-factory.h:
5482 media-factory: make lock macro
5484 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
5486 * gst/rtsp-server/rtsp-client.c:
5487 rtsp-server: Remove unused variable and dead assignment
5489 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
5491 * examples/test-launch.c:
5492 * examples/test-mp4.c:
5493 * examples/test-ogg.c:
5494 * examples/test-readme.c:
5495 * examples/test-sdp.c:
5496 * examples/test-video.c:
5497 examples: Run gst-indent
5499 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
5501 * gst/rtsp-server/rtsp-client.c:
5502 * gst/rtsp-server/rtsp-media-factory.c:
5503 * gst/rtsp-server/rtsp-media-mapping.c:
5504 * gst/rtsp-server/rtsp-media.c:
5505 * gst/rtsp-server/rtsp-params.c:
5506 * gst/rtsp-server/rtsp-sdp.c:
5507 * gst/rtsp-server/rtsp-server.c:
5508 * gst/rtsp-server/rtsp-session-pool.c:
5509 * gst/rtsp-server/rtsp-session.c:
5510 rtsp-server: Run gst-indent
5511 Since it wasn't using the upstream common previously, there was no
5512 indentation check before commiting.
5514 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
5516 * gst/rtsp-server/rtsp-media-mapping.h:
5517 * gst/rtsp-server/rtsp-media.c:
5518 * gst/rtsp-server/rtsp-media.h:
5519 * gst/rtsp-server/rtsp-sdp.c:
5520 * gst/rtsp-server/rtsp-session-pool.h:
5521 * gst/rtsp-server/rtsp-session.c:
5522 * gst/rtsp-server/rtsp-session.h:
5523 rtsp-server: Some more doc fixups
5525 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5528 Makefile: Add cruft-cleaning support
5530 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5535 * docs/libs/Makefile.am:
5536 * docs/libs/gst-rtsp-server-docs.sgml:
5537 * docs/libs/gst-rtsp-server-sections.txt:
5538 * docs/libs/gst-rtsp-server.types:
5539 * docs/version.entities.in:
5540 docs: Add gtk-doc build system
5542 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5544 * gst/rtsp-server/Makefile.am:
5545 Makefile.am: Use standard GIR make behaviour
5547 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5551 autogen/configure: Bring more in sync to standard gst module behaviour
5553 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5555 * gst/rtsp-server/rtsp-media.c:
5556 media: warn and fail when gstrtpbin is not found
5558 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5561 configure: open 0.11 branch
5563 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
5567 Add common submodule
5569 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
5572 * common/Makefile.am:
5573 * common/c-to-xml.py:
5575 * common/coverage/coverage-report-entry.pl:
5576 * common/coverage/coverage-report.pl:
5577 * common/coverage/coverage-report.xsl:
5578 * common/coverage/lcov.mak:
5579 * common/gettext.patch:
5580 * common/glib-gen.mak:
5581 * common/gst-autogen.sh:
5582 * common/gst-xmlinspect.py:
5584 * common/gstdoc-scangobj:
5585 * common/gtk-doc-plugins.mak:
5586 * common/gtk-doc.mak:
5587 * common/m4/.gitignore:
5588 * common/m4/Makefile.am:
5590 * common/m4/as-ac-expand.m4:
5591 * common/m4/as-auto-alt.m4:
5592 * common/m4/as-compiler-flag.m4:
5593 * common/m4/as-compiler.m4:
5594 * common/m4/as-docbook.m4:
5595 * common/m4/as-libtool-tags.m4:
5596 * common/m4/as-libtool.m4:
5597 * common/m4/as-python.m4:
5598 * common/m4/as-scrub-include.m4:
5599 * common/m4/as-version.m4:
5600 * common/m4/ax_create_stdint_h.m4:
5601 * common/m4/check.m4:
5602 * common/m4/glib-gettext.m4:
5603 * common/m4/gst-arch.m4:
5604 * common/m4/gst-args.m4:
5605 * common/m4/gst-check.m4:
5606 * common/m4/gst-debuginfo.m4:
5607 * common/m4/gst-default.m4:
5608 * common/m4/gst-doc.m4:
5609 * common/m4/gst-error.m4:
5610 * common/m4/gst-feature.m4:
5611 * common/m4/gst-function.m4:
5612 * common/m4/gst-gettext.m4:
5613 * common/m4/gst-glib2.m4:
5614 * common/m4/gst-libxml2.m4:
5615 * common/m4/gst-plugindir.m4:
5616 * common/m4/gst-valgrind.m4:
5617 * common/m4/gtk-doc.m4:
5618 * common/m4/introspection.m4:
5620 * common/mangle-tmpl.py:
5621 * common/plugins.xsl:
5623 * common/release.mak:
5624 * common/scangobj-merge.py:
5625 * common/upload.mak:
5626 common: Remove static version
5628 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
5630 * common/m4/introspection.m4:
5631 Update introspection.m4 to match usage
5633 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5637 Remove old stuff from the README
5639 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5644 === release 0.10.7 ===
5646 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5651 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5653 * examples/test-ogg.c:
5654 test-ogg: remove parsers
5655 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
5656 buffers with timestamps. Using the parsers also seems to break things.
5658 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
5660 * bindings/vala/gst-rtsp-server-0.10.vapi:
5661 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
5662 Updated Vala bindings
5664 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
5666 * common/m4/introspection.m4:
5668 * gst/rtsp-server/Makefile.am:
5669 Added initial gobject-introspection support
5671 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5673 * gst/rtsp-server/rtsp-media-factory.c:
5674 media-factory: don't use host for shared hash key
5675 When we generate the key to share made between connections, don't include the
5676 host used to connect so that we can share media even if between clients that
5677 connected with localhost and ones with the ip address.
5679 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5681 * bindings/vala/Makefile.am:
5682 build: fix distcheck
5684 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5686 * bindings/vala/gst-rtsp-server-0.10.vapi:
5687 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
5688 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
5689 Update Vala bindings
5691 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5693 * bindings/vala/Makefile.am:
5695 Fix configure checks and installation location for Vala bindings
5698 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5703 === release 0.10.6 ===
5705 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5708 configure: release 0.10.6
5710 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5712 * gst/rtsp-server/rtsp-media.c:
5713 media: help the compiler a little
5715 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5717 * gst/rtsp-server/rtsp-media.c:
5718 * gst/rtsp-server/rtsp-media.h:
5719 * gst/rtsp-server/rtsp-session.c:
5720 media: cleanup media transport before freeing
5721 Cleanup the media transport data before freeing. In particular, remove the qdata
5722 from the rtpsource object.
5724 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5726 * gst/rtsp-server/rtsp-media-factory.c:
5727 * gst/rtsp-server/rtsp-media-factory.h:
5728 * gst/rtsp-server/rtsp-media.c:
5729 * gst/rtsp-server/rtsp-media.h:
5730 media-factory: add eos-shutdown property
5731 Add an eos-shutdown property that will send an EOS to the pipeline before
5732 shutting it down. This allows for nice cleanup in case of a muxer.
5735 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5737 * gst/rtsp-server/rtsp-media.c:
5738 * gst/rtsp-server/rtsp-media.h:
5739 media: use multiudpsink send-duplicates when we can
5740 If we have a new enough multiudpsink with the send-duplicates property, use this
5741 instead of doing our own filtering. Our custom filtering code should eventually
5742 be removed when we can depend on a released -good.
5744 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5746 * gst/rtsp-server/rtsp-media.c:
5747 media: don't leak destinations
5748 Refactor and cleanup the destinations array when the stream is destroyed.
5750 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5752 * gst/rtsp-server/rtsp-media.c:
5753 * gst/rtsp-server/rtsp-media.h:
5754 media: don't add udp addresses multiple times
5755 Keep track of the udp addresses we added to udpsink and never add the same udp
5756 destination twice. This avoids duplicate packets when using multicast.
5758 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5760 * gst/rtsp-server/rtsp-server.c:
5761 server: disable use of SO_LINGER
5762 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
5763 server close()s the connection.
5765 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5767 * gst/rtsp-server/rtsp-server.c:
5768 server: use 5 second linger period in SO_LINGER
5769 Wait 5 seconds before clearing the send buffers and reseting the connection with
5770 the client when we do a close. This should be enough time to get the message to
5774 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
5776 * gst/rtsp-server/rtsp-server.c:
5777 server: use SO_LINGER
5778 SO_LINGER on the socket will make sure that any pending data on the socket is
5779 flushed ASAP and that the socket connection is reset. This makes sure that the
5780 socket can be reused immediately.
5783 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5786 README: add blurb about shared media factories
5788 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
5790 * gst/rtsp-server/rtsp-media.c:
5791 Add stdlib.h for atoi()
5793 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5795 * bindings/python/Makefile.am:
5796 * bindings/vala/Makefile.am:
5797 build: distcheck fixes
5798 Fix 'make distcheck', somewhat (it still fails because it tries to
5799 install files into /usr/share/vala/vapi/ irrespective of the
5802 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5805 configure: bump core/base requirements to released version
5806 Makes things less confusing for people.
5808 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5811 configure: fail if GStreamer core/base requirements are not met
5813 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5815 * gst/rtsp-server/rtsp-client.c:
5816 client: improve client cleanups
5817 Make sure the session does not timeout when using TCP. We need to do this
5818 because quicktime player does not send RTCP for some reason in tunneled
5820 Refactor some cleanup code.
5823 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5825 * gst/rtsp-server/rtsp-session.c:
5826 * gst/rtsp-server/rtsp-session.h:
5827 session: add support for prevent session timeouts
5828 Add an atomix counter to prevent session timeouts when we are, for example,
5831 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5833 * gst/rtsp-server/rtsp-client.c:
5834 client: fix unlink on session timeouts
5835 When our session times out, make sure we unlink all streams in this
5837 Remove the tunnelid when closing the connection.
5839 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5841 * gst/rtsp-server/rtsp-session.c:
5842 session: small cleanups
5844 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5846 * gst/rtsp-server/rtsp-client.c:
5847 client: handle lost_tunnel callbacks
5848 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
5849 hashtable so that we can reuse it for when the client reopens the POST
5851 Close the connection after a TEARDOWN.
5852 Make sure or watchid is cleared when the watch is removed.
5855 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5857 * gst/rtsp-server/rtsp-client.c:
5858 * gst/rtsp-server/rtsp-media.c:
5859 * gst/rtsp-server/rtsp-sdp.c:
5860 rtsp-server: add more support for multicast
5862 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5865 * gst/rtsp-server/rtsp-media.c:
5866 * gst/rtsp-server/rtsp-media.h:
5867 media: allow configuration of allowed lower transport
5869 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5871 * gst/rtsp-server/rtsp-client.h:
5872 * gst/rtsp-server/rtsp-media.c:
5873 * gst/rtsp-server/rtsp-media.h:
5874 * gst/rtsp-server/rtsp-sdp.c:
5875 * gst/rtsp-server/rtsp-sdp.h:
5876 * gst/rtsp-server/rtsp-server.c:
5877 rtsp: keep track of server ip and ipv6
5878 Keep track of how the client connected to the server and setup the udp ports
5879 with the same protocol.
5880 Copy the server ip address in the SDP so that clients can send RTCP back to
5883 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5885 * gst/rtsp-server/rtsp-session.c:
5888 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5890 * gst/rtsp-server/rtsp-client.c:
5891 client: use right size for malloc
5893 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5895 * gst/rtsp-server/rtsp-server.c:
5896 server: comment ipv6 server listening address
5898 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5900 * gst/rtsp-server/rtsp-media.c:
5901 media: allow for ipv6 sockets
5903 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5905 * gst/rtsp-server/rtsp-server.c:
5906 * gst/rtsp-server/rtsp-server.h:
5907 server: rework server part
5908 Allow setting a bind address, make sure we can deal with ipv6.
5909 Remove the port property and change with the service property.
5911 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5913 * gst/rtsp-server/rtsp-media.h:
5914 media: update comments a little
5916 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5918 * gst/rtsp-server/rtsp-client.c:
5919 client: make content-base better
5920 Use the URI formatting functions to make a content-base. Also make sure that
5921 there is a trailing / at the end.
5923 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5925 * gst/rtsp-server/rtsp-client.c:
5926 client: guard against invalid paths
5928 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5930 * examples/test-video.c:
5931 test: catch server bind errors
5933 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
5935 * gst/rtsp-server/rtsp-media.c:
5936 rtspmedia: emit "unprepared" if _prepare fails.
5937 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
5938 media object is removed from its factory's cache.
5940 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5942 * gst/rtsp-server/rtsp-media.c:
5943 media: collect media position when seek completes
5945 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
5947 * gst/rtsp-server/rtsp-client.c:
5948 client: call unlink_streams in client finalize
5951 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5953 * gst/rtsp-server/rtsp-media.c:
5954 media: limit the time to wait to something huge
5955 Avoid waiting forever but limit the timeout to 20 seconds.
5957 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5959 * gst/rtsp-server/rtsp-sdp.c:
5960 sdp: reindent and check for prepared status
5962 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5964 * gst/rtsp-server/rtsp-media.c:
5965 * gst/rtsp-server/rtsp-media.h:
5966 * gst/rtsp-server/rtsp-session.c:
5967 media: avoid doing _get_state() for state changes
5968 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
5969 until the media is prerolled or in error. This avoids doing a blocking call of
5970 gst_element_get_state() that can cause lockups when there is an error.
5973 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5975 * gst/rtsp-server/rtsp-media.c:
5978 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5980 * gst/rtsp-server/rtsp-media-factory.c:
5981 media-factory: better error handling
5982 Improve the error handling a bit.
5984 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5986 * gst/rtsp-server/rtsp-client.c:
5987 client: rework transport parsing
5988 Rework the transport parsing code so that we can ignore transports we don't
5989 support instead of just picking the first one we can parse.
5990 Configure a (for now hardcoded) destination for multicast transports.
5992 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5994 * gst/rtsp-server/rtsp-media.c:
5995 media: set multicast sink parameters
5996 Disable loop and automatic multicast join on the udpsink elements.
5997 Add some more debug info.
5998 Reset some state variables in the right place.
5999 Use the right port numbers for multicast.
6001 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6003 * gst/rtsp-server/rtsp-session.c:
6004 session: handle transport setup correctly
6005 Handle UDP, MCAST and TCP transport negotiation more correctly.
6006 Store the server session SSRC in the transport.
6008 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6010 * gst/rtsp-server/rtsp-client.c:
6011 rtsp-client: implement error_full
6012 Implement error_full to avoid some segfaults when the rtspconnection calls it.
6015 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6018 * gst/rtsp-server/rtsp-client.c:
6019 * gst/rtsp-server/rtsp-server.c:
6020 docs: update docs and comments
6022 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
6024 * gst/rtsp-server/rtsp-sdp.c:
6025 sdp: make server work better when behind a proxy
6027 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6029 * gst/rtsp-server/rtsp-client.c:
6030 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
6032 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6034 * gst/rtsp-server/rtsp-client.c:
6035 * gst/rtsp-server/rtsp-media-factory.c:
6036 * gst/rtsp-server/rtsp-media-mapping.c:
6037 * gst/rtsp-server/rtsp-media.c:
6038 * gst/rtsp-server/rtsp-server.c:
6039 * gst/rtsp-server/rtsp-session-pool.c:
6040 * gst/rtsp-server/rtsp-session.c:
6041 Use GStreamer's debugging subsystem
6043 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6045 * gst/rtsp-server/rtsp-media-factory.c:
6046 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
6048 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6053 === release 0.10.5 ===
6055 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6060 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6063 configure: bump required versions
6065 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
6067 * gst/rtsp-server/rtsp-client.c:
6068 client: call weak-unref on client->sessions from finalize
6071 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6073 * gst/rtsp-server/rtsp-media.c:
6074 media: Fixed crasher where caps got unref'ed too often
6076 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6079 * pkgconfig/.gitignore:
6080 * pkgconfig/Makefile.am:
6081 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
6082 Added pkg-config file to use gst-rtsp-server uninstalled
6084 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6086 * gst/rtsp-server/rtsp-media.c:
6087 media: add some docs
6089 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
6091 * gst/rtsp-server/rtsp-client.c:
6092 rtsp: Use gst_rtsp_watch_send_message().
6093 Use gst_rtsp_watch_send_message() since the old API which used
6094 gst_rtsp_watch_queue_message() has been deprecated.
6096 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6101 === release 0.10.4 ===
6103 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6108 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6110 * gst/rtsp-server/rtsp-client.c:
6111 * gst/rtsp-server/rtsp-session.c:
6112 * gst/rtsp-server/rtsp-session.h:
6113 rtsp: allocate channels in TCP mode
6114 When the client does not provide us with channels in TCP mode, allocate channels
6117 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6119 * gst/rtsp-server/rtsp-client.c:
6120 client: don't crash when tunnelid is missing
6121 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
6122 don't crash but return an error response to the client.
6125 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6127 * bindings/vala/gst-rtsp-server-0.10.vapi:
6128 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6129 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6130 bindings: update vala bindings with new method
6132 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6134 * gst/rtsp-server/rtsp-session-pool.c:
6135 * gst/rtsp-server/rtsp-session-pool.h:
6136 sessionpool: add function to filter sessions
6137 Add generic function to retrieve/remove sessions.
6139 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6142 configure: bump core/base requirements to release
6144 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6146 * gst/rtsp-server/rtsp-media.c:
6147 media: fix indentation
6149 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6151 * gst/rtsp-server/rtsp-media.c:
6152 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
6154 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6156 * gst/rtsp-server/rtsp-media.c:
6157 set state and remove elements of media in for loop
6159 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
6161 * bindings/vala/gst-rtsp-server-0.10.vapi:
6162 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6163 Added gst_rtsp_media_remove_elements function to Vala bindings
6165 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
6167 * gst/rtsp-server/rtsp-media.c:
6168 * gst/rtsp-server/rtsp-media.h:
6169 Added gst_rtsp_media_remove_elements function
6171 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
6173 * gst/rtsp-server/rtsp-media.c:
6174 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
6176 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6178 * bindings/vala/gst-rtsp-server-0.10.vapi:
6179 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6180 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6181 Updated Vala bindings
6183 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6185 * gst/rtsp-server/rtsp-media.c:
6186 * gst/rtsp-server/rtsp-media.h:
6187 Added vmethod unprepare to GstRTSPMedia
6188 The default implementation sets the state of the pipeline to GST_STATE_NULL
6190 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6192 * gst/rtsp-server/rtsp-media-factory.c:
6193 * gst/rtsp-server/rtsp-media-factory.h:
6194 Made collect_streams function public
6196 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6198 * gst/rtsp-server/rtsp-media-factory.c:
6199 * gst/rtsp-server/rtsp-media-factory.h:
6200 * gst/rtsp-server/rtsp-media.c:
6201 Added vmethod create_pipeline to GstRTSPMediaFactory
6202 The pipeline is created in this method and the GstRTSPMedia's element is added to it
6204 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6206 * gst/rtsp-server/rtsp-client.c:
6207 client: use g_source_destroy()
6208 We need to use g_source_destroy() because we might have added the source to a
6209 different main context than the default one.
6211 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6213 * gst/rtsp-server/Makefile.am:
6214 * gst/rtsp-server/rtsp-client.c:
6215 * gst/rtsp-server/rtsp-params.c:
6216 * gst/rtsp-server/rtsp-params.h:
6217 rtsp: prepare for handling GET/SET_PARAMETER
6218 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
6220 Fix return codes of handlers.
6222 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6224 * gst/rtsp-server/rtsp-media.c:
6225 media: don't leak session pads
6227 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6229 * gst/rtsp-server/rtsp-media.c:
6230 media: clean up the messages a bit
6232 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6234 * gst/rtsp-server/rtsp-sdp.c:
6235 sdp: warn and skip streams without media
6237 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6239 * bindings/vala/gst-rtsp-server-0.10.vapi:
6240 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6241 vala: Fixed typo in header file of RTSPMediaStream
6243 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6245 * gst/rtsp-server/rtsp-media.c:
6248 Make dumping RTCP stats configurable
6250 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6252 * gst/rtsp-server/rtsp-media.c:
6253 media: be less verbose and leak less
6255 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6257 * gst/rtsp-server/rtsp-media.c:
6258 media: don't leak the destination address
6260 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6262 * gst/rtsp-server/rtsp-client.c:
6263 * gst/rtsp-server/rtsp-media.c:
6264 * gst/rtsp-server/rtsp-media.h:
6265 * gst/rtsp-server/rtsp-session.c:
6266 * gst/rtsp-server/rtsp-session.h:
6267 rtsp: use RTCP to keep the session alive
6268 Use the RTCP rtcp-from stats field to find the associated session and use this
6269 to keep the session alive.
6271 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6273 * gst/rtsp-server/rtsp-session.c:
6274 session: add 5sec to the real session timeout
6275 Allow the session to live 5sec longer before really timing out. This should give
6276 clients some extra time to keep the session active.
6278 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6280 * gst/rtsp-server/rtsp-client.c:
6281 client: replay OK to GET/SET_PARAMETER
6282 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
6283 so that we return OK for those requests.
6285 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6287 * gst/rtsp-server/rtsp-media.c:
6288 * gst/rtsp-server/rtsp-media.h:
6289 media: keep track of active transports
6290 Keep track of which transport is active to avoid closing the connection too
6292 Remove the destination transport also when going to NULL.
6293 Print some stats about the SDES and other RTCP messages we receive from the
6296 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6298 * examples/.gitignore:
6299 * examples/Makefile.am:
6300 * examples/test-sdp.c:
6301 example: add SDP relay example
6303 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6305 * gst/rtsp-server/rtsp-media.c:
6306 media: also count active TCP connections
6308 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6310 * gst/rtsp-server/rtsp-media-factory.c:
6311 * gst/rtsp-server/rtsp-media.c:
6312 * gst/rtsp-server/rtsp-media.h:
6313 rtsp: add support for dynamic elements
6314 Add support for dynamic elements.
6315 Don't set live pipelines back to paused.
6317 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6319 * gst/rtsp-server/rtsp-sdp.c:
6320 sdp: don't add encoding name when absent in caps
6322 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6324 * gst/rtsp-server/rtsp-client.c:
6325 client: warn when we can't do RTP-Info
6327 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6329 * gst/rtsp-server/rtsp-media-factory.c:
6330 factory: factor out the stream construction
6332 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6334 * gst/rtsp-server/rtsp-client.c:
6335 client: only add RTP-Info when we have the info
6336 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
6339 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6344 === release 0.10.3 ===
6346 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6350 - Fixes a bug where it put the wrong verion in pkgconfig
6351 - Link RTP and RTCP sources
6353 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6355 * gst/rtsp-server/rtsp-media.c:
6356 * gst/rtsp-server/rtsp-media.h:
6357 media: link the RTP udpsrc to the session manager
6358 Link the RTP udpsrc and the appsrc to the session manager so that they don't
6359 shut down when the client sends a packet to open firewalls.
6361 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6363 * pkgconfig/gst-rtsp-server.pc.in:
6364 Don't use hard-coded version number in pkg-config file
6366 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6371 === release 0.10.2 ===
6373 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6378 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6381 * common/m4/.gitignore:
6382 * examples/.gitignore:
6383 * pkgconfig/.gitignore:
6384 add some .gitignore files
6386 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6388 * gst/rtsp-server/rtsp-media.c:
6389 media: seek to key frames
6391 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6393 * gst/rtsp-server/rtsp-media.c:
6394 media: emit the unprepared signal by id
6395 Emit the unprepared signal by id instead of name and set the media as
6398 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6400 * gst/rtsp-server/rtsp-media.c:
6401 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
6403 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6405 * gst/rtsp-server/rtsp-server.c:
6406 Added finalize function to GstRTPSPServer to unref session pool and media mapping
6408 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6410 * bindings/vala/gst-rtsp-server-0.10.vapi:
6411 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6412 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6413 Updated vala bindings
6415 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6417 * gst/rtsp-server/Makefile.am:
6418 * gst/rtsp-server/rtsp-client.c:
6419 * gst/rtsp-server/rtsp-media.c:
6420 server: use appsink and appsrc with the API
6421 Use the appsink/appsrc API instead of the signals for higher
6424 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6426 * examples/test-ogg.c:
6427 tests: set the payload type correctly
6429 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6431 * gst/rtsp-server/rtsp-media-factory.c:
6432 factory: connect to the unprepare signal
6433 Connect to the unprepare signal for non-reusable media so that we can remove
6434 them from the cache.
6436 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6438 * gst/rtsp-server/rtsp-media.c:
6439 * gst/rtsp-server/rtsp-media.h:
6440 media: add signal to notify of unprepare
6442 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6444 * gst/rtsp-server/rtsp-media.c:
6445 * gst/rtsp-server/rtsp-media.h:
6446 media: more work on making the media shared
6447 Add a reusable flag to medias, indicating that they can be reused after a state
6451 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6453 * examples/test-readme.c:
6454 examples: mark the example as shared for testing
6456 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6458 * gst/rtsp-server/rtsp-media.c:
6459 * gst/rtsp-server/rtsp-media.h:
6460 client: support shared media
6461 Always perform the state actions even if the target state of the pipeline is
6462 already correct, we still want to add/remove the transports when we are dealing
6464 Keep a counter of the number of active transports for a media so that we can use
6465 this to perform a state change when needed.
6466 Perform a state change of the pipeline only when the first transport was added
6467 or when there are no active transports.
6469 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6471 * gst/rtsp-server/rtsp-client.c:
6472 client: fix refcounting crasher
6473 Don't need to remove the weak refs in the finalize methods, they are already
6474 removed in the dispose.
6475 Don't register the callback with a DestroyNofity.
6477 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6479 * gst/rtsp-server/rtsp-client.c:
6480 Fix rtsp client refcount management in TCP mode.
6481 Don't unref a client ref we never had. Fixes an unref
6482 of an already-free client object after a client
6483 teardown request for me.
6485 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6487 * gst/rtsp-server/rtsp-session.c:
6488 docs: fix typo in API docs
6490 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6492 * gst/rtsp-server/rtsp-media.c:
6494 Keep the udp sources in playing even if we go to paused. unlock the sources when
6496 Add some more debug info.
6497 Only seek when we need to.
6498 Keep track of the position when we go to paused.
6500 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6502 * gst/rtsp-server/rtsp-client.c:
6503 * gst/rtsp-server/rtsp-media.c:
6504 * gst/rtsp-server/rtsp-media.h:
6505 Add beginnings of seeking.
6506 Parse the Range header and perform a seek on the pipeline for the requested
6507 position. It's disabled currently until I figure out what's going wrong.
6509 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6511 * gst/rtsp-server/rtsp-client.c:
6512 allow pause requests for now.
6515 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6517 * gst/rtsp-server/rtsp-client.c:
6518 Remove weak ref on the session in teardown
6519 We need to remove our weakref from the session when we do a teardown because
6520 else we close the TCP connection prematurely.
6522 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6524 * gst/rtsp-server/rtsp-client.c:
6525 * gst/rtsp-server/rtsp-client.h:
6526 * gst/rtsp-server/rtsp-session-pool.c:
6527 Do some more session cleanup
6528 Make session timeout kill the TCP connection that currently watches the
6530 Remove the client timeout property.
6532 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6534 * gst/rtsp-server/rtsp-client.c:
6535 * gst/rtsp-server/rtsp-client.h:
6536 * gst/rtsp-server/rtsp-media.c:
6537 * gst/rtsp-server/rtsp-media.h:
6538 * gst/rtsp-server/rtsp-server.c:
6539 * gst/rtsp-server/rtsp-session.c:
6540 * gst/rtsp-server/rtsp-session.h:
6542 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
6545 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6547 * examples/Makefile.am:
6548 * examples/test-launch.c:
6549 Add example server that takes launch lines
6550 Add an example server that streams any -launch line.
6552 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6554 * examples/test-readme.c:
6555 * gst/rtsp-server/rtsp-client.c:
6556 * gst/rtsp-server/rtsp-media.c:
6557 * gst/rtsp-server/rtsp-media.h:
6558 Add support for live streams
6559 Add support for live streams and ranges
6560 Start on handling TCP data transfer.
6562 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6564 * gst/rtsp-server/rtsp-media.c:
6565 Free the pipeline before other things
6568 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6570 * gst/rtsp-server/rtsp-client.c:
6571 Only free the pending tunnel if there is one
6574 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6576 * gst/rtsp-server/rtsp-client.c:
6577 * gst/rtsp-server/rtsp-client.h:
6578 * gst/rtsp-server/rtsp-media.c:
6579 rtsp-server: Add support for tunneling
6580 Add support for tunneling over HTTP.
6581 Use new connection methods to retrieve the url.
6582 Dispatch messages based on the message type instead of blindly
6583 assuming it's always a request.
6584 Keep track of the watch id so that we can remove it later.
6585 Set the media pipeline to NULL before unreffing the pipeline.
6587 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6589 * gst/rtsp-server/rtsp-client.c:
6590 * gst/rtsp-server/rtsp-client.h:
6591 Fix for channel -> watch rename in gstreamer
6592 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
6594 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6596 * gst/rtsp-server/rtsp-client.c:
6597 * gst/rtsp-server/rtsp-client.h:
6599 Use the async RTSP channels instead of spawning a new thread for each client.
6600 If a sessionid is specified in a request, fail if we don't have the session.
6602 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6604 * gst/rtsp-server/rtsp-media.c:
6605 Add better debug info
6606 Add some better debug info.
6608 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6610 * examples/test-video.c:
6612 Add support for session timeouts in the example.
6614 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6616 * gst/rtsp-server/rtsp-session-pool.c:
6617 * gst/rtsp-server/rtsp-session-pool.h:
6618 Pass GTimeVal around for performance reasons
6619 Get the current time only once and pass it around so that sessions don't have to
6620 get the current time anymore.
6621 Add experimental support for a GSource that dispatches when the session needs to
6624 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6626 * gst/rtsp-server/rtsp-session.c:
6627 * gst/rtsp-server/rtsp-session.h:
6628 Add better support for session timeouts
6629 Add a method to request the number of milliseconds when a session will timeout.
6631 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6633 * gst/rtsp-server/rtsp-media.c:
6634 * gst/rtsp-server/rtsp-media.h:
6635 Add suport for RTP manager monitoring
6636 Add the first stage in monitoring the rtp manager.
6637 Make sure we don't update the state to something we don't want.
6639 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6641 * gst/rtsp-server/rtsp-client.c:
6642 Add support for session keepalive
6643 Get and update the session timeout for all requests. get the session as early as
6646 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6648 * gst/rtsp-server/rtsp-media-factory.h:
6649 * gst/rtsp-server/rtsp-media.c:
6650 * gst/rtsp-server/rtsp-media.h:
6651 Handle media bus messages
6652 Handle media bus messages in a custom mainloop and dispatch them to the
6653 RTSPMedia objects. Let the default implementation handle some common messages.
6655 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6657 * gst/rtsp-server/rtsp-client.c:
6658 * gst/rtsp-server/rtsp-session-pool.c:
6659 * gst/rtsp-server/rtsp-session.c:
6660 Some more session timeout handling
6661 Move the session header setting code to a central place so that we always add
6662 the timeout parameter too.
6663 Handle timeouts by running the session cleanup code.
6664 Stop media before cleaning up.
6666 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6668 * gst/rtsp-server/rtsp-client.c:
6669 * gst/rtsp-server/rtsp-client.h:
6670 Add timeout property
6671 Add a timeout property ot the client and make the other properties into GObject
6674 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6676 * gst/rtsp-server/rtsp-session-pool.c:
6677 Use getters and setters in property code
6678 Use the getters and setters for the timeout property instead of locking
6681 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6683 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
6685 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6687 * gst/rtsp-server/rtsp-session-pool.c:
6688 * gst/rtsp-server/rtsp-session-pool.h:
6689 * gst/rtsp-server/rtsp-session.c:
6690 * gst/rtsp-server/rtsp-session.h:
6691 Add more timeout stuff
6692 Add method to check if a session is expired.
6693 Add method to perform cleanup on a session pool.
6695 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6697 * gst/rtsp-server/rtsp-client.c:
6698 * gst/rtsp-server/rtsp-session-pool.c:
6699 * gst/rtsp-server/rtsp-session-pool.h:
6700 * gst/rtsp-server/rtsp-session.c:
6701 * gst/rtsp-server/rtsp-session.h:
6702 Add beginnings of session timeouts and limits
6703 Add the timeout value to the Session header for unusual timeout values.
6704 Allow us to configure a limit to the amount of active sessions in a pool. Set a
6705 limit on the amount of retry we do after a sessionid collision.
6706 Add properties to the sessionid and the timeout of a session. Keep track of
6707 creation time and last access time for sessions.
6709 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6711 * gst/rtsp-server/rtsp-client.c:
6712 * gst/rtsp-server/rtsp-media.c:
6713 * gst/rtsp-server/rtsp-media.h:
6714 * gst/rtsp-server/rtsp-sdp.c:
6715 * gst/rtsp-server/rtsp-session-pool.c:
6716 * gst/rtsp-server/rtsp-session.c:
6717 * gst/rtsp-server/rtsp-session.h:
6718 Cleanup of sessions and more
6719 Fix the refcounting of media and sessions in the client. Properly clean up the
6720 session data when the client performs a teardown.
6721 Add Server header to responses.
6722 Allow for multiple uri setups in one session.
6723 Add Range header to the PLAY response and add the range attribute to the SDP
6725 Fix the session pool remove method, it used the wrong key in the hashtable. Also
6726 give the ownership of the sessionid to the session object.
6728 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6730 * gst/rtsp-server/rtsp-server.c:
6731 * gst/rtsp-server/rtsp-server.h:
6733 Rename the 'server_port' variable to simply 'port'.
6735 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6738 * gst/rtsp-server/rtsp-client.c:
6739 * gst/rtsp-server/rtsp-media.c:
6740 * gst/rtsp-server/rtsp-media.h:
6741 * gst/rtsp-server/rtsp-session.c:
6742 * gst/rtsp-server/rtsp-session.h:
6743 Rework the way we handle transports for streams
6744 Make the media accept an array of transports for the streams that we have
6745 configured for the play/pause requests.
6746 Implement server states for a client and its media.
6747 Require 0.10.22.1 (git HEAD) of gstreamer.
6749 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6751 * gst/rtsp-server/rtsp-client.c:
6752 * gst/rtsp-server/rtsp-media-factory.c:
6753 Drop const from functions dealing with urls
6754 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
6755 have the right const in them.
6757 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6759 * gst/rtsp-server/rtsp-client.c:
6760 * gst/rtsp-server/rtsp-media.c:
6761 * gst/rtsp-server/rtsp-sdp.c:
6765 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6767 * gst/rtsp-server/rtsp-client.c:
6768 * gst/rtsp-server/rtsp-media-factory.c:
6769 * gst/rtsp-server/rtsp-media.c:
6770 * gst/rtsp-server/rtsp-media.h:
6772 Don't keep a reference to the GstRTSPMedia in the stream.
6773 Free more things when freeing the GstRTSPMedia.
6775 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6778 * gst/rtsp-server/rtsp-media-factory.c:
6779 * gst/rtsp-server/rtsp-media-factory.h:
6780 * gst/rtsp-server/rtsp-media.c:
6781 * gst/rtsp-server/rtsp-media.h:
6782 * gst/rtsp-server/rtsp-server.c:
6783 * gst/rtsp-server/rtsp-server.h:
6784 More docs and small cleanups
6785 Add some more docs and update the README
6786 Cleanup some method names.
6787 Remove an unneeded idx field in the GstRTSPMediaStream
6789 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6792 * examples/Makefile.am:
6793 * examples/test-readme.c:
6794 Add a README and more example code
6795 Add a README file that contains a small introduction on how to use the server
6796 along with the example code explained in the readme.
6798 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6800 * gst/rtsp-server/rtsp-media.c:
6801 * gst/rtsp-server/rtsp-server.c:
6802 Fix some leaks and change default port
6803 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
6804 we finished the initial preroll. If we keep them locked, setting the pipeline to
6805 NULL will not stop and clean up the sources correctly.
6806 Change the default RTSP port to 8554 aka the official alternative RTSP port.
6808 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6810 * gst/rtsp-server/rtsp-session.c:
6811 * gst/rtsp-server/rtsp-session.h:
6812 Cleanups to the session object
6813 Remove some unneeded variables in the session state of a stream such as the
6814 owner media and the server transport.
6815 Get the configuration of a media stream in a session based on the media_stream
6816 in the original object instead of our cached index.
6817 Free more data in the finalize method.
6819 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6821 * gst/rtsp-server/rtsp-client.c:
6822 * gst/rtsp-server/rtsp-client.h:
6823 Cleanups and reuse media from DESCRIBE
6824 Handle thread create errors.
6825 Rename some internal methods to better match what they actually do.
6826 Handle misconfiguration of session_pool and media_mapping gracefully.
6827 Cache the DESCRIBE media and uri in the client connection and reuse them when
6828 we receive a SETUP request in the same connection for the same uri.
6829 Cleanup the client connection object.
6831 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6833 * gst/rtsp-server/rtsp-media-factory.c:
6834 * gst/rtsp-server/rtsp-media-factory.h:
6835 * gst/rtsp-server/rtsp-media.c:
6836 * gst/rtsp-server/rtsp-media.h:
6837 Add shared properties to media and factory
6838 Add the shared property to media.
6839 Implement some simple caching in the factory depending on if the media is shared
6842 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6844 * gst/rtsp-server/rtsp-client.c:
6845 Add a little comment
6846 Add some comment about the content-base header.
6848 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6850 * examples/Makefile.am:
6852 * examples/test-mp4.c:
6853 * examples/test-ogg.c:
6854 * examples/test-video.c:
6855 * gst/rtsp-server/Makefile.am:
6856 * gst/rtsp-server/rtsp-client.c:
6857 * gst/rtsp-server/rtsp-client.h:
6858 * gst/rtsp-server/rtsp-media-factory.c:
6859 * gst/rtsp-server/rtsp-media-factory.h:
6860 * gst/rtsp-server/rtsp-media.c:
6861 * gst/rtsp-server/rtsp-media.h:
6862 * gst/rtsp-server/rtsp-sdp.c:
6863 * gst/rtsp-server/rtsp-sdp.h:
6864 * gst/rtsp-server/rtsp-server.c:
6865 * gst/rtsp-server/rtsp-server.h:
6866 * gst/rtsp-server/rtsp-session.c:
6867 * gst/rtsp-server/rtsp-session.h:
6868 Reorganize things, prepare for media sharing
6869 Added various other test server examples
6870 Move the SDP message generation to a separate helper.
6871 Refactor common code for finding the session.
6872 Add content-base for realplayer compatibility
6873 Clean up request uris before processing for better vlc compatibility.
6874 Move prerolling and pipeline construction to the RTSPMedia object.
6875 Use multiudpsink for future pipeline reuse.
6877 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6883 === release 0.10.1 ===
6885 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6891 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6893 * bindings/vala/Makefile.am:
6895 Add more directories and files to the dist.
6897 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6899 * bindings/python/Makefile.am:
6900 * bindings/python/rtspserver.override:
6901 Fixed compile error of python bindings
6903 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6905 * bindings/vala/gst-rtsp-server-0.10.vapi:
6906 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6907 Marked values as nullable accordingly
6909 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6911 * bindings/vala/gst-rtsp-server-0.10.vapi:
6912 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
6913 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6914 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6915 Updated Vala bindings
6917 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6919 * gst/rtsp-server/rtsp-client.c:
6920 * gst/rtsp-server/rtsp-media-mapping.c:
6921 * gst/rtsp-server/rtsp-media-mapping.h:
6922 * gst/rtsp-server/rtsp-media.h:
6923 * gst/rtsp-server/rtsp-session-pool.h:
6924 Cleanups and doc updates
6925 Add some more documentation and do some minor cleanups here and there.
6927 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6929 * gst/rtsp-server/rtsp-client.c:
6930 * gst/rtsp-server/rtsp-media-factory.c:
6931 * gst/rtsp-server/rtsp-media-factory.h:
6932 * gst/rtsp-server/rtsp-media.c:
6933 * gst/rtsp-server/rtsp-media.h:
6934 * gst/rtsp-server/rtsp-session.c:
6935 * gst/rtsp-server/rtsp-session.h:
6937 Rename GstRTSPMediaBin to GstRTSPMedia
6938 Parse the request url into a GstRTSPUri object and pass this object to the
6939 various handlers and methods that require the uri.
6941 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6945 Add some more docs and remove some old code from the example.
6947 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6949 * gst/rtsp-server/rtsp-client.c:
6950 Handle state change failures better
6951 Handle state change failures better when changing the state of the pipeline to
6954 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6956 * gst/rtsp-server/rtsp-media-factory.c:
6957 * gst/rtsp-server/rtsp-media-factory.h:
6958 Make element creation more extendible
6959 Add get_element vmethod to the default MediaFactory so that subclasses can just
6960 override that method and still use the default logic for making a MediaBin from
6963 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6966 * gst/rtsp-server/Makefile.am:
6967 * gst/rtsp-server/rtsp-client.c:
6968 * gst/rtsp-server/rtsp-client.h:
6969 * gst/rtsp-server/rtsp-media-factory.c:
6970 * gst/rtsp-server/rtsp-media-factory.h:
6971 * gst/rtsp-server/rtsp-media-mapping.c:
6972 * gst/rtsp-server/rtsp-media-mapping.h:
6973 * gst/rtsp-server/rtsp-media.c:
6974 * gst/rtsp-server/rtsp-media.h:
6975 * gst/rtsp-server/rtsp-server.c:
6976 * gst/rtsp-server/rtsp-server.h:
6977 * gst/rtsp-server/rtsp-session.c:
6978 * gst/rtsp-server/rtsp-session.h:
6979 Make the server handle arbitrary pipelines
6980 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
6981 The GstMediaBin object has a handle to a bin with elements and to a list of
6982 GstMediaStream objects that this bin produces.
6983 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
6984 with methods to register and remove those mappings.
6985 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
6986 used by the server instance.
6987 Modify the example application so that it shows how to create custom pipelines
6988 attached to a specific mount point.
6989 Various misc cleanps.
6991 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6993 * gst/rtsp-server/rtsp-server.c:
6994 * gst/rtsp-server/rtsp-server.h:
6995 Allow setting a custom media factory for a server
6997 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6999 * gst/rtsp-server/rtsp-client.c:
7000 * gst/rtsp-server/rtsp-client.h:
7001 Allow setting a custom media factory for a client.
7003 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7005 * gst/rtsp-server/Makefile.am:
7006 Add Makefile entry for the media factory
7008 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7010 * gst/rtsp-server/rtsp-media-factory.c:
7011 * gst/rtsp-server/rtsp-media-factory.h:
7012 Add media factory to map urls to media pipeline objects.
7014 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7016 * gst/rtsp-server/rtsp-media.c:
7017 * gst/rtsp-server/rtsp-media.h:
7018 Add comments. Remove unused field
7020 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7022 * gst/rtsp-server/rtsp-session-pool.c:
7023 * gst/rtsp-server/rtsp-session-pool.h:
7024 Allow custom session pools to override the session id allocation algorithms Add some comments.
7026 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7028 * gst/rtsp-server/rtsp-session.h:
7031 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7033 * gst/rtsp-server/rtsp-client.c:
7034 * gst/rtsp-server/rtsp-client.h:
7035 Move the connection code in one place Add some comments
7037 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7039 * gst/rtsp-server/rtsp-server.c:
7040 * gst/rtsp-server/rtsp-server.h:
7041 Make vmethod to create and accept new clients. Add some docs.
7043 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7045 * gst/rtsp-server/rtsp-server.c:
7046 * gst/rtsp-server/rtsp-server.h:
7047 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
7049 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7051 * gst/rtsp-server/rtsp-client.c:
7052 * gst/rtsp-server/rtsp-client.h:
7053 Name the parameters more appropriately.
7055 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7057 * gst/rtsp-server/rtsp-session-pool.c:
7058 Do some more cleanup of the session pool.
7060 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7062 * gst/rtsp-server/Makefile.am:
7063 * gst/rtsp-server/rtsp-client.c:
7064 Check if return value of gst_rtsp_session_get_media is not NULL
7066 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7068 * gst/rtsp-server/Makefile.am:
7069 Install rtsp-session and rtsp-session-pool headers
7071 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7076 * bindings/python/Makefile.am:
7077 * bindings/python/arg-types.py:
7078 * bindings/python/codegen/Makefile.am:
7079 * bindings/python/codegen/__init__.py:
7080 * bindings/python/codegen/argtypes.py:
7081 * bindings/python/codegen/code-coverage.py:
7082 * bindings/python/codegen/codegen.py:
7083 * bindings/python/codegen/definitions.py:
7084 * bindings/python/codegen/defsparser.py:
7085 * bindings/python/codegen/docextract.py:
7086 * bindings/python/codegen/docgen.py:
7087 * bindings/python/codegen/fileprefix.override:
7088 * bindings/python/codegen/fileprefixmodule.c:
7089 * bindings/python/codegen/h2def.py:
7090 * bindings/python/codegen/mergedefs.py:
7091 * bindings/python/codegen/mkskel.py:
7092 * bindings/python/codegen/override.py:
7093 * bindings/python/codegen/reversewrapper.py:
7094 * bindings/python/codegen/scmexpr.py:
7095 * bindings/python/rtspserver-types.defs:
7096 * bindings/python/rtspserver.defs:
7097 * bindings/python/rtspserver.override:
7098 * bindings/python/rtspservermodule.c:
7100 Add python bindings.
7102 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7104 * bindings/Makefile.am:
7106 Don't go into python dir when requirements for python bindings are missing
7108 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7110 * bindings/Makefile.am:
7111 * bindings/vala/Makefile.am:
7113 Install Vala bindings if vala is available
7115 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7117 * bindings/vala/gst-rtsp-server-0.10.deps:
7118 * bindings/vala/gst-rtsp-server-0.10.vapi:
7119 * bindings/vala/gst-rtsp-server.vapi:
7120 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
7121 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
7122 * bindings/vala/packages/gst-rtsp-server-0.10.files:
7123 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7124 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7125 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
7126 * bindings/vala/packages/gst-rtsp-server.deps:
7127 * bindings/vala/packages/gst-rtsp-server.excludes:
7128 * bindings/vala/packages/gst-rtsp-server.files:
7129 * bindings/vala/packages/gst-rtsp-server.gi:
7130 * bindings/vala/packages/gst-rtsp-server.metadata:
7131 * bindings/vala/packages/gst-rtsp-server.namespace:
7132 Regenerated Vala bindings
7134 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7136 * bindings/vala/gst-rtsp-server.vapi:
7137 * bindings/vala/packages/gst-rtsp-server.metadata:
7138 Fixed typo in included headers for vala bindings
7140 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7144 * pkgconfig/Makefile.am:
7145 * pkgconfig/gst-rtsp-server.pc.in:
7146 Added pkgconfig file
7148 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
7150 * bindings/vala/gst-rtsp-server.vapi:
7151 * bindings/vala/packages/gst-rtsp-server.excludes:
7152 * bindings/vala/packages/gst-rtsp-server.gi:
7153 * bindings/vala/packages/gst-rtsp-server.metadata:
7154 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
7156 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
7158 * bindings/vala/gst-rtsp-server.vapi:
7159 * bindings/vala/packages/gst-rtsp-server.deps:
7160 * bindings/vala/packages/gst-rtsp-server.files:
7161 * bindings/vala/packages/gst-rtsp-server.gi:
7162 * bindings/vala/packages/gst-rtsp-server.metadata:
7163 * bindings/vala/packages/gst-rtsp-server.namespace:
7166 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
7168 * gst/rtsp-server/rtsp-session.c:
7169 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
7171 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
7173 * examples/Makefile.am:
7174 * gst/rtsp-server/Makefile.am:
7175 Put GStreamer version in library name
7177 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7179 * examples/Makefile.am:
7180 * gst/rtsp-server/Makefile.am:
7181 Fix some issues to pass distcheck
7183 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7185 * gst/rtsp-server/rtsp-server.c:
7186 Added port property to GstRTSPServer class.
7188 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7193 * examples/Makefile.am:
7196 * gst/rtsp-server/Makefile.am:
7197 * gst/rtsp-server/rtsp-client.c:
7198 * gst/rtsp-server/rtsp-client.h:
7199 * gst/rtsp-server/rtsp-media.c:
7200 * gst/rtsp-server/rtsp-media.h:
7201 * gst/rtsp-server/rtsp-server.c:
7202 * gst/rtsp-server/rtsp-server.h:
7203 * gst/rtsp-server/rtsp-session-pool.c:
7204 * gst/rtsp-server/rtsp-session-pool.h:
7205 * gst/rtsp-server/rtsp-session.c:
7206 * gst/rtsp-server/rtsp-session.h:
7209 * src/rtsp-client.c:
7210 * src/rtsp-client.h:
7213 * src/rtsp-server.c:
7214 * src/rtsp-server.h:
7215 * src/rtsp-session-pool.c:
7216 * src/rtsp-session-pool.h:
7217 * src/rtsp-session.c:
7218 * src/rtsp-session.h:
7219 Split in library and example program
7221 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
7223 * src/rtsp-client.h:
7224 Removed obsolete variable
7226 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
7228 * src/rtsp-client.c:
7229 * src/rtsp-client.h:
7230 Removed pipeline variable GstRTSPClient, because it's only used in one function
7232 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7235 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
7237 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
7239 * src/rtsp-session.c:
7240 Initialize some more vars.
7242 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
7244 * src/rtsp-session.c:
7245 Initialize variable to avoid compiler warning.
7247 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
7250 Add a reasonable generic .gitignore