3 2014-05-21 Sebastian Dröge <slomo@coaxion.net>
8 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
11 Automatic update of common submodule
12 From 211fa5f to 1f5d3c3
14 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
16 * gst/rtsp-server/rtsp-client.c:
17 client: store TCP ports in transport
18 Store the TCP ports in the transport when we are doing RTSP over TCP.
19 This way, we can easily get to the ports from the transport.
20 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
22 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
24 * gst/rtsp-server/rtsp-stream.c:
25 stream: add signals for new RTP/RTCP encoders
26 New signals to allow the user to configure the dynamically created
28 https://bugzilla.gnome.org/show_bug.cgi?id=730228
30 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
32 * gst/rtsp-server/rtsp-media.c:
33 * gst/rtsp-server/rtsp-media.h:
34 media: Make suspend()/unsuspend() virtual
35 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
37 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
39 * gst/rtsp-server/rtsp-client.c:
40 client: fix send-message signal marshaller
41 Use generic marshalling for the send-message signal. It has
42 two POINTER arguments, not just one.
43 https://bugzilla.gnome.org/show_bug.cgi?id=729900
45 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
47 * tests/check/gst/media.c:
48 tests: add and remove pads only once
49 In this test we simulate a dynamic pad by watching the caps event.
50 Because of renegotiation in the base payloader now, this caps is sent
51 multiple times but we can only deal with 1 invocation, use a variable to
52 only 'add and remove' the pad once.
54 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
56 * tests/check/gst/rtspserver.c:
57 tests: add unit test for correct handling of Require headers
58 https://bugzilla.gnome.org/show_bug.cgi?id=729426
60 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
62 * gst/rtsp-server/rtsp-client.c:
63 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
64 Servers must handle Require headers and must report a failure
65 if they don't handle any of the Required options, see RFC 2326,
66 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
67 https://bugzilla.gnome.org/show_bug.cgi?id=729426
69 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
76 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
82 * gst-rtsp-server.doap:
85 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
88 Automatic update of common submodule
89 From bcb1518 to 211fa5f
91 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
96 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
98 * tests/check/gst/sessionmedia.c:
99 tests: fix memory leak in sessionmedia unit test
101 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
103 * gst/rtsp-server/rtsp-client.c:
104 client: emit a signal before sending a message
105 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
107 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
109 * gst/rtsp-server/rtsp-client.c:
110 client: pass context to send_message
111 Pass the current context to send_message, we will need it later.
113 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
115 * gst/rtsp-server/rtsp-client.c:
116 client: fix typo in comment
118 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
120 * gst/rtsp-server/rtsp-media.c:
121 media: Do not stop thread twice if default_prepare() fails
123 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
125 * gst/rtsp-server/rtsp-client.c:
126 client: set the watch to flushing before going to NULL
127 First set the watch to flushing so that we unblock any current and
128 future attempt to send data on the watch, Then set the pipeline to
130 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
132 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
134 * gst/rtsp-server/rtsp-session-pool.c:
135 * tests/check/gst/sessionpool.c:
136 rtsp-session-pool: Fixes annotation
137 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
138 in the sessionpool test.
139 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
141 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
143 * gst/rtsp-server/rtsp-media.c:
144 * gst/rtsp-server/rtsp-media.h:
145 media: make media_prepare virtual
146 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
148 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
150 * gst/rtsp-server/rtsp-media.c:
151 * tests/check/gst/media.c:
152 media: stop the thread in more error cases
154 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
156 * gst/rtsp-server/rtsp-media.c:
157 * tests/check/gst/media.c:
158 media: allow NULL as the thread
159 Use the default context whan passing a NULL thread.
161 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
163 * gst/rtsp-server/rtsp-client.c:
164 rtsp-client: indent cleanup
165 Coverity was moaning about unreachable code, and I think it was just
166 confused by { being before the label. We'll see if it pops up again.
169 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
171 * gst/rtsp-server/rtsp-client.c:
172 * gst/rtsp-server/rtsp-media.c:
173 client: Add drop-backlog property
174 When we have too many messages queued for a client (currently hardcoded
175 to 100) we overflow and drop the messages. Add a drop-backlog property
176 to control this behaviour. Setting this property to FALSE will retry
177 to send the messages to the client by waiting for more room in the
179 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
181 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
183 * gst/rtsp-server/rtsp-client.c:
184 client: support for POST before GET when setting up a tunnel
186 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
188 * gst/rtsp-server/rtsp-client.c:
189 client: remove watch of the second client after http tunnel setup
190 The second client will be freed after the HTTP tunnel has been set up.
191 Make sure it's RTSP watch is never dispatched again.
192 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
194 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
196 * gst/rtsp-server/rtsp-media.c:
197 * tests/check/gst/media.c:
198 media: Make media_prepare() fail if port allocation fails
199 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
201 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
203 * tests/check/gst/media.c:
204 media test: cleanup the thread pool in tests
206 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
208 * gst/rtsp-server/rtsp-media.c:
209 * tests/check/gst/media.c:
210 rtsp-media: Unblock blocked streams in unprepare
211 The streams will be blocked when a live media is prepared.
212 The streams should be unblocked in gst_rtsp_media_unprepare.
213 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
215 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
217 * gst/rtsp-server/rtsp-media.c:
218 media: release the state lock when going to NULL
219 Set our state to UNPREPARING and release the state-lock before
220 setting the pipeline to the NULL state. This way, any pad-added
221 callback will be able to take the state-lock and check that we are now
222 unpreparing instead of deadlocking.
223 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
225 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
227 * gst/rtsp-server/rtsp-media.c:
228 media: protect status with lock
229 Make sure we only update the status with the lock.
231 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
233 * gst/rtsp-server/rtsp-client.c:
234 * gst/rtsp-server/rtsp-sdp.c:
235 rtsp: update for MIKEY API changes
237 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
239 * gst/rtsp-server/rtsp-client.c:
240 client: parse the mikey response from the client
241 Parse the mikey response from the client and update the policy for
244 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
246 * gst/rtsp-server/rtsp-stream.c:
247 * gst/rtsp-server/rtsp-stream.h:
248 stream: add method to set crypto info
249 Make a method to configure the crypto information of a stream.
250 Set udpsrc in READY instead of PAUSED so that we can configure caps
253 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
255 * gst/rtsp-server/rtsp-client.c:
256 client: cleanup error paths
258 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
260 * gst/rtsp-server/rtsp-media.c:
263 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
265 * examples/test-video.c:
266 test: enable SRTP only on RTSPS
267 We only want to enable SRTP when doing rtsp over TLS so that we can
268 exchange the keys in a secure way.
270 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
272 * examples/test-video.c:
273 test: print an error on failure
275 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
278 * examples/test-video.c:
279 * gst/rtsp-server/rtsp-sdp.c:
280 * gst/rtsp-server/rtsp-stream.c:
281 * tests/check/Makefile.am:
282 stream: add SRTP support
283 Install srtp encoder and decoder elements in rtpbin
286 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
288 * tests/check/Makefile.am:
289 * tests/check/gst/sessionpool.c:
290 tests: Add unit tests for sessionpool
291 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
293 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
295 * tests/check/gst/threadpool.c:
296 tests: Improve code coverage of rtsp-threadpool tests
297 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
299 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
301 * tests/check/gst/sessionmedia.c:
302 tests: Improve code coverage for rtsp-session-media
303 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
305 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
307 gobject-introspection: Add annotations to support language bindings
308 In addition a few cosmetic changes:
309 * Adjust the order of arguments
310 * Fix typo: occured -> occurred
311 * Fix indentation after Return:-clauses
312 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
314 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
316 * gst/rtsp-server/rtsp-stream.c:
317 rtsp-stream: Don't mix IPv4 and IPv6 addresses
318 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
320 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
322 * gst/rtsp-server/rtsp-stream.c:
323 stream: take caps after the session manager
324 Take the caps for the SDP after they leave the rtpbin so that we can
325 also get the properties added by rtpbin elements.
327 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
329 * gst/rtsp-server/rtsp-stream.c:
330 stream: release lock while pushing out packets
331 Keep a cache of the transports and use this to iterate the transport
332 while pushing packets. This allows us to release the lock early.
333 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
335 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
337 * gst/rtsp-server/rtsp-client.c:
338 * gst/rtsp-server/rtsp-client.h:
339 rtsp-client: vmethod for modifying tunnel GET response
340 Add a vmethod tunnel_http_response where the response to the HTTP GET
341 for tunneled connections can be modified.
342 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
344 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
346 * gst/rtsp-server/rtsp-sdp.c:
347 sdp: make 1 media line per profile
348 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
349 line in the SDP for each profile. The client is then supposed to pick
350 one of the profiles in the SETUP request. Because the m= lines have the
351 same pt, the client also knows that only 1 option is possible.
353 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
355 * gst/rtsp-server/rtsp-media-factory.c:
356 * gst/rtsp-server/rtsp-media-factory.h:
357 * gst/rtsp-server/rtsp-media.c:
358 factory: add profile property and pass to media and streams
360 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
362 * examples/test-multicast.c:
363 * gst/rtsp-server/rtsp-sdp.c:
364 sdp: pass multicast connection for multicast-only stream
365 Pass the multicast address of the stream in the connection info in the
366 SDP so that clients try a multicast connection first.
367 Only allow multicast connections in the test-multicast example. Also
368 increase the TTL a little.
370 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
373 .gitignore: Ignore gcov intermediate files
374 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
376 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
378 * gst/rtsp-server/rtsp-stream.c:
379 stream: release some locks in error cases
381 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
383 docs: Enable and fix gtk-doc warnings
384 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
385 * addresspool/mediafactory: Add missing annotation colon
386 * stream: Annotate return value
387 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
389 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
392 Automatic update of common submodule
393 From fe1672e to bcb1518
395 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
398 Automatic update of common submodule
399 From 1a07da9 to fe1672e
401 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
403 * examples/Makefile.am:
404 examples: use LDADD for libs instead of LDFLAGS
406 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
409 configure: make sure releases are in .doap file
411 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
413 * examples/test-cgroups.c:
414 examples: test-cgroups: don't put code with side effects into g_assert()
415 The g_assert() might get compiled out with the right
416 compiler/preprocessor flags.
418 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
420 * examples/.gitignore:
421 examples: add cgroup test binary to .gitignore
423 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
425 * examples/test-cgroups.c:
426 examples: fix cgroup test build
427 Fixes build failure caused by compiler warning:
428 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
430 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
433 .gitignore: ignore temp files created in the course of 'make check'
435 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
437 * gst/rtsp-server/rtsp-media.c:
438 rtsp-media: don't loose frames handling new PLAY request
439 If client supplied a range check if the range specifies the start point.
440 If not, then do an accurate seek to the current position. If a start
441 point was specified do do a key unit seek to make sure the streaming
442 starts with decodeable frames.
443 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
445 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
447 * gst/rtsp-server/rtsp-media.c:
448 Revert "media: only flush when setting a new start position"
449 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
450 We need to do the flush in all cases, demuxer block currently for
453 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
455 * gst/rtsp-server/rtsp-media.c:
456 media: only flush when setting a new start position
457 Only flush the pipeline when we change the start position with
459 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
461 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
463 * gst/rtsp-server/rtsp-stream.c:
464 stream: set ttl-mc before adding the socket
465 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
466 never be set on socket.
467 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
469 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
471 * gst/rtsp-server/rtsp-media.c:
472 media: stop thread if media is already prepared
473 in gst_rtsp_media_prepare() the thread is not used if media is already
474 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
476 https://bugzilla.gnome.org/show_bug.cgi?id=724182
478 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
481 build: Ship gst-rtsp-server.doap file
483 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
485 * tests/check/gst/rtspserver.c:
486 tests: Fix another compiler warning with gcc
488 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
490 * gst/rtsp-server/rtsp-client.c:
491 * gst/rtsp-server/rtsp-mount-points.c:
492 * gst/rtsp-server/rtsp-stream.c:
493 * tests/check/gst/client.c:
494 rtsp-server: Fix lots of compiler warnings with clang
496 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
499 * gst-rtsp-server.doap:
501 configure: Synchronise with the configure scripts of the other modules
503 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
506 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
508 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
510 * gst/rtsp-server/rtsp-media.c:
511 * gst/rtsp-server/rtsp-stream.c:
512 Revert "rtsp-server: support build against last stable release"
513 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
514 Let us require 1.2.3 now, which is going to be released in a few
517 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
519 * gst/rtsp-server/rtsp-session-media.c:
520 * gst/rtsp-server/rtsp-stream-transport.c:
521 session: improve RTP-Info
522 Ignore streams that can't generate RTP-Info instead of failing.
523 Don't return the empty string when all streams are unconfigured but
524 return NULL so that we don't generate and empty RTP-Info header.
525 Improve docs a little.
527 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
529 * gst/rtsp-server/rtsp-session-media.c:
530 Don't free rtpinfo GString when it is NULL
531 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
533 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
535 * gst/rtsp-server/rtsp-media.c:
536 media: only set keyframe flag when modifying start
537 Only set the keyframe flag when we modify the start position. The
538 keyframe flag should probably be ignored when no change is requested but
539 until we can claim this is all documented properly and all demuxer
540 implement this, avoid setting the flag.
541 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
543 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
545 * gst/rtsp-server/rtsp-thread-pool.c:
546 thread-pool: Unref source after mainloop has quit to avoid races in GLib
547 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
549 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
551 * gst/rtsp-server/rtsp-stream.c:
552 stream: handle NULL seqnum and rtptime arguments
554 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
556 * gst/rtsp-server/rtsp-thread-pool.c:
557 * tests/check/gst/threadpool.c:
558 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
559 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
561 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
563 * gst/rtsp-server/rtsp-stream.c:
564 stream: add fallback for missing stats property
565 Use a fallback when the payloader does not have a stats property
566 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
568 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
571 Automatic update of common submodule
572 From f7bc1c3 to 1a07da9
574 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
576 * gst/rtsp-server/rtsp-stream.c:
577 stream: don't leak stats structure
578 Don't leak the stats structure and deal with NULL stats.
580 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
582 * gst/rtsp-server/rtsp-stream.c:
583 stream: Get rtpinfo properties atomically from payloader
584 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
586 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
588 * gst/rtsp-server/rtsp-media.c:
589 media: refactor state change functions and signals
590 Make functions to set the target state and the pipeline state and emit
591 the signals from those functions.
593 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
595 * gst/rtsp-server/rtsp-media.c:
596 * gst/rtsp-server/rtsp-media.h:
597 media: add signal to notify of pending state changes
599 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
601 * gst/rtsp-server/rtsp-media.c:
602 * gst/rtsp-server/rtsp-stream.c:
603 rtsp-server: support build against last stable release
604 Until 1.2.3 is out with the new get_type function and we
607 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
609 * gst/rtsp-server/rtsp-stream.c:
610 stream: fix compilation
612 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
614 * gst/rtsp-server/rtsp-media.c:
615 * gst/rtsp-server/rtsp-media.h:
616 * gst/rtsp-server/rtsp-stream.c:
617 * gst/rtsp-server/rtsp-stream.h:
618 stream: add property to configure profiles
620 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
622 * gst/rtsp-server/rtsp-client.c:
623 client: let stream check supported transport
624 Delegate the check if a transport is allowed to the stream.
625 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
627 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
629 * gst/rtsp-server/rtsp-stream.c:
630 * gst/rtsp-server/rtsp-stream.h:
631 stream: add method to check supported transport
632 Add a method to check if a transport is supported
634 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
637 configure.ac: Only check for gstreamer-check, not check
638 We include check in gstreamer-check since quite some time now.
640 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
642 * gst/rtsp-server/rtsp-session-media.c:
643 * gst/rtsp-server/rtsp-stream-transport.c:
644 * gst/rtsp-server/rtsp-stream.c:
645 * gst/rtsp-server/rtsp-stream.h:
646 stream: return clock-rate from get_rtpinfo
647 And use it to correct the rtptime to the requested start-time.
648 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
650 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
652 * gst/rtsp-server/rtsp-session-media.c:
653 * gst/rtsp-server/rtsp-stream-transport.c:
654 * gst/rtsp-server/rtsp-stream-transport.h:
655 session-media: calculate start-time
657 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
659 * gst/rtsp-server/rtsp-stream-transport.c:
660 * gst/rtsp-server/rtsp-stream.c:
661 * gst/rtsp-server/rtsp-stream.h:
662 stream: also return the running-time
663 Return the running-time in the rtpinfo as well.
665 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
667 * gst/rtsp-server/rtsp-client.c:
668 * gst/rtsp-server/rtsp-session-media.c:
669 * gst/rtsp-server/rtsp-session-media.h:
670 * gst/rtsp-server/rtsp-stream-transport.c:
671 * gst/rtsp-server/rtsp-stream-transport.h:
672 session-media: let the session-media make the RTPInfo
673 Add method to create the RTPInfo for a stream-transport.
674 Add method to create the RTPInfo for all stream-transports in a
676 Use the session-media RTPInfo code in client. This allows us to refactor
677 another method to link the TCP callbacks.
679 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
681 mount-points: sort sequence before g_sequence_lookup
682 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
683 sort sequence if dirty, otherwise lookup will fail.
684 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
686 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
689 configure: rename package from gst-rtsp to gst-rtsp-server
690 To match git module name and avoid confusion with the
691 rtsp lib in gst-plugins-base and rtsp plugin in -good.
693 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
696 configure: bump core/base/good requirement to 1.2.0
697 Bump to released stable version and make implicit
698 requirements explicit.
700 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
705 Fix broken gettext setup which is not used anyway
707 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
710 Automatic update of common submodule
711 From dbedaa0 to d48bed3
713 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
715 * gst/rtsp-server/rtsp-client.c:
716 * gst/rtsp-server/rtsp-media.c:
717 * gst/rtsp-server/rtsp-media.h:
718 media: add setup_sdp vmethod
719 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
720 gst_rtsp_media_setup_sdp.
721 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
723 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
725 * gst/rtsp-server/rtsp-stream.c:
726 rtsp-stream: Check return value of sscanf
727 streamid is only valid if sscanf matched something.
729 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
731 * gst/rtsp-server/rtsp-client.c:
732 rtsp-client: Fix iteration
733 Wouldn't even enter the code block otherwise (i++ was used as the check
734 and not the postfix).
736 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
738 * gst/rtsp-server/rtsp-client.c:
739 * gst/rtsp-server/rtsp-client.h:
740 client: add vmethod to configure media and streams
741 Implement a vmethod that can be used to configure the media and the
742 streams based on the current context. Handle the blocksize handling in
744 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
746 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
749 Make git ignore more unit test binaries
751 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
753 * gst/rtsp-server/rtsp-address-pool.h:
754 * gst/rtsp-server/rtsp-auth.h:
755 * gst/rtsp-server/rtsp-client.h:
756 * gst/rtsp-server/rtsp-context.h:
757 * gst/rtsp-server/rtsp-media-factory-uri.h:
758 * gst/rtsp-server/rtsp-media-factory.h:
759 * gst/rtsp-server/rtsp-media.h:
760 * gst/rtsp-server/rtsp-mount-points.h:
761 * gst/rtsp-server/rtsp-server.h:
762 * gst/rtsp-server/rtsp-session-media.h:
763 * gst/rtsp-server/rtsp-session-pool.h:
764 * gst/rtsp-server/rtsp-session.h:
765 * gst/rtsp-server/rtsp-stream-transport.h:
766 * gst/rtsp-server/rtsp-stream.h:
767 * gst/rtsp-server/rtsp-thread-pool.h:
768 * gst/rtsp-server/rtsp-token.h:
769 rtsp-server: add padding to many public structures
770 Not mini objects though, since they are not subclassable
771 anyway, nor kept on the stack or inlined in a structure.
773 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
775 media: add new create_rtpbin vmethod
776 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
777 https://bugzilla.gnome.org/show_bug.cgi?id=719734
779 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
781 * tests/check/gst/media.c:
782 tests: fix memory leak, free test's thread pool
783 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
785 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
787 * gst/rtsp-server/rtsp-stream-transport.c:
788 stream-transport: free url in finalize
790 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
792 * gst/rtsp-server/rtsp-media.c:
793 media: also do state change in suspended state
795 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
797 * gst/rtsp-server/rtsp-client.c:
798 * gst/rtsp-server/rtsp-media.c:
799 media: also handle prepare and range in suspended state
800 When we are suspended, we are already prepared.
801 We can get the range in the suspended state.
803 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
805 * tests/check/Makefile.am:
806 * tests/check/gst/sessionmedia.c:
807 check: add test for uri in setup
808 Added unit tests for the new functionality in GstRTSPStreamTransport.
809 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
811 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
813 * gst/rtsp-server/rtsp-client.c:
814 client: store setup uri and use in PLAY response
815 Store the uri used when doing the setup and use that in the PLAY
817 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
819 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
821 * gst/rtsp-server/rtsp-stream-transport.c:
822 * gst/rtsp-server/rtsp-stream-transport.h:
823 stream-transport: add method to get/set url
825 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
827 * gst/rtsp-server/rtsp-client.c:
828 client: suspend after SDP and unsuspend before PLAYING
829 Based on patches by Ognyan Tonchev <ognyan@axis.com>
830 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
832 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
834 * gst/rtsp-server/rtsp-media-factory.c:
835 * gst/rtsp-server/rtsp-media-factory.h:
836 * gst/rtsp-server/rtsp-media.c:
837 * gst/rtsp-server/rtsp-media.h:
838 * gst/rtsp-server/rtsp-session-media.c:
839 * gst/rtsp-server/rtsp-session.c:
840 * tests/check/gst/media.c:
841 * tests/check/gst/mediafactory.c:
842 media: add suspend modes
843 Add support for different suspend modes. The stream is suspended right after
844 producing the SDP and after PAUSE. Different suspend modes are available that
845 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
846 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
847 state and RESET will bring the pipeline to the NULL state.
848 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
849 this means that the pipeline needs to be prerolled again.
850 Base on patches by Ognyan Tonchev <ognyan@axis.com>
851 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
853 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
855 * gst/rtsp-server/rtsp-media.c:
856 media: start live streams in blocked state
857 Start live streams in the blocked state and make them preroll using the
858 messages. This ensure that no data is played by the sink until we explicitly
859 unblock the stream right before going to PLAYING.
860 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
862 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
864 * gst/rtsp-server/rtsp-media.c:
865 media: refactor starting and waiting for preroll
866 Based on patches from Ognyan Tonchev <ognyan@axis.com>
867 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
869 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
871 * gst/rtsp-server/rtsp-stream.c:
872 * gst/rtsp-server/rtsp-stream.h:
873 stream: add API to block streams
874 Add an API to block on the streams and make it post a message.
875 Based on patch by Ognyan Tonchev <ognyan@axis.com>
876 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
878 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
880 * docs/libs/Makefile.am:
881 docs: Specify the override file
882 Even if it's empty (for now) it avoids make distcheck complaining
884 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
886 * gst/rtsp-server/rtsp-media.c:
887 media: move default implementations to where they are used
889 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
891 * gst/rtsp-server/rtsp-media.c:
892 media: take the right lock in gst_rtsp_media_set_pipeline_state()
893 We need to take the state_lock when calling this method.
895 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
897 * gst/rtsp-server/rtsp-media.c:
898 media: handle add-added on non-bins too
899 Handle dynamic payloaders that are not bins, as used in the unit-test.
901 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
903 * gst/rtsp-server/rtsp-media-factory.c:
904 * gst/rtsp-server/rtsp-media-factory.h:
905 * gst/rtsp-server/rtsp-media.c:
906 rtsp-media/-factory: Fix request pad name comments
907 These must be escaped for gtk-doc to parse the comments without warnings.
909 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
911 rtsp-media: remove transports if media is in error status
912 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
913 trying to change to GST_STATE_NULL and media is in error status, we
914 remove all transports.
915 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
917 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
919 * gst/rtsp-server/rtsp-media.c:
920 rtsp-media: use element metadata to find payloader
921 Use the element metadata to find the payloader instead of checking
923 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
925 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
927 rtsp-stream: add getter for payload type
928 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
929 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
930 element and create the stream with this one instead of the dynpay%d
932 https://bugzilla.gnome.org/show_bug.cgi?id=712396
934 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
936 * gst/rtsp-server/rtsp-client.c:
937 * gst/rtsp-server/rtsp-context.h:
938 * gst/rtsp-server/rtsp-media.c:
939 * gst/rtsp-server/rtsp-mount-points.c:
940 * gst/rtsp-server/rtsp-server.c:
941 * gst/rtsp-server/rtsp-token.c:
942 rtsp-*: Refer to NULL as a constant in comments
944 https://bugzilla.gnome.org/show_bug.cgi?id=714988
946 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
948 rtsp-*: Fix type name typos in comments
949 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
950 * rtsp-auth: Refer to part of constant name as text
951 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
952 * rtsp-session-media: Fix GstRTSPSessionMedia typo
953 * rtsp-stream: Fix typo when refering to GstBin
954 https://bugzilla.gnome.org/show_bug.cgi?id=714988
956 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
959 * docs/libs/gst-rtsp-server-docs.sgml:
960 * docs/libs/gst-rtsp-server-sections.txt:
961 docs: Improve documentation
962 * Include annotation-glossary to quiet gtk-doc
963 * Rename remaining ClientState -> Context
964 * Rename object hierarchy file
965 * Remove stale chapter references
966 * Add missing function and object references
967 * Include missing GstRTSPAddressPoolResult
968 https://bugzilla.gnome.org/show_bug.cgi?id=714988
970 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
972 * gst/rtsp-server/rtsp-client.c:
973 * gst/rtsp-server/rtsp-server.c:
974 * gst/rtsp-server/rtsp-session-pool.c:
975 * gst/rtsp-server/rtsp-session.c:
976 * gst/rtsp-server/rtsp-stream.c:
977 rtsp-server: sprinkle some allow-none annotations for g-i
979 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
981 * gst/rtsp-server/rtsp-stream.c:
982 * gst/rtsp-server/rtsp-stream.h:
983 stream: add method to filter transports
984 Add a method to safely iterate and collect the stream transports
985 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
987 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
989 * gst/rtsp-server/rtsp-client.c:
990 * gst/rtsp-server/rtsp-server.c:
991 * gst/rtsp-server/rtsp-session-pool.c:
992 * gst/rtsp-server/rtsp-session.c:
993 rtsp: allow NULL func in filters
994 Passing a null function make the filters return a list of
997 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
999 * gst/rtsp-server/rtsp-address-pool.c:
1000 * tests/check/gst/addresspool.c:
1001 address-pool: fix address increment
1002 Use a guint instead of guint8 to increment the address. It's still not
1003 completely correct because a guint might not be able to hold the complete
1004 address range, but that's an enhacement for later.
1005 Add unit test to test improved behaviour.
1006 https://bugzilla.gnome.org/show_bug.cgi?id=708237
1008 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
1010 * gst/rtsp-server/rtsp-client.c:
1011 * tests/check/gst/client.c:
1012 client: allow absolute path in requests
1013 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
1015 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
1017 * gst/rtsp-server/rtsp-client.c:
1018 * gst/rtsp-server/rtsp-client.h:
1019 client: make make_path_from_uri a vmethod
1021 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
1023 * docs/libs/gst-rtsp-server-sections.txt:
1024 * gst/rtsp-server/rtsp-stream.c:
1025 * gst/rtsp-server/rtsp-stream.h:
1026 * tests/check/Makefile.am:
1027 * tests/check/gst/stream.c:
1028 stream: Add functions to get rtp and rtcp sockets
1029 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
1031 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
1033 * gst/rtsp-server/rtsp-context.c:
1034 * gst/rtsp-server/rtsp-context.h:
1035 context: defing a GType for the context
1036 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
1038 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
1040 * gst/rtsp-server/Makefile.am:
1041 * gst/rtsp-server/rtsp-auth.c:
1042 * gst/rtsp-server/rtsp-context.c:
1043 * gst/rtsp-server/rtsp-media.c:
1044 * gst/rtsp-server/rtsp-mount-points.c:
1045 * gst/rtsp-server/rtsp-server.h:
1046 * gst/rtsp-server/rtsp-session-media.c:
1047 * gst/rtsp-server/rtsp-session.c:
1048 * gst/rtsp-server/rtsp-stream.c:
1049 Fixed several GIR warnings
1051 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
1053 * gst/rtsp-server/rtsp-auth.c:
1056 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1058 * tests/check/Makefile.am:
1059 * tests/check/gst/token.c:
1060 tests: Add unit tests for token
1061 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
1063 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1065 * gst/rtsp-server/rtsp-token.c:
1066 token: Validate args for gst_rtsp_token_is_allowed
1067 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
1069 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1071 * gst/rtsp-server/rtsp-token.c:
1072 token: Fix bug when creating empty token
1073 We always want to have a valid GstStructure in the token.
1074 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
1076 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
1078 * gst/rtsp-server/rtsp-thread-pool.c:
1079 thread-pool: avoid race in shutdown
1080 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
1081 don't actually stop the mainloop ever. Solve this race by adding an idle source
1082 to the mainloop that calls the _quit. This way we immediately exit the mainloop
1083 if quit was called before we started it.
1085 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1087 * tests/check/Makefile.am:
1088 * tests/check/gst/permissions.c:
1089 tests: Add unit tests for permissions
1090 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
1092 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1094 * tests/check/gst/mediafactory.c:
1095 tests: Test mediafactory permissions
1096 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
1098 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1100 * gst/rtsp-server/rtsp-permissions.c:
1101 permissions: Fix refcounting when adding/removing roles
1102 Previously a role that was removed was unreffed twice, and when
1103 replacing an existing role the replaced role was freed while still being
1104 referenced. Both bugs are now fixed.
1105 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
1107 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1109 * tests/check/gst/media.c:
1110 * tests/check/gst/mediafactory.c:
1111 * tests/check/gst/rtspserver.c:
1112 tests: Check gst_rtsp_url_parse return value
1113 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
1115 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
1118 Automatic update of common submodule
1119 From 865aa20 to dbedaa0
1121 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
1123 * gst/rtsp-server/rtsp-server.c:
1124 rtsp-server: Fix socket leak
1125 https://bugzilla.gnome.org/show_bug.cgi?id=710088
1127 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
1129 * gst/rtsp-server/rtsp-session-pool.c:
1130 rtsp-session-pool: Make sure session IDs are properly URI-escaped
1131 https://bugzilla.gnome.org/show_bug.cgi?id=643812
1133 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
1135 * examples/.gitignore:
1136 * examples/test-video.c:
1137 examples: fix compilation when WITH_AUTH is defined
1138 https://bugzilla.gnome.org/show_bug.cgi?id=710228
1140 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
1143 gitignore: Add new test binary
1145 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
1147 * tests/check/Makefile.am:
1148 * tests/check/gst/threadpool.c:
1149 thread-pool: Add unit test for the thread pools
1150 https://bugzilla.gnome.org/show_bug.cgi?id=710228
1152 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
1154 * gst/rtsp-server/rtsp-thread-pool.c:
1155 thread-pool: Fix thread leak when reusing threads
1156 https://bugzilla.gnome.org/show_bug.cgi?id=709730
1158 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
1160 * gst/rtsp-server/rtsp-server.c:
1161 * tests/check/gst/rtspserver.c:
1162 tests: fixed racy behavior in rtspserver tests
1163 https://bugzilla.gnome.org/show_bug.cgi?id=710078
1165 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1167 * tests/check/gst/addresspool.c:
1168 tests: Improve address pool unit tests
1169 Add a range with mixed IPV4 and IPV6 addresses to pool.
1170 Get an IPV4 address from an IPV6-only pool.
1171 Get an IPV6 address from an IPV4-only pool.
1172 Reserve a IPV6 address from an IPV4-only pool.
1173 Check for unicast addresses in multicast-only pool.
1174 Check for unicast addresses in uni-/multicast-mixed pool.
1175 https://bugzilla.gnome.org/show_bug.cgi?id=710128
1177 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1179 * gst/rtsp-server/rtsp-client.c:
1180 client: append query string in PAUSE/PLAY/TEARDOWN as well
1182 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
1184 * gst/rtsp-server/rtsp-client.c:
1185 client: Add query to control path
1186 If the SETUP url contains a query it must be appended to the control
1187 path so that it matches any already created stream in the media. The
1188 query will also be appended to the session media path.
1190 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1192 * gst/rtsp-server/rtsp-media.c:
1193 rtsp-media: remove old line
1195 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
1197 * gst/rtsp-server/rtsp-stream.c:
1198 stream: Correct control comparison
1199 https://bugzilla.gnome.org/show_bug.cgi?id=709176
1201 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1203 * gst/rtsp-server/rtsp-media.c:
1204 media: Check dynamically if the pipeline supports seeking
1205 We should not depend on whether or not the pipeline state change
1206 returned NO_PREROLL or not. A media could dynamically change its
1207 element and switch from seekable to non seekable so it's best to test
1208 the seekable nature of the pipeline dynamically when we try to do a seek.
1210 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1212 * gst/rtsp-server/rtsp-media.c:
1213 media: Return FALSE if seeking is not supported
1215 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1217 * gst/rtsp-server/rtsp-media.c:
1218 rtsp-media: don't seek accurate by default
1219 Accurate seeking is perhaps a little overkill in the most common situation and
1220 causes some formats (mp3) over slow media to seek extremely slowly.
1222 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
1224 * tests/check/gst/rtspserver.c:
1225 tests: fix unit test
1226 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
1228 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
1230 * gst/rtsp-server/rtsp-client.c:
1231 client: Reply 400 if media cannot be constructed
1232 Reply 400 Bad Request instead of 503 Service Unavailable if media
1233 cannot be constructed in SETUP.
1234 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
1236 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
1238 * gst/rtsp-server/rtsp-client.c:
1239 client: Send setup reply once only
1240 If find_media() failed in handle_setup_request() two replies was sent.
1241 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
1243 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
1246 Automatic update of common submodule
1247 From 6b03ba7 to 865aa20
1249 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
1251 * gst/rtsp-server/rtsp-server.c:
1252 server: Emit client-connected signal earlier
1253 Emit client-connected before the client ref is given to a GSource,
1254 otherwise client-connected can be emitted after the client object has
1257 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
1259 * gst/rtsp-server/rtsp-address-pool.c:
1260 * gst/rtsp-server/rtsp-address-pool.h:
1261 * gst/rtsp-server/rtsp-stream.c:
1262 * tests/check/gst/addresspool.c:
1263 addresspool: return reason of failure
1264 Let gst_rtsp_address_pool_reserve_address() return the reason why
1265 the address could not be reserved.
1266 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
1268 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
1271 autogen.sh: Sync behaviour with other GStreamer modules
1272 Allows building from outside of tree amongst other things
1274 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
1277 Automatic update of common submodule
1278 From b613661 to 6b03ba7
1280 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
1283 Automatic update of common submodule
1284 From 74a6857 to b613661
1286 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
1289 Automatic update of common submodule
1290 From 01a7a46 to 74a6857
1292 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
1294 * gst/rtsp-server/rtsp-client.c:
1295 client: Do not read beyond end of path string
1296 If the setup was done without a control url, make sure we don't try to read the
1297 non-existing control string and crash.
1299 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1301 * gst/rtsp-server/rtsp-client.c:
1302 client: Fix RTPInfo header
1303 Refactor the method to make the content_base.
1304 Use the content-base and the control url to construct the RTPInfo
1307 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1309 * gst/rtsp-server/rtsp-client.c:
1310 client: map url to path only in describe
1311 Only map the request url to a path in the DESCRIBE method. The SDP then
1312 contains the base and control urls that should be used to SETUP/PAUSE/
1313 PLAY/TEARDOWN the media.
1315 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1317 * gst/rtsp-server/rtsp-client.c:
1318 Revert "client: map URL to path in requests"
1319 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
1320 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
1321 contains the base and control urls which are used in the SETUP, PLAY,
1322 PAUSE and TEARDOWN requests.
1324 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1326 * gst/rtsp-server/rtsp-client.c:
1327 client: map URL to path in requests
1329 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1331 * gst/rtsp-server/rtsp-client.c:
1332 * gst/rtsp-server/rtsp-mount-points.c:
1333 * gst/rtsp-server/rtsp-mount-points.h:
1334 mount-points: make vmethod to make path from uri
1335 Make a vmethod to transform an url into a path. The path is then used to lookup
1336 the factory. This makes it possible to also use other bits of the url, such as
1337 the query parameters, to locate the factory.
1339 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
1341 * gst/rtsp-server/rtsp-thread-pool.c:
1342 * gst/rtsp-server/rtsp-thread-pool.h:
1343 thread-pool: Add cleanup to wait for the threadpool to finish
1344 Also fix race condition if two threads are asking for the first
1345 thread from the thread pool at once. This would case two internal
1346 GThreadPools to be created.
1347 https://bugzilla.gnome.org/show_bug.cgi?id=707753
1349 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
1351 * gst/rtsp-server/rtsp-client.c:
1352 * tests/check/gst/client.c:
1353 client: free threadpool
1354 https://bugzilla.gnome.org/show_bug.cgi?id=707638
1356 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
1358 * tests/check/gst/mountpoints.c:
1359 mountpoints tests: unref matched factories
1360 https://bugzilla.gnome.org/show_bug.cgi?id=707638
1362 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
1364 * tests/check/gst/media.c:
1365 media tests: unref thread pool and caps
1366 https://bugzilla.gnome.org/show_bug.cgi?id=707638
1368 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
1370 * gst/rtsp-server/rtsp-auth.c:
1371 * gst/rtsp-server/rtsp-media-factory.c:
1372 * gst/rtsp-server/rtsp-media.c:
1373 auth, media, media-factory: unref permissions
1374 https://bugzilla.gnome.org/show_bug.cgi?id=707638
1376 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1378 * examples/Makefile.am:
1379 Makefile: add rule for appsrc example
1381 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1383 * examples/test-appsrc.c:
1384 tests: add appsrc example
1385 Add an example on how to use appsrc to feed the server pipeline with data.
1387 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
1389 * gst/rtsp-server/rtsp-client.c:
1390 rtsp-client: remove query part from content-base string
1391 Make sure that after the control url has been resolved, it's
1392 not a part of the query-string.
1393 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
1395 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1397 * gst/rtsp-server/rtsp-client.c:
1398 client: don't check url in response
1399 There is no url or method in the response to check
1401 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1403 * gst/rtsp-server/rtsp-client.c:
1404 * gst/rtsp-server/rtsp-client.h:
1405 Add handle-response signal for when we receive a GET_PARAMETER response
1407 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1409 * gst/rtsp-server/rtsp-server.c:
1410 Fix gst_rtsp_server_client_filter, using wrong variable type
1412 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
1414 * gst/rtsp-server/rtsp-media-factory-uri.c:
1415 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
1416 For AAC we need to check for framed=true instead of parsed=true.
1417 https://bugzilla.gnome.org/show_bug.cgi?id=701384
1419 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1421 * gst/rtsp-server/rtsp-stream.c:
1422 stream: optimize pipeline for protocols
1423 When TCP is not an allowed protocol for the stream, avoid creating the
1424 appsrc/appsink/queue and tee elements.
1426 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1428 * gst/rtsp-server/rtsp-media.c:
1429 media: set protocols on streams
1431 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1433 * gst/rtsp-server/rtsp-client.c:
1434 client: use protocols supported by stream
1436 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1438 * gst/rtsp-server/rtsp-media-factory.c:
1439 * gst/rtsp-server/rtsp-media.c:
1440 * gst/rtsp-server/rtsp-stream.c:
1441 media-factory: allow all protocols
1443 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1445 * gst/rtsp-server/rtsp-media.c:
1446 media: configure protocols in new streams
1448 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1450 * gst/rtsp-server/rtsp-stream.c:
1451 * gst/rtsp-server/rtsp-stream.h:
1452 stream: add protocols property
1454 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1456 * gst/rtsp-server/rtsp-media.c:
1457 rtsp-media: send state in "new-state" signal
1458 https://bugzilla.gnome.org/show_bug.cgi?id=705110
1460 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
1463 build: add subdir-objects to AM_INIT_AUTOMAKE
1464 Fixes warnings with automake 1.14
1465 https://bugzilla.gnome.org/show_bug.cgi?id=705350
1467 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1469 * docs/libs/gst-rtsp-server-sections.txt:
1470 * gst/rtsp-server/rtsp-client.c:
1471 * gst/rtsp-server/rtsp-server.c:
1472 * gst/rtsp-server/rtsp-server.h:
1473 server: add method to iterate clients of server
1475 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1477 * gst/rtsp-server/rtsp-media.c:
1478 * gst/rtsp-server/rtsp-media.h:
1479 Add vmethod for rtsp-media subclass to access rtpbin
1481 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1483 * gst/rtsp-server/rtsp-client.h:
1484 small documentation fix
1486 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1488 * gst/rtsp-server/rtsp-client.c:
1489 Do not take range header if range is invalid
1491 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1493 * docs/libs/gst-rtsp-server-sections.txt:
1494 * gst/rtsp-server/rtsp-media.c:
1495 media: add docs for new method
1497 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1499 * gst/rtsp-server/rtsp-media.c:
1500 * gst/rtsp-server/rtsp-media.h:
1501 Add API to rtsp-media set the pipeline's state
1503 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1505 * gst/rtsp-server/rtsp-media.c:
1506 Update current position/duration when gst_rtsp_media_get_range_string is called
1508 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1510 * examples/test-cgroups.c:
1511 tests: add some more docs
1513 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1515 * examples/test-cgroups.c:
1516 * gst/rtsp-server/Makefile.am:
1517 * gst/rtsp-server/rtsp-auth.c:
1518 * gst/rtsp-server/rtsp-auth.h:
1519 * gst/rtsp-server/rtsp-client.c:
1520 * gst/rtsp-server/rtsp-client.h:
1521 * gst/rtsp-server/rtsp-context.c:
1522 * gst/rtsp-server/rtsp-context.h:
1523 * gst/rtsp-server/rtsp-params.c:
1524 * gst/rtsp-server/rtsp-params.h:
1525 * gst/rtsp-server/rtsp-server.c:
1526 * gst/rtsp-server/rtsp-thread-pool.c:
1527 * gst/rtsp-server/rtsp-thread-pool.h:
1528 * tests/check/gst/client.c:
1529 ClientState -> Context
1530 Rename the clientstate to context and put the code in a separate file.
1532 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1534 * examples/test-auth.c:
1535 * gst/rtsp-server/rtsp-auth.c:
1536 * gst/rtsp-server/rtsp-auth.h:
1537 auth: add support for default token
1538 The default token is used when the user is not authenticated and can be used to
1539 give minimal permissions.
1541 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1543 * examples/test-auth.c:
1544 * gst/rtsp-server/rtsp-auth.c:
1545 auth: use defines when possible
1547 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1549 * gst/rtsp-server/rtsp-address-pool.c:
1550 address-pool: improve docs
1552 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1554 * gst/rtsp-server/rtsp-permissions.c:
1555 permissions: add the role to the copy
1557 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
1559 * gst/rtsp-server/rtsp-permissions.c:
1560 permissions: Also copy the roles
1562 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
1564 * gst/rtsp-server/rtsp-permissions.c:
1565 permissions: Make it build
1567 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1569 * gst/rtsp-server/rtsp-address-pool.h:
1572 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1574 * docs/libs/gst-rtsp-server-sections.txt:
1575 * gst/rtsp-server/rtsp-auth.c:
1576 * gst/rtsp-server/rtsp-auth.h:
1577 * gst/rtsp-server/rtsp-media.c:
1578 * gst/rtsp-server/rtsp-session-media.c:
1579 * gst/rtsp-server/rtsp-stream-transport.c:
1580 * gst/rtsp-server/rtsp-stream-transport.h:
1581 * gst/rtsp-server/rtsp-stream.c:
1582 * tests/check/gst/client.c:
1585 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1587 * docs/libs/gst-rtsp-server-sections.txt:
1588 * gst/rtsp-server/rtsp-address-pool.c:
1589 * gst/rtsp-server/rtsp-address-pool.h:
1590 * tests/check/gst/addresspool.c:
1591 * tests/check/gst/rtspserver.c:
1592 address-pool: cleanups
1593 Remove redundant method, improve docs.
1595 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1597 * docs/libs/gst-rtsp-server-sections.txt:
1598 * gst/rtsp-server/rtsp-auth.h:
1599 * gst/rtsp-server/rtsp-permissions.c:
1600 * gst/rtsp-server/rtsp-permissions.h:
1601 * gst/rtsp-server/rtsp-token.c:
1604 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1606 * gst/rtsp-server/rtsp-permissions.c:
1607 permissions: implement _remove_role
1609 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1611 * gst/rtsp-server/rtsp-permissions.c:
1612 permissions: update docs
1614 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1616 * tests/check/gst/client.c:
1617 tests: simplify tests
1618 Client settings are now disabled by default so we don't need an auth
1619 module to disable them.
1621 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1623 * gst/rtsp-server/rtsp-auth.c:
1624 auth: add default authorizations
1625 When no auth module is specified, use our table of defaults to look up the
1626 default value of the check instead of always allowing everything. This was
1627 we can disallow client settings by default.
1629 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1632 README: update readme
1634 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1636 * gst/rtsp-server/rtsp-thread-pool.c:
1637 * gst/rtsp-server/rtsp-thread-pool.h:
1638 thread-pool: add more docs
1640 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1642 * gst/rtsp-server/rtsp-thread-pool.c:
1643 * gst/rtsp-server/rtsp-thread-pool.h:
1644 thread-pool: fix race in thread reuse
1645 If we try to reuse a thread right after we made it stop, we end up using a
1646 stopped thread. Catch this case and only reuse threads that are not stopping.
1648 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1650 * gst/rtsp-server/rtsp-server.c:
1651 server: add small debug
1653 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1655 * tests/check/gst/client.c:
1657 Add some permissions to media so we can use the auth and enable
1660 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1662 * gst/rtsp-server/rtsp-client.c:
1663 client: support pushed context in handle_request
1664 If we already have a pushed state, reuse it and add our own things. This makes
1665 it easier to write tests.
1667 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1669 * gst/rtsp-server/rtsp-auth.c:
1670 auth: don't auth on methods
1671 Don't authorize on methods anymore but on the resources that we
1672 try to access, this is more flexible.
1673 Move the authorization checks to where they are needed and let the
1674 check return the response on error.
1676 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1678 * gst/rtsp-server/rtsp-mount-points.c:
1679 mount-points: add some debug
1681 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1683 * tests/check/gst/client.c:
1684 tests: almost fix test
1686 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1688 * gst/rtsp-server/rtsp-auth.c:
1689 * gst/rtsp-server/rtsp-auth.h:
1690 * gst/rtsp-server/rtsp-client.c:
1691 * gst/rtsp-server/rtsp-client.h:
1692 * gst/rtsp-server/rtsp-server.c:
1693 * gst/rtsp-server/rtsp-server.h:
1694 auth: let the auth module check client_settings
1695 Let the auth module decide if client settings are allowed for the
1698 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1700 * gst/rtsp-server/rtsp-token.c:
1701 * gst/rtsp-server/rtsp-token.h:
1702 token: add method to check boolean permission
1704 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1706 * examples/test-auth.c:
1707 * examples/test-cgroups.c:
1708 * gst/rtsp-server/rtsp-token.c:
1709 * gst/rtsp-server/rtsp-token.h:
1710 token: simplify token constructor
1711 Use variable arguments to make easier API.
1713 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1715 * examples/test-auth.c:
1716 * examples/test-cgroups.c:
1717 * gst/rtsp-server/rtsp-media-factory.c:
1718 * gst/rtsp-server/rtsp-media-factory.h:
1719 media-factory: add convenience API for factory
1721 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1723 * examples/test-auth.c:
1724 * examples/test-cgroups.c:
1725 * gst/rtsp-server/rtsp-permissions.c:
1726 * gst/rtsp-server/rtsp-permissions.h:
1727 permissions: simplify API a little
1728 Avoid passing GstStructure in the add_role method, use varargs instead
1729 to construct the structure behind the scenes. We can then also use the
1730 structure name as the role and simplify some more logic.
1732 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1734 * gst/rtsp-server/rtsp-auth.c:
1737 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1739 * gst/rtsp-server/rtsp-auth.c:
1740 * gst/rtsp-server/rtsp-auth.h:
1741 * gst/rtsp-server/rtsp-client.c:
1742 auth: handle unauthorized response
1743 Move handling of the unauthorized response to the auth module, it can add
1744 the appropriate headers to request authorization for the required method
1745 much better than the client.
1747 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1749 * gst/rtsp-server/rtsp-client.c:
1750 * gst/rtsp-server/rtsp-client.h:
1751 client: allow for sending any message, not only requests
1752 Change the _send_request() method to _send_message() so that we
1753 can both send requests and replies.
1755 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1757 * docs/libs/gst-rtsp-server-sections.txt:
1758 * gst/rtsp-server/rtsp-server.h:
1761 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1763 * examples/test-video.c:
1764 * gst/rtsp-server/rtsp-auth.c:
1765 * gst/rtsp-server/rtsp-auth.h:
1766 * gst/rtsp-server/rtsp-server.c:
1767 * gst/rtsp-server/rtsp-server.h:
1768 auth: move TLS handling to auth module
1769 Remove the TLS settings on the server and move it to the auth module because
1770 that is where security related bits go.
1772 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1774 * gst/rtsp-server/rtsp-client.c:
1775 * gst/rtsp-server/rtsp-client.h:
1776 client: add state push/pop
1778 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1780 * gst/rtsp-server/rtsp-client.c:
1781 * gst/rtsp-server/rtsp-client.h:
1782 client: add connection to state
1784 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1786 * gst/rtsp-server/rtsp-mount-points.c:
1787 mount-points: fix debug
1789 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1791 * tests/check/gst/media.c:
1792 tests: fix media test
1794 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1796 * gst/rtsp-server/rtsp-thread-pool.c:
1797 thread-pool: we don't require a state
1799 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1801 * gst/rtsp-server/rtsp-server.c:
1802 server: let context ref the server
1803 So that we don't risk losing the server object early anc crash.
1805 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1807 * tests/check/gst/client.c:
1808 tests: fix client test
1810 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1813 * docs/libs/gst-rtsp-server-docs.sgml:
1814 * docs/libs/gst-rtsp-server-sections.txt:
1815 * gst/rtsp-server/rtsp-address-pool.c:
1816 * gst/rtsp-server/rtsp-auth.c:
1817 * gst/rtsp-server/rtsp-client.c:
1818 * gst/rtsp-server/rtsp-client.h:
1819 * gst/rtsp-server/rtsp-media-factory-uri.c:
1820 * gst/rtsp-server/rtsp-media-factory.c:
1821 * gst/rtsp-server/rtsp-media-factory.h:
1822 * gst/rtsp-server/rtsp-media.c:
1823 * gst/rtsp-server/rtsp-mount-points.c:
1824 * gst/rtsp-server/rtsp-params.c:
1825 * gst/rtsp-server/rtsp-permissions.c:
1826 * gst/rtsp-server/rtsp-sdp.c:
1827 * gst/rtsp-server/rtsp-server.c:
1828 * gst/rtsp-server/rtsp-server.h:
1829 * gst/rtsp-server/rtsp-session-media.c:
1830 * gst/rtsp-server/rtsp-session-pool.c:
1831 * gst/rtsp-server/rtsp-session.c:
1832 * gst/rtsp-server/rtsp-stream-transport.c:
1833 * gst/rtsp-server/rtsp-stream.c:
1834 * gst/rtsp-server/rtsp-thread-pool.c:
1835 * gst/rtsp-server/rtsp-token.c:
1838 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1840 * gst/rtsp-server/rtsp-session-pool.c:
1841 * gst/rtsp-server/rtsp-session-pool.h:
1842 session-pool: make vmethod to create a session
1843 Make a vmethod to create a sessions so that subclasses can create
1844 custom session objects
1846 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1848 * gst/rtsp-server/rtsp-auth.c:
1849 * gst/rtsp-server/rtsp-media-factory.h:
1850 * gst/rtsp-server/rtsp-media.h:
1851 * gst/rtsp-server/rtsp-mount-points.h:
1852 * gst/rtsp-server/rtsp-session-pool.h:
1853 * gst/rtsp-server/rtsp-stream.h:
1856 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1858 * docs/libs/gst-rtsp-server-docs.sgml:
1859 * docs/libs/gst-rtsp-server-sections.txt:
1860 * gst/rtsp-server/rtsp-address-pool.c:
1861 * gst/rtsp-server/rtsp-address-pool.h:
1862 * gst/rtsp-server/rtsp-auth.c:
1863 * gst/rtsp-server/rtsp-client.h:
1864 * gst/rtsp-server/rtsp-media-factory.h:
1865 * gst/rtsp-server/rtsp-media.c:
1866 * gst/rtsp-server/rtsp-media.h:
1867 * gst/rtsp-server/rtsp-permissions.c:
1868 * gst/rtsp-server/rtsp-permissions.h:
1869 * gst/rtsp-server/rtsp-server.h:
1870 * gst/rtsp-server/rtsp-session-media.c:
1871 * gst/rtsp-server/rtsp-session-media.h:
1872 * gst/rtsp-server/rtsp-session-pool.h:
1873 * gst/rtsp-server/rtsp-session.h:
1874 * gst/rtsp-server/rtsp-stream-transport.h:
1875 * gst/rtsp-server/rtsp-stream.c:
1876 * gst/rtsp-server/rtsp-thread-pool.h:
1879 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1882 * examples/Makefile.am:
1883 configure: compile cgroup example conditionally
1884 Only compile the cgroup example when we have libcgroup
1886 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1889 * examples/Makefile.am:
1890 * examples/test-cgroups.c:
1891 examples: add cgroups example
1893 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1895 * tests/check/gst/rtspserver.c:
1896 tests: fix compilation
1898 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1900 * gst/rtsp-server/rtsp-thread-pool.c:
1901 thread-pool: fix vmethod invocation
1903 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1905 * gst/rtsp-server/rtsp-thread-pool.c:
1906 * gst/rtsp-server/rtsp-thread-pool.h:
1907 thread-pool: store thread type in thread
1909 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1911 * gst/rtsp-server/rtsp-client.c:
1912 client: pass thread from pool to media _prepare
1913 Get a thread from the configured threadpool and pass it to the prepare method of
1916 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1918 * gst/rtsp-server/rtsp-media.c:
1919 * gst/rtsp-server/rtsp-media.h:
1920 media: Accept a thread in _prepare
1921 Remove out own threadpool handling and use the provided thread and
1922 maincontext for the bus messages and the state changes.
1924 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1926 * gst/rtsp-server/rtsp-server.c:
1927 server: configure client thread pool
1929 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1931 * gst/rtsp-server/rtsp-client.c:
1932 * gst/rtsp-server/rtsp-client.h:
1933 client: add method to configure thread pool
1935 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1937 * gst/rtsp-server/rtsp-client.h:
1938 * gst/rtsp-server/rtsp-server.c:
1939 * gst/rtsp-server/rtsp-server.h:
1940 server: use thread pool
1941 Use the thread pool instead of doing our own thing.
1943 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1945 * gst/rtsp-server/Makefile.am:
1946 * gst/rtsp-server/rtsp-thread-pool.c:
1947 * gst/rtsp-server/rtsp-thread-pool.h:
1948 thread-pool: add object to manage threads
1949 Add an object to manage the client and media threads.
1951 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1953 * gst/rtsp-server/rtsp-auth.c:
1954 auth: debug authorization check
1956 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1958 * gst/rtsp-server/rtsp-media.c:
1959 media: start media pipeline in context
1960 Start the media pipeline in the provided context (or our default one
1961 when NULL). This makes sure that we run the bus thread in this context and that
1962 all media threads are children of this context.
1964 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1966 * gst/rtsp-server/rtsp-media-factory.c:
1967 factory: pass permissions to media by default
1969 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1971 * examples/test-auth.c:
1972 test: add permissions to auth test
1973 Ass some permissions to the media factory in the test.
1975 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1977 * gst/rtsp-server/rtsp-auth.c:
1978 * gst/rtsp-server/rtsp-auth.h:
1979 * gst/rtsp-server/rtsp-client.c:
1980 auth: simplify auth checks
1981 Remove client from methods, it's now in the state
1982 Perform the check specified by the string, use the information from the
1983 thread local context.
1985 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1987 * gst/rtsp-server/rtsp-client.c:
1988 * gst/rtsp-server/rtsp-client.h:
1989 client: add state to current thread
1990 Add the client to the ClientState object.
1991 Place the ClientState on the current thread.
1993 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1995 * gst/rtsp-server/rtsp-media-factory.c:
1996 * gst/rtsp-server/rtsp-media-factory.h:
1997 * gst/rtsp-server/rtsp-media.c:
1998 * gst/rtsp-server/rtsp-media.h:
1999 media: make it possible to set permissions
2000 Make it possible to set permissions on media and media factory objects
2002 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2004 * gst/rtsp-server/Makefile.am:
2005 * gst/rtsp-server/rtsp-permissions.c:
2006 * gst/rtsp-server/rtsp-permissions.h:
2007 permissions: add permissions object
2008 Add a mini object to store permissions based on a role.
2010 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2012 * examples/test-auth.c:
2013 * gst/rtsp-server/rtsp-auth.c:
2014 * gst/rtsp-server/rtsp-auth.h:
2015 * gst/rtsp-server/rtsp-client.c:
2016 auth: add auth checks
2017 Add an enum with auth checks and implement the checks in the auth object.
2018 Perform the checks from the client.
2020 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2022 * examples/test-auth.c:
2023 * gst/rtsp-server/rtsp-auth.c:
2024 * gst/rtsp-server/rtsp-auth.h:
2025 * gst/rtsp-server/rtsp-client.h:
2026 auth: use the token after authentication
2027 After we authenticated a user, keep the Token around in the state.
2029 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2031 * gst/rtsp-server/rtsp-client.c:
2032 * gst/rtsp-server/rtsp-media.c:
2033 * gst/rtsp-server/rtsp-media.h:
2034 * tests/check/gst/media.c:
2035 media: add optional context for bus messages
2036 Add an optional mainloop to _prepare that will handle the bus messages instead
2037 of always using the shared mainloop.
2039 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2041 * gst/rtsp-server/Makefile.am:
2042 * gst/rtsp-server/rtsp-token.c:
2043 * gst/rtsp-server/rtsp-token.h:
2044 token: add authorization token
2045 Add a simply miniobject that contains the authorizations. The object contains a
2046 GstStructure that hold all authorization fields. When a user is authenticated,
2047 the auth module will create a Token for the user. The token is then used to
2048 check what operations the user is allowed to do and various other configuration
2051 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2053 * examples/test-auth.c:
2054 * gst/rtsp-server/rtsp-auth.c:
2055 * gst/rtsp-server/rtsp-auth.h:
2056 * gst/rtsp-server/rtsp-client.c:
2057 * gst/rtsp-server/rtsp-client.h:
2058 * gst/rtsp-server/rtsp-media-factory.c:
2059 * gst/rtsp-server/rtsp-media-factory.h:
2060 * gst/rtsp-server/rtsp-media.c:
2061 * gst/rtsp-server/rtsp-media.h:
2062 auth: remove auth from media and factory
2063 Remove the auth object from media and factory. We want to have the RTSPClient
2064 authenticate and authorize resources, there is no need to place another auth
2065 manager on the media/factory.
2067 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2069 * examples/test-auth.c:
2070 * gst/rtsp-server/rtsp-auth.c:
2071 * gst/rtsp-server/rtsp-auth.h:
2072 * gst/rtsp-server/rtsp-client.h:
2073 auth: add support for multiple basic auth tokens
2074 Make it possible to add multiple basic authorisation tokens to one authorization
2075 object. Associate with each token an authorization group that will define what
2076 capabilities are allowed.
2078 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2080 * gst/rtsp-server/rtsp-client.c:
2081 client: error out on non-aggregate control
2082 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
2084 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2086 * gst/rtsp-server/rtsp-client.c:
2087 client: rework setup request a little
2088 Cache the media in DESCRIBE based on the longest matching path with the uri
2089 that we can find in the mount points.
2090 Rework the setup request a little to get the media from the session or from
2091 the longest matching path, this way we can derive the control string as
2092 everything after the path instead of hardcoding it.
2093 Find the stream based on the control string and only open a session when all
2096 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2098 * gst/rtsp-server/rtsp-media.c:
2099 * gst/rtsp-server/rtsp-media.h:
2100 media: add method to find a stream by control url
2102 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2104 * gst/rtsp-server/rtsp-stream.c:
2105 * gst/rtsp-server/rtsp-stream.h:
2106 stream: add method to check control url of stream
2108 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2110 * gst/rtsp-server/rtsp-client.c:
2111 * gst/rtsp-server/rtsp-session-media.c:
2112 * gst/rtsp-server/rtsp-session-media.h:
2113 * gst/rtsp-server/rtsp-session.c:
2114 * gst/rtsp-server/rtsp-session.h:
2115 session: use path matching for session media
2116 Use a path string instead of a uri to lookup session media in the sessions. Also
2117 use path matching to find the largest possible path that matches.
2119 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2121 * gst/rtsp-server/rtsp-client.c:
2122 * gst/rtsp-server/rtsp-mount-points.c:
2123 * gst/rtsp-server/rtsp-mount-points.h:
2124 * tests/check/gst/mountpoints.c:
2125 mount-points: remove useless vmethod
2126 Making lookups in the mount points should not be done with a URL, if there is a
2127 mapping to be done from URL to mount points, we'll need to do it somewhere
2130 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2132 * gst/rtsp-server/rtsp-mount-points.c:
2133 * gst/rtsp-server/rtsp-mount-points.h:
2134 * tests/check/gst/mountpoints.c:
2135 mount-points: improve mount point searching
2136 Use a GSequence to keep track of the mount points.
2137 Match a URL to the longest matching registered mount point. This should be the
2138 URL to perform aggreagate control and the remainder is the stream specific
2140 Add some unit tests for this.
2142 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
2144 * gst/rtsp-server/Makefile.am:
2145 rtsp-server: Allow building of static library
2147 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2149 * tests/check/gst/mediafactory.c:
2150 tests: fix compilation
2152 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2154 * gst/rtsp-server/rtsp-sdp.c:
2155 sdp: get control string from stream
2156 Use the control string as configured in the stream.
2158 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2160 * gst/rtsp-server/rtsp-stream.c:
2161 * gst/rtsp-server/rtsp-stream.h:
2162 stream: add methods and property to set control string
2164 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2166 * gst/rtsp-server/rtsp-client.c:
2168 Rename variables for clarity
2169 Keep media in state when we can
2171 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2173 * gst/rtsp-server/rtsp-client.c:
2174 * gst/rtsp-server/rtsp-stream.c:
2175 * gst/rtsp-server/rtsp-stream.h:
2176 stream: add more support for IPv6
2177 Rename _get_address to _get_multicast_address in GstRTSPStream to
2178 make it clear that this function only deals with multicast.
2179 Make it possible to have both an IPv4 and IPv6 multicast address on
2180 a stream. Give the client an IPv4 or IPv6 address depending on the
2181 address it used to connect to the server.
2182 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
2184 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2186 * gst/rtsp-server/rtsp-client.c:
2189 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2191 * gst/rtsp-server/rtsp-stream.c:
2192 stream: handle failed port allocation
2193 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
2194 can't allocate any family at all. Also keep track of what port families we
2196 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
2198 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2200 * gst/rtsp-server/rtsp-stream.c:
2201 stream: improve docs
2203 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2205 * gst/rtsp-server/rtsp-stream-transport.c:
2206 stream-transport: remove old if 0 block
2208 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
2210 * tests/check/gst/client.c:
2212 gst_rtsp_client_get_uri() has been removed
2213 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
2215 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2217 * gst/rtsp-server/rtsp-client.c:
2218 * gst/rtsp-server/rtsp-client.h:
2219 client: add method to filter managed sessions
2220 Add a method to filter the sessions managed by this client connection.
2221 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
2223 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2225 * gst/rtsp-server/rtsp-client.c:
2226 * gst/rtsp-server/rtsp-client.h:
2227 client: remove _get_uri() method
2228 Remove the get_uri() method on the client. A client has no uri, the uri
2229 property is an internal property to manage the last cached media for
2232 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2234 * gst/rtsp-server/rtsp-media-factory.h:
2235 media-factory: fix typo
2237 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
2239 * gst/rtsp-server/rtsp-media.c:
2240 rtsp-media: Do not leak the query in default_query_stop
2241 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
2243 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2245 * gst/rtsp-server/rtsp-media.c:
2246 media: don't unlock when conversion fails
2247 Don't unlock the state lock when conversion fails because it was not locked.
2249 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2251 * gst/rtsp-server/rtsp-media.c:
2252 * gst/rtsp-server/rtsp-media.h:
2253 Add query_position and query_stop vmethods to rtsp-media
2255 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2257 * gst/rtsp-server/rtsp-media.c:
2258 Fix typo in property install for rtsp-media's time-provider
2260 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2262 * gst/rtsp-server/rtsp-client.c:
2263 * gst/rtsp-server/rtsp-client.h:
2264 client: clean some variables
2265 Clean some variables and add some guards to _send_request()
2267 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2269 * gst/rtsp-server/rtsp-client.c:
2270 * gst/rtsp-server/rtsp-client.h:
2271 Add gst_rtsp_client_send_request API
2272 This makes it possible to send arbitrary messages to a client, such as
2273 SET_PARAMETER or GET_PARAMETER
2275 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2277 * gst/rtsp-server/rtsp-media.c:
2278 * gst/rtsp-server/rtsp-media.h:
2279 media: add _get_element() method
2280 Add method to get the element used when creating the media.
2281 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
2283 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2285 * gst/rtsp-server/rtsp-media.c:
2288 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
2290 * gst/rtsp-server/rtsp-stream.c:
2291 * gst/rtsp-server/rtsp-stream.h:
2292 stream: allow access to the rtp session
2293 https://bugzilla.gnome.org/show_bug.cgi?id=703004
2295 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
2297 * gst/rtsp-server/rtsp-stream.c:
2298 * gst/rtsp-server/rtsp-stream.h:
2299 dscp qos support in gst-rtsp-stream
2300 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
2302 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2304 * tests/check/gst/rtspserver.c:
2306 Actually do what the comment says. Also keep the old code around, not sure what
2307 should happen when you get a 454 from a TEARDOWN, does it close the connection?
2308 it currently doesn't.
2310 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2312 * gst/rtsp-server/rtsp-client.c:
2313 client: also watch newly created session
2314 When we newly created a session, start watching it immediately instead of
2315 on the next request.
2317 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
2319 * tests/check/gst/client.c:
2320 tests: add unit test for new-session
2321 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
2323 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2325 * gst/rtsp-server/rtsp-client.c:
2326 client: emit new-session when new session is created
2327 Only emit new-session when we created a new session for a client, not when a
2328 client picked up a previous session.
2329 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
2331 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
2333 * gst/rtsp-server/rtsp-client.c:
2334 client: handle asterisk as path in requests
2335 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
2337 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2339 * gst/rtsp-server/rtsp-media.c:
2340 media: handle segment query format mismatch
2341 It's possible that the segment query returns with a different format than what
2342 we asked for, handle this case also.
2344 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
2346 * gst/rtsp-server/rtsp-media.c:
2347 media: use segment stop in collect_media_stats
2348 Use segment stop instead of duration as range end point.
2349 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
2351 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
2353 * gst/rtsp-server/rtsp-media.c:
2354 * tests/check/gst/media.c:
2355 rtsp-media: Do not leak the element in take_pipeline
2356 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
2358 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
2360 * gst/rtsp-server/rtsp-client.c:
2361 * gst/rtsp-server/rtsp-client.h:
2362 rtsp-client: Make configure_client_transport virtual
2363 This patch makes configure_client_transport virtual. The functionality is
2364 needed to handle some weird clients sending multicast transport settings as url
2366 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
2368 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
2370 * gst/rtsp-server/rtsp-client.c:
2371 * gst/rtsp-server/rtsp-client.h:
2372 rtsp-client: Make param_set and param_get virtual
2373 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
2375 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
2377 * gst/rtsp-server/rtsp-client.c:
2378 * gst/rtsp-server/rtsp-media.c:
2379 * gst/rtsp-server/rtsp-media.h:
2380 media: convert_range replaces get_range_times
2381 get_range_times worked for handling UTC ranges for seeks, but we also
2382 need to convert back from NPT to the requested unit in
2383 get_range_string. convert_range is now used for both.
2384 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
2386 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2388 * gst/rtsp-server/rtsp-client.c:
2389 * gst/rtsp-server/rtsp-sdp.c:
2390 * gst/rtsp-server/rtsp-sdp.h:
2391 sdp: cleanup sdp info
2392 We don't need to pass the proto, we can more easily check a boolean.
2393 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
2395 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
2397 * gst/rtsp-server/rtsp-sdp.c:
2398 use 0.0.0.0 or :: for c= line instead of server address
2400 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
2402 * gst/rtsp-server/rtsp-client.c:
2403 use local address, not remote, in SDP
2404 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
2406 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2409 Automatic update of common submodule
2410 From 098c0d7 to 01a7a46
2412 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
2414 * gst/rtsp-server/rtsp-media.c:
2415 * gst/rtsp-server/rtsp-media.h:
2416 media: possibility to override range time conversion
2417 Make it possible to override the conversion from GstRTSPTimeRange to
2418 GstClockTimes, that is done before seeking on the media
2419 pipeline. Overriding can be useful for UTC ranges, where the default
2420 conversion gives nanoseconds since 1900.
2421 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
2423 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
2425 * gst/rtsp-server/rtsp-server.c:
2426 * gst/rtsp-server/rtsp-server.h:
2427 rtsp-server: Expose the use_client_settings API
2428 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
2430 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
2432 * gst/rtsp-server/rtsp-client.c:
2433 * gst/rtsp-server/rtsp-stream.c:
2434 * gst/rtsp-server/rtsp-stream.h:
2435 rtspstream: handle both ipv4 and ipv6 clients
2436 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
2438 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2440 * gst/rtsp-server/rtsp-sdp.c:
2441 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
2442 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
2443 We already have a way to place extra attributes in the SDP by using a string
2444 property with prefix x- or a- in the caps.
2446 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2448 * gst/rtsp-server/rtsp-sdp.c:
2449 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
2450 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
2451 We already have a way to place extra attributes in the SDP, just make a string
2452 property in the payloader with a- or x- prefix.
2454 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2456 * gst/rtsp-server/rtsp-sdp.c:
2457 rtsp: place a- and x- properties as attributes
2458 application/x-rtp has properties with a- and x- prefixes that should be
2459 placed as attributes in the SDP for the media instead of being added to the
2462 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2464 * examples/Makefile.am:
2465 * examples/test-video.c:
2466 example: add TLS example
2468 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2470 * gst/rtsp-server/rtsp-server.c:
2471 * gst/rtsp-server/rtsp-server.h:
2472 server: add support for TLS
2473 Add methods to set and get a TLS certificate.
2474 Add vmethod to configure a new connection. By default, configure the TLS
2475 certificate in a new connection if needed.
2477 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2479 * gst/rtsp-server/rtsp-server.c:
2480 * gst/rtsp-server/rtsp-server.h:
2481 server: remove accept_client vmethod
2482 This vmethod is not very useful so remove it.
2484 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2486 * gst/rtsp-server/rtsp-server.c:
2487 server: don't crash on NULL GError
2489 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
2491 * gst/rtsp-server/rtsp-session-pool.c:
2492 rtsp-session-pool: corrected session timeout detection
2493 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
2495 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2497 * gst/rtsp-server/rtsp-client.c:
2498 client: improve debug
2500 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2502 * gst/rtsp-server/rtsp-client.c:
2503 * gst/rtsp-server/rtsp-client.h:
2504 * gst/rtsp-server/rtsp-server.c:
2505 server: refactor connection setup
2506 Let the server accept the socket connection and construct a GstRTSPConnection
2507 from it. Remove the code from the client and let the client only deal with
2508 a fully configure GstRTSPConnection object.
2509 We will need this later when the server will configure the connection for
2512 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2514 * gst/rtsp-server/rtsp-stream.c:
2515 stream: keep the transport object alive
2516 Keep the transport object alive while we have it as qdata on the
2519 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
2521 * gst/rtsp-server/rtsp-client.c:
2522 * gst/rtsp-server/rtsp-server.c:
2523 rtsp-server: Do not crash on nmapping of server
2524 * generate error when gst_rtsp_connection_accept fails
2525 * do not stop accepting incoming connections because
2526 accepting a client fails
2527 https://bugzilla.gnome.org/show_bug.cgi?id=701072
2529 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
2531 * gst/rtsp-server/rtsp-client.c:
2532 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
2533 https://bugzilla.gnome.org/show_bug.cgi?id=700953
2535 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
2537 * gst/rtsp-server/rtsp-sdp.c:
2538 rtsp-sdp: Parse framerate caps field and set SDP attribute
2539 The SDP attribute and its format is described in RFC4566.
2540 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
2542 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
2544 * gst/rtsp-server/rtsp-sdp.c:
2545 rtsp-sdp: Parse width/height from caps and set SDP attribute
2546 The SDP attribute and its format is described in RFC6064.
2547 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
2549 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
2551 * gst/rtsp-server/rtsp-sdp.c:
2552 * tests/check/gst/client.c:
2553 rtsp-sdp: add bandwidth line
2554 https://bugzilla.gnome.org/show_bug.cgi?id=699220
2556 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2559 Automatic update of common submodule
2560 From 5edcd85 to 098c0d7
2562 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
2564 * tests/check/gst/media.c:
2565 tests: add dynamic payloader prepare/unprepare check
2567 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2569 * gst/rtsp-server/rtsp-media.c:
2570 media: release lock when removing fakesink
2572 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2574 * gst/rtsp-server/rtsp-stream.c:
2575 stream: set elements to NULL before removing
2576 When removing a stream, set the elements to NULL first. This avoids
2577 element-is-not-in-NULL-state errors when we dispose the elements.
2579 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
2582 Automatic update of common submodule
2583 From 3cb3d3c to 5edcd85
2585 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2587 * gst/rtsp-server/rtsp-media.c:
2588 * gst/rtsp-server/rtsp-media.h:
2589 media: listen to pad-removed signals
2590 Listen to the pad-removed signal and remove the stream associated with the
2592 Add signal to be notified of the removed pad.
2593 Remove the fakesink in unprepare()
2594 Fix signatures of the signal methods
2596 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2598 * examples/test-sdp.c:
2599 tests: add example of reusable pipelines
2601 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
2603 * gst/rtsp-server/rtsp-stream.c:
2604 * gst/rtsp-server/rtsp-stream.h:
2605 stream: add method to get the srcpad
2607 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
2609 * tests/check/gst/media.c:
2610 check: add media prepare/unprepare test
2611 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2613 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
2615 * gst/rtsp-server/rtsp-media.c:
2616 media: disconnect from signal handlers in unprepare()
2617 We connected to the pad-added and no-more-pads signals in prepare() so
2618 we need to disconnect from them in unprepare().
2619 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2621 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
2623 * gst/rtsp-server/rtsp-media.c:
2624 media: don't free streams array
2625 Don't free the streams array in the unprepare() method, they were not
2627 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2629 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
2631 * gst/rtsp-server/rtsp-media.c:
2632 media: don't unref the pipeline in unprepare
2633 Unprepare() should undo what prepare() does. Because the pipeline is
2634 not created in prepare(), we should not unref it in unprepare()
2636 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
2638 * gst/rtsp-server/rtsp-stream.c:
2639 stream: clear session and caps for reuse
2640 Set the session and caps to NULL after unref otherwise we might unref
2642 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2644 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
2646 * gst/rtsp-server/rtsp-client.c:
2647 client: send out teardown signal before tearing down
2648 The advantage is that in the signal handler you get direct access to
2649 information about what streams are about to get torn down (in the
2650 GstRTSPClientState).
2651 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
2653 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
2655 * gst/rtsp-server/rtsp-client.c:
2656 * gst/rtsp-server/rtsp-client.h:
2657 client: expose connection
2658 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
2660 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
2663 Automatic update of common submodule
2664 From aed87ae to 3cb3d3c
2666 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2668 * gst/rtsp-server/rtsp-media.c:
2669 * gst/rtsp-server/rtsp-media.h:
2670 * gst/rtsp-server/rtsp-session-media.c:
2671 * gst/rtsp-server/rtsp-session-media.h:
2672 media: add method to get the base_time of the pipeline
2673 Together with a shared clock, this base-time could eventually be sent to
2674 the client so that it can reconstruct the exact running-time of the clock
2677 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2679 * gst/rtsp-server/Makefile.am:
2680 * gst/rtsp-server/rtsp-media.c:
2681 * gst/rtsp-server/rtsp-media.h:
2682 * gst/rtsp-server/rtsp-sdp.c:
2683 media: add GstNetTimeProvider support
2684 Add a property to let the media provide a GstNetTimeProvider for its clock.
2685 Make methods to get the clock and nettimeprovider
2686 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
2687 provider and also the current time of the clock. This should make it possible
2688 for (GStreamer) clients to slave their clock to the server clock.
2690 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
2693 Automatic update of common submodule
2694 From 04c7a1e to aed87ae
2696 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2698 * gst/rtsp-server/rtsp-media.c:
2699 media: wait for buffering to complete
2700 Wait for buffering to complete before changing the state to the target state.
2702 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2704 * gst/rtsp-server/rtsp-media.c:
2705 media: small cleanup
2707 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
2709 * tests/check/gst/rtspserver.c:
2710 tests: remove extra unref in test_setup_non_existing_stream
2711 The unref is not needed anymore, teardown runs without it.
2712 https://bugzilla.gnome.org/show_bug.cgi?id=696542
2714 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
2716 * tests/check/gst/rtspserver.c:
2717 tests: GSocketService cleanup in test_bind_already_in_use
2718 Use g_socket_service_stop so the rtspserver test stops listening for
2719 incoming connections in test_bind_already_in_use.
2720 https://bugzilla.gnome.org/show_bug.cgi?id=696541
2722 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
2724 * gst/rtsp-server/rtsp-media-factory.c:
2725 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
2726 Instead use a GWeakRef which is safe to use
2727 This is a known GLib bug, see:
2728 https://bugzilla.gnome.org/show_bug.cgi?id=667145
2730 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
2732 * gst/rtsp-server/rtsp-client.c:
2733 * gst/rtsp-server/rtsp-media.c:
2734 * gst/rtsp-server/rtsp-media.h:
2735 * gst/rtsp-server/rtsp-sdp.c:
2736 * tests/check/gst/media.c:
2737 * tests/check/gst/rtspserver.c:
2738 rtsp-media/client: Reply to PLAY request with same type of Range
2739 Remember the type of Range from the PLAY request and use the same type for
2742 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
2744 * gst/rtsp-server/rtsp-client.c:
2745 * gst/rtsp-server/rtsp-client.h:
2746 * tests/check/gst/client.c:
2747 rtsp-client: expose uri
2749 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
2751 * tests/check/gst/mediafactory.c:
2752 tests: Hold ref while creating second media
2753 To test if the media aren't shared, make sure we keep the first one while creating a second
2754 otherwise the same memory address may be reused.
2756 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
2759 configure: remove out-of-date comment
2761 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
2764 .gitignore: ignore more build files
2766 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
2768 * tests/check/Makefile.am:
2769 tests: use right _LIBS variable for gst-plugins-base libs
2771 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2773 * tests/check/Makefile.am:
2774 check: add librtp to libs
2776 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
2778 * tests/check/gst/rtspserver.c:
2779 tests: Add test to check selecting a port the server will send from
2781 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
2783 * tests/check/gst/rtspserver.c:
2784 tests: Make sure packets are actually received
2786 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
2788 * gst/rtsp-server/rtsp-stream.c:
2789 stream: Select unicast address from pool if appropriate
2791 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
2793 * gst/rtsp-server/rtsp-stream.c:
2794 stream: Properties are always there in Gst 1.0
2796 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
2798 * tests/check/gst/addresspool.c:
2799 tests: Add tests for unicast addresses in pool
2801 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
2803 * gst/rtsp-server/rtsp-address-pool.c:
2804 * tests/check/gst/addresspool.c:
2805 address-pool: Verify that multicast addresses are used for multicast and vice-versa
2807 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
2809 * docs/libs/gst-rtsp-server-sections.txt:
2810 * gst/rtsp-server/rtsp-address-pool.c:
2811 * gst/rtsp-server/rtsp-address-pool.h:
2812 * gst/rtsp-server/rtsp-stream.c:
2813 * tests/check/gst/addresspool.c:
2814 address-pool: Add unicast addresses
2816 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
2819 * gst/rtsp-server/rtsp-server.c:
2820 * tests/check/gst/rtspserver.c:
2821 rtsp-server: Limit the number of threads per server instance
2822 If we exceed the maximum, just round robin the clients over the existing
2825 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
2827 * gst/rtsp-server/rtsp-server.c:
2828 rtsp-server: No need to store the GMainContext in the client context
2830 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
2832 * tests/check/gst/rtspserver.c:
2833 tests: Add test for client disconnection
2835 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
2837 * tests/check/gst/rtspserver.c:
2838 tests: Test client and session timeouts with multiple threads
2840 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
2842 * gst/rtsp-server/rtsp-address-pool.c:
2843 * gst/rtsp-server/rtsp-auth.c:
2844 * gst/rtsp-server/rtsp-client.c:
2845 * gst/rtsp-server/rtsp-media-factory-uri.c:
2846 * gst/rtsp-server/rtsp-media-factory.c:
2847 * gst/rtsp-server/rtsp-media.c:
2848 * gst/rtsp-server/rtsp-mount-points.c:
2849 * gst/rtsp-server/rtsp-server.c:
2850 * gst/rtsp-server/rtsp-session-media.c:
2851 * gst/rtsp-server/rtsp-session-pool.c:
2852 * gst/rtsp-server/rtsp-session.c:
2853 Document locking and its order
2855 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
2857 * tests/check/gst/rtspserver.c:
2858 tests: Test that slow DESCRIBE don't block other clients
2860 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
2862 * tests/check/gst/client.c:
2863 tests: Add tests for client-requested multicast address
2865 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
2867 * docs/libs/gst-rtsp-server-sections.txt:
2868 docs: Put the various functions in the right sections
2870 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
2872 * docs/libs/gst-rtsp-server-docs.sgml:
2873 * docs/libs/gst-rtsp-server-sections.txt:
2874 * gst/rtsp-server/rtsp-address-pool.c:
2875 * gst/rtsp-server/rtsp-address-pool.h:
2876 docs: Generate docs for GstRTSPAddressPool
2878 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
2880 * gst/rtsp-server/rtsp-client.c:
2881 * gst/rtsp-server/rtsp-stream.c:
2882 * gst/rtsp-server/rtsp-stream.h:
2883 client: Check client provided addresses against the address pool
2885 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
2887 * gst/rtsp-server/rtsp-address-pool.c:
2888 * gst/rtsp-server/rtsp-address-pool.h:
2889 * tests/check/gst/addresspool.c:
2890 address-pool: Add API to request a specific address from the pool
2891 Also add relevant unit tests.
2893 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
2895 * tests/check/gst/mediafactory.c:
2896 tests: Check the passing around of a RTSPAddressPool
2897 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
2898 way down to the stream.
2900 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
2902 * tests/check/gst/addresspool.c:
2903 tests: Add more tests for the address pool
2905 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
2907 * gst/rtsp-server/rtsp-address-pool.c:
2908 address-pool: Fix off by one error
2909 When splitting a port range, the port after a skip is not part of range.
2911 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
2914 Automatic update of common submodule
2915 From 2de221c to 04c7a1e
2917 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
2920 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
2921 AM_CONFIG_HEADER was removed in automake 1.13
2922 https://bugzilla.gnome.org/show_bug.cgi?id=693368
2924 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
2927 Automatic update of common submodule
2928 From a942293 to 2de221c
2930 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2932 * gst/rtsp-server/rtsp-client.c:
2933 client: make sure the watch exists while sending data
2934 Protect the send_func with a lock. This allows us to wait for sending
2935 to complete before changing the send_func and user_data. We add an
2936 extra ref to the watch to make sure that it remains valid during
2938 When closing the connection, set the send_func to NULL
2939 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
2941 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2943 * tests/check/Makefile.am:
2944 tests: use GST_*_1_0 environment variables everywhere
2945 The _1_0 suffixed environment variables override the
2946 non-suffixed ones, so if we're in an environment that
2947 sets the _1_0 suffixed ones, such as jhbuild, we need
2948 to set those to make sure ours actually always get
2951 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2954 Automatic update of common submodule
2955 From acb04d9 to a942293
2957 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2959 * gst/rtsp-server/rtsp-client.c:
2960 rtsp-client: set the client backlog
2961 Set the client backlog to a reasonable default
2963 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
2965 * gst/rtsp-server/rtsp-media.c:
2966 rtsp-media: Make the element a constructor parameter
2967 https://bugzilla.gnome.org/show_bug.cgi?id=689594
2969 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2971 * docs/libs/Makefile.am:
2972 docs: Link with gcov library when gcov is enabled
2973 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
2975 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2977 * gst/rtsp-server/rtsp-media.c:
2978 media: match prepare with unprepare
2979 Really unprepare when there were an equal amount of prepare calls.
2981 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2983 * gst/rtsp-server/rtsp-media.c:
2984 media: media has to be unprepared in finalize
2985 Because unprepare takes away the last ref on the media.
2987 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2989 * gst/rtsp-server/rtsp-client.c:
2990 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
2991 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
2992 We can't use the refcount to trigger unprepare because it is the unprepare call
2993 that removes the last refcount after all messages are consumed. What we should
2994 probably do is make a prepared refcount and only unprepare when the refcount
2997 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2999 * gst/rtsp-server/rtsp-media.c:
3000 media: let the source unref the last media ref
3001 the last ref to the media is held by the source so we don't need to add more ref
3002 and unrefs, we simply destroy the media when the source is gone.
3004 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3006 * gst/rtsp-server/rtsp-media.c:
3007 media: improve debug
3009 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3011 * gst/rtsp-server/rtsp-media.c:
3013 Make sure we are in the right state when collecting the position and duration.
3014 Only make ourselves PREPARED when we were previously PREPARING.
3016 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3018 * gst/rtsp-server/rtsp-media.c:
3019 media: use g_object_ref/unref for GObjects
3021 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
3023 * gst/rtsp-server/rtsp-client.c:
3024 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
3025 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
3026 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
3027 isn't being used anymore.
3029 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
3031 * gst/rtsp-server/rtsp-media.c:
3032 Fix compiler warning
3034 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
3036 * gst/rtsp-server/rtsp-media-factory-uri.c:
3037 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
3039 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3041 * gst/rtsp-server/rtsp-session-media.h:
3044 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3046 * gst/rtsp-server/rtsp-media.c:
3047 * tests/check/gst/media.c:
3048 media: avoid element leak
3050 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3052 * gst/rtsp-server/rtsp-media.c:
3053 media: require an element in media constructor
3055 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3057 * gst/rtsp-server/rtsp-client.c:
3058 Revert "client: TEARDOWN brings that state to Init again"
3059 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
3060 The object is already disposed, there is no point in setting the state.
3062 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3064 * gst/rtsp-server/rtsp-client.c:
3065 client: TEARDOWN brings that state to Init again
3067 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3069 * docs/libs/gst-rtsp-server-sections.txt:
3070 * examples/test-auth.c:
3071 * gst/rtsp-server/rtsp-auth.c:
3072 * gst/rtsp-server/rtsp-auth.h:
3073 * gst/rtsp-server/rtsp-client.c:
3074 * gst/rtsp-server/rtsp-client.h:
3075 * gst/rtsp-server/rtsp-media-factory-uri.c:
3076 * gst/rtsp-server/rtsp-media-factory-uri.h:
3077 * gst/rtsp-server/rtsp-media-factory.c:
3078 * gst/rtsp-server/rtsp-media-factory.h:
3079 * gst/rtsp-server/rtsp-media.c:
3080 * gst/rtsp-server/rtsp-media.h:
3081 * gst/rtsp-server/rtsp-mount-points.c:
3082 * gst/rtsp-server/rtsp-mount-points.h:
3083 * gst/rtsp-server/rtsp-sdp.c:
3084 * gst/rtsp-server/rtsp-server.c:
3085 * gst/rtsp-server/rtsp-server.h:
3086 * gst/rtsp-server/rtsp-session-media.c:
3087 * gst/rtsp-server/rtsp-session-media.h:
3088 * gst/rtsp-server/rtsp-session-pool.c:
3089 * gst/rtsp-server/rtsp-session-pool.h:
3090 * gst/rtsp-server/rtsp-session.c:
3091 * gst/rtsp-server/rtsp-session.h:
3092 * gst/rtsp-server/rtsp-stream-transport.c:
3093 * gst/rtsp-server/rtsp-stream-transport.h:
3094 * gst/rtsp-server/rtsp-stream.c:
3095 * gst/rtsp-server/rtsp-stream.h:
3096 * tests/check/gst/media.c:
3097 rtsp: make object details private
3098 Make all object details private
3099 Add methods to access private bits
3101 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3103 * tests/check/Makefile.am:
3104 * tests/check/gst/media.c:
3105 tests: add media tests
3107 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3109 * gst/rtsp-server/rtsp-media.c:
3110 media: check if prepared for some methods
3111 Check that the media object is prepared before doing seek and getting the
3112 current position etc.
3113 Add some g_return checks.
3115 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3117 * tests/check/Makefile.am:
3118 * tests/check/gst/mediafactory.c:
3119 tests: add mediafactory test
3121 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3123 * gst/rtsp-server/rtsp-stream.c:
3124 stream: improve debug
3126 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3128 * gst/rtsp-server/rtsp-media.c:
3129 * gst/rtsp-server/rtsp-media.h:
3130 media: unref pipeline in finalize to avoid leaking it
3132 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3134 * gst/rtsp-server/rtsp-media-factory-uri.c:
3135 * gst/rtsp-server/rtsp-media.c:
3136 rtsp: use gst_object_unref on GstObjects
3138 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3140 * gst/rtsp-server/rtsp-media-factory.c:
3141 media-factory: require an url
3143 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3145 * examples/test-uri.c:
3146 examples: fix include
3148 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3150 * gst/rtsp-server/rtsp-server.h:
3151 server: remove unused include
3153 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3155 * tests/check/Makefile.am:
3156 * tests/check/gst/mountpoints.c:
3157 tests: add test for mountpoints
3159 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3161 * gst/rtsp-server/rtsp-client.c:
3162 client: fix factory leak
3163 Keep the factory in the state object only for authorization checks and make
3164 sure we unref it on failure. Also don't keep invalid objects in the state
3167 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3169 * gst/rtsp-server/rtsp-mount-points.c:
3170 mounts: add g_return_if guards
3172 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3174 * tests/check/gst/client.c:
3175 tests: add more tests
3177 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3179 * gst/rtsp-server/rtsp-client.c:
3180 client: improve debug
3182 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3184 * gst/rtsp-server/rtsp-client.c:
3185 client: improve debug and fix leaks
3186 Cleanup the uri and session when there is a bad request.
3188 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3193 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3195 * tests/check/gst/client.c:
3196 test: add test for session in options request
3198 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3200 * gst/rtsp-server/rtsp-client.c:
3201 client: use 454 when session can't be found
3202 We should use 454 when a session can't be found because there was no session
3203 pool configured in the server. This is not a server configuration problem
3204 because the server on which the request is done might not be the same one that
3205 will keep the sessions for us and so it does not need to support sessions.
3207 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3209 * gst/rtsp-server/rtsp-client.c:
3210 client: only free connection when there is one
3211 It's possible that the client doesn't have a connection when we try to free it.
3213 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3215 * tests/check/Makefile.am:
3216 * tests/check/gst/client.c:
3217 tests: add unit test for the client object
3219 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3221 * gst/rtsp-server/rtsp-client.c:
3222 client: small cleanup
3224 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3226 * gst/rtsp-server/rtsp-client.h:
3227 client: remove unused include
3229 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3231 * gst/rtsp-server/rtsp-client.c:
3232 client: fix compilation
3234 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3236 * gst/rtsp-server/rtsp-client.c:
3237 client: call destroy without the lock
3239 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3241 * gst/rtsp-server/rtsp-client.c:
3242 * gst/rtsp-server/rtsp-client.h:
3243 client: make the client usable without a socket
3244 Make a method to let the client handle a message and a callback when the client
3245 wants us to send a response message back. This makes it possible to also use the
3246 client object without the sockets, which should make it easier to test.
3248 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3250 * gst/rtsp-server/rtsp-client.c:
3251 * gst/rtsp-server/rtsp-client.h:
3252 client: small cleanup
3254 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3256 * docs/libs/gst-rtsp-server-sections.txt:
3257 * gst/rtsp-server/rtsp-client.c:
3258 * gst/rtsp-server/rtsp-client.h:
3259 * gst/rtsp-server/rtsp-server.c:
3260 client: remove reference to server
3261 We don't need to keep a ref to the server
3263 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3265 * gst/rtsp-server/rtsp-client.c:
3266 * gst/rtsp-server/rtsp-client.h:
3268 Also add some g_return_if()
3270 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3272 * gst/rtsp-server/rtsp-client.c:
3273 client: log more errors
3275 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3277 * gst/rtsp-server/rtsp-client.c:
3278 client: fix compilation
3280 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3282 * gst/rtsp-server/rtsp-client.c:
3283 * gst/rtsp-server/rtsp-client.h:
3284 client: add generic close-after-send support
3285 Add a property to send_response() to close the connection after the response has
3286 been sent to the client.
3288 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3291 * docs/libs/gst-rtsp-server-docs.sgml:
3292 * docs/libs/gst-rtsp-server-sections.txt:
3293 * docs/libs/gst-rtsp-server.types:
3294 * examples/test-auth.c:
3295 * examples/test-launch.c:
3296 * examples/test-mp4.c:
3297 * examples/test-multicast.c:
3298 * examples/test-multicast2.c:
3299 * examples/test-ogg.c:
3300 * examples/test-readme.c:
3301 * examples/test-sdp.c:
3302 * examples/test-uri.c:
3303 * examples/test-video.c:
3304 * gst/rtsp-server/Makefile.am:
3305 * gst/rtsp-server/rtsp-auth.h:
3306 * gst/rtsp-server/rtsp-client.c:
3307 * gst/rtsp-server/rtsp-client.h:
3308 * gst/rtsp-server/rtsp-media-mapping.c:
3309 * gst/rtsp-server/rtsp-media-mapping.h:
3310 * gst/rtsp-server/rtsp-mount-points.c:
3311 * gst/rtsp-server/rtsp-mount-points.h:
3312 * gst/rtsp-server/rtsp-server.c:
3313 * gst/rtsp-server/rtsp-server.h:
3314 * gst/rtsp-server/rtsp-session-media.c:
3315 * gst/rtsp-server/rtsp-session-pool.c:
3316 * gst/rtsp-server/rtsp-session-pool.h:
3317 * tests/check/gst/rtspserver.c:
3318 MediaMapping -> MountPoints
3319 Describes better what the object manages.
3321 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3324 configure: bump required version of -base
3326 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3328 * gst/rtsp-server/rtsp-media.c:
3331 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3333 * gst/rtsp-server/rtsp-media.c:
3334 * gst/rtsp-server/rtsp-media.h:
3335 media: support more Range formats
3336 Use the new -base methods to convert the Range string into a seek start and stop
3339 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3341 * examples/test-launch.c:
3342 examples: fix whitespace
3344 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3346 * examples/test-auth.c:
3347 test-auth: add example of how to remove sessions
3348 Add an example of the session filter api.
3350 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3352 * examples/test-uri.c:
3353 test-uri: remove mapping example
3355 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3357 * examples/test-uri.c:
3358 test-uri: fix callback signature
3360 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3362 * gst/rtsp-server/rtsp-media-factory.c:
3363 factory: keep ref to factory while media active
3364 While the media from a factory is alive, keep a ref to the factory.
3365 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
3367 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3369 * gst/rtsp-server/rtsp-media-factory-uri.c:
3370 factory-uri: add some debug
3372 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3374 * gst/rtsp-server/rtsp-stream.c:
3375 stream: set udp sources to PLAYING
3376 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
3377 so that it doesn't cause our pipeline to produce ASYNC-DONE.
3379 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3381 * gst/rtsp-server/rtsp-media-factory-uri.c:
3382 factory-uri: take ref to factory
3383 Take a ref to the factory that we place in our list.
3385 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3387 * tests/Makefile.am:
3388 * tests/test-reuse.c:
3389 test: add test for server reuse
3390 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
3392 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
3394 * gst/rtsp-server/rtsp-server.c:
3395 server: start and stop multiple times
3396 Stop listening on the RTSP port when the GSource is removed, so clients
3397 can't connect and the server can be started again.
3398 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
3400 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3402 * gst/rtsp-server/rtsp-server.c:
3403 server: fix small leak
3405 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3407 * gst/rtsp-server/rtsp-media.c:
3408 media: unref source in finish_unprepare
3409 The source is created in prepare, unref it in finish_unprepare.
3410 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
3412 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
3414 * gst/rtsp-server/rtsp-client.c:
3415 * gst/rtsp-server/rtsp-media.c:
3416 rtsp-media: remove bus watch before finalizing
3417 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
3418 * An extra media ref is added for the bus watch. This extra ref is unreffed by
3419 the GDestroyNotify function.
3420 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
3421 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
3422 gst_rtsp_media_unprepare before unreffing the media.
3423 This way, the bus watch will be removed before the media is finalized.
3424 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
3426 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
3428 * gst/rtsp-server/rtsp-client.c:
3429 * gst/rtsp-server/rtsp-client.h:
3430 client: wait until the TEARDOWN response is sent to close the connection
3431 Responses can be sent async so we need to wait until the TEARDOWN response has
3432 been written before we close the connection to the client. This avoids the risk
3433 of writing/polling closed sockets.
3434 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
3436 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
3438 * gst/rtsp-server/rtsp-stream.c:
3439 rtsp-stream: plug socket leak
3440 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
3442 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
3445 Automatic update of common submodule
3446 From 6bb6951 to a72faea
3448 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
3450 * gst/rtsp-server/rtsp-media-factory-uri.c:
3451 rtsp-server: don't use deprecated API
3453 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
3455 * gst/rtsp-server/rtsp-client.c:
3456 rtsp-client: fix unused-but-set-variable compiler warning
3457 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
3459 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3462 * docs/libs/gst-rtsp-server-sections.txt:
3463 * gst/rtsp-server/rtsp-client.c:
3466 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3468 * examples/Makefile.am:
3469 * examples/test-multicast2.c:
3470 examples: add another multicast example
3471 Add an example for how to configure separate multicast ranges for each media
3474 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3476 * examples/test-multicast.c:
3479 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3481 * gst/rtsp-server/rtsp-client.c:
3482 * gst/rtsp-server/rtsp-media.c:
3483 * gst/rtsp-server/rtsp-session-media.c:
3484 * gst/rtsp-server/rtsp-session-media.h:
3485 * gst/rtsp-server/rtsp-stream-transport.c:
3486 * gst/rtsp-server/rtsp-stream-transport.h:
3487 stream: use the address managed by the stream
3488 Use the address managed by the stream for multicast. This allows us to have 1
3489 multicast address for each stream.
3490 Because the address is now managed by the stream we don't have to pass it around
3492 Set the address pool on the streams.
3494 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3496 * gst/rtsp-server/rtsp-client.c:
3497 * gst/rtsp-server/rtsp-media.c:
3498 * gst/rtsp-server/rtsp-stream.c:
3501 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3503 * gst/rtsp-server/rtsp-media.c:
3504 * gst/rtsp-server/rtsp-media.h:
3505 media: add signal for new streams
3506 This allows applications to listen for new streams and configure properties on
3507 them, like the address pool.
3509 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3511 * gst/rtsp-server/rtsp-media.c:
3512 media: configure address pool in new streams
3514 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3516 * gst/rtsp-server/rtsp-stream.c:
3517 * gst/rtsp-server/rtsp-stream.h:
3518 stream: add methods to deal with address pool
3519 Add methods to get and set the address pool for the stream
3520 Add method to allocate and get the multicast addresses for this stream.
3522 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3524 * docs/libs/gst-rtsp-server-sections.txt:
3525 * gst/rtsp-server/rtsp-media.c:
3526 * gst/rtsp-server/rtsp-media.h:
3527 media: remove MTU property
3528 It is a stream property
3530 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3532 * gst/rtsp-server/rtsp-client.c:
3533 client: set blocksize only on stream
3534 Set the blocksize only on the current stream.
3536 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3538 * gst/rtsp-server/rtsp-stream.c:
3539 stream: share src and sink sockets
3540 the allocated socket is in the used-socket property, not socket.
3542 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3544 * gst/rtsp-server/rtsp-address-pool.c:
3545 * gst/rtsp-server/rtsp-address-pool.h:
3546 * gst/rtsp-server/rtsp-client.c:
3547 * gst/rtsp-server/rtsp-session-media.c:
3548 * gst/rtsp-server/rtsp-session-media.h:
3549 * gst/rtsp-server/rtsp-stream-transport.c:
3550 * gst/rtsp-server/rtsp-stream-transport.h:
3551 * tests/check/gst/addresspool.c:
3552 rtsp: make address-pool return an address object
3553 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
3554 store more info in the structure and allows us to more easily return the address
3555 to the right pool when no longer needed.
3556 Pass the address to the StreamTransport so that we can return it to the pool
3557 when the stream transport is freed or changed.
3559 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3561 * examples/Makefile.am:
3562 * examples/test-multicast.c:
3563 examples: add multicast example
3564 Show how to set up the multicast address pool so that media can be
3565 server with multicast.
3567 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3569 * gst/rtsp-server/rtsp-client.c:
3570 * gst/rtsp-server/rtsp-media-factory.c:
3571 * gst/rtsp-server/rtsp-media-factory.h:
3572 * gst/rtsp-server/rtsp-media.c:
3573 * gst/rtsp-server/rtsp-media.h:
3574 rtsp: use AddressPool
3575 Remove the multicast_group property.
3576 Use the configured addresspool to allocate multicast addresses.
3578 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3580 * gst/rtsp-server/rtsp-address-pool.c:
3581 * gst/rtsp-server/rtsp-address-pool.h:
3582 address-pool: add clear method
3584 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3586 * gst/rtsp-server/rtsp-address-pool.c:
3587 address-pool: small cleanups
3589 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3591 * tests/check/Makefile.am:
3592 * tests/check/gst/addresspool.c:
3593 tests: add addresspool unit test
3595 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3597 * gst/rtsp-server/Makefile.am:
3598 * gst/rtsp-server/rtsp-address-pool.c:
3599 * gst/rtsp-server/rtsp-address-pool.h:
3600 address-pool: add object to manage multicast addresses
3601 Make an object that can manage a rage of multicast addresses and ports.
3603 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3605 * gst/rtsp-server/rtsp-server.c:
3606 server: set default max-threads property
3608 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3610 * gst/rtsp-server/rtsp-media.c:
3611 media: wait for concurrent _prepare
3612 If a prepare is busy, wait for the result.
3614 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3616 * gst/rtsp-server/rtsp-media.c:
3617 media: add lock around message handler
3618 We don't want to dispatch messages while we are still processing the result of
3621 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3623 * gst/rtsp-server/rtsp-media.c:
3624 * gst/rtsp-server/rtsp-media.h:
3625 media: add lock to protect state changes
3627 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3629 * gst/rtsp-server/rtsp-stream.c:
3630 * gst/rtsp-server/rtsp-stream.h:
3633 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3635 * gst/rtsp-server/rtsp-stream-transport.c:
3636 * gst/rtsp-server/rtsp-stream-transport.h:
3637 * gst/rtsp-server/rtsp-stream.c:
3638 stream-transport: add keep-alive method
3640 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3642 * gst/rtsp-server/rtsp-stream-transport.c:
3643 * gst/rtsp-server/rtsp-stream-transport.h:
3644 * gst/rtsp-server/rtsp-stream.c:
3645 stream-transport: add method to handle RTP/RTCP
3646 Call new methods instead of poking into the structures directly.
3648 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3650 * gst/rtsp-server/rtsp-session-media.c:
3651 * gst/rtsp-server/rtsp-session-media.h:
3652 session-media: add locking
3654 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3656 * gst/rtsp-server/rtsp-session.c:
3657 * gst/rtsp-server/rtsp-session.h:
3658 session: add locking
3660 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3662 * gst/rtsp-server/rtsp-server.c:
3663 server: free old socket
3665 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3667 * gst/rtsp-server/rtsp-media-mapping.c:
3668 * gst/rtsp-server/rtsp-media-mapping.h:
3669 mapping: add locking
3671 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3673 * gst/rtsp-server/rtsp-media-factory.c:
3674 media-factory: add locking
3676 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3678 * gst/rtsp-server/rtsp-auth.c:
3679 * gst/rtsp-server/rtsp-auth.h:
3682 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3684 * gst/rtsp-server/rtsp-server.c:
3685 * gst/rtsp-server/rtsp-server.h:
3686 server: add max-thread property
3688 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3690 * gst/rtsp-server/rtsp-server.c:
3691 * gst/rtsp-server/rtsp-server.h:
3692 server: use a threadpool for the mainloops
3694 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3696 * gst/rtsp-server/rtsp-client.c:
3697 * gst/rtsp-server/rtsp-client.h:
3698 client: rename method
3699 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
3700 don't really create the client from the socket, we use the socket for the
3703 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3705 * gst/rtsp-server/rtsp-client.c:
3706 * gst/rtsp-server/rtsp-client.h:
3707 * gst/rtsp-server/rtsp-server.c:
3708 server: rework maincontext handling in clients
3709 Make a separate method to attach a client to a MainContext.
3710 Let the server decide in what GMainContext the client will operate and give this
3711 context to the client in attach. Then the server can later decide to use a
3712 separate thread for each client or just use the mainthread.
3714 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3716 * gst/rtsp-server/rtsp-client.c:
3717 * gst/rtsp-server/rtsp-session.c:
3718 * gst/rtsp-server/rtsp-session.h:
3719 session: move session header code in session object
3721 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
3725 * examples/test-auth.c:
3726 * examples/test-launch.c:
3727 * examples/test-mp4.c:
3728 * examples/test-ogg.c:
3729 * examples/test-readme.c:
3730 * examples/test-sdp.c:
3731 * examples/test-uri.c:
3732 * examples/test-video.c:
3733 * gst/rtsp-server/rtsp-auth.c:
3734 * gst/rtsp-server/rtsp-auth.h:
3735 * gst/rtsp-server/rtsp-client.c:
3736 * gst/rtsp-server/rtsp-client.h:
3737 * gst/rtsp-server/rtsp-media-factory-uri.c:
3738 * gst/rtsp-server/rtsp-media-factory-uri.h:
3739 * gst/rtsp-server/rtsp-media-factory.c:
3740 * gst/rtsp-server/rtsp-media-factory.h:
3741 * gst/rtsp-server/rtsp-media-mapping.c:
3742 * gst/rtsp-server/rtsp-media-mapping.h:
3743 * gst/rtsp-server/rtsp-media.c:
3744 * gst/rtsp-server/rtsp-media.h:
3745 * gst/rtsp-server/rtsp-params.c:
3746 * gst/rtsp-server/rtsp-params.h:
3747 * gst/rtsp-server/rtsp-sdp.c:
3748 * gst/rtsp-server/rtsp-sdp.h:
3749 * gst/rtsp-server/rtsp-server.c:
3750 * gst/rtsp-server/rtsp-server.h:
3751 * gst/rtsp-server/rtsp-session-media.c:
3752 * gst/rtsp-server/rtsp-session-media.h:
3753 * gst/rtsp-server/rtsp-session-pool.c:
3754 * gst/rtsp-server/rtsp-session-pool.h:
3755 * gst/rtsp-server/rtsp-session.c:
3756 * gst/rtsp-server/rtsp-session.h:
3757 * gst/rtsp-server/rtsp-stream-transport.c:
3758 * gst/rtsp-server/rtsp-stream-transport.h:
3759 * gst/rtsp-server/rtsp-stream.c:
3760 * gst/rtsp-server/rtsp-stream.h:
3761 * tests/check/gst/rtspserver.c:
3762 * tests/test-cleanup.c:
3765 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
3767 * gst/rtsp-server/rtsp-media.c:
3768 * gst/rtsp-server/rtsp-session-media.c:
3769 * gst/rtsp-server/rtsp-session.c:
3770 rtsp-server: added annotations to indicate type of ownership transfer of return values
3771 https://bugzilla.gnome.org/show_bug.cgi?id=680777
3773 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
3776 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
3778 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
3781 * bindings/Makefile.am:
3782 * bindings/vala/Makefile.am:
3783 * bindings/vala/gst-rtsp-server-0.10.deps:
3784 * bindings/vala/gst-rtsp-server-0.10.vapi:
3785 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
3786 * bindings/vala/packages/gst-rtsp-server-0.10.files:
3787 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
3788 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
3789 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
3791 bindings: remove vala bindings
3792 They'll be reunited with the other GStreamer bindings
3793 https://bugzilla.gnome.org/show_bug.cgi?id=680777
3795 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3797 * gst/rtsp-server/rtsp-client.c:
3798 * gst/rtsp-server/rtsp-session-media.c:
3799 * gst/rtsp-server/rtsp-session-media.h:
3800 * gst/rtsp-server/rtsp-stream-transport.c:
3801 * gst/rtsp-server/rtsp-stream-transport.h:
3802 rtsp: only create transport when needed
3803 Only create the StreamTransport when configured.
3805 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3807 * gst/rtsp-server/rtsp-client.c:
3808 client: small cleanup
3810 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3812 * gst/rtsp-server/rtsp-client.c:
3813 * gst/rtsp-server/rtsp-client.h:
3814 * gst/rtsp-server/rtsp-stream-transport.c:
3815 * gst/rtsp-server/rtsp-stream-transport.h:
3816 rtsp: refactor configuration of transport
3817 Move the configuration of the transport to a place where it makes
3820 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3822 * gst/rtsp-server/rtsp-client.c:
3823 client: refactor transport parsing
3825 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3827 * gst/rtsp-server/rtsp-client.c:
3828 client: refuse to change the MTU on shared media
3829 If we change the MTU of chared media, it changes for all clients.
3830 We don't want to set the MTU to something large for clients that
3833 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3835 * examples/test-mp4.c:
3836 * gst/rtsp-server/rtsp-media.c:
3837 small fixes to docs and debug
3839 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3841 * gst/rtsp-server/rtsp-stream.c:
3842 stream: transports must already have been removed
3844 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3846 * gst/rtsp-server/rtsp-media.c:
3847 * gst/rtsp-server/rtsp-stream.c:
3848 * gst/rtsp-server/rtsp-stream.h:
3849 stream: improve join and leave of the pipeline
3851 Do the cleanup properly
3854 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3856 * gst/rtsp-server/rtsp-media.c:
3857 media: move unprepare below default implementation
3858 Makes it easier to find the default implementation
3860 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3862 * gst/rtsp-server/rtsp-media.c:
3863 media: signal unprepared when we actually finish
3865 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3867 * gst/rtsp-server/rtsp-media.c:
3868 media: no need to unlock, unprepare does that when needed
3870 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3872 * docs/libs/gst-rtsp-server-sections.txt:
3873 * gst/rtsp-server/rtsp-media-factory.h:
3874 * gst/rtsp-server/rtsp-media-mapping.c:
3875 * gst/rtsp-server/rtsp-media.h:
3876 * gst/rtsp-server/rtsp-params.c:
3877 * gst/rtsp-server/rtsp-server.c:
3878 * gst/rtsp-server/rtsp-session-pool.h:
3879 * gst/rtsp-server/rtsp-session.c:
3880 * gst/rtsp-server/rtsp-session.h:
3881 * gst/rtsp-server/rtsp-stream-transport.h:
3882 * gst/rtsp-server/rtsp-stream.h:
3885 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3887 * gst/rtsp-server/rtsp-client.c:
3888 * gst/rtsp-server/rtsp-media-mapping.h:
3889 * gst/rtsp-server/rtsp-media.c:
3890 * gst/rtsp-server/rtsp-media.h:
3891 * gst/rtsp-server/rtsp-server.h:
3892 * gst/rtsp-server/rtsp-stream.c:
3893 * gst/rtsp-server/rtsp-stream.h:
3894 rtsp: fix MTU setting
3895 Fix setting of the MTU. There is no need for a vmethod.
3897 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3902 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3905 configure: bump version number after refactoring
3907 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3909 * gst/rtsp-server/Makefile.am:
3910 * gst/rtsp-server/rtsp-client.c:
3911 * gst/rtsp-server/rtsp-client.h:
3912 * gst/rtsp-server/rtsp-media-factory-uri.c:
3913 * gst/rtsp-server/rtsp-media-factory.c:
3914 * gst/rtsp-server/rtsp-media-factory.h:
3915 * gst/rtsp-server/rtsp-media.c:
3916 * gst/rtsp-server/rtsp-media.h:
3917 * gst/rtsp-server/rtsp-sdp.c:
3918 * gst/rtsp-server/rtsp-session-media.c:
3919 * gst/rtsp-server/rtsp-session-media.h:
3920 * gst/rtsp-server/rtsp-session.c:
3921 * gst/rtsp-server/rtsp-session.h:
3922 * gst/rtsp-server/rtsp-stream-transport.c:
3923 * gst/rtsp-server/rtsp-stream-transport.h:
3924 * gst/rtsp-server/rtsp-stream.c:
3925 * gst/rtsp-server/rtsp-stream.h:
3926 rtsp: massive refactoring
3927 Make GObjects from the remaining simple structures.
3928 Remove GstRTSPSessionStream, it's not needed.
3929 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
3930 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
3931 a GstRTSPStream should be transported to a client.
3932 Rename GstRTSPMediaFactory::get_element -> create_element because that
3933 more accurately describes what it does.
3934 Make nice methods instead of poking in the structures.
3935 Move some methods inside the relevant object source code.
3936 Use GPtrArray to store objects instead of plain arrays, it is more
3937 natural and allows us to more easily clean up.
3938 Move the allocation of udp ports to the Stream object. The Stream object
3939 contains the elements needed to stream the media to a client.
3940 Improve the prepare and unprepare methods. Unprepare should now undo
3941 everything prepare did. Improve also async unprepare when doing EOS on
3942 shutdown. Make sure we always unprepare correctly.
3944 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
3946 * gst/rtsp-server/rtsp-client.c:
3947 rtsp-client: Unref server address clients connected to
3948 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
3950 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
3952 * gst/rtsp-server/rtsp-server.c:
3953 rtsp-server: don't ref server socket if it is NULL
3954 Fixes test_bind_already_in_use unit test again after commit 6a497440.
3955 https://bugzilla.gnome.org/show_bug.cgi?id=686644
3957 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
3959 * tests/check/Makefile.am:
3960 tests: Add libgio link dependency
3961 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
3963 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
3965 * gst/rtsp-server/rtsp-media-mapping.c:
3966 * gst/rtsp-server/rtsp-media-mapping.h:
3967 rtsp-media-mapping: rename find_media vfunc to find_factory
3968 The virtual method and class method should have the same name
3969 so it is correctly represented in GIR file
3970 https://bugzilla.gnome.org/show_bug.cgi?id=680777
3972 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
3974 * gst/rtsp-server/rtsp-auth.c:
3975 * gst/rtsp-server/rtsp-client.c:
3976 * gst/rtsp-server/rtsp-media-factory-uri.c:
3977 * gst/rtsp-server/rtsp-media-factory.c:
3978 * gst/rtsp-server/rtsp-media-mapping.c:
3979 * gst/rtsp-server/rtsp-media.c:
3980 * gst/rtsp-server/rtsp-server.c:
3981 * gst/rtsp-server/rtsp-session-pool.c:
3982 * gst/rtsp-server/rtsp-session.c:
3983 rtsp-server: fixed comments and GIR annotations
3984 https://bugzilla.gnome.org/show_bug.cgi?id=680777
3986 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
3988 * gst/rtsp-server/rtsp-media-mapping.c:
3989 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
3991 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
3993 * gst/rtsp-server/rtsp-server.c:
3994 rtsp-server: allow binding on port 0 (binds on a random port)
3996 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
3998 * gst/rtsp-server/rtsp-server.c:
3999 * gst/rtsp-server/rtsp-server.h:
4000 rtsp-server: add bound-port property
4001 bound-port can be used to retrieve the port number when the server is bound on
4002 port 0, which binds on a random port.
4004 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
4006 * gst/rtsp-server/rtsp-media-factory.c:
4007 * gst/rtsp-server/rtsp-media-factory.h:
4008 rtsp-media-factory: make ::get_element overridable by GI bindings
4009 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
4010 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
4011 as the invoker for ::get_element(), making it overridable by GI generated
4014 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
4016 * gst/rtsp-server/rtsp-media-factory-uri.c:
4017 rtsp-media-factory-uri: don't autoplug parsers in a loop
4018 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
4021 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
4023 * gst/rtsp-server/Makefile.am:
4024 Explicitly link against gio. Fix link error on mac.
4026 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
4028 * gst/rtsp-server/rtsp-session.c:
4029 session: add ttl to the transport header in SETUP
4030 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
4032 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
4034 * gst/rtsp-server/rtsp-client.c:
4035 * gst/rtsp-server/rtsp-client.h:
4036 * gst/rtsp-server/rtsp-media.c:
4037 client: Use client transport settings for multicast if allowed.
4038 This patch makes it possible for the client to send transport settings for
4039 multicast (destination && ttl). Client settings must be explicitly allowed or
4040 the server will use its own settings.
4041 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
4043 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
4046 Automatic update of common submodule
4047 From 6c0b52c to 6bb6951
4049 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
4051 * gst/rtsp-server/rtsp-client.c:
4052 rtsp-client: do not destroy the rtsp watch
4053 Don't destroy the client watch while dispatching. The rtsp watch is
4054 automatically destroyed after the rtsp watch function closed() has
4056 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
4058 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
4061 Automatic update of common submodule
4062 From 4f962f7 to 6c0b52c
4064 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
4066 * gst/rtsp-server/rtsp-media.c:
4067 media: fix check for seekability
4069 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4071 * gst/rtsp-server/rtsp-client.c:
4072 client: use more GIO
4073 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
4075 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4077 * gst/rtsp-server/rtsp-server.c:
4078 server: remove obsolete includes
4080 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4082 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
4083 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
4084 be available in "on_new_ssrc". The transports are added in
4085 gst_rtsp_media_set_state when going to PLAYING state. However,
4086 "on_new_ssrc" might be called before this happens.
4087 https://bugzilla.gnome.org/show_bug.cgi?id=683304
4089 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4091 * gst/rtsp-server/rtsp-client.c:
4092 * gst/rtsp-server/rtsp-client.h:
4093 rtsp-client: add signals for rtsp requests (fixes #683287)
4095 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4097 * gst/rtsp-server/rtsp-client.c:
4098 * gst/rtsp-server/rtsp-client.h:
4099 add new-session signal to rtsp-client (fixes #683058)
4101 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
4104 Automatic update of common submodule
4105 From 668acee to 4f962f7
4107 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
4109 * gst/rtsp-server/rtsp-server.c:
4110 * tests/check/gst/rtspserver.c:
4111 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
4112 Do not assume that *error is set in g_socket_address_enumerator_next.
4113 Added test_bind_already_in_use unit-test.
4114 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
4116 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
4119 Automatic update of common submodule
4120 From 94ccf4c to 668acee
4122 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
4124 * gst/rtsp-server/rtsp-client.c:
4125 * gst/rtsp-server/rtsp-client.h:
4126 rtsp-client: make create_sdp virtual method
4127 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
4129 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4132 Automatic update of common submodule
4133 From 98e386f to 94ccf4c
4135 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4137 * gst/rtsp-server/rtsp-client.c:
4140 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
4142 * gst/rtsp-server/rtsp-client.c:
4143 * gst/rtsp-server/rtsp-client.h:
4144 * gst/rtsp-server/rtsp-server.c:
4145 * gst/rtsp-server/rtsp-server.h:
4146 rtsp-server: use an existing socket to establish HTTP tunnel
4147 Make it possible to transfer a socket from an HTTP server to be used as
4148 an RTSP over HTTP tunnel.
4150 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
4152 * gst/rtsp-server/rtsp-client.c:
4153 * gst/rtsp-server/rtsp-media.c:
4154 * gst/rtsp-server/rtsp-media.h:
4155 rtsp: Handle the blocksize parameter
4156 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
4158 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
4160 * tests/check/Makefile.am:
4161 * tests/check/gst/rtspserver.c:
4162 Have unit test get header from source dir, not installed dir
4163 This makes compilation of unit tests work in a build directory other
4164 than the source directory.
4165 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
4167 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
4169 * gst/rtsp-server/rtsp-media.c:
4170 rtsp-media: update for gst_element_make_from_uri() changes
4172 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
4175 * tests/Makefile.am:
4176 * tests/check/Makefile.am:
4177 * tests/check/gst/rtspserver.c:
4179 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
4181 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
4183 * gst/rtsp-server/rtsp-media.c:
4184 rtsp-media: don't collect media stats when going to NULL
4185 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
4187 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4189 * gst/rtsp-server/rtsp-client.c:
4190 client: don't leak transports
4192 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
4194 * gst/rtsp-server/rtsp-client.c:
4195 rtsp-client: free transport on no_stream in SETUP handler
4197 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
4199 * gst/rtsp-server/rtsp-client.c:
4200 rtsp-client: changed session media iteration
4201 In client_unlink_session: now don't iterate in session->medias
4202 list where items are removed by gst_rtsp_session_release_media.
4203 Instead, repeatedly remove the first item.
4205 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
4207 * gst/rtsp-server/rtsp-client.c:
4208 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
4209 GstRTSPSessionMedia is not a GObject type. When the
4210 GstRTSPSession is freed, it will free the media.
4212 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
4214 * gst/rtsp-server/rtsp-media-factory.c:
4215 factory: plug pad leak in collect_streams
4216 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
4217 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
4218 will take one reference, and the other reference will otherwise
4221 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
4224 configure: suppress some warnings when debug is disabled
4225 Warnings about unused variables should be suppressed if core has the
4226 debug system disabled.
4227 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
4229 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4231 * docs/libs/Makefile.am:
4232 docs: fix build in uninstalled setup
4233 Include gst-plugins-base libs properly.
4235 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
4237 * docs/libs/gst-rtsp-server.types:
4238 docs: include headers defining rtsp-server object types
4239 Fixes compiler warnings during docs build.
4240 https://bugzilla.gnome.org/show_bug.cgi?id=676824
4242 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
4245 configure: Add warning flags for compiler when configuring
4246 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
4248 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4251 Automatic update of common submodule
4252 From 03a0e57 to 98e386f
4254 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4257 Automatic update of common submodule
4258 From 1fab359 to 03a0e57
4260 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
4262 * gst/rtsp-server/rtsp-client.c:
4263 client: fix GSocketAddress leak in gst_rtsp_client_accept
4264 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
4266 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4269 Automatic update of common submodule
4270 From f1b5a96 to 1fab359
4272 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4275 Automatic update of common submodule
4276 From 92b7266 to f1b5a96
4278 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4281 Automatic update of common submodule
4282 From ec1c4a8 to 92b7266
4284 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4287 Automatic update of common submodule
4288 From 3429ba6 to ec1c4a8
4290 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
4292 * gst/rtsp-server/rtsp-auth.c:
4293 * gst/rtsp-server/rtsp-client.c:
4294 * gst/rtsp-server/rtsp-media-factory-uri.c:
4295 * gst/rtsp-server/rtsp-server.c:
4296 rtsp: fix compiler warnings
4297 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
4299 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4302 Automatic update of common submodule
4303 From dc70203 to 3429ba6
4305 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4307 * gst/rtsp-server/rtsp-client.c:
4308 * gst/rtsp-server/rtsp-media-factory.c:
4309 * gst/rtsp-server/rtsp-media-factory.h:
4310 * gst/rtsp-server/rtsp-media.c:
4311 * gst/rtsp-server/rtsp-media.h:
4312 * gst/rtsp-server/rtsp-server.c:
4313 * gst/rtsp-server/rtsp-server.h:
4314 * gst/rtsp-server/rtsp-session-pool.c:
4315 * gst/rtsp-server/rtsp-session-pool.h:
4316 rtsp-server: port to new thread API
4318 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4321 Automatic update of common submodule
4322 From 6db25be to dc70203
4324 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4326 * gst/rtsp-server/rtsp-auth.c:
4327 * gst/rtsp-server/rtsp-auth.h:
4328 * gst/rtsp-server/rtsp-client.c:
4329 rtsp-server: Fix compilation and compiler warnings
4331 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4335 * gst/rtsp-server/Makefile.am:
4336 configure: Modernize autotools setup a bit
4337 Also we now only create tar.bz2 and tar.xz tarballs.
4339 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4342 Automatic update of common submodule
4343 From 464fe15 to 6db25be
4345 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4348 Automatic update of common submodule
4349 From 7fda524 to 464fe15
4351 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4354 * docs/libs/Makefile.am:
4355 * docs/version.entities.in:
4357 * gst/rtsp-server/Makefile.am:
4358 * pkgconfig/Makefile.am:
4359 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4360 * pkgconfig/gstreamer-rtsp-server.pc.in:
4361 * tests/Makefile.am:
4362 rtsp-server: Update versioning
4364 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4366 Merge remote-tracking branch 'origin/0.10'
4368 gst/rtsp-server/rtsp-session-pool.c
4370 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4372 * gst/rtsp-server/rtsp-session-pool.c:
4373 rtsp-server: Don't use deprecated GLib API
4375 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4377 Replace master with 0.11
4379 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4381 Merge branch 'master' into 0.11
4383 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4385 Merge branch 'master' into 0.11
4387 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
4390 A couple minor typo fixes
4392 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4394 * gst/rtsp-server/rtsp-media.c:
4395 media: fix state of the appqueue
4397 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4399 * gst/rtsp-server/rtsp-media-factory-uri.c:
4400 factory: use videoconvert
4402 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4404 * gst/rtsp-server/rtsp-media-factory-uri.c:
4405 factory: change to new style caps
4407 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4409 * gst/rtsp-server/rtsp-client.c:
4410 * gst/rtsp-server/rtsp-client.h:
4411 * gst/rtsp-server/rtsp-media-factory-uri.c:
4412 * gst/rtsp-server/rtsp-media.c:
4413 * gst/rtsp-server/rtsp-server.c:
4414 * gst/rtsp-server/rtsp-server.h:
4415 * gst/rtsp-server/rtsp-session-pool.c:
4416 rtsp-server: port to GIO
4419 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4422 configure: fix build
4424 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4427 docs: fix for gst_rtsp_server_set_port() -> _set_service()
4428 https://bugzilla.gnome.org/show_bug.cgi?id=666548
4430 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4433 * examples/Makefile.am:
4434 First rule of gst-rtsp-server club: don't talk about gst-phonon
4436 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4439 * pkgconfig/Makefile.am:
4440 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
4441 * pkgconfig/gst-rtsp-server.pc.in:
4442 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4443 * pkgconfig/gstreamer-rtsp-server.pc.in:
4444 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
4445 For consistency with all other modules.
4447 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4449 * gst/rtsp-server/rtsp-client.c:
4450 rtsp-client: update for new map API
4452 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4455 * bindings/Makefile.am:
4456 * bindings/python/Makefile.am:
4457 * bindings/python/arg-types.py:
4458 * bindings/python/codegen/Makefile.am:
4459 * bindings/python/codegen/__init__.py:
4460 * bindings/python/codegen/argtypes.py:
4461 * bindings/python/codegen/code-coverage.py:
4462 * bindings/python/codegen/codegen.py:
4463 * bindings/python/codegen/definitions.py:
4464 * bindings/python/codegen/defsparser.py:
4465 * bindings/python/codegen/docextract.py:
4466 * bindings/python/codegen/docgen.py:
4467 * bindings/python/codegen/fileprefix.override:
4468 * bindings/python/codegen/fileprefixmodule.c:
4469 * bindings/python/codegen/h2def.py:
4470 * bindings/python/codegen/mergedefs.py:
4471 * bindings/python/codegen/mkskel.py:
4472 * bindings/python/codegen/override.py:
4473 * bindings/python/codegen/reversewrapper.py:
4474 * bindings/python/codegen/scmexpr.py:
4475 * bindings/python/rtspserver-types.defs:
4476 * bindings/python/rtspserver.defs:
4477 * bindings/python/rtspserver.override:
4478 * bindings/python/rtspservermodule.c:
4479 * bindings/python/test.py:
4481 python: remove pygst-based python bindings
4482 pygi is the future, apparently.
4484 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
4487 Automatic update of common submodule
4488 From c463bc0 to 7fda524
4490 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4493 Automatic update of common submodule
4494 From 2a59016 to c463bc0
4496 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4499 Automatic update of common submodule
4500 From 0807187 to 2a59016
4502 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4505 Automatic update of common submodule
4506 From 11f0cd5 to 0807187
4508 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4510 * examples/test-auth.c:
4511 example: update for new caps
4513 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4515 * examples/test-video.c:
4516 * gst/rtsp-server/rtsp-client.c:
4517 * gst/rtsp-server/rtsp-media-factory-uri.c:
4518 * gst/rtsp-server/rtsp-media.c:
4519 * gst/rtsp-server/rtsp-media.h:
4520 * gst/rtsp-server/rtsp-session.c:
4521 * gst/rtsp-server/rtsp-session.h:
4522 rtsp-server: port some more to 0.11
4524 Remove bufferlist stuff
4526 Add queue before appsink now that preroll-queue-len is gone.
4527 Update for request pad changes.
4529 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4531 Merge branch 'master' into 0.11
4533 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
4535 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
4536 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
4537 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
4539 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
4541 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
4542 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
4543 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
4545 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4547 Merge branch 'master' into 0.11
4549 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4551 * gst/rtsp-server/rtsp-media.c:
4552 * gst/rtsp-server/rtsp-media.h:
4553 media: add a seekable boolean
4554 Maintain the seekable state with a new variable instead of reusing the
4557 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
4559 * gst/rtsp-server/rtsp-media.c:
4560 Disallow seek in live media
4562 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4564 Merge branch 'master' into 0.11
4566 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
4568 * gst/rtsp-server/rtsp-server.c:
4569 #ifdef statements for windows socket creation were missing
4571 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
4574 Automatic update of common submodule
4575 From a39eb83 to 11f0cd5
4577 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
4580 Automatic update of common submodule
4581 From 605cd9a to a39eb83
4583 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4585 Merge branch 'master' into 0.11
4587 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4589 * gst/rtsp-server/rtsp-client.c:
4590 client: use method to access property
4592 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4594 * gst/rtsp-server/rtsp-media-factory.c:
4595 * gst/rtsp-server/rtsp-media-factory.h:
4596 media-factory: add protocols property
4597 Add a property to configure the allowed protocols in the media created from the
4600 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4602 * gst/rtsp-server/rtsp-media-factory.c:
4603 * gst/rtsp-server/rtsp-media-factory.h:
4604 media-factory: add media-configure signal
4605 Add signal to allow the application to configure the media after it was created
4608 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4610 * gst/rtsp-server/rtsp-client.c:
4611 client: use method to access property
4613 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4615 * gst/rtsp-server/rtsp-media-factory.c:
4616 * gst/rtsp-server/rtsp-media-factory.h:
4617 media-factory: add protocols property
4618 Add a property to configure the allowed protocols in the media created from the
4621 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4623 * gst/rtsp-server/rtsp-media-factory.c:
4624 * gst/rtsp-server/rtsp-media-factory.h:
4625 media-factory: add media-configure signal
4626 Add signal to allow the application to configure the media after it was created
4629 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4631 Merge branch 'master' into 0.11
4633 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4635 * gst/rtsp-server/rtsp-client.c:
4636 client: use media multicast group
4638 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4640 * gst/rtsp-server/rtsp-media-factory.h:
4641 * gst/rtsp-server/rtsp-server.h:
4642 * gst/rtsp-server/rtsp-session-pool.h:
4643 * gst/rtsp-server/rtsp-session.h:
4646 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
4648 * gst/rtsp-server/rtsp-client.c:
4649 * gst/rtsp-server/rtsp-sdp.h:
4650 sdp: copy and free the server ip address
4651 Copy and free the server ip address to make memory management easier later.
4653 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4655 * gst/rtsp-server/rtsp-media-factory.c:
4656 media-factory: configure multicast in media
4658 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4660 * gst/rtsp-server/rtsp-media.c:
4661 * gst/rtsp-server/rtsp-media.h:
4662 media: add property for multicast group
4663 Add a property to configure the multicast group in the media.
4664 Based on patches from Marc Leeman and Robert Krakora.
4666 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4668 * gst/rtsp-server/rtsp-media-factory.c:
4669 * gst/rtsp-server/rtsp-media-factory.h:
4670 media-factory: add property for multicast group
4671 Add a property to configure the multicast group in the media factory.
4672 Based on patches from Marc Leeman and Robert Krakora.
4674 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4676 * gst/rtsp-server/rtsp-client.c:
4677 client: do configuration of transport in one place
4678 Move the configuration of the transport destination address to where we also
4679 configure the other bits.
4681 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4683 * gst/rtsp-server/rtsp-client.c:
4684 client: use media multicast group
4686 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4688 * gst/rtsp-server/rtsp-media-factory.h:
4689 * gst/rtsp-server/rtsp-server.h:
4690 * gst/rtsp-server/rtsp-session-pool.h:
4691 * gst/rtsp-server/rtsp-session.h:
4694 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
4696 * gst/rtsp-server/rtsp-client.c:
4697 * gst/rtsp-server/rtsp-sdp.h:
4698 sdp: copy and free the server ip address
4699 Copy and free the server ip address to make memory management easier later.
4701 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4703 * gst/rtsp-server/rtsp-media-factory.c:
4704 media-factory: configure multicast in media
4706 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4708 * gst/rtsp-server/rtsp-media.c:
4709 * gst/rtsp-server/rtsp-media.h:
4710 media: add property for multicast group
4711 Add a property to configure the multicast group in the media.
4712 Based on patches from Marc Leeman and Robert Krakora.
4714 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4716 * gst/rtsp-server/rtsp-media-factory.c:
4717 * gst/rtsp-server/rtsp-media-factory.h:
4718 media-factory: add property for multicast group
4719 Add a property to configure the multicast group in the media factory.
4720 Based on patches from Marc Leeman and Robert Krakora.
4722 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4724 * gst/rtsp-server/rtsp-client.c:
4725 client: do configuration of transport in one place
4726 Move the configuration of the transport destination address to where we also
4727 configure the other bits.
4729 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4731 Merge branch 'master' into 0.11
4733 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
4735 * gst/rtsp-server/rtsp-client.c:
4736 client: destroy pipeline on client disconnect with no prior TEARDOWN.
4737 The problem occurs when the client abruptly closes the connection without
4738 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
4739 server is where the pipeline gets torn down. Since this handler is not called,
4740 the pipeline remains and is up and running. Subsequent clients get their own
4741 pipelines and if the do not issue TEARDOWNs then those pipelines will also
4742 remain up and running. This is a resource leak.
4744 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4746 Merge branch 'master' into 0.11
4748 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
4750 * gst/rtsp-server/rtsp-media-factory.c:
4751 * gst/rtsp-server/rtsp-media-factory.h:
4752 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
4753 For example, it can be used to retrieve source elements like appsrc, in a more
4754 convenient way than subclassing get_element.
4756 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4758 Merge branch 'master' into 0.11
4760 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
4762 * gst/rtsp-server/rtsp-server.c:
4763 rtsp-server: hold on to reference while using object
4765 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4767 * gst/rtsp-server/rtsp-media.c:
4770 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4773 configure: use unstable api
4775 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
4777 * gst/rtsp-server/rtsp-client.c:
4778 client: fix reference counting
4780 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
4782 * gst/rtsp-server/rtsp-client.c:
4783 * gst/rtsp-server/rtsp-media.c:
4784 fix compiler warnings about unused variables
4786 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
4788 * examples/test-launch.c:
4789 * examples/test-readme.c:
4790 * examples/test-uri.c:
4791 * examples/test-video.c:
4792 examples: tell rtsp uri when ready
4794 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
4797 Automatic update of common submodule
4798 From 69b981f to 605cd9a
4800 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4802 * gst/rtsp-server/rtsp-client.c:
4803 client: update for buffer API change
4805 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4807 * gst/rtsp-server/Makefile.am:
4808 Makefile.am: 0.10 => @GST_MAJORMINOR@
4810 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4812 * gst/rtsp-server/rtsp-media-factory-uri.c:
4813 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
4815 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4817 * gst/rtsp-server/.gitignore:
4818 .gitignore: 0.10 => 0.11
4820 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4822 * gst/rtsp-server/Makefile.am:
4823 Makefile.am: 0.10 => @GST_MAJORMINOR@
4825 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4827 Merge branch 'master' into 0.11
4829 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
4832 Automatic update of common submodule
4833 From 9e5bbd5 to 69b981f
4835 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
4838 Automatic update of common submodule
4839 From fd35073 to 9e5bbd5
4841 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
4844 Automatic update of common submodule
4845 From 46dfcea to fd35073
4847 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4849 * gst/rtsp-server/rtsp-media-factory-uri.c:
4850 * gst/rtsp-server/rtsp-media.c:
4851 media: port to new caps API
4853 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4855 Merge branch 'master' into 0.11
4857 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
4859 * bindings/vala/gst-rtsp-server-0.10.vapi:
4860 Updated Vala bindings.
4861 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
4863 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
4865 * gst/rtsp-server/rtsp-server.c:
4866 * gst/rtsp-server/rtsp-server.h:
4867 Add a signal for newly connected clients.
4868 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
4870 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
4872 * bindings/python/rtspserver.override:
4873 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
4875 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4877 * gst/rtsp-server/Makefile.am:
4878 * gst/rtsp-server/rtsp-client.c:
4879 * gst/rtsp-server/rtsp-funnel.c:
4880 * gst/rtsp-server/rtsp-funnel.h:
4881 * gst/rtsp-server/rtsp-media.c:
4882 rtsp-server: port to 0.11
4884 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4889 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4891 Merge branch 'master' into 0.11
4896 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4899 Automatic update of common submodule
4900 From c3cafe1 to 46dfcea
4902 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
4904 * bindings/python/Makefile.am:
4905 * bindings/python/rtspserver.defs:
4906 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
4908 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
4910 * bindings/python/arg-types.py:
4911 python bindings: add GstRTSPUrlParam
4912 Needed to implement MediaFactory virtual proxies
4914 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
4916 * bindings/python/arg-types.py:
4917 python bindings: fix returning GstRTSPUrl types
4919 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
4921 * bindings/python/arg-types.py:
4922 python bindings: add arg type for GstRTSPUrl
4924 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
4926 * bindings/python/rtspserver.defs:
4927 python bindings: fix the definition of MediaFactory.collect_stream
4929 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
4932 Automatic update of common submodule
4933 From 1ccbe09 to c3cafe1
4935 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4938 Automatic update of common submodule
4939 From 193b717 to 1ccbe09
4941 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
4944 Automatic update of common submodule
4945 From b77e2bf to 193b717
4947 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4950 build: Include lcov.mak to allow test coverage report generation
4952 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4955 Automatic update of common submodule
4956 From d8814b6 to b77e2bf
4958 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4961 Automatic update of common submodule
4962 From 6aaa286 to d8814b6
4964 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
4967 Automatic update of common submodule
4968 From 6aec6b9 to 6aaa286
4970 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
4973 autogen: wingo signed comment
4975 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
4977 * gst/rtsp-server/rtsp-session-pool.c:
4978 session: use full charset for RTSP session ID
4979 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
4980 session ID more difficult.
4981 https://bugzilla.gnome.org/show_bug.cgi?id=643812
4983 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4985 * gst/rtsp-server/Makefile.am:
4986 rtsp-server: Don't install the funnel header
4988 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
4991 Automatic update of common submodule
4992 From 1de7f6a to 6aec6b9
4994 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4997 configure: require core/base 0.10.31
4998 Needed at least for gst_plugin_feature_rank_compare_func().
5000 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
5003 Automatic update of common submodule
5004 From f94d739 to 1de7f6a
5006 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5008 * gst/rtsp-server/rtsp-media.c:
5009 media: remove more unused code
5011 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5013 * gst/rtsp-server/rtsp-media.c:
5014 * gst/rtsp-server/rtsp-media.h:
5015 media: remove duplicate filtering
5016 Remove the duplicate filtering code now that we have a released -good version.
5017 Give a warning instead.
5019 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5021 * gst/rtsp-server/rtsp-media-factory.c:
5022 * gst/rtsp-server/rtsp-media.c:
5023 media: fix default buffer size
5025 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5027 * gst/rtsp-server/rtsp-media-factory.c:
5028 * gst/rtsp-server/rtsp-media-factory.h:
5029 media-factory: add property to configure the buffer-size
5030 Add a property to configure the kernel UDP buffer size.
5032 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5034 * gst/rtsp-server/rtsp-media.c:
5035 * gst/rtsp-server/rtsp-media.h:
5036 media: add property to configure kernel buffer sizes
5037 Add a property to configure the kernel UDP buffer size.
5039 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5042 configure: set PYGOBJECT_REQ before using it
5043 https://bugzilla.gnome.org/show_bug.cgi?id=640641
5045 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5048 docs: recursive into sub-directories on 'make upload'
5050 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5052 * docs/libs/gst-rtsp-server-docs.sgml:
5053 * docs/version.entities.in:
5054 docs: mention full version these docs are for, not just major-minor
5056 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5061 === release 0.10.8 ===
5063 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5068 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5070 * gst/rtsp-server/rtsp-server.c:
5071 rtsp-server: clarify docs a little
5073 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5075 * gst/rtsp-server/rtsp-media.c:
5076 media: init debug category before starting thread
5078 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5080 * gst/rtsp-server/rtsp-auth.c:
5081 auth: add realm to make it more spec compliant
5083 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5085 * gst/rtsp-server/rtsp-server.c:
5086 * gst/rtsp-server/rtsp-server.h:
5089 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5091 * examples/test-video.c:
5092 example: improve example docs a little
5094 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5096 * gst/rtsp-server/rtsp-server.c:
5097 server: ensure the watch has a ref to the server
5099 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5101 * gst/rtsp-server/rtsp-server.c:
5102 server: simpify channel function
5104 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5106 * gst/rtsp-server/rtsp-server.c:
5107 * gst/rtsp-server/rtsp-server.h:
5108 server: simplify management of channel and source
5109 We don't need to keep around the channel and source objects. Let the mainloop
5110 and the source manage the source and channel respectively.
5112 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5118 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5121 * tests/Makefile.am:
5122 * tests/test-cleanup.c:
5123 tests: add tests directory and cleanup test
5125 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5127 * gst/rtsp-server/rtsp-media-factory-uri.c:
5128 * gst/rtsp-server/rtsp-media-factory.c:
5129 * gst/rtsp-server/rtsp-media-mapping.c:
5130 * gst/rtsp-server/rtsp-media.c:
5131 * gst/rtsp-server/rtsp-session-pool.c:
5132 * gst/rtsp-server/rtsp-session.c:
5133 server: improve debugging in various objects
5135 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5137 * gst/rtsp-server/rtsp-server.c:
5138 server: chain up to the parent finalize
5140 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
5142 * bindings/python/rtspserver-types.defs:
5143 * bindings/python/rtspserver.defs:
5144 * bindings/python/rtspserver.override:
5145 * bindings/python/test.py:
5146 gst-rtsp-server: update python bindings
5148 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5150 * gst/rtsp-server/rtsp-client.c:
5151 client: use the response from the clientstate
5152 Create the response object only once and store in the client state.
5153 Make all methods use the state response,
5155 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5157 * gst/rtsp-server/rtsp-server.c:
5158 server: use signal to keep track of clients
5159 Keep track of all the clients that the server creates and remove them when they
5160 fire the 'closed' signal.
5162 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5164 * gst/rtsp-server/rtsp-client.c:
5165 * gst/rtsp-server/rtsp-client.h:
5166 client: emit signal when closing
5168 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5170 * examples/.gitignore:
5171 * examples/Makefile.am:
5172 * examples/test-auth.c:
5173 * examples/test-video.c:
5174 * gst/rtsp-server/rtsp-auth.c:
5175 * gst/rtsp-server/rtsp-auth.h:
5176 * gst/rtsp-server/rtsp-client.c:
5177 * gst/rtsp-server/rtsp-media-factory.c:
5178 * gst/rtsp-server/rtsp-media.c:
5179 * gst/rtsp-server/rtsp-media.h:
5180 * gst/rtsp-server/rtsp-session-pool.h:
5181 * gst/rtsp-server/rtsp-session.h:
5182 media: enable per factory authorisations
5183 Allow for adding a GstRTSPAuth on the factory and media level and check
5184 permissions when accessing the factory.
5185 Add hints to the auth methods for future more fine grained authorisation.
5186 Add example application for per factory authentication.
5188 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5190 * gst/rtsp-server/rtsp-auth.c:
5191 * gst/rtsp-server/rtsp-auth.h:
5192 * gst/rtsp-server/rtsp-client.c:
5193 * gst/rtsp-server/rtsp-client.h:
5194 * gst/rtsp-server/rtsp-params.c:
5195 * gst/rtsp-server/rtsp-params.h:
5196 rtsp-server: Pass ClientState structure arround
5197 Pass the collected information for the ongoing request in a GstRTSPClientState
5198 structure that we can then pass around to simplify the method arguments. This
5199 will also be handy when we implement logging functionality.
5201 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5203 * gst/rtsp-server/rtsp-media-factory.c:
5204 * gst/rtsp-server/rtsp-media-factory.h:
5205 media-factory: add methods to configure authorisation
5207 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5209 * gst/rtsp-server/rtsp-client.c:
5210 client: unref auth in finalize
5212 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5214 * gst/rtsp-server/rtsp-server.c:
5215 server: unref auth in finalize
5217 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5219 * docs/libs/gst-rtsp-server-docs.sgml:
5220 * docs/libs/gst-rtsp-server-sections.txt:
5221 * docs/libs/gst-rtsp-server.types:
5224 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5226 * gst/rtsp-server/rtsp-server.c:
5227 * gst/rtsp-server/rtsp-server.h:
5228 server: separate create and accept
5229 Create separate create and accept methods so that subclasses can create custom
5231 Configure the server in the client object and prepare for keeping track of
5234 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5236 * gst/rtsp-server/rtsp-client.c:
5237 * gst/rtsp-server/rtsp-client.h:
5238 client: add support for setting the server.
5239 Add support for keeping a ref to the server that started this client
5242 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5244 * gst/rtsp-server/rtsp-auth.c:
5245 auth: fix memleak and add some docs
5246 Fix a memleak of the basic auth token.
5247 Add docs for the helper function
5249 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5251 * gst/rtsp-server/rtsp-auth.c:
5252 * gst/rtsp-server/rtsp-auth.h:
5253 * gst/rtsp-server/rtsp-client.c:
5254 client: delegate setup of auth to the manager
5255 Delegate the configuration of the authentication tokens to the manager object
5258 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5260 * examples/test-video.c:
5261 * gst/rtsp-server/Makefile.am:
5262 * gst/rtsp-server/rtsp-auth.c:
5263 * gst/rtsp-server/rtsp-auth.h:
5264 * gst/rtsp-server/rtsp-client.c:
5265 * gst/rtsp-server/rtsp-client.h:
5266 * gst/rtsp-server/rtsp-server.c:
5267 * gst/rtsp-server/rtsp-server.h:
5268 auth: add authentication object
5269 Add an object that can check the authorization of requests.
5270 Implement basic authentication.
5271 Add example authentication to test-video
5273 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5275 * gst/rtsp-server/rtsp-server.c:
5276 * gst/rtsp-server/rtsp-server.h:
5277 server: move includes back
5278 the includes are needed for sockaddr_in.
5280 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5282 * gst/rtsp-server/rtsp-client.c:
5283 * gst/rtsp-server/rtsp-client.h:
5284 * gst/rtsp-server/rtsp-server.c:
5285 * gst/rtsp-server/rtsp-server.h:
5286 rtsp: move network includes where they are needed
5288 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
5290 * gst/rtsp-server/rtsp-media.h:
5291 rtsp-media.h: Minor corrections in comments.
5294 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
5297 Automatic update of common submodule
5298 From e572c87 to f94d739
5300 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5304 * docs/libs/.gitignore:
5305 * examples/.gitignore:
5306 * gst/rtsp-server/.gitignore:
5309 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5311 * docs/libs/Makefile.am:
5312 docs: We don't build ps/pdf for API reference docs
5314 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5317 Automatic update of common submodule
5318 From ccbaa85 to e572c87
5320 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5323 Automatic update of common submodule
5324 From 46445ad to ccbaa85
5326 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5328 * gst/rtsp-server/Makefile.am:
5329 * gst/rtsp-server/fs-funnel.c:
5330 * gst/rtsp-server/fs-funnel.h:
5331 * gst/rtsp-server/rtsp-funnel.c:
5332 * gst/rtsp-server/rtsp-funnel.h:
5333 * gst/rtsp-server/rtsp-media.c:
5334 funnel: rename fsfunnel to rtspfunnel
5335 Rename the funnel to avoid conflicts with the farsight one.
5337 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5339 * gst/rtsp-server/Makefile.am:
5340 * gst/rtsp-server/fs-funnel.c:
5341 * gst/rtsp-server/fs-funnel.h:
5342 * gst/rtsp-server/rtsp-media.c:
5343 rtsp-media: add and use fsfunnel
5344 Add a copy of fsfunnel to the build because input-selector removed the (broken)
5345 select-all property that we need.
5347 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5349 * gst/rtsp-server/Makefile.am:
5350 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
5351 Use PKG_CONFIG_PATH specified at configure time (if any) as well
5352 for the g-ir-compiler, rather than just assuming the env var has
5355 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5362 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
5364 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5367 * gst/rtsp-server/Makefile.am:
5368 gobject-introspection: fix g-i build for uninstalled setup
5369 Requires gst-plugins-base git (> 0.10.31.2).
5371 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5373 * examples/test-uri.c:
5374 examples: add some more options and comments
5376 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5378 * gst/rtsp-server/rtsp-media-factory-uri.c:
5379 factory-uri: use right property type
5381 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5383 * gst/rtsp-server/rtsp-media-factory-uri.c:
5384 factory-uri: attempt to configure buffer-lists
5385 Attempt to configure buffer lists in the payloader for improved performance.
5387 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5389 * gst/rtsp-server/rtsp-media.c:
5390 media: attempt to configure bigger UDP buffers
5391 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
5392 send buffers with high bitrate streams.
5394 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
5396 * gst/rtsp-server/rtsp-client.c:
5397 client: use the socket length from getsockname
5398 Use the length returned by getsockname to perform the getnameinfo call because
5399 the size can depend on the socket type and platform.
5402 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5404 * docs/libs/gst-rtsp-server-docs.sgml:
5405 * docs/libs/gst-rtsp-server-sections.txt:
5406 docs: add uri factory to the docs
5408 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5410 * gst/rtsp-server/rtsp-client.c:
5411 * gst/rtsp-server/rtsp-media.h:
5414 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5416 * gst/rtsp-server/rtsp-client.c:
5417 * gst/rtsp-server/rtsp-media.c:
5418 * gst/rtsp-server/rtsp-media.h:
5419 * gst/rtsp-server/rtsp-session.c:
5420 * gst/rtsp-server/rtsp-session.h:
5421 rtsp-server: add support for buffer lists
5422 Add support for sending bufferlists received from appsink.
5425 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5427 * gst/rtsp-server/rtsp-client.c:
5428 * gst/rtsp-server/rtsp-media.c:
5429 * gst/rtsp-server/rtsp-media.h:
5430 * gst/rtsp-server/rtsp-sdp.c:
5431 media: make method to retrieve the play range
5432 Make a method to retrieve the playback range so that we can conditionally create
5433 a different range for the SDP and the PLAY requests.
5435 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5437 * gst/rtsp-server/rtsp-media.c:
5438 * gst/rtsp-server/rtsp-media.h:
5439 media: add signal to notify of state changes
5441 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5443 * gst/rtsp-server/rtsp-client.h:
5444 client: cleanup headers
5446 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5448 * gst/rtsp-server/rtsp-client.c:
5451 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5453 * gst/rtsp-server/rtsp-media-factory-uri.c:
5454 * gst/rtsp-server/rtsp-media-factory-uri.h:
5455 factory-uri: add support for gstpay
5456 Add an option to prefer gstpay over decoder + raw payloader.
5458 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5460 * gst/rtsp-server/rtsp-media-factory-uri.c:
5461 * gst/rtsp-server/rtsp-media-factory-uri.h:
5462 factory-uri: rework the autoplugger.
5463 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
5466 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5468 * gst/rtsp-server/rtsp-media-factory-uri.c:
5469 factory-uri: use better factory filter
5470 Make better payloader filter based on autoplug rank and RTP use case.
5472 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5475 Automatic update of common submodule
5476 From 169462a to 46445ad
5478 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5480 * gst/rtsp-server/rtsp-server.c:
5481 server: set SO_REUSEADDR before bind
5482 Set the SO_REUSEADDR _before_ bind() to make it actually work.
5484 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5486 * gst/rtsp-server/rtsp-media.c:
5487 * gst/rtsp-server/rtsp-media.h:
5488 media: emit prepared signal when prepared
5489 Make a 'prepared' signal and emit it when we successfully prepared the element.
5490 This signal can be used to configure the media object after it has been prepared
5493 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
5496 Automatic update of common submodule
5497 From 011bcc8 to 169462a
5499 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
5501 python an optional dependency
5502 * configure.ac: Move up valgrind and g-i checks. Make the python
5503 dependency optional, as it was before.
5505 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5507 Merge branch 'master' into 0.11
5512 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5514 * gst/rtsp-server/rtsp-media.c:
5515 media: update range when active clients changed
5516 When we changed the number of active clients, update the current range
5517 information because we want the second client connecting to a shared resource
5518 continue from where the stream currently.
5520 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5522 * gst/rtsp-server/rtsp-media-factory-uri.c:
5523 * gst/rtsp-server/rtsp-media-factory-uri.h:
5524 factory-uri: add colorspace and fix pt
5525 Rework the way we pass data to the autoplugger.
5526 When we have raw caps, plug a converter element to make pluggin to raw
5527 payloaders more successful.
5528 Make sure all dynamically plugged payloaders have a unique payload types.
5530 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5532 * examples/Makefile.am:
5533 * examples/test-uri.c:
5534 example: add example of the uri factory
5536 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5538 * gst/rtsp-server/Makefile.am:
5539 * gst/rtsp-server/rtsp-media-factory-uri.c:
5540 * gst/rtsp-server/rtsp-media-factory-uri.h:
5541 * gst/rtsp-server/rtsp-server.h:
5542 factory-uri: add a factory to stream any URI
5543 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
5546 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5548 * gst/rtsp-server/rtsp-media.c:
5549 * gst/rtsp-server/rtsp-media.h:
5550 media: ignore spurious ASYNC_DONE messages
5551 When we are dynamically adding pads, the addition of the udpsrc elements will
5552 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
5553 the real ASYNC_DONE when everything is prerolled.
5555 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5557 * gst/rtsp-server/rtsp-media-factory.c:
5558 * gst/rtsp-server/rtsp-media-factory.h:
5559 media-factory: make lock macro
5561 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
5563 * gst/rtsp-server/rtsp-client.c:
5564 rtsp-server: Remove unused variable and dead assignment
5566 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
5568 * examples/test-launch.c:
5569 * examples/test-mp4.c:
5570 * examples/test-ogg.c:
5571 * examples/test-readme.c:
5572 * examples/test-sdp.c:
5573 * examples/test-video.c:
5574 examples: Run gst-indent
5576 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
5578 * gst/rtsp-server/rtsp-client.c:
5579 * gst/rtsp-server/rtsp-media-factory.c:
5580 * gst/rtsp-server/rtsp-media-mapping.c:
5581 * gst/rtsp-server/rtsp-media.c:
5582 * gst/rtsp-server/rtsp-params.c:
5583 * gst/rtsp-server/rtsp-sdp.c:
5584 * gst/rtsp-server/rtsp-server.c:
5585 * gst/rtsp-server/rtsp-session-pool.c:
5586 * gst/rtsp-server/rtsp-session.c:
5587 rtsp-server: Run gst-indent
5588 Since it wasn't using the upstream common previously, there was no
5589 indentation check before commiting.
5591 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
5593 * gst/rtsp-server/rtsp-media-mapping.h:
5594 * gst/rtsp-server/rtsp-media.c:
5595 * gst/rtsp-server/rtsp-media.h:
5596 * gst/rtsp-server/rtsp-sdp.c:
5597 * gst/rtsp-server/rtsp-session-pool.h:
5598 * gst/rtsp-server/rtsp-session.c:
5599 * gst/rtsp-server/rtsp-session.h:
5600 rtsp-server: Some more doc fixups
5602 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5605 Makefile: Add cruft-cleaning support
5607 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5612 * docs/libs/Makefile.am:
5613 * docs/libs/gst-rtsp-server-docs.sgml:
5614 * docs/libs/gst-rtsp-server-sections.txt:
5615 * docs/libs/gst-rtsp-server.types:
5616 * docs/version.entities.in:
5617 docs: Add gtk-doc build system
5619 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5621 * gst/rtsp-server/Makefile.am:
5622 Makefile.am: Use standard GIR make behaviour
5624 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5628 autogen/configure: Bring more in sync to standard gst module behaviour
5630 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5632 * gst/rtsp-server/rtsp-media.c:
5633 media: warn and fail when gstrtpbin is not found
5635 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5638 configure: open 0.11 branch
5640 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
5644 Add common submodule
5646 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
5649 * common/Makefile.am:
5650 * common/c-to-xml.py:
5652 * common/coverage/coverage-report-entry.pl:
5653 * common/coverage/coverage-report.pl:
5654 * common/coverage/coverage-report.xsl:
5655 * common/coverage/lcov.mak:
5656 * common/gettext.patch:
5657 * common/glib-gen.mak:
5658 * common/gst-autogen.sh:
5659 * common/gst-xmlinspect.py:
5661 * common/gstdoc-scangobj:
5662 * common/gtk-doc-plugins.mak:
5663 * common/gtk-doc.mak:
5664 * common/m4/.gitignore:
5665 * common/m4/Makefile.am:
5667 * common/m4/as-ac-expand.m4:
5668 * common/m4/as-auto-alt.m4:
5669 * common/m4/as-compiler-flag.m4:
5670 * common/m4/as-compiler.m4:
5671 * common/m4/as-docbook.m4:
5672 * common/m4/as-libtool-tags.m4:
5673 * common/m4/as-libtool.m4:
5674 * common/m4/as-python.m4:
5675 * common/m4/as-scrub-include.m4:
5676 * common/m4/as-version.m4:
5677 * common/m4/ax_create_stdint_h.m4:
5678 * common/m4/check.m4:
5679 * common/m4/glib-gettext.m4:
5680 * common/m4/gst-arch.m4:
5681 * common/m4/gst-args.m4:
5682 * common/m4/gst-check.m4:
5683 * common/m4/gst-debuginfo.m4:
5684 * common/m4/gst-default.m4:
5685 * common/m4/gst-doc.m4:
5686 * common/m4/gst-error.m4:
5687 * common/m4/gst-feature.m4:
5688 * common/m4/gst-function.m4:
5689 * common/m4/gst-gettext.m4:
5690 * common/m4/gst-glib2.m4:
5691 * common/m4/gst-libxml2.m4:
5692 * common/m4/gst-plugindir.m4:
5693 * common/m4/gst-valgrind.m4:
5694 * common/m4/gtk-doc.m4:
5695 * common/m4/introspection.m4:
5697 * common/mangle-tmpl.py:
5698 * common/plugins.xsl:
5700 * common/release.mak:
5701 * common/scangobj-merge.py:
5702 * common/upload.mak:
5703 common: Remove static version
5705 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
5707 * common/m4/introspection.m4:
5708 Update introspection.m4 to match usage
5710 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5714 Remove old stuff from the README
5716 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5721 === release 0.10.7 ===
5723 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5728 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5730 * examples/test-ogg.c:
5731 test-ogg: remove parsers
5732 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
5733 buffers with timestamps. Using the parsers also seems to break things.
5735 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
5737 * bindings/vala/gst-rtsp-server-0.10.vapi:
5738 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
5739 Updated Vala bindings
5741 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
5743 * common/m4/introspection.m4:
5745 * gst/rtsp-server/Makefile.am:
5746 Added initial gobject-introspection support
5748 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5750 * gst/rtsp-server/rtsp-media-factory.c:
5751 media-factory: don't use host for shared hash key
5752 When we generate the key to share made between connections, don't include the
5753 host used to connect so that we can share media even if between clients that
5754 connected with localhost and ones with the ip address.
5756 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5758 * bindings/vala/Makefile.am:
5759 build: fix distcheck
5761 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5763 * bindings/vala/gst-rtsp-server-0.10.vapi:
5764 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
5765 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
5766 Update Vala bindings
5768 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5770 * bindings/vala/Makefile.am:
5772 Fix configure checks and installation location for Vala bindings
5775 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5780 === release 0.10.6 ===
5782 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5785 configure: release 0.10.6
5787 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5789 * gst/rtsp-server/rtsp-media.c:
5790 media: help the compiler a little
5792 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5794 * gst/rtsp-server/rtsp-media.c:
5795 * gst/rtsp-server/rtsp-media.h:
5796 * gst/rtsp-server/rtsp-session.c:
5797 media: cleanup media transport before freeing
5798 Cleanup the media transport data before freeing. In particular, remove the qdata
5799 from the rtpsource object.
5801 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5803 * gst/rtsp-server/rtsp-media-factory.c:
5804 * gst/rtsp-server/rtsp-media-factory.h:
5805 * gst/rtsp-server/rtsp-media.c:
5806 * gst/rtsp-server/rtsp-media.h:
5807 media-factory: add eos-shutdown property
5808 Add an eos-shutdown property that will send an EOS to the pipeline before
5809 shutting it down. This allows for nice cleanup in case of a muxer.
5812 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5814 * gst/rtsp-server/rtsp-media.c:
5815 * gst/rtsp-server/rtsp-media.h:
5816 media: use multiudpsink send-duplicates when we can
5817 If we have a new enough multiudpsink with the send-duplicates property, use this
5818 instead of doing our own filtering. Our custom filtering code should eventually
5819 be removed when we can depend on a released -good.
5821 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5823 * gst/rtsp-server/rtsp-media.c:
5824 media: don't leak destinations
5825 Refactor and cleanup the destinations array when the stream is destroyed.
5827 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5829 * gst/rtsp-server/rtsp-media.c:
5830 * gst/rtsp-server/rtsp-media.h:
5831 media: don't add udp addresses multiple times
5832 Keep track of the udp addresses we added to udpsink and never add the same udp
5833 destination twice. This avoids duplicate packets when using multicast.
5835 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5837 * gst/rtsp-server/rtsp-server.c:
5838 server: disable use of SO_LINGER
5839 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
5840 server close()s the connection.
5842 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5844 * gst/rtsp-server/rtsp-server.c:
5845 server: use 5 second linger period in SO_LINGER
5846 Wait 5 seconds before clearing the send buffers and reseting the connection with
5847 the client when we do a close. This should be enough time to get the message to
5851 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
5853 * gst/rtsp-server/rtsp-server.c:
5854 server: use SO_LINGER
5855 SO_LINGER on the socket will make sure that any pending data on the socket is
5856 flushed ASAP and that the socket connection is reset. This makes sure that the
5857 socket can be reused immediately.
5860 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5863 README: add blurb about shared media factories
5865 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
5867 * gst/rtsp-server/rtsp-media.c:
5868 Add stdlib.h for atoi()
5870 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5872 * bindings/python/Makefile.am:
5873 * bindings/vala/Makefile.am:
5874 build: distcheck fixes
5875 Fix 'make distcheck', somewhat (it still fails because it tries to
5876 install files into /usr/share/vala/vapi/ irrespective of the
5879 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5882 configure: bump core/base requirements to released version
5883 Makes things less confusing for people.
5885 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5888 configure: fail if GStreamer core/base requirements are not met
5890 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5892 * gst/rtsp-server/rtsp-client.c:
5893 client: improve client cleanups
5894 Make sure the session does not timeout when using TCP. We need to do this
5895 because quicktime player does not send RTCP for some reason in tunneled
5897 Refactor some cleanup code.
5900 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5902 * gst/rtsp-server/rtsp-session.c:
5903 * gst/rtsp-server/rtsp-session.h:
5904 session: add support for prevent session timeouts
5905 Add an atomix counter to prevent session timeouts when we are, for example,
5908 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5910 * gst/rtsp-server/rtsp-client.c:
5911 client: fix unlink on session timeouts
5912 When our session times out, make sure we unlink all streams in this
5914 Remove the tunnelid when closing the connection.
5916 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5918 * gst/rtsp-server/rtsp-session.c:
5919 session: small cleanups
5921 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5923 * gst/rtsp-server/rtsp-client.c:
5924 client: handle lost_tunnel callbacks
5925 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
5926 hashtable so that we can reuse it for when the client reopens the POST
5928 Close the connection after a TEARDOWN.
5929 Make sure or watchid is cleared when the watch is removed.
5932 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5934 * gst/rtsp-server/rtsp-client.c:
5935 * gst/rtsp-server/rtsp-media.c:
5936 * gst/rtsp-server/rtsp-sdp.c:
5937 rtsp-server: add more support for multicast
5939 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5942 * gst/rtsp-server/rtsp-media.c:
5943 * gst/rtsp-server/rtsp-media.h:
5944 media: allow configuration of allowed lower transport
5946 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5948 * gst/rtsp-server/rtsp-client.h:
5949 * gst/rtsp-server/rtsp-media.c:
5950 * gst/rtsp-server/rtsp-media.h:
5951 * gst/rtsp-server/rtsp-sdp.c:
5952 * gst/rtsp-server/rtsp-sdp.h:
5953 * gst/rtsp-server/rtsp-server.c:
5954 rtsp: keep track of server ip and ipv6
5955 Keep track of how the client connected to the server and setup the udp ports
5956 with the same protocol.
5957 Copy the server ip address in the SDP so that clients can send RTCP back to
5960 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5962 * gst/rtsp-server/rtsp-session.c:
5965 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5967 * gst/rtsp-server/rtsp-client.c:
5968 client: use right size for malloc
5970 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5972 * gst/rtsp-server/rtsp-server.c:
5973 server: comment ipv6 server listening address
5975 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5977 * gst/rtsp-server/rtsp-media.c:
5978 media: allow for ipv6 sockets
5980 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5982 * gst/rtsp-server/rtsp-server.c:
5983 * gst/rtsp-server/rtsp-server.h:
5984 server: rework server part
5985 Allow setting a bind address, make sure we can deal with ipv6.
5986 Remove the port property and change with the service property.
5988 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5990 * gst/rtsp-server/rtsp-media.h:
5991 media: update comments a little
5993 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5995 * gst/rtsp-server/rtsp-client.c:
5996 client: make content-base better
5997 Use the URI formatting functions to make a content-base. Also make sure that
5998 there is a trailing / at the end.
6000 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6002 * gst/rtsp-server/rtsp-client.c:
6003 client: guard against invalid paths
6005 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6007 * examples/test-video.c:
6008 test: catch server bind errors
6010 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
6012 * gst/rtsp-server/rtsp-media.c:
6013 rtspmedia: emit "unprepared" if _prepare fails.
6014 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
6015 media object is removed from its factory's cache.
6017 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6019 * gst/rtsp-server/rtsp-media.c:
6020 media: collect media position when seek completes
6022 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
6024 * gst/rtsp-server/rtsp-client.c:
6025 client: call unlink_streams in client finalize
6028 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6030 * gst/rtsp-server/rtsp-media.c:
6031 media: limit the time to wait to something huge
6032 Avoid waiting forever but limit the timeout to 20 seconds.
6034 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6036 * gst/rtsp-server/rtsp-sdp.c:
6037 sdp: reindent and check for prepared status
6039 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6041 * gst/rtsp-server/rtsp-media.c:
6042 * gst/rtsp-server/rtsp-media.h:
6043 * gst/rtsp-server/rtsp-session.c:
6044 media: avoid doing _get_state() for state changes
6045 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
6046 until the media is prerolled or in error. This avoids doing a blocking call of
6047 gst_element_get_state() that can cause lockups when there is an error.
6050 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6052 * gst/rtsp-server/rtsp-media.c:
6055 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6057 * gst/rtsp-server/rtsp-media-factory.c:
6058 media-factory: better error handling
6059 Improve the error handling a bit.
6061 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6063 * gst/rtsp-server/rtsp-client.c:
6064 client: rework transport parsing
6065 Rework the transport parsing code so that we can ignore transports we don't
6066 support instead of just picking the first one we can parse.
6067 Configure a (for now hardcoded) destination for multicast transports.
6069 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6071 * gst/rtsp-server/rtsp-media.c:
6072 media: set multicast sink parameters
6073 Disable loop and automatic multicast join on the udpsink elements.
6074 Add some more debug info.
6075 Reset some state variables in the right place.
6076 Use the right port numbers for multicast.
6078 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6080 * gst/rtsp-server/rtsp-session.c:
6081 session: handle transport setup correctly
6082 Handle UDP, MCAST and TCP transport negotiation more correctly.
6083 Store the server session SSRC in the transport.
6085 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6087 * gst/rtsp-server/rtsp-client.c:
6088 rtsp-client: implement error_full
6089 Implement error_full to avoid some segfaults when the rtspconnection calls it.
6092 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6095 * gst/rtsp-server/rtsp-client.c:
6096 * gst/rtsp-server/rtsp-server.c:
6097 docs: update docs and comments
6099 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
6101 * gst/rtsp-server/rtsp-sdp.c:
6102 sdp: make server work better when behind a proxy
6104 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6106 * gst/rtsp-server/rtsp-client.c:
6107 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
6109 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6111 * gst/rtsp-server/rtsp-client.c:
6112 * gst/rtsp-server/rtsp-media-factory.c:
6113 * gst/rtsp-server/rtsp-media-mapping.c:
6114 * gst/rtsp-server/rtsp-media.c:
6115 * gst/rtsp-server/rtsp-server.c:
6116 * gst/rtsp-server/rtsp-session-pool.c:
6117 * gst/rtsp-server/rtsp-session.c:
6118 Use GStreamer's debugging subsystem
6120 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6122 * gst/rtsp-server/rtsp-media-factory.c:
6123 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
6125 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6130 === release 0.10.5 ===
6132 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6137 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6140 configure: bump required versions
6142 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
6144 * gst/rtsp-server/rtsp-client.c:
6145 client: call weak-unref on client->sessions from finalize
6148 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6150 * gst/rtsp-server/rtsp-media.c:
6151 media: Fixed crasher where caps got unref'ed too often
6153 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6156 * pkgconfig/.gitignore:
6157 * pkgconfig/Makefile.am:
6158 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
6159 Added pkg-config file to use gst-rtsp-server uninstalled
6161 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6163 * gst/rtsp-server/rtsp-media.c:
6164 media: add some docs
6166 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
6168 * gst/rtsp-server/rtsp-client.c:
6169 rtsp: Use gst_rtsp_watch_send_message().
6170 Use gst_rtsp_watch_send_message() since the old API which used
6171 gst_rtsp_watch_queue_message() has been deprecated.
6173 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6178 === release 0.10.4 ===
6180 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6185 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6187 * gst/rtsp-server/rtsp-client.c:
6188 * gst/rtsp-server/rtsp-session.c:
6189 * gst/rtsp-server/rtsp-session.h:
6190 rtsp: allocate channels in TCP mode
6191 When the client does not provide us with channels in TCP mode, allocate channels
6194 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6196 * gst/rtsp-server/rtsp-client.c:
6197 client: don't crash when tunnelid is missing
6198 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
6199 don't crash but return an error response to the client.
6202 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6204 * bindings/vala/gst-rtsp-server-0.10.vapi:
6205 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6206 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6207 bindings: update vala bindings with new method
6209 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6211 * gst/rtsp-server/rtsp-session-pool.c:
6212 * gst/rtsp-server/rtsp-session-pool.h:
6213 sessionpool: add function to filter sessions
6214 Add generic function to retrieve/remove sessions.
6216 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6219 configure: bump core/base requirements to release
6221 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6223 * gst/rtsp-server/rtsp-media.c:
6224 media: fix indentation
6226 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6228 * gst/rtsp-server/rtsp-media.c:
6229 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
6231 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6233 * gst/rtsp-server/rtsp-media.c:
6234 set state and remove elements of media in for loop
6236 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
6238 * bindings/vala/gst-rtsp-server-0.10.vapi:
6239 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6240 Added gst_rtsp_media_remove_elements function to Vala bindings
6242 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
6244 * gst/rtsp-server/rtsp-media.c:
6245 * gst/rtsp-server/rtsp-media.h:
6246 Added gst_rtsp_media_remove_elements function
6248 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
6250 * gst/rtsp-server/rtsp-media.c:
6251 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
6253 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6255 * bindings/vala/gst-rtsp-server-0.10.vapi:
6256 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6257 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6258 Updated Vala bindings
6260 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6262 * gst/rtsp-server/rtsp-media.c:
6263 * gst/rtsp-server/rtsp-media.h:
6264 Added vmethod unprepare to GstRTSPMedia
6265 The default implementation sets the state of the pipeline to GST_STATE_NULL
6267 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6269 * gst/rtsp-server/rtsp-media-factory.c:
6270 * gst/rtsp-server/rtsp-media-factory.h:
6271 Made collect_streams function public
6273 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6275 * gst/rtsp-server/rtsp-media-factory.c:
6276 * gst/rtsp-server/rtsp-media-factory.h:
6277 * gst/rtsp-server/rtsp-media.c:
6278 Added vmethod create_pipeline to GstRTSPMediaFactory
6279 The pipeline is created in this method and the GstRTSPMedia's element is added to it
6281 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6283 * gst/rtsp-server/rtsp-client.c:
6284 client: use g_source_destroy()
6285 We need to use g_source_destroy() because we might have added the source to a
6286 different main context than the default one.
6288 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6290 * gst/rtsp-server/Makefile.am:
6291 * gst/rtsp-server/rtsp-client.c:
6292 * gst/rtsp-server/rtsp-params.c:
6293 * gst/rtsp-server/rtsp-params.h:
6294 rtsp: prepare for handling GET/SET_PARAMETER
6295 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
6297 Fix return codes of handlers.
6299 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6301 * gst/rtsp-server/rtsp-media.c:
6302 media: don't leak session pads
6304 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6306 * gst/rtsp-server/rtsp-media.c:
6307 media: clean up the messages a bit
6309 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6311 * gst/rtsp-server/rtsp-sdp.c:
6312 sdp: warn and skip streams without media
6314 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6316 * bindings/vala/gst-rtsp-server-0.10.vapi:
6317 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6318 vala: Fixed typo in header file of RTSPMediaStream
6320 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6322 * gst/rtsp-server/rtsp-media.c:
6325 Make dumping RTCP stats configurable
6327 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6329 * gst/rtsp-server/rtsp-media.c:
6330 media: be less verbose and leak less
6332 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6334 * gst/rtsp-server/rtsp-media.c:
6335 media: don't leak the destination address
6337 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6339 * gst/rtsp-server/rtsp-client.c:
6340 * gst/rtsp-server/rtsp-media.c:
6341 * gst/rtsp-server/rtsp-media.h:
6342 * gst/rtsp-server/rtsp-session.c:
6343 * gst/rtsp-server/rtsp-session.h:
6344 rtsp: use RTCP to keep the session alive
6345 Use the RTCP rtcp-from stats field to find the associated session and use this
6346 to keep the session alive.
6348 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6350 * gst/rtsp-server/rtsp-session.c:
6351 session: add 5sec to the real session timeout
6352 Allow the session to live 5sec longer before really timing out. This should give
6353 clients some extra time to keep the session active.
6355 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6357 * gst/rtsp-server/rtsp-client.c:
6358 client: replay OK to GET/SET_PARAMETER
6359 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
6360 so that we return OK for those requests.
6362 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6364 * gst/rtsp-server/rtsp-media.c:
6365 * gst/rtsp-server/rtsp-media.h:
6366 media: keep track of active transports
6367 Keep track of which transport is active to avoid closing the connection too
6369 Remove the destination transport also when going to NULL.
6370 Print some stats about the SDES and other RTCP messages we receive from the
6373 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6375 * examples/.gitignore:
6376 * examples/Makefile.am:
6377 * examples/test-sdp.c:
6378 example: add SDP relay example
6380 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6382 * gst/rtsp-server/rtsp-media.c:
6383 media: also count active TCP connections
6385 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6387 * gst/rtsp-server/rtsp-media-factory.c:
6388 * gst/rtsp-server/rtsp-media.c:
6389 * gst/rtsp-server/rtsp-media.h:
6390 rtsp: add support for dynamic elements
6391 Add support for dynamic elements.
6392 Don't set live pipelines back to paused.
6394 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6396 * gst/rtsp-server/rtsp-sdp.c:
6397 sdp: don't add encoding name when absent in caps
6399 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6401 * gst/rtsp-server/rtsp-client.c:
6402 client: warn when we can't do RTP-Info
6404 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6406 * gst/rtsp-server/rtsp-media-factory.c:
6407 factory: factor out the stream construction
6409 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6411 * gst/rtsp-server/rtsp-client.c:
6412 client: only add RTP-Info when we have the info
6413 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
6416 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6421 === release 0.10.3 ===
6423 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6427 - Fixes a bug where it put the wrong verion in pkgconfig
6428 - Link RTP and RTCP sources
6430 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6432 * gst/rtsp-server/rtsp-media.c:
6433 * gst/rtsp-server/rtsp-media.h:
6434 media: link the RTP udpsrc to the session manager
6435 Link the RTP udpsrc and the appsrc to the session manager so that they don't
6436 shut down when the client sends a packet to open firewalls.
6438 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6440 * pkgconfig/gst-rtsp-server.pc.in:
6441 Don't use hard-coded version number in pkg-config file
6443 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6448 === release 0.10.2 ===
6450 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6455 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6458 * common/m4/.gitignore:
6459 * examples/.gitignore:
6460 * pkgconfig/.gitignore:
6461 add some .gitignore files
6463 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6465 * gst/rtsp-server/rtsp-media.c:
6466 media: seek to key frames
6468 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6470 * gst/rtsp-server/rtsp-media.c:
6471 media: emit the unprepared signal by id
6472 Emit the unprepared signal by id instead of name and set the media as
6475 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6477 * gst/rtsp-server/rtsp-media.c:
6478 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
6480 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6482 * gst/rtsp-server/rtsp-server.c:
6483 Added finalize function to GstRTPSPServer to unref session pool and media mapping
6485 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6487 * bindings/vala/gst-rtsp-server-0.10.vapi:
6488 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6489 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6490 Updated vala bindings
6492 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6494 * gst/rtsp-server/Makefile.am:
6495 * gst/rtsp-server/rtsp-client.c:
6496 * gst/rtsp-server/rtsp-media.c:
6497 server: use appsink and appsrc with the API
6498 Use the appsink/appsrc API instead of the signals for higher
6501 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6503 * examples/test-ogg.c:
6504 tests: set the payload type correctly
6506 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6508 * gst/rtsp-server/rtsp-media-factory.c:
6509 factory: connect to the unprepare signal
6510 Connect to the unprepare signal for non-reusable media so that we can remove
6511 them from the cache.
6513 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6515 * gst/rtsp-server/rtsp-media.c:
6516 * gst/rtsp-server/rtsp-media.h:
6517 media: add signal to notify of unprepare
6519 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6521 * gst/rtsp-server/rtsp-media.c:
6522 * gst/rtsp-server/rtsp-media.h:
6523 media: more work on making the media shared
6524 Add a reusable flag to medias, indicating that they can be reused after a state
6528 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6530 * examples/test-readme.c:
6531 examples: mark the example as shared for testing
6533 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6535 * gst/rtsp-server/rtsp-media.c:
6536 * gst/rtsp-server/rtsp-media.h:
6537 client: support shared media
6538 Always perform the state actions even if the target state of the pipeline is
6539 already correct, we still want to add/remove the transports when we are dealing
6541 Keep a counter of the number of active transports for a media so that we can use
6542 this to perform a state change when needed.
6543 Perform a state change of the pipeline only when the first transport was added
6544 or when there are no active transports.
6546 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6548 * gst/rtsp-server/rtsp-client.c:
6549 client: fix refcounting crasher
6550 Don't need to remove the weak refs in the finalize methods, they are already
6551 removed in the dispose.
6552 Don't register the callback with a DestroyNofity.
6554 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6556 * gst/rtsp-server/rtsp-client.c:
6557 Fix rtsp client refcount management in TCP mode.
6558 Don't unref a client ref we never had. Fixes an unref
6559 of an already-free client object after a client
6560 teardown request for me.
6562 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6564 * gst/rtsp-server/rtsp-session.c:
6565 docs: fix typo in API docs
6567 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6569 * gst/rtsp-server/rtsp-media.c:
6571 Keep the udp sources in playing even if we go to paused. unlock the sources when
6573 Add some more debug info.
6574 Only seek when we need to.
6575 Keep track of the position when we go to paused.
6577 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6579 * gst/rtsp-server/rtsp-client.c:
6580 * gst/rtsp-server/rtsp-media.c:
6581 * gst/rtsp-server/rtsp-media.h:
6582 Add beginnings of seeking.
6583 Parse the Range header and perform a seek on the pipeline for the requested
6584 position. It's disabled currently until I figure out what's going wrong.
6586 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6588 * gst/rtsp-server/rtsp-client.c:
6589 allow pause requests for now.
6592 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6594 * gst/rtsp-server/rtsp-client.c:
6595 Remove weak ref on the session in teardown
6596 We need to remove our weakref from the session when we do a teardown because
6597 else we close the TCP connection prematurely.
6599 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6601 * gst/rtsp-server/rtsp-client.c:
6602 * gst/rtsp-server/rtsp-client.h:
6603 * gst/rtsp-server/rtsp-session-pool.c:
6604 Do some more session cleanup
6605 Make session timeout kill the TCP connection that currently watches the
6607 Remove the client timeout property.
6609 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6611 * gst/rtsp-server/rtsp-client.c:
6612 * gst/rtsp-server/rtsp-client.h:
6613 * gst/rtsp-server/rtsp-media.c:
6614 * gst/rtsp-server/rtsp-media.h:
6615 * gst/rtsp-server/rtsp-server.c:
6616 * gst/rtsp-server/rtsp-session.c:
6617 * gst/rtsp-server/rtsp-session.h:
6619 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
6622 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6624 * examples/Makefile.am:
6625 * examples/test-launch.c:
6626 Add example server that takes launch lines
6627 Add an example server that streams any -launch line.
6629 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6631 * examples/test-readme.c:
6632 * gst/rtsp-server/rtsp-client.c:
6633 * gst/rtsp-server/rtsp-media.c:
6634 * gst/rtsp-server/rtsp-media.h:
6635 Add support for live streams
6636 Add support for live streams and ranges
6637 Start on handling TCP data transfer.
6639 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6641 * gst/rtsp-server/rtsp-media.c:
6642 Free the pipeline before other things
6645 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6647 * gst/rtsp-server/rtsp-client.c:
6648 Only free the pending tunnel if there is one
6651 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6653 * gst/rtsp-server/rtsp-client.c:
6654 * gst/rtsp-server/rtsp-client.h:
6655 * gst/rtsp-server/rtsp-media.c:
6656 rtsp-server: Add support for tunneling
6657 Add support for tunneling over HTTP.
6658 Use new connection methods to retrieve the url.
6659 Dispatch messages based on the message type instead of blindly
6660 assuming it's always a request.
6661 Keep track of the watch id so that we can remove it later.
6662 Set the media pipeline to NULL before unreffing the pipeline.
6664 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6666 * gst/rtsp-server/rtsp-client.c:
6667 * gst/rtsp-server/rtsp-client.h:
6668 Fix for channel -> watch rename in gstreamer
6669 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
6671 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6673 * gst/rtsp-server/rtsp-client.c:
6674 * gst/rtsp-server/rtsp-client.h:
6676 Use the async RTSP channels instead of spawning a new thread for each client.
6677 If a sessionid is specified in a request, fail if we don't have the session.
6679 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6681 * gst/rtsp-server/rtsp-media.c:
6682 Add better debug info
6683 Add some better debug info.
6685 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6687 * examples/test-video.c:
6689 Add support for session timeouts in the example.
6691 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6693 * gst/rtsp-server/rtsp-session-pool.c:
6694 * gst/rtsp-server/rtsp-session-pool.h:
6695 Pass GTimeVal around for performance reasons
6696 Get the current time only once and pass it around so that sessions don't have to
6697 get the current time anymore.
6698 Add experimental support for a GSource that dispatches when the session needs to
6701 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6703 * gst/rtsp-server/rtsp-session.c:
6704 * gst/rtsp-server/rtsp-session.h:
6705 Add better support for session timeouts
6706 Add a method to request the number of milliseconds when a session will timeout.
6708 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6710 * gst/rtsp-server/rtsp-media.c:
6711 * gst/rtsp-server/rtsp-media.h:
6712 Add suport for RTP manager monitoring
6713 Add the first stage in monitoring the rtp manager.
6714 Make sure we don't update the state to something we don't want.
6716 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6718 * gst/rtsp-server/rtsp-client.c:
6719 Add support for session keepalive
6720 Get and update the session timeout for all requests. get the session as early as
6723 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6725 * gst/rtsp-server/rtsp-media-factory.h:
6726 * gst/rtsp-server/rtsp-media.c:
6727 * gst/rtsp-server/rtsp-media.h:
6728 Handle media bus messages
6729 Handle media bus messages in a custom mainloop and dispatch them to the
6730 RTSPMedia objects. Let the default implementation handle some common messages.
6732 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6734 * gst/rtsp-server/rtsp-client.c:
6735 * gst/rtsp-server/rtsp-session-pool.c:
6736 * gst/rtsp-server/rtsp-session.c:
6737 Some more session timeout handling
6738 Move the session header setting code to a central place so that we always add
6739 the timeout parameter too.
6740 Handle timeouts by running the session cleanup code.
6741 Stop media before cleaning up.
6743 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6745 * gst/rtsp-server/rtsp-client.c:
6746 * gst/rtsp-server/rtsp-client.h:
6747 Add timeout property
6748 Add a timeout property ot the client and make the other properties into GObject
6751 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6753 * gst/rtsp-server/rtsp-session-pool.c:
6754 Use getters and setters in property code
6755 Use the getters and setters for the timeout property instead of locking
6758 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6760 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
6762 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6764 * gst/rtsp-server/rtsp-session-pool.c:
6765 * gst/rtsp-server/rtsp-session-pool.h:
6766 * gst/rtsp-server/rtsp-session.c:
6767 * gst/rtsp-server/rtsp-session.h:
6768 Add more timeout stuff
6769 Add method to check if a session is expired.
6770 Add method to perform cleanup on a session pool.
6772 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6774 * gst/rtsp-server/rtsp-client.c:
6775 * gst/rtsp-server/rtsp-session-pool.c:
6776 * gst/rtsp-server/rtsp-session-pool.h:
6777 * gst/rtsp-server/rtsp-session.c:
6778 * gst/rtsp-server/rtsp-session.h:
6779 Add beginnings of session timeouts and limits
6780 Add the timeout value to the Session header for unusual timeout values.
6781 Allow us to configure a limit to the amount of active sessions in a pool. Set a
6782 limit on the amount of retry we do after a sessionid collision.
6783 Add properties to the sessionid and the timeout of a session. Keep track of
6784 creation time and last access time for sessions.
6786 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6788 * gst/rtsp-server/rtsp-client.c:
6789 * gst/rtsp-server/rtsp-media.c:
6790 * gst/rtsp-server/rtsp-media.h:
6791 * gst/rtsp-server/rtsp-sdp.c:
6792 * gst/rtsp-server/rtsp-session-pool.c:
6793 * gst/rtsp-server/rtsp-session.c:
6794 * gst/rtsp-server/rtsp-session.h:
6795 Cleanup of sessions and more
6796 Fix the refcounting of media and sessions in the client. Properly clean up the
6797 session data when the client performs a teardown.
6798 Add Server header to responses.
6799 Allow for multiple uri setups in one session.
6800 Add Range header to the PLAY response and add the range attribute to the SDP
6802 Fix the session pool remove method, it used the wrong key in the hashtable. Also
6803 give the ownership of the sessionid to the session object.
6805 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6807 * gst/rtsp-server/rtsp-server.c:
6808 * gst/rtsp-server/rtsp-server.h:
6810 Rename the 'server_port' variable to simply 'port'.
6812 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6815 * gst/rtsp-server/rtsp-client.c:
6816 * gst/rtsp-server/rtsp-media.c:
6817 * gst/rtsp-server/rtsp-media.h:
6818 * gst/rtsp-server/rtsp-session.c:
6819 * gst/rtsp-server/rtsp-session.h:
6820 Rework the way we handle transports for streams
6821 Make the media accept an array of transports for the streams that we have
6822 configured for the play/pause requests.
6823 Implement server states for a client and its media.
6824 Require 0.10.22.1 (git HEAD) of gstreamer.
6826 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6828 * gst/rtsp-server/rtsp-client.c:
6829 * gst/rtsp-server/rtsp-media-factory.c:
6830 Drop const from functions dealing with urls
6831 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
6832 have the right const in them.
6834 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6836 * gst/rtsp-server/rtsp-client.c:
6837 * gst/rtsp-server/rtsp-media.c:
6838 * gst/rtsp-server/rtsp-sdp.c:
6842 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6844 * gst/rtsp-server/rtsp-client.c:
6845 * gst/rtsp-server/rtsp-media-factory.c:
6846 * gst/rtsp-server/rtsp-media.c:
6847 * gst/rtsp-server/rtsp-media.h:
6849 Don't keep a reference to the GstRTSPMedia in the stream.
6850 Free more things when freeing the GstRTSPMedia.
6852 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6855 * gst/rtsp-server/rtsp-media-factory.c:
6856 * gst/rtsp-server/rtsp-media-factory.h:
6857 * gst/rtsp-server/rtsp-media.c:
6858 * gst/rtsp-server/rtsp-media.h:
6859 * gst/rtsp-server/rtsp-server.c:
6860 * gst/rtsp-server/rtsp-server.h:
6861 More docs and small cleanups
6862 Add some more docs and update the README
6863 Cleanup some method names.
6864 Remove an unneeded idx field in the GstRTSPMediaStream
6866 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6869 * examples/Makefile.am:
6870 * examples/test-readme.c:
6871 Add a README and more example code
6872 Add a README file that contains a small introduction on how to use the server
6873 along with the example code explained in the readme.
6875 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6877 * gst/rtsp-server/rtsp-media.c:
6878 * gst/rtsp-server/rtsp-server.c:
6879 Fix some leaks and change default port
6880 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
6881 we finished the initial preroll. If we keep them locked, setting the pipeline to
6882 NULL will not stop and clean up the sources correctly.
6883 Change the default RTSP port to 8554 aka the official alternative RTSP port.
6885 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6887 * gst/rtsp-server/rtsp-session.c:
6888 * gst/rtsp-server/rtsp-session.h:
6889 Cleanups to the session object
6890 Remove some unneeded variables in the session state of a stream such as the
6891 owner media and the server transport.
6892 Get the configuration of a media stream in a session based on the media_stream
6893 in the original object instead of our cached index.
6894 Free more data in the finalize method.
6896 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6898 * gst/rtsp-server/rtsp-client.c:
6899 * gst/rtsp-server/rtsp-client.h:
6900 Cleanups and reuse media from DESCRIBE
6901 Handle thread create errors.
6902 Rename some internal methods to better match what they actually do.
6903 Handle misconfiguration of session_pool and media_mapping gracefully.
6904 Cache the DESCRIBE media and uri in the client connection and reuse them when
6905 we receive a SETUP request in the same connection for the same uri.
6906 Cleanup the client connection object.
6908 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6910 * gst/rtsp-server/rtsp-media-factory.c:
6911 * gst/rtsp-server/rtsp-media-factory.h:
6912 * gst/rtsp-server/rtsp-media.c:
6913 * gst/rtsp-server/rtsp-media.h:
6914 Add shared properties to media and factory
6915 Add the shared property to media.
6916 Implement some simple caching in the factory depending on if the media is shared
6919 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6921 * gst/rtsp-server/rtsp-client.c:
6922 Add a little comment
6923 Add some comment about the content-base header.
6925 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6927 * examples/Makefile.am:
6929 * examples/test-mp4.c:
6930 * examples/test-ogg.c:
6931 * examples/test-video.c:
6932 * gst/rtsp-server/Makefile.am:
6933 * gst/rtsp-server/rtsp-client.c:
6934 * gst/rtsp-server/rtsp-client.h:
6935 * gst/rtsp-server/rtsp-media-factory.c:
6936 * gst/rtsp-server/rtsp-media-factory.h:
6937 * gst/rtsp-server/rtsp-media.c:
6938 * gst/rtsp-server/rtsp-media.h:
6939 * gst/rtsp-server/rtsp-sdp.c:
6940 * gst/rtsp-server/rtsp-sdp.h:
6941 * gst/rtsp-server/rtsp-server.c:
6942 * gst/rtsp-server/rtsp-server.h:
6943 * gst/rtsp-server/rtsp-session.c:
6944 * gst/rtsp-server/rtsp-session.h:
6945 Reorganize things, prepare for media sharing
6946 Added various other test server examples
6947 Move the SDP message generation to a separate helper.
6948 Refactor common code for finding the session.
6949 Add content-base for realplayer compatibility
6950 Clean up request uris before processing for better vlc compatibility.
6951 Move prerolling and pipeline construction to the RTSPMedia object.
6952 Use multiudpsink for future pipeline reuse.
6954 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6960 === release 0.10.1 ===
6962 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6968 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6970 * bindings/vala/Makefile.am:
6972 Add more directories and files to the dist.
6974 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6976 * bindings/python/Makefile.am:
6977 * bindings/python/rtspserver.override:
6978 Fixed compile error of python bindings
6980 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6982 * bindings/vala/gst-rtsp-server-0.10.vapi:
6983 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6984 Marked values as nullable accordingly
6986 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6988 * bindings/vala/gst-rtsp-server-0.10.vapi:
6989 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
6990 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6991 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6992 Updated Vala bindings
6994 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6996 * gst/rtsp-server/rtsp-client.c:
6997 * gst/rtsp-server/rtsp-media-mapping.c:
6998 * gst/rtsp-server/rtsp-media-mapping.h:
6999 * gst/rtsp-server/rtsp-media.h:
7000 * gst/rtsp-server/rtsp-session-pool.h:
7001 Cleanups and doc updates
7002 Add some more documentation and do some minor cleanups here and there.
7004 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7006 * gst/rtsp-server/rtsp-client.c:
7007 * gst/rtsp-server/rtsp-media-factory.c:
7008 * gst/rtsp-server/rtsp-media-factory.h:
7009 * gst/rtsp-server/rtsp-media.c:
7010 * gst/rtsp-server/rtsp-media.h:
7011 * gst/rtsp-server/rtsp-session.c:
7012 * gst/rtsp-server/rtsp-session.h:
7014 Rename GstRTSPMediaBin to GstRTSPMedia
7015 Parse the request url into a GstRTSPUri object and pass this object to the
7016 various handlers and methods that require the uri.
7018 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7022 Add some more docs and remove some old code from the example.
7024 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7026 * gst/rtsp-server/rtsp-client.c:
7027 Handle state change failures better
7028 Handle state change failures better when changing the state of the pipeline to
7031 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7033 * gst/rtsp-server/rtsp-media-factory.c:
7034 * gst/rtsp-server/rtsp-media-factory.h:
7035 Make element creation more extendible
7036 Add get_element vmethod to the default MediaFactory so that subclasses can just
7037 override that method and still use the default logic for making a MediaBin from
7040 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7043 * gst/rtsp-server/Makefile.am:
7044 * gst/rtsp-server/rtsp-client.c:
7045 * gst/rtsp-server/rtsp-client.h:
7046 * gst/rtsp-server/rtsp-media-factory.c:
7047 * gst/rtsp-server/rtsp-media-factory.h:
7048 * gst/rtsp-server/rtsp-media-mapping.c:
7049 * gst/rtsp-server/rtsp-media-mapping.h:
7050 * gst/rtsp-server/rtsp-media.c:
7051 * gst/rtsp-server/rtsp-media.h:
7052 * gst/rtsp-server/rtsp-server.c:
7053 * gst/rtsp-server/rtsp-server.h:
7054 * gst/rtsp-server/rtsp-session.c:
7055 * gst/rtsp-server/rtsp-session.h:
7056 Make the server handle arbitrary pipelines
7057 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
7058 The GstMediaBin object has a handle to a bin with elements and to a list of
7059 GstMediaStream objects that this bin produces.
7060 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
7061 with methods to register and remove those mappings.
7062 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
7063 used by the server instance.
7064 Modify the example application so that it shows how to create custom pipelines
7065 attached to a specific mount point.
7066 Various misc cleanps.
7068 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7070 * gst/rtsp-server/rtsp-server.c:
7071 * gst/rtsp-server/rtsp-server.h:
7072 Allow setting a custom media factory for a server
7074 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7076 * gst/rtsp-server/rtsp-client.c:
7077 * gst/rtsp-server/rtsp-client.h:
7078 Allow setting a custom media factory for a client.
7080 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7082 * gst/rtsp-server/Makefile.am:
7083 Add Makefile entry for the media factory
7085 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7087 * gst/rtsp-server/rtsp-media-factory.c:
7088 * gst/rtsp-server/rtsp-media-factory.h:
7089 Add media factory to map urls to media pipeline objects.
7091 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7093 * gst/rtsp-server/rtsp-media.c:
7094 * gst/rtsp-server/rtsp-media.h:
7095 Add comments. Remove unused field
7097 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7099 * gst/rtsp-server/rtsp-session-pool.c:
7100 * gst/rtsp-server/rtsp-session-pool.h:
7101 Allow custom session pools to override the session id allocation algorithms Add some comments.
7103 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7105 * gst/rtsp-server/rtsp-session.h:
7108 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7110 * gst/rtsp-server/rtsp-client.c:
7111 * gst/rtsp-server/rtsp-client.h:
7112 Move the connection code in one place Add some comments
7114 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7116 * gst/rtsp-server/rtsp-server.c:
7117 * gst/rtsp-server/rtsp-server.h:
7118 Make vmethod to create and accept new clients. Add some docs.
7120 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7122 * gst/rtsp-server/rtsp-server.c:
7123 * gst/rtsp-server/rtsp-server.h:
7124 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
7126 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7128 * gst/rtsp-server/rtsp-client.c:
7129 * gst/rtsp-server/rtsp-client.h:
7130 Name the parameters more appropriately.
7132 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7134 * gst/rtsp-server/rtsp-session-pool.c:
7135 Do some more cleanup of the session pool.
7137 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7139 * gst/rtsp-server/Makefile.am:
7140 * gst/rtsp-server/rtsp-client.c:
7141 Check if return value of gst_rtsp_session_get_media is not NULL
7143 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7145 * gst/rtsp-server/Makefile.am:
7146 Install rtsp-session and rtsp-session-pool headers
7148 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7153 * bindings/python/Makefile.am:
7154 * bindings/python/arg-types.py:
7155 * bindings/python/codegen/Makefile.am:
7156 * bindings/python/codegen/__init__.py:
7157 * bindings/python/codegen/argtypes.py:
7158 * bindings/python/codegen/code-coverage.py:
7159 * bindings/python/codegen/codegen.py:
7160 * bindings/python/codegen/definitions.py:
7161 * bindings/python/codegen/defsparser.py:
7162 * bindings/python/codegen/docextract.py:
7163 * bindings/python/codegen/docgen.py:
7164 * bindings/python/codegen/fileprefix.override:
7165 * bindings/python/codegen/fileprefixmodule.c:
7166 * bindings/python/codegen/h2def.py:
7167 * bindings/python/codegen/mergedefs.py:
7168 * bindings/python/codegen/mkskel.py:
7169 * bindings/python/codegen/override.py:
7170 * bindings/python/codegen/reversewrapper.py:
7171 * bindings/python/codegen/scmexpr.py:
7172 * bindings/python/rtspserver-types.defs:
7173 * bindings/python/rtspserver.defs:
7174 * bindings/python/rtspserver.override:
7175 * bindings/python/rtspservermodule.c:
7177 Add python bindings.
7179 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7181 * bindings/Makefile.am:
7183 Don't go into python dir when requirements for python bindings are missing
7185 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7187 * bindings/Makefile.am:
7188 * bindings/vala/Makefile.am:
7190 Install Vala bindings if vala is available
7192 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7194 * bindings/vala/gst-rtsp-server-0.10.deps:
7195 * bindings/vala/gst-rtsp-server-0.10.vapi:
7196 * bindings/vala/gst-rtsp-server.vapi:
7197 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
7198 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
7199 * bindings/vala/packages/gst-rtsp-server-0.10.files:
7200 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7201 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7202 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
7203 * bindings/vala/packages/gst-rtsp-server.deps:
7204 * bindings/vala/packages/gst-rtsp-server.excludes:
7205 * bindings/vala/packages/gst-rtsp-server.files:
7206 * bindings/vala/packages/gst-rtsp-server.gi:
7207 * bindings/vala/packages/gst-rtsp-server.metadata:
7208 * bindings/vala/packages/gst-rtsp-server.namespace:
7209 Regenerated Vala bindings
7211 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7213 * bindings/vala/gst-rtsp-server.vapi:
7214 * bindings/vala/packages/gst-rtsp-server.metadata:
7215 Fixed typo in included headers for vala bindings
7217 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7221 * pkgconfig/Makefile.am:
7222 * pkgconfig/gst-rtsp-server.pc.in:
7223 Added pkgconfig file
7225 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
7227 * bindings/vala/gst-rtsp-server.vapi:
7228 * bindings/vala/packages/gst-rtsp-server.excludes:
7229 * bindings/vala/packages/gst-rtsp-server.gi:
7230 * bindings/vala/packages/gst-rtsp-server.metadata:
7231 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
7233 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
7235 * bindings/vala/gst-rtsp-server.vapi:
7236 * bindings/vala/packages/gst-rtsp-server.deps:
7237 * bindings/vala/packages/gst-rtsp-server.files:
7238 * bindings/vala/packages/gst-rtsp-server.gi:
7239 * bindings/vala/packages/gst-rtsp-server.metadata:
7240 * bindings/vala/packages/gst-rtsp-server.namespace:
7243 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
7245 * gst/rtsp-server/rtsp-session.c:
7246 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
7248 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
7250 * examples/Makefile.am:
7251 * gst/rtsp-server/Makefile.am:
7252 Put GStreamer version in library name
7254 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7256 * examples/Makefile.am:
7257 * gst/rtsp-server/Makefile.am:
7258 Fix some issues to pass distcheck
7260 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7262 * gst/rtsp-server/rtsp-server.c:
7263 Added port property to GstRTSPServer class.
7265 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7270 * examples/Makefile.am:
7273 * gst/rtsp-server/Makefile.am:
7274 * gst/rtsp-server/rtsp-client.c:
7275 * gst/rtsp-server/rtsp-client.h:
7276 * gst/rtsp-server/rtsp-media.c:
7277 * gst/rtsp-server/rtsp-media.h:
7278 * gst/rtsp-server/rtsp-server.c:
7279 * gst/rtsp-server/rtsp-server.h:
7280 * gst/rtsp-server/rtsp-session-pool.c:
7281 * gst/rtsp-server/rtsp-session-pool.h:
7282 * gst/rtsp-server/rtsp-session.c:
7283 * gst/rtsp-server/rtsp-session.h:
7286 * src/rtsp-client.c:
7287 * src/rtsp-client.h:
7290 * src/rtsp-server.c:
7291 * src/rtsp-server.h:
7292 * src/rtsp-session-pool.c:
7293 * src/rtsp-session-pool.h:
7294 * src/rtsp-session.c:
7295 * src/rtsp-session.h:
7296 Split in library and example program
7298 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
7300 * src/rtsp-client.h:
7301 Removed obsolete variable
7303 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
7305 * src/rtsp-client.c:
7306 * src/rtsp-client.h:
7307 Removed pipeline variable GstRTSPClient, because it's only used in one function
7309 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7312 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
7314 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
7316 * src/rtsp-session.c:
7317 Initialize some more vars.
7319 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
7321 * src/rtsp-session.c:
7322 Initialize variable to avoid compiler warning.
7324 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
7327 Add a reasonable generic .gitignore