3 2014-06-28 Sebastian Dröge <slomo@coaxion.net>
8 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
10 * gst/rtsp-server/rtsp-stream.c:
11 stream: crypto can be NULL
13 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
15 * gst/rtsp-server/rtsp-client.c:
16 * gst/rtsp-server/rtsp-media.c:
17 * gst/rtsp-server/rtsp-mount-points.c:
18 introspection: add missing allow-none annotations
19 https://bugzilla.gnome.org/show_bug.cgi?id=730952
21 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
23 * gst/rtsp-server/rtsp-address-pool.c:
24 * gst/rtsp-server/rtsp-media.c:
25 * gst/rtsp-server/rtsp-session-media.c:
26 * gst/rtsp-server/rtsp-session-pool.c:
27 * gst/rtsp-server/rtsp-stream-transport.c:
28 * gst/rtsp-server/rtsp-stream.c:
29 * gst/rtsp-server/rtsp-token.c:
30 introspection: add (nullable) annotations to return values
31 https://bugzilla.gnome.org/show_bug.cgi?id=730952
33 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
35 * gst/rtsp-server/rtsp-client.c:
36 * gst/rtsp-server/rtsp-stream.c:
37 gi: improve annotations
38 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
40 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
42 * gst/rtsp-server/rtsp-client.c:
43 * gst/rtsp-server/rtsp-media-factory.c:
44 * gst/rtsp-server/rtsp-media.c:
45 * gst/rtsp-server/rtsp-server.c:
46 signals: use generic marshal function
47 Use the generic C marshal function.
48 Use more explicit type instead of G_TYPE_POINTER
50 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
52 * gst/rtsp-server/rtsp-context.h:
53 context: add type macro
55 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
57 * gst/rtsp-server/rtsp-client.c:
58 * gst/rtsp-server/rtsp-sdp.c:
59 * gst/rtsp-server/rtsp-sdp.h:
60 sdp: hide key length defines
61 They don't have a namespace.
63 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
70 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
76 * gst-rtsp-server.doap:
79 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
81 * gst/rtsp-server/rtsp-client.c:
82 * gst/rtsp-server/rtsp-sdp.c:
83 * gst/rtsp-server/rtsp-sdp.h:
84 mikey: add different key length parameters
85 Add encryption and authentication key length parameters to MIKEY. For
86 the encoders, the key lengths are obtained from the cipher and auth
87 algorithms set in the caps. For the decoders, they are obtained while
88 parsing the key management from the client.
89 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
91 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
93 * tests/check/gst/stream.c:
94 stream tests: Make sure we get right multicast address from stream
95 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
97 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
99 * gst/rtsp-server/rtsp-client.c:
100 client: ref the context until rtsp watch is alive
101 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
103 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
105 * gst/rtsp-server/rtsp-client.c:
106 client: Destroy the rtsp watch after connection close
108 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
110 * gst/rtsp-server/rtsp-media.c:
111 media: fix confusing comment
113 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
115 * gst/rtsp-server/rtsp-session.c:
116 rtsp-session: Timeout in header.
117 Adding the possbilty to always have timout in header.
118 This is configurabe with setting "timeout-always-visible".
119 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
121 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
126 === release 1.3.2 ===
128 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
135 * gst-rtsp-server.doap:
138 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
141 Automatic update of common submodule
142 From 211fa5f to 1f5d3c3
144 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
146 * gst/rtsp-server/rtsp-client.c:
147 client: store TCP ports in transport
148 Store the TCP ports in the transport when we are doing RTSP over TCP.
149 This way, we can easily get to the ports from the transport.
150 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
152 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
154 * gst/rtsp-server/rtsp-stream.c:
155 stream: add signals for new RTP/RTCP encoders
156 New signals to allow the user to configure the dynamically created
158 https://bugzilla.gnome.org/show_bug.cgi?id=730228
160 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
162 * gst/rtsp-server/rtsp-media.c:
163 * gst/rtsp-server/rtsp-media.h:
164 media: Make suspend()/unsuspend() virtual
165 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
167 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
169 * gst/rtsp-server/rtsp-client.c:
170 client: fix send-message signal marshaller
171 Use generic marshalling for the send-message signal. It has
172 two POINTER arguments, not just one.
173 https://bugzilla.gnome.org/show_bug.cgi?id=729900
175 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
177 * tests/check/gst/media.c:
178 tests: add and remove pads only once
179 In this test we simulate a dynamic pad by watching the caps event.
180 Because of renegotiation in the base payloader now, this caps is sent
181 multiple times but we can only deal with 1 invocation, use a variable to
182 only 'add and remove' the pad once.
184 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
186 * tests/check/gst/rtspserver.c:
187 tests: add unit test for correct handling of Require headers
188 https://bugzilla.gnome.org/show_bug.cgi?id=729426
190 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
192 * gst/rtsp-server/rtsp-client.c:
193 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
194 Servers must handle Require headers and must report a failure
195 if they don't handle any of the Required options, see RFC 2326,
196 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
197 https://bugzilla.gnome.org/show_bug.cgi?id=729426
199 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
204 === release 1.3.1 ===
206 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
212 * gst-rtsp-server.doap:
215 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
218 Automatic update of common submodule
219 From bcb1518 to 211fa5f
221 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
226 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
228 * tests/check/gst/sessionmedia.c:
229 tests: fix memory leak in sessionmedia unit test
231 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
233 * gst/rtsp-server/rtsp-client.c:
234 client: emit a signal before sending a message
235 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
237 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
239 * gst/rtsp-server/rtsp-client.c:
240 client: pass context to send_message
241 Pass the current context to send_message, we will need it later.
243 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
245 * gst/rtsp-server/rtsp-client.c:
246 client: fix typo in comment
248 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
250 * gst/rtsp-server/rtsp-media.c:
251 media: Do not stop thread twice if default_prepare() fails
253 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
255 * gst/rtsp-server/rtsp-client.c:
256 client: set the watch to flushing before going to NULL
257 First set the watch to flushing so that we unblock any current and
258 future attempt to send data on the watch, Then set the pipeline to
260 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
262 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
264 * gst/rtsp-server/rtsp-session-pool.c:
265 * tests/check/gst/sessionpool.c:
266 rtsp-session-pool: Fixes annotation
267 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
268 in the sessionpool test.
269 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
271 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
273 * gst/rtsp-server/rtsp-media.c:
274 * gst/rtsp-server/rtsp-media.h:
275 media: make media_prepare virtual
276 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
278 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
280 * gst/rtsp-server/rtsp-media.c:
281 * tests/check/gst/media.c:
282 media: stop the thread in more error cases
284 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
286 * gst/rtsp-server/rtsp-media.c:
287 * tests/check/gst/media.c:
288 media: allow NULL as the thread
289 Use the default context whan passing a NULL thread.
291 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
293 * gst/rtsp-server/rtsp-client.c:
294 rtsp-client: indent cleanup
295 Coverity was moaning about unreachable code, and I think it was just
296 confused by { being before the label. We'll see if it pops up again.
299 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
301 * gst/rtsp-server/rtsp-client.c:
302 * gst/rtsp-server/rtsp-media.c:
303 client: Add drop-backlog property
304 When we have too many messages queued for a client (currently hardcoded
305 to 100) we overflow and drop the messages. Add a drop-backlog property
306 to control this behaviour. Setting this property to FALSE will retry
307 to send the messages to the client by waiting for more room in the
309 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
311 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
313 * gst/rtsp-server/rtsp-client.c:
314 client: support for POST before GET when setting up a tunnel
316 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
318 * gst/rtsp-server/rtsp-client.c:
319 client: remove watch of the second client after http tunnel setup
320 The second client will be freed after the HTTP tunnel has been set up.
321 Make sure it's RTSP watch is never dispatched again.
322 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
324 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
326 * gst/rtsp-server/rtsp-media.c:
327 * tests/check/gst/media.c:
328 media: Make media_prepare() fail if port allocation fails
329 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
331 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
333 * tests/check/gst/media.c:
334 media test: cleanup the thread pool in tests
336 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
338 * gst/rtsp-server/rtsp-media.c:
339 * tests/check/gst/media.c:
340 rtsp-media: Unblock blocked streams in unprepare
341 The streams will be blocked when a live media is prepared.
342 The streams should be unblocked in gst_rtsp_media_unprepare.
343 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
345 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
347 * gst/rtsp-server/rtsp-media.c:
348 media: release the state lock when going to NULL
349 Set our state to UNPREPARING and release the state-lock before
350 setting the pipeline to the NULL state. This way, any pad-added
351 callback will be able to take the state-lock and check that we are now
352 unpreparing instead of deadlocking.
353 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
355 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
357 * gst/rtsp-server/rtsp-media.c:
358 media: protect status with lock
359 Make sure we only update the status with the lock.
361 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
363 * gst/rtsp-server/rtsp-client.c:
364 * gst/rtsp-server/rtsp-sdp.c:
365 rtsp: update for MIKEY API changes
367 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
369 * gst/rtsp-server/rtsp-client.c:
370 client: parse the mikey response from the client
371 Parse the mikey response from the client and update the policy for
374 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
376 * gst/rtsp-server/rtsp-stream.c:
377 * gst/rtsp-server/rtsp-stream.h:
378 stream: add method to set crypto info
379 Make a method to configure the crypto information of a stream.
380 Set udpsrc in READY instead of PAUSED so that we can configure caps
383 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
385 * gst/rtsp-server/rtsp-client.c:
386 client: cleanup error paths
388 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
390 * gst/rtsp-server/rtsp-media.c:
393 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
395 * examples/test-video.c:
396 test: enable SRTP only on RTSPS
397 We only want to enable SRTP when doing rtsp over TLS so that we can
398 exchange the keys in a secure way.
400 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
402 * examples/test-video.c:
403 test: print an error on failure
405 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
408 * examples/test-video.c:
409 * gst/rtsp-server/rtsp-sdp.c:
410 * gst/rtsp-server/rtsp-stream.c:
411 * tests/check/Makefile.am:
412 stream: add SRTP support
413 Install srtp encoder and decoder elements in rtpbin
416 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
418 * tests/check/Makefile.am:
419 * tests/check/gst/sessionpool.c:
420 tests: Add unit tests for sessionpool
421 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
423 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
425 * tests/check/gst/threadpool.c:
426 tests: Improve code coverage of rtsp-threadpool tests
427 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
429 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
431 * tests/check/gst/sessionmedia.c:
432 tests: Improve code coverage for rtsp-session-media
433 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
435 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
437 gobject-introspection: Add annotations to support language bindings
438 In addition a few cosmetic changes:
439 * Adjust the order of arguments
440 * Fix typo: occured -> occurred
441 * Fix indentation after Return:-clauses
442 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
444 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
446 * gst/rtsp-server/rtsp-stream.c:
447 rtsp-stream: Don't mix IPv4 and IPv6 addresses
448 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
450 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
452 * gst/rtsp-server/rtsp-stream.c:
453 stream: take caps after the session manager
454 Take the caps for the SDP after they leave the rtpbin so that we can
455 also get the properties added by rtpbin elements.
457 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
459 * gst/rtsp-server/rtsp-stream.c:
460 stream: release lock while pushing out packets
461 Keep a cache of the transports and use this to iterate the transport
462 while pushing packets. This allows us to release the lock early.
463 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
465 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
467 * gst/rtsp-server/rtsp-client.c:
468 * gst/rtsp-server/rtsp-client.h:
469 rtsp-client: vmethod for modifying tunnel GET response
470 Add a vmethod tunnel_http_response where the response to the HTTP GET
471 for tunneled connections can be modified.
472 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
474 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
476 * gst/rtsp-server/rtsp-sdp.c:
477 sdp: make 1 media line per profile
478 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
479 line in the SDP for each profile. The client is then supposed to pick
480 one of the profiles in the SETUP request. Because the m= lines have the
481 same pt, the client also knows that only 1 option is possible.
483 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
485 * gst/rtsp-server/rtsp-media-factory.c:
486 * gst/rtsp-server/rtsp-media-factory.h:
487 * gst/rtsp-server/rtsp-media.c:
488 factory: add profile property and pass to media and streams
490 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
492 * examples/test-multicast.c:
493 * gst/rtsp-server/rtsp-sdp.c:
494 sdp: pass multicast connection for multicast-only stream
495 Pass the multicast address of the stream in the connection info in the
496 SDP so that clients try a multicast connection first.
497 Only allow multicast connections in the test-multicast example. Also
498 increase the TTL a little.
500 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
503 .gitignore: Ignore gcov intermediate files
504 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
506 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
508 * gst/rtsp-server/rtsp-stream.c:
509 stream: release some locks in error cases
511 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
513 docs: Enable and fix gtk-doc warnings
514 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
515 * addresspool/mediafactory: Add missing annotation colon
516 * stream: Annotate return value
517 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
519 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
522 Automatic update of common submodule
523 From fe1672e to bcb1518
525 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
528 Automatic update of common submodule
529 From 1a07da9 to fe1672e
531 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
533 * examples/Makefile.am:
534 examples: use LDADD for libs instead of LDFLAGS
536 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
539 configure: make sure releases are in .doap file
541 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
543 * examples/test-cgroups.c:
544 examples: test-cgroups: don't put code with side effects into g_assert()
545 The g_assert() might get compiled out with the right
546 compiler/preprocessor flags.
548 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
550 * examples/.gitignore:
551 examples: add cgroup test binary to .gitignore
553 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
555 * examples/test-cgroups.c:
556 examples: fix cgroup test build
557 Fixes build failure caused by compiler warning:
558 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
560 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
563 .gitignore: ignore temp files created in the course of 'make check'
565 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
567 * gst/rtsp-server/rtsp-media.c:
568 rtsp-media: don't loose frames handling new PLAY request
569 If client supplied a range check if the range specifies the start point.
570 If not, then do an accurate seek to the current position. If a start
571 point was specified do do a key unit seek to make sure the streaming
572 starts with decodeable frames.
573 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
575 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
577 * gst/rtsp-server/rtsp-media.c:
578 Revert "media: only flush when setting a new start position"
579 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
580 We need to do the flush in all cases, demuxer block currently for
583 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
585 * gst/rtsp-server/rtsp-media.c:
586 media: only flush when setting a new start position
587 Only flush the pipeline when we change the start position with
589 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
591 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
593 * gst/rtsp-server/rtsp-stream.c:
594 stream: set ttl-mc before adding the socket
595 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
596 never be set on socket.
597 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
599 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
601 * gst/rtsp-server/rtsp-media.c:
602 media: stop thread if media is already prepared
603 in gst_rtsp_media_prepare() the thread is not used if media is already
604 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
606 https://bugzilla.gnome.org/show_bug.cgi?id=724182
608 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
611 build: Ship gst-rtsp-server.doap file
613 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
615 * tests/check/gst/rtspserver.c:
616 tests: Fix another compiler warning with gcc
618 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
620 * gst/rtsp-server/rtsp-client.c:
621 * gst/rtsp-server/rtsp-mount-points.c:
622 * gst/rtsp-server/rtsp-stream.c:
623 * tests/check/gst/client.c:
624 rtsp-server: Fix lots of compiler warnings with clang
626 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
629 * gst-rtsp-server.doap:
631 configure: Synchronise with the configure scripts of the other modules
633 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
636 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
638 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
640 * gst/rtsp-server/rtsp-media.c:
641 * gst/rtsp-server/rtsp-stream.c:
642 Revert "rtsp-server: support build against last stable release"
643 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
644 Let us require 1.2.3 now, which is going to be released in a few
647 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
649 * gst/rtsp-server/rtsp-session-media.c:
650 * gst/rtsp-server/rtsp-stream-transport.c:
651 session: improve RTP-Info
652 Ignore streams that can't generate RTP-Info instead of failing.
653 Don't return the empty string when all streams are unconfigured but
654 return NULL so that we don't generate and empty RTP-Info header.
655 Improve docs a little.
657 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
659 * gst/rtsp-server/rtsp-session-media.c:
660 Don't free rtpinfo GString when it is NULL
661 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
663 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
665 * gst/rtsp-server/rtsp-media.c:
666 media: only set keyframe flag when modifying start
667 Only set the keyframe flag when we modify the start position. The
668 keyframe flag should probably be ignored when no change is requested but
669 until we can claim this is all documented properly and all demuxer
670 implement this, avoid setting the flag.
671 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
673 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
675 * gst/rtsp-server/rtsp-thread-pool.c:
676 thread-pool: Unref source after mainloop has quit to avoid races in GLib
677 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
679 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
681 * gst/rtsp-server/rtsp-stream.c:
682 stream: handle NULL seqnum and rtptime arguments
684 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
686 * gst/rtsp-server/rtsp-thread-pool.c:
687 * tests/check/gst/threadpool.c:
688 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
689 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
691 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
693 * gst/rtsp-server/rtsp-stream.c:
694 stream: add fallback for missing stats property
695 Use a fallback when the payloader does not have a stats property
696 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
698 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
701 Automatic update of common submodule
702 From f7bc1c3 to 1a07da9
704 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
706 * gst/rtsp-server/rtsp-stream.c:
707 stream: don't leak stats structure
708 Don't leak the stats structure and deal with NULL stats.
710 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
712 * gst/rtsp-server/rtsp-stream.c:
713 stream: Get rtpinfo properties atomically from payloader
714 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
716 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
718 * gst/rtsp-server/rtsp-media.c:
719 media: refactor state change functions and signals
720 Make functions to set the target state and the pipeline state and emit
721 the signals from those functions.
723 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
725 * gst/rtsp-server/rtsp-media.c:
726 * gst/rtsp-server/rtsp-media.h:
727 media: add signal to notify of pending state changes
729 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
731 * gst/rtsp-server/rtsp-media.c:
732 * gst/rtsp-server/rtsp-stream.c:
733 rtsp-server: support build against last stable release
734 Until 1.2.3 is out with the new get_type function and we
737 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
739 * gst/rtsp-server/rtsp-stream.c:
740 stream: fix compilation
742 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
744 * gst/rtsp-server/rtsp-media.c:
745 * gst/rtsp-server/rtsp-media.h:
746 * gst/rtsp-server/rtsp-stream.c:
747 * gst/rtsp-server/rtsp-stream.h:
748 stream: add property to configure profiles
750 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
752 * gst/rtsp-server/rtsp-client.c:
753 client: let stream check supported transport
754 Delegate the check if a transport is allowed to the stream.
755 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
757 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
759 * gst/rtsp-server/rtsp-stream.c:
760 * gst/rtsp-server/rtsp-stream.h:
761 stream: add method to check supported transport
762 Add a method to check if a transport is supported
764 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
767 configure.ac: Only check for gstreamer-check, not check
768 We include check in gstreamer-check since quite some time now.
770 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
772 * gst/rtsp-server/rtsp-session-media.c:
773 * gst/rtsp-server/rtsp-stream-transport.c:
774 * gst/rtsp-server/rtsp-stream.c:
775 * gst/rtsp-server/rtsp-stream.h:
776 stream: return clock-rate from get_rtpinfo
777 And use it to correct the rtptime to the requested start-time.
778 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
780 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
782 * gst/rtsp-server/rtsp-session-media.c:
783 * gst/rtsp-server/rtsp-stream-transport.c:
784 * gst/rtsp-server/rtsp-stream-transport.h:
785 session-media: calculate start-time
787 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
789 * gst/rtsp-server/rtsp-stream-transport.c:
790 * gst/rtsp-server/rtsp-stream.c:
791 * gst/rtsp-server/rtsp-stream.h:
792 stream: also return the running-time
793 Return the running-time in the rtpinfo as well.
795 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
797 * gst/rtsp-server/rtsp-client.c:
798 * gst/rtsp-server/rtsp-session-media.c:
799 * gst/rtsp-server/rtsp-session-media.h:
800 * gst/rtsp-server/rtsp-stream-transport.c:
801 * gst/rtsp-server/rtsp-stream-transport.h:
802 session-media: let the session-media make the RTPInfo
803 Add method to create the RTPInfo for a stream-transport.
804 Add method to create the RTPInfo for all stream-transports in a
806 Use the session-media RTPInfo code in client. This allows us to refactor
807 another method to link the TCP callbacks.
809 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
811 mount-points: sort sequence before g_sequence_lookup
812 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
813 sort sequence if dirty, otherwise lookup will fail.
814 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
816 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
819 configure: rename package from gst-rtsp to gst-rtsp-server
820 To match git module name and avoid confusion with the
821 rtsp lib in gst-plugins-base and rtsp plugin in -good.
823 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
826 configure: bump core/base/good requirement to 1.2.0
827 Bump to released stable version and make implicit
828 requirements explicit.
830 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
835 Fix broken gettext setup which is not used anyway
837 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
840 Automatic update of common submodule
841 From dbedaa0 to d48bed3
843 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
845 * gst/rtsp-server/rtsp-client.c:
846 * gst/rtsp-server/rtsp-media.c:
847 * gst/rtsp-server/rtsp-media.h:
848 media: add setup_sdp vmethod
849 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
850 gst_rtsp_media_setup_sdp.
851 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
853 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
855 * gst/rtsp-server/rtsp-stream.c:
856 rtsp-stream: Check return value of sscanf
857 streamid is only valid if sscanf matched something.
859 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
861 * gst/rtsp-server/rtsp-client.c:
862 rtsp-client: Fix iteration
863 Wouldn't even enter the code block otherwise (i++ was used as the check
864 and not the postfix).
866 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
868 * gst/rtsp-server/rtsp-client.c:
869 * gst/rtsp-server/rtsp-client.h:
870 client: add vmethod to configure media and streams
871 Implement a vmethod that can be used to configure the media and the
872 streams based on the current context. Handle the blocksize handling in
874 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
876 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
879 Make git ignore more unit test binaries
881 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
883 * gst/rtsp-server/rtsp-address-pool.h:
884 * gst/rtsp-server/rtsp-auth.h:
885 * gst/rtsp-server/rtsp-client.h:
886 * gst/rtsp-server/rtsp-context.h:
887 * gst/rtsp-server/rtsp-media-factory-uri.h:
888 * gst/rtsp-server/rtsp-media-factory.h:
889 * gst/rtsp-server/rtsp-media.h:
890 * gst/rtsp-server/rtsp-mount-points.h:
891 * gst/rtsp-server/rtsp-server.h:
892 * gst/rtsp-server/rtsp-session-media.h:
893 * gst/rtsp-server/rtsp-session-pool.h:
894 * gst/rtsp-server/rtsp-session.h:
895 * gst/rtsp-server/rtsp-stream-transport.h:
896 * gst/rtsp-server/rtsp-stream.h:
897 * gst/rtsp-server/rtsp-thread-pool.h:
898 * gst/rtsp-server/rtsp-token.h:
899 rtsp-server: add padding to many public structures
900 Not mini objects though, since they are not subclassable
901 anyway, nor kept on the stack or inlined in a structure.
903 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
905 media: add new create_rtpbin vmethod
906 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
907 https://bugzilla.gnome.org/show_bug.cgi?id=719734
909 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
911 * tests/check/gst/media.c:
912 tests: fix memory leak, free test's thread pool
913 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
915 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
917 * gst/rtsp-server/rtsp-stream-transport.c:
918 stream-transport: free url in finalize
920 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
922 * gst/rtsp-server/rtsp-media.c:
923 media: also do state change in suspended state
925 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
927 * gst/rtsp-server/rtsp-client.c:
928 * gst/rtsp-server/rtsp-media.c:
929 media: also handle prepare and range in suspended state
930 When we are suspended, we are already prepared.
931 We can get the range in the suspended state.
933 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
935 * tests/check/Makefile.am:
936 * tests/check/gst/sessionmedia.c:
937 check: add test for uri in setup
938 Added unit tests for the new functionality in GstRTSPStreamTransport.
939 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
941 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
943 * gst/rtsp-server/rtsp-client.c:
944 client: store setup uri and use in PLAY response
945 Store the uri used when doing the setup and use that in the PLAY
947 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
949 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
951 * gst/rtsp-server/rtsp-stream-transport.c:
952 * gst/rtsp-server/rtsp-stream-transport.h:
953 stream-transport: add method to get/set url
955 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
957 * gst/rtsp-server/rtsp-client.c:
958 client: suspend after SDP and unsuspend before PLAYING
959 Based on patches by Ognyan Tonchev <ognyan@axis.com>
960 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
962 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
964 * gst/rtsp-server/rtsp-media-factory.c:
965 * gst/rtsp-server/rtsp-media-factory.h:
966 * gst/rtsp-server/rtsp-media.c:
967 * gst/rtsp-server/rtsp-media.h:
968 * gst/rtsp-server/rtsp-session-media.c:
969 * gst/rtsp-server/rtsp-session.c:
970 * tests/check/gst/media.c:
971 * tests/check/gst/mediafactory.c:
972 media: add suspend modes
973 Add support for different suspend modes. The stream is suspended right after
974 producing the SDP and after PAUSE. Different suspend modes are available that
975 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
976 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
977 state and RESET will bring the pipeline to the NULL state.
978 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
979 this means that the pipeline needs to be prerolled again.
980 Base on patches by Ognyan Tonchev <ognyan@axis.com>
981 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
983 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
985 * gst/rtsp-server/rtsp-media.c:
986 media: start live streams in blocked state
987 Start live streams in the blocked state and make them preroll using the
988 messages. This ensure that no data is played by the sink until we explicitly
989 unblock the stream right before going to PLAYING.
990 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
992 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
994 * gst/rtsp-server/rtsp-media.c:
995 media: refactor starting and waiting for preroll
996 Based on patches from Ognyan Tonchev <ognyan@axis.com>
997 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
999 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
1001 * gst/rtsp-server/rtsp-stream.c:
1002 * gst/rtsp-server/rtsp-stream.h:
1003 stream: add API to block streams
1004 Add an API to block on the streams and make it post a message.
1005 Based on patch by Ognyan Tonchev <ognyan@axis.com>
1006 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
1008 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
1010 * docs/libs/Makefile.am:
1011 docs: Specify the override file
1012 Even if it's empty (for now) it avoids make distcheck complaining
1014 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
1016 * gst/rtsp-server/rtsp-media.c:
1017 media: move default implementations to where they are used
1019 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
1021 * gst/rtsp-server/rtsp-media.c:
1022 media: take the right lock in gst_rtsp_media_set_pipeline_state()
1023 We need to take the state_lock when calling this method.
1025 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
1027 * gst/rtsp-server/rtsp-media.c:
1028 media: handle add-added on non-bins too
1029 Handle dynamic payloaders that are not bins, as used in the unit-test.
1031 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1033 * gst/rtsp-server/rtsp-media-factory.c:
1034 * gst/rtsp-server/rtsp-media-factory.h:
1035 * gst/rtsp-server/rtsp-media.c:
1036 rtsp-media/-factory: Fix request pad name comments
1037 These must be escaped for gtk-doc to parse the comments without warnings.
1039 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
1041 rtsp-media: remove transports if media is in error status
1042 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
1043 trying to change to GST_STATE_NULL and media is in error status, we
1044 remove all transports.
1045 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
1047 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
1049 * gst/rtsp-server/rtsp-media.c:
1050 rtsp-media: use element metadata to find payloader
1051 Use the element metadata to find the payloader instead of checking
1053 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
1055 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
1057 rtsp-stream: add getter for payload type
1058 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
1059 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
1060 element and create the stream with this one instead of the dynpay%d
1062 https://bugzilla.gnome.org/show_bug.cgi?id=712396
1064 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1066 * gst/rtsp-server/rtsp-client.c:
1067 * gst/rtsp-server/rtsp-context.h:
1068 * gst/rtsp-server/rtsp-media.c:
1069 * gst/rtsp-server/rtsp-mount-points.c:
1070 * gst/rtsp-server/rtsp-server.c:
1071 * gst/rtsp-server/rtsp-token.c:
1072 rtsp-*: Refer to NULL as a constant in comments
1074 https://bugzilla.gnome.org/show_bug.cgi?id=714988
1076 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1078 rtsp-*: Fix type name typos in comments
1079 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
1080 * rtsp-auth: Refer to part of constant name as text
1081 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
1082 * rtsp-session-media: Fix GstRTSPSessionMedia typo
1083 * rtsp-stream: Fix typo when refering to GstBin
1084 https://bugzilla.gnome.org/show_bug.cgi?id=714988
1086 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1089 * docs/libs/gst-rtsp-server-docs.sgml:
1090 * docs/libs/gst-rtsp-server-sections.txt:
1091 docs: Improve documentation
1092 * Include annotation-glossary to quiet gtk-doc
1093 * Rename remaining ClientState -> Context
1094 * Rename object hierarchy file
1095 * Remove stale chapter references
1096 * Add missing function and object references
1097 * Include missing GstRTSPAddressPoolResult
1098 https://bugzilla.gnome.org/show_bug.cgi?id=714988
1100 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
1102 * gst/rtsp-server/rtsp-client.c:
1103 * gst/rtsp-server/rtsp-server.c:
1104 * gst/rtsp-server/rtsp-session-pool.c:
1105 * gst/rtsp-server/rtsp-session.c:
1106 * gst/rtsp-server/rtsp-stream.c:
1107 rtsp-server: sprinkle some allow-none annotations for g-i
1109 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
1111 * gst/rtsp-server/rtsp-stream.c:
1112 * gst/rtsp-server/rtsp-stream.h:
1113 stream: add method to filter transports
1114 Add a method to safely iterate and collect the stream transports
1115 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
1117 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
1119 * gst/rtsp-server/rtsp-client.c:
1120 * gst/rtsp-server/rtsp-server.c:
1121 * gst/rtsp-server/rtsp-session-pool.c:
1122 * gst/rtsp-server/rtsp-session.c:
1123 rtsp: allow NULL func in filters
1124 Passing a null function make the filters return a list of
1127 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
1129 * gst/rtsp-server/rtsp-address-pool.c:
1130 * tests/check/gst/addresspool.c:
1131 address-pool: fix address increment
1132 Use a guint instead of guint8 to increment the address. It's still not
1133 completely correct because a guint might not be able to hold the complete
1134 address range, but that's an enhacement for later.
1135 Add unit test to test improved behaviour.
1136 https://bugzilla.gnome.org/show_bug.cgi?id=708237
1138 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
1140 * gst/rtsp-server/rtsp-client.c:
1141 * tests/check/gst/client.c:
1142 client: allow absolute path in requests
1143 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
1145 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
1147 * gst/rtsp-server/rtsp-client.c:
1148 * gst/rtsp-server/rtsp-client.h:
1149 client: make make_path_from_uri a vmethod
1151 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
1153 * docs/libs/gst-rtsp-server-sections.txt:
1154 * gst/rtsp-server/rtsp-stream.c:
1155 * gst/rtsp-server/rtsp-stream.h:
1156 * tests/check/Makefile.am:
1157 * tests/check/gst/stream.c:
1158 stream: Add functions to get rtp and rtcp sockets
1159 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
1161 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
1163 * gst/rtsp-server/rtsp-context.c:
1164 * gst/rtsp-server/rtsp-context.h:
1165 context: defing a GType for the context
1166 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
1168 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
1170 * gst/rtsp-server/Makefile.am:
1171 * gst/rtsp-server/rtsp-auth.c:
1172 * gst/rtsp-server/rtsp-context.c:
1173 * gst/rtsp-server/rtsp-media.c:
1174 * gst/rtsp-server/rtsp-mount-points.c:
1175 * gst/rtsp-server/rtsp-server.h:
1176 * gst/rtsp-server/rtsp-session-media.c:
1177 * gst/rtsp-server/rtsp-session.c:
1178 * gst/rtsp-server/rtsp-stream.c:
1179 Fixed several GIR warnings
1181 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
1183 * gst/rtsp-server/rtsp-auth.c:
1186 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1188 * tests/check/Makefile.am:
1189 * tests/check/gst/token.c:
1190 tests: Add unit tests for token
1191 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
1193 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1195 * gst/rtsp-server/rtsp-token.c:
1196 token: Validate args for gst_rtsp_token_is_allowed
1197 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
1199 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1201 * gst/rtsp-server/rtsp-token.c:
1202 token: Fix bug when creating empty token
1203 We always want to have a valid GstStructure in the token.
1204 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
1206 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
1208 * gst/rtsp-server/rtsp-thread-pool.c:
1209 thread-pool: avoid race in shutdown
1210 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
1211 don't actually stop the mainloop ever. Solve this race by adding an idle source
1212 to the mainloop that calls the _quit. This way we immediately exit the mainloop
1213 if quit was called before we started it.
1215 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1217 * tests/check/Makefile.am:
1218 * tests/check/gst/permissions.c:
1219 tests: Add unit tests for permissions
1220 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
1222 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1224 * tests/check/gst/mediafactory.c:
1225 tests: Test mediafactory permissions
1226 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
1228 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1230 * gst/rtsp-server/rtsp-permissions.c:
1231 permissions: Fix refcounting when adding/removing roles
1232 Previously a role that was removed was unreffed twice, and when
1233 replacing an existing role the replaced role was freed while still being
1234 referenced. Both bugs are now fixed.
1235 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
1237 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1239 * tests/check/gst/media.c:
1240 * tests/check/gst/mediafactory.c:
1241 * tests/check/gst/rtspserver.c:
1242 tests: Check gst_rtsp_url_parse return value
1243 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
1245 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
1248 Automatic update of common submodule
1249 From 865aa20 to dbedaa0
1251 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
1253 * gst/rtsp-server/rtsp-server.c:
1254 rtsp-server: Fix socket leak
1255 https://bugzilla.gnome.org/show_bug.cgi?id=710088
1257 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
1259 * gst/rtsp-server/rtsp-session-pool.c:
1260 rtsp-session-pool: Make sure session IDs are properly URI-escaped
1261 https://bugzilla.gnome.org/show_bug.cgi?id=643812
1263 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
1265 * examples/.gitignore:
1266 * examples/test-video.c:
1267 examples: fix compilation when WITH_AUTH is defined
1268 https://bugzilla.gnome.org/show_bug.cgi?id=710228
1270 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
1273 gitignore: Add new test binary
1275 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
1277 * tests/check/Makefile.am:
1278 * tests/check/gst/threadpool.c:
1279 thread-pool: Add unit test for the thread pools
1280 https://bugzilla.gnome.org/show_bug.cgi?id=710228
1282 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
1284 * gst/rtsp-server/rtsp-thread-pool.c:
1285 thread-pool: Fix thread leak when reusing threads
1286 https://bugzilla.gnome.org/show_bug.cgi?id=709730
1288 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
1290 * gst/rtsp-server/rtsp-server.c:
1291 * tests/check/gst/rtspserver.c:
1292 tests: fixed racy behavior in rtspserver tests
1293 https://bugzilla.gnome.org/show_bug.cgi?id=710078
1295 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1297 * tests/check/gst/addresspool.c:
1298 tests: Improve address pool unit tests
1299 Add a range with mixed IPV4 and IPV6 addresses to pool.
1300 Get an IPV4 address from an IPV6-only pool.
1301 Get an IPV6 address from an IPV4-only pool.
1302 Reserve a IPV6 address from an IPV4-only pool.
1303 Check for unicast addresses in multicast-only pool.
1304 Check for unicast addresses in uni-/multicast-mixed pool.
1305 https://bugzilla.gnome.org/show_bug.cgi?id=710128
1307 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1309 * gst/rtsp-server/rtsp-client.c:
1310 client: append query string in PAUSE/PLAY/TEARDOWN as well
1312 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
1314 * gst/rtsp-server/rtsp-client.c:
1315 client: Add query to control path
1316 If the SETUP url contains a query it must be appended to the control
1317 path so that it matches any already created stream in the media. The
1318 query will also be appended to the session media path.
1320 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1322 * gst/rtsp-server/rtsp-media.c:
1323 rtsp-media: remove old line
1325 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
1327 * gst/rtsp-server/rtsp-stream.c:
1328 stream: Correct control comparison
1329 https://bugzilla.gnome.org/show_bug.cgi?id=709176
1331 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1333 * gst/rtsp-server/rtsp-media.c:
1334 media: Check dynamically if the pipeline supports seeking
1335 We should not depend on whether or not the pipeline state change
1336 returned NO_PREROLL or not. A media could dynamically change its
1337 element and switch from seekable to non seekable so it's best to test
1338 the seekable nature of the pipeline dynamically when we try to do a seek.
1340 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1342 * gst/rtsp-server/rtsp-media.c:
1343 media: Return FALSE if seeking is not supported
1345 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1347 * gst/rtsp-server/rtsp-media.c:
1348 rtsp-media: don't seek accurate by default
1349 Accurate seeking is perhaps a little overkill in the most common situation and
1350 causes some formats (mp3) over slow media to seek extremely slowly.
1352 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
1354 * tests/check/gst/rtspserver.c:
1355 tests: fix unit test
1356 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
1358 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
1360 * gst/rtsp-server/rtsp-client.c:
1361 client: Reply 400 if media cannot be constructed
1362 Reply 400 Bad Request instead of 503 Service Unavailable if media
1363 cannot be constructed in SETUP.
1364 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
1366 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
1368 * gst/rtsp-server/rtsp-client.c:
1369 client: Send setup reply once only
1370 If find_media() failed in handle_setup_request() two replies was sent.
1371 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
1373 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
1376 Automatic update of common submodule
1377 From 6b03ba7 to 865aa20
1379 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
1381 * gst/rtsp-server/rtsp-server.c:
1382 server: Emit client-connected signal earlier
1383 Emit client-connected before the client ref is given to a GSource,
1384 otherwise client-connected can be emitted after the client object has
1387 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
1389 * gst/rtsp-server/rtsp-address-pool.c:
1390 * gst/rtsp-server/rtsp-address-pool.h:
1391 * gst/rtsp-server/rtsp-stream.c:
1392 * tests/check/gst/addresspool.c:
1393 addresspool: return reason of failure
1394 Let gst_rtsp_address_pool_reserve_address() return the reason why
1395 the address could not be reserved.
1396 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
1398 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
1401 autogen.sh: Sync behaviour with other GStreamer modules
1402 Allows building from outside of tree amongst other things
1404 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
1407 Automatic update of common submodule
1408 From b613661 to 6b03ba7
1410 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
1413 Automatic update of common submodule
1414 From 74a6857 to b613661
1416 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
1419 Automatic update of common submodule
1420 From 01a7a46 to 74a6857
1422 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
1424 * gst/rtsp-server/rtsp-client.c:
1425 client: Do not read beyond end of path string
1426 If the setup was done without a control url, make sure we don't try to read the
1427 non-existing control string and crash.
1429 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1431 * gst/rtsp-server/rtsp-client.c:
1432 client: Fix RTPInfo header
1433 Refactor the method to make the content_base.
1434 Use the content-base and the control url to construct the RTPInfo
1437 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1439 * gst/rtsp-server/rtsp-client.c:
1440 client: map url to path only in describe
1441 Only map the request url to a path in the DESCRIBE method. The SDP then
1442 contains the base and control urls that should be used to SETUP/PAUSE/
1443 PLAY/TEARDOWN the media.
1445 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1447 * gst/rtsp-server/rtsp-client.c:
1448 Revert "client: map URL to path in requests"
1449 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
1450 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
1451 contains the base and control urls which are used in the SETUP, PLAY,
1452 PAUSE and TEARDOWN requests.
1454 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1456 * gst/rtsp-server/rtsp-client.c:
1457 client: map URL to path in requests
1459 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1461 * gst/rtsp-server/rtsp-client.c:
1462 * gst/rtsp-server/rtsp-mount-points.c:
1463 * gst/rtsp-server/rtsp-mount-points.h:
1464 mount-points: make vmethod to make path from uri
1465 Make a vmethod to transform an url into a path. The path is then used to lookup
1466 the factory. This makes it possible to also use other bits of the url, such as
1467 the query parameters, to locate the factory.
1469 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
1471 * gst/rtsp-server/rtsp-thread-pool.c:
1472 * gst/rtsp-server/rtsp-thread-pool.h:
1473 thread-pool: Add cleanup to wait for the threadpool to finish
1474 Also fix race condition if two threads are asking for the first
1475 thread from the thread pool at once. This would case two internal
1476 GThreadPools to be created.
1477 https://bugzilla.gnome.org/show_bug.cgi?id=707753
1479 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
1481 * gst/rtsp-server/rtsp-client.c:
1482 * tests/check/gst/client.c:
1483 client: free threadpool
1484 https://bugzilla.gnome.org/show_bug.cgi?id=707638
1486 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
1488 * tests/check/gst/mountpoints.c:
1489 mountpoints tests: unref matched factories
1490 https://bugzilla.gnome.org/show_bug.cgi?id=707638
1492 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
1494 * tests/check/gst/media.c:
1495 media tests: unref thread pool and caps
1496 https://bugzilla.gnome.org/show_bug.cgi?id=707638
1498 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
1500 * gst/rtsp-server/rtsp-auth.c:
1501 * gst/rtsp-server/rtsp-media-factory.c:
1502 * gst/rtsp-server/rtsp-media.c:
1503 auth, media, media-factory: unref permissions
1504 https://bugzilla.gnome.org/show_bug.cgi?id=707638
1506 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1508 * examples/Makefile.am:
1509 Makefile: add rule for appsrc example
1511 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1513 * examples/test-appsrc.c:
1514 tests: add appsrc example
1515 Add an example on how to use appsrc to feed the server pipeline with data.
1517 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
1519 * gst/rtsp-server/rtsp-client.c:
1520 rtsp-client: remove query part from content-base string
1521 Make sure that after the control url has been resolved, it's
1522 not a part of the query-string.
1523 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
1525 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1527 * gst/rtsp-server/rtsp-client.c:
1528 client: don't check url in response
1529 There is no url or method in the response to check
1531 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1533 * gst/rtsp-server/rtsp-client.c:
1534 * gst/rtsp-server/rtsp-client.h:
1535 Add handle-response signal for when we receive a GET_PARAMETER response
1537 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1539 * gst/rtsp-server/rtsp-server.c:
1540 Fix gst_rtsp_server_client_filter, using wrong variable type
1542 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
1544 * gst/rtsp-server/rtsp-media-factory-uri.c:
1545 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
1546 For AAC we need to check for framed=true instead of parsed=true.
1547 https://bugzilla.gnome.org/show_bug.cgi?id=701384
1549 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1551 * gst/rtsp-server/rtsp-stream.c:
1552 stream: optimize pipeline for protocols
1553 When TCP is not an allowed protocol for the stream, avoid creating the
1554 appsrc/appsink/queue and tee elements.
1556 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1558 * gst/rtsp-server/rtsp-media.c:
1559 media: set protocols on streams
1561 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1563 * gst/rtsp-server/rtsp-client.c:
1564 client: use protocols supported by stream
1566 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1568 * gst/rtsp-server/rtsp-media-factory.c:
1569 * gst/rtsp-server/rtsp-media.c:
1570 * gst/rtsp-server/rtsp-stream.c:
1571 media-factory: allow all protocols
1573 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1575 * gst/rtsp-server/rtsp-media.c:
1576 media: configure protocols in new streams
1578 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1580 * gst/rtsp-server/rtsp-stream.c:
1581 * gst/rtsp-server/rtsp-stream.h:
1582 stream: add protocols property
1584 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1586 * gst/rtsp-server/rtsp-media.c:
1587 rtsp-media: send state in "new-state" signal
1588 https://bugzilla.gnome.org/show_bug.cgi?id=705110
1590 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
1593 build: add subdir-objects to AM_INIT_AUTOMAKE
1594 Fixes warnings with automake 1.14
1595 https://bugzilla.gnome.org/show_bug.cgi?id=705350
1597 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1599 * docs/libs/gst-rtsp-server-sections.txt:
1600 * gst/rtsp-server/rtsp-client.c:
1601 * gst/rtsp-server/rtsp-server.c:
1602 * gst/rtsp-server/rtsp-server.h:
1603 server: add method to iterate clients of server
1605 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1607 * gst/rtsp-server/rtsp-media.c:
1608 * gst/rtsp-server/rtsp-media.h:
1609 Add vmethod for rtsp-media subclass to access rtpbin
1611 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1613 * gst/rtsp-server/rtsp-client.h:
1614 small documentation fix
1616 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1618 * gst/rtsp-server/rtsp-client.c:
1619 Do not take range header if range is invalid
1621 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1623 * docs/libs/gst-rtsp-server-sections.txt:
1624 * gst/rtsp-server/rtsp-media.c:
1625 media: add docs for new method
1627 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1629 * gst/rtsp-server/rtsp-media.c:
1630 * gst/rtsp-server/rtsp-media.h:
1631 Add API to rtsp-media set the pipeline's state
1633 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1635 * gst/rtsp-server/rtsp-media.c:
1636 Update current position/duration when gst_rtsp_media_get_range_string is called
1638 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1640 * examples/test-cgroups.c:
1641 tests: add some more docs
1643 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1645 * examples/test-cgroups.c:
1646 * gst/rtsp-server/Makefile.am:
1647 * gst/rtsp-server/rtsp-auth.c:
1648 * gst/rtsp-server/rtsp-auth.h:
1649 * gst/rtsp-server/rtsp-client.c:
1650 * gst/rtsp-server/rtsp-client.h:
1651 * gst/rtsp-server/rtsp-context.c:
1652 * gst/rtsp-server/rtsp-context.h:
1653 * gst/rtsp-server/rtsp-params.c:
1654 * gst/rtsp-server/rtsp-params.h:
1655 * gst/rtsp-server/rtsp-server.c:
1656 * gst/rtsp-server/rtsp-thread-pool.c:
1657 * gst/rtsp-server/rtsp-thread-pool.h:
1658 * tests/check/gst/client.c:
1659 ClientState -> Context
1660 Rename the clientstate to context and put the code in a separate file.
1662 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1664 * examples/test-auth.c:
1665 * gst/rtsp-server/rtsp-auth.c:
1666 * gst/rtsp-server/rtsp-auth.h:
1667 auth: add support for default token
1668 The default token is used when the user is not authenticated and can be used to
1669 give minimal permissions.
1671 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1673 * examples/test-auth.c:
1674 * gst/rtsp-server/rtsp-auth.c:
1675 auth: use defines when possible
1677 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1679 * gst/rtsp-server/rtsp-address-pool.c:
1680 address-pool: improve docs
1682 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1684 * gst/rtsp-server/rtsp-permissions.c:
1685 permissions: add the role to the copy
1687 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
1689 * gst/rtsp-server/rtsp-permissions.c:
1690 permissions: Also copy the roles
1692 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
1694 * gst/rtsp-server/rtsp-permissions.c:
1695 permissions: Make it build
1697 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1699 * gst/rtsp-server/rtsp-address-pool.h:
1702 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1704 * docs/libs/gst-rtsp-server-sections.txt:
1705 * gst/rtsp-server/rtsp-auth.c:
1706 * gst/rtsp-server/rtsp-auth.h:
1707 * gst/rtsp-server/rtsp-media.c:
1708 * gst/rtsp-server/rtsp-session-media.c:
1709 * gst/rtsp-server/rtsp-stream-transport.c:
1710 * gst/rtsp-server/rtsp-stream-transport.h:
1711 * gst/rtsp-server/rtsp-stream.c:
1712 * tests/check/gst/client.c:
1715 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1717 * docs/libs/gst-rtsp-server-sections.txt:
1718 * gst/rtsp-server/rtsp-address-pool.c:
1719 * gst/rtsp-server/rtsp-address-pool.h:
1720 * tests/check/gst/addresspool.c:
1721 * tests/check/gst/rtspserver.c:
1722 address-pool: cleanups
1723 Remove redundant method, improve docs.
1725 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1727 * docs/libs/gst-rtsp-server-sections.txt:
1728 * gst/rtsp-server/rtsp-auth.h:
1729 * gst/rtsp-server/rtsp-permissions.c:
1730 * gst/rtsp-server/rtsp-permissions.h:
1731 * gst/rtsp-server/rtsp-token.c:
1734 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1736 * gst/rtsp-server/rtsp-permissions.c:
1737 permissions: implement _remove_role
1739 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1741 * gst/rtsp-server/rtsp-permissions.c:
1742 permissions: update docs
1744 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1746 * tests/check/gst/client.c:
1747 tests: simplify tests
1748 Client settings are now disabled by default so we don't need an auth
1749 module to disable them.
1751 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1753 * gst/rtsp-server/rtsp-auth.c:
1754 auth: add default authorizations
1755 When no auth module is specified, use our table of defaults to look up the
1756 default value of the check instead of always allowing everything. This was
1757 we can disallow client settings by default.
1759 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1762 README: update readme
1764 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1766 * gst/rtsp-server/rtsp-thread-pool.c:
1767 * gst/rtsp-server/rtsp-thread-pool.h:
1768 thread-pool: add more docs
1770 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1772 * gst/rtsp-server/rtsp-thread-pool.c:
1773 * gst/rtsp-server/rtsp-thread-pool.h:
1774 thread-pool: fix race in thread reuse
1775 If we try to reuse a thread right after we made it stop, we end up using a
1776 stopped thread. Catch this case and only reuse threads that are not stopping.
1778 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1780 * gst/rtsp-server/rtsp-server.c:
1781 server: add small debug
1783 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1785 * tests/check/gst/client.c:
1787 Add some permissions to media so we can use the auth and enable
1790 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1792 * gst/rtsp-server/rtsp-client.c:
1793 client: support pushed context in handle_request
1794 If we already have a pushed state, reuse it and add our own things. This makes
1795 it easier to write tests.
1797 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1799 * gst/rtsp-server/rtsp-auth.c:
1800 auth: don't auth on methods
1801 Don't authorize on methods anymore but on the resources that we
1802 try to access, this is more flexible.
1803 Move the authorization checks to where they are needed and let the
1804 check return the response on error.
1806 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1808 * gst/rtsp-server/rtsp-mount-points.c:
1809 mount-points: add some debug
1811 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1813 * tests/check/gst/client.c:
1814 tests: almost fix test
1816 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1818 * gst/rtsp-server/rtsp-auth.c:
1819 * gst/rtsp-server/rtsp-auth.h:
1820 * gst/rtsp-server/rtsp-client.c:
1821 * gst/rtsp-server/rtsp-client.h:
1822 * gst/rtsp-server/rtsp-server.c:
1823 * gst/rtsp-server/rtsp-server.h:
1824 auth: let the auth module check client_settings
1825 Let the auth module decide if client settings are allowed for the
1828 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1830 * gst/rtsp-server/rtsp-token.c:
1831 * gst/rtsp-server/rtsp-token.h:
1832 token: add method to check boolean permission
1834 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1836 * examples/test-auth.c:
1837 * examples/test-cgroups.c:
1838 * gst/rtsp-server/rtsp-token.c:
1839 * gst/rtsp-server/rtsp-token.h:
1840 token: simplify token constructor
1841 Use variable arguments to make easier API.
1843 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1845 * examples/test-auth.c:
1846 * examples/test-cgroups.c:
1847 * gst/rtsp-server/rtsp-media-factory.c:
1848 * gst/rtsp-server/rtsp-media-factory.h:
1849 media-factory: add convenience API for factory
1851 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1853 * examples/test-auth.c:
1854 * examples/test-cgroups.c:
1855 * gst/rtsp-server/rtsp-permissions.c:
1856 * gst/rtsp-server/rtsp-permissions.h:
1857 permissions: simplify API a little
1858 Avoid passing GstStructure in the add_role method, use varargs instead
1859 to construct the structure behind the scenes. We can then also use the
1860 structure name as the role and simplify some more logic.
1862 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1864 * gst/rtsp-server/rtsp-auth.c:
1867 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1869 * gst/rtsp-server/rtsp-auth.c:
1870 * gst/rtsp-server/rtsp-auth.h:
1871 * gst/rtsp-server/rtsp-client.c:
1872 auth: handle unauthorized response
1873 Move handling of the unauthorized response to the auth module, it can add
1874 the appropriate headers to request authorization for the required method
1875 much better than the client.
1877 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1879 * gst/rtsp-server/rtsp-client.c:
1880 * gst/rtsp-server/rtsp-client.h:
1881 client: allow for sending any message, not only requests
1882 Change the _send_request() method to _send_message() so that we
1883 can both send requests and replies.
1885 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1887 * docs/libs/gst-rtsp-server-sections.txt:
1888 * gst/rtsp-server/rtsp-server.h:
1891 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1893 * examples/test-video.c:
1894 * gst/rtsp-server/rtsp-auth.c:
1895 * gst/rtsp-server/rtsp-auth.h:
1896 * gst/rtsp-server/rtsp-server.c:
1897 * gst/rtsp-server/rtsp-server.h:
1898 auth: move TLS handling to auth module
1899 Remove the TLS settings on the server and move it to the auth module because
1900 that is where security related bits go.
1902 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1904 * gst/rtsp-server/rtsp-client.c:
1905 * gst/rtsp-server/rtsp-client.h:
1906 client: add state push/pop
1908 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1910 * gst/rtsp-server/rtsp-client.c:
1911 * gst/rtsp-server/rtsp-client.h:
1912 client: add connection to state
1914 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1916 * gst/rtsp-server/rtsp-mount-points.c:
1917 mount-points: fix debug
1919 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1921 * tests/check/gst/media.c:
1922 tests: fix media test
1924 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1926 * gst/rtsp-server/rtsp-thread-pool.c:
1927 thread-pool: we don't require a state
1929 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1931 * gst/rtsp-server/rtsp-server.c:
1932 server: let context ref the server
1933 So that we don't risk losing the server object early anc crash.
1935 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1937 * tests/check/gst/client.c:
1938 tests: fix client test
1940 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1943 * docs/libs/gst-rtsp-server-docs.sgml:
1944 * docs/libs/gst-rtsp-server-sections.txt:
1945 * gst/rtsp-server/rtsp-address-pool.c:
1946 * gst/rtsp-server/rtsp-auth.c:
1947 * gst/rtsp-server/rtsp-client.c:
1948 * gst/rtsp-server/rtsp-client.h:
1949 * gst/rtsp-server/rtsp-media-factory-uri.c:
1950 * gst/rtsp-server/rtsp-media-factory.c:
1951 * gst/rtsp-server/rtsp-media-factory.h:
1952 * gst/rtsp-server/rtsp-media.c:
1953 * gst/rtsp-server/rtsp-mount-points.c:
1954 * gst/rtsp-server/rtsp-params.c:
1955 * gst/rtsp-server/rtsp-permissions.c:
1956 * gst/rtsp-server/rtsp-sdp.c:
1957 * gst/rtsp-server/rtsp-server.c:
1958 * gst/rtsp-server/rtsp-server.h:
1959 * gst/rtsp-server/rtsp-session-media.c:
1960 * gst/rtsp-server/rtsp-session-pool.c:
1961 * gst/rtsp-server/rtsp-session.c:
1962 * gst/rtsp-server/rtsp-stream-transport.c:
1963 * gst/rtsp-server/rtsp-stream.c:
1964 * gst/rtsp-server/rtsp-thread-pool.c:
1965 * gst/rtsp-server/rtsp-token.c:
1968 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1970 * gst/rtsp-server/rtsp-session-pool.c:
1971 * gst/rtsp-server/rtsp-session-pool.h:
1972 session-pool: make vmethod to create a session
1973 Make a vmethod to create a sessions so that subclasses can create
1974 custom session objects
1976 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1978 * gst/rtsp-server/rtsp-auth.c:
1979 * gst/rtsp-server/rtsp-media-factory.h:
1980 * gst/rtsp-server/rtsp-media.h:
1981 * gst/rtsp-server/rtsp-mount-points.h:
1982 * gst/rtsp-server/rtsp-session-pool.h:
1983 * gst/rtsp-server/rtsp-stream.h:
1986 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1988 * docs/libs/gst-rtsp-server-docs.sgml:
1989 * docs/libs/gst-rtsp-server-sections.txt:
1990 * gst/rtsp-server/rtsp-address-pool.c:
1991 * gst/rtsp-server/rtsp-address-pool.h:
1992 * gst/rtsp-server/rtsp-auth.c:
1993 * gst/rtsp-server/rtsp-client.h:
1994 * gst/rtsp-server/rtsp-media-factory.h:
1995 * gst/rtsp-server/rtsp-media.c:
1996 * gst/rtsp-server/rtsp-media.h:
1997 * gst/rtsp-server/rtsp-permissions.c:
1998 * gst/rtsp-server/rtsp-permissions.h:
1999 * gst/rtsp-server/rtsp-server.h:
2000 * gst/rtsp-server/rtsp-session-media.c:
2001 * gst/rtsp-server/rtsp-session-media.h:
2002 * gst/rtsp-server/rtsp-session-pool.h:
2003 * gst/rtsp-server/rtsp-session.h:
2004 * gst/rtsp-server/rtsp-stream-transport.h:
2005 * gst/rtsp-server/rtsp-stream.c:
2006 * gst/rtsp-server/rtsp-thread-pool.h:
2009 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2012 * examples/Makefile.am:
2013 configure: compile cgroup example conditionally
2014 Only compile the cgroup example when we have libcgroup
2016 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2019 * examples/Makefile.am:
2020 * examples/test-cgroups.c:
2021 examples: add cgroups example
2023 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2025 * tests/check/gst/rtspserver.c:
2026 tests: fix compilation
2028 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2030 * gst/rtsp-server/rtsp-thread-pool.c:
2031 thread-pool: fix vmethod invocation
2033 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2035 * gst/rtsp-server/rtsp-thread-pool.c:
2036 * gst/rtsp-server/rtsp-thread-pool.h:
2037 thread-pool: store thread type in thread
2039 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2041 * gst/rtsp-server/rtsp-client.c:
2042 client: pass thread from pool to media _prepare
2043 Get a thread from the configured threadpool and pass it to the prepare method of
2046 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2048 * gst/rtsp-server/rtsp-media.c:
2049 * gst/rtsp-server/rtsp-media.h:
2050 media: Accept a thread in _prepare
2051 Remove out own threadpool handling and use the provided thread and
2052 maincontext for the bus messages and the state changes.
2054 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2056 * gst/rtsp-server/rtsp-server.c:
2057 server: configure client thread pool
2059 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2061 * gst/rtsp-server/rtsp-client.c:
2062 * gst/rtsp-server/rtsp-client.h:
2063 client: add method to configure thread pool
2065 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2067 * gst/rtsp-server/rtsp-client.h:
2068 * gst/rtsp-server/rtsp-server.c:
2069 * gst/rtsp-server/rtsp-server.h:
2070 server: use thread pool
2071 Use the thread pool instead of doing our own thing.
2073 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2075 * gst/rtsp-server/Makefile.am:
2076 * gst/rtsp-server/rtsp-thread-pool.c:
2077 * gst/rtsp-server/rtsp-thread-pool.h:
2078 thread-pool: add object to manage threads
2079 Add an object to manage the client and media threads.
2081 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2083 * gst/rtsp-server/rtsp-auth.c:
2084 auth: debug authorization check
2086 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2088 * gst/rtsp-server/rtsp-media.c:
2089 media: start media pipeline in context
2090 Start the media pipeline in the provided context (or our default one
2091 when NULL). This makes sure that we run the bus thread in this context and that
2092 all media threads are children of this context.
2094 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2096 * gst/rtsp-server/rtsp-media-factory.c:
2097 factory: pass permissions to media by default
2099 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2101 * examples/test-auth.c:
2102 test: add permissions to auth test
2103 Ass some permissions to the media factory in the test.
2105 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2107 * gst/rtsp-server/rtsp-auth.c:
2108 * gst/rtsp-server/rtsp-auth.h:
2109 * gst/rtsp-server/rtsp-client.c:
2110 auth: simplify auth checks
2111 Remove client from methods, it's now in the state
2112 Perform the check specified by the string, use the information from the
2113 thread local context.
2115 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2117 * gst/rtsp-server/rtsp-client.c:
2118 * gst/rtsp-server/rtsp-client.h:
2119 client: add state to current thread
2120 Add the client to the ClientState object.
2121 Place the ClientState on the current thread.
2123 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2125 * gst/rtsp-server/rtsp-media-factory.c:
2126 * gst/rtsp-server/rtsp-media-factory.h:
2127 * gst/rtsp-server/rtsp-media.c:
2128 * gst/rtsp-server/rtsp-media.h:
2129 media: make it possible to set permissions
2130 Make it possible to set permissions on media and media factory objects
2132 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2134 * gst/rtsp-server/Makefile.am:
2135 * gst/rtsp-server/rtsp-permissions.c:
2136 * gst/rtsp-server/rtsp-permissions.h:
2137 permissions: add permissions object
2138 Add a mini object to store permissions based on a role.
2140 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2142 * examples/test-auth.c:
2143 * gst/rtsp-server/rtsp-auth.c:
2144 * gst/rtsp-server/rtsp-auth.h:
2145 * gst/rtsp-server/rtsp-client.c:
2146 auth: add auth checks
2147 Add an enum with auth checks and implement the checks in the auth object.
2148 Perform the checks from the client.
2150 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2152 * examples/test-auth.c:
2153 * gst/rtsp-server/rtsp-auth.c:
2154 * gst/rtsp-server/rtsp-auth.h:
2155 * gst/rtsp-server/rtsp-client.h:
2156 auth: use the token after authentication
2157 After we authenticated a user, keep the Token around in the state.
2159 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2161 * gst/rtsp-server/rtsp-client.c:
2162 * gst/rtsp-server/rtsp-media.c:
2163 * gst/rtsp-server/rtsp-media.h:
2164 * tests/check/gst/media.c:
2165 media: add optional context for bus messages
2166 Add an optional mainloop to _prepare that will handle the bus messages instead
2167 of always using the shared mainloop.
2169 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2171 * gst/rtsp-server/Makefile.am:
2172 * gst/rtsp-server/rtsp-token.c:
2173 * gst/rtsp-server/rtsp-token.h:
2174 token: add authorization token
2175 Add a simply miniobject that contains the authorizations. The object contains a
2176 GstStructure that hold all authorization fields. When a user is authenticated,
2177 the auth module will create a Token for the user. The token is then used to
2178 check what operations the user is allowed to do and various other configuration
2181 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2183 * examples/test-auth.c:
2184 * gst/rtsp-server/rtsp-auth.c:
2185 * gst/rtsp-server/rtsp-auth.h:
2186 * gst/rtsp-server/rtsp-client.c:
2187 * gst/rtsp-server/rtsp-client.h:
2188 * gst/rtsp-server/rtsp-media-factory.c:
2189 * gst/rtsp-server/rtsp-media-factory.h:
2190 * gst/rtsp-server/rtsp-media.c:
2191 * gst/rtsp-server/rtsp-media.h:
2192 auth: remove auth from media and factory
2193 Remove the auth object from media and factory. We want to have the RTSPClient
2194 authenticate and authorize resources, there is no need to place another auth
2195 manager on the media/factory.
2197 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2199 * examples/test-auth.c:
2200 * gst/rtsp-server/rtsp-auth.c:
2201 * gst/rtsp-server/rtsp-auth.h:
2202 * gst/rtsp-server/rtsp-client.h:
2203 auth: add support for multiple basic auth tokens
2204 Make it possible to add multiple basic authorisation tokens to one authorization
2205 object. Associate with each token an authorization group that will define what
2206 capabilities are allowed.
2208 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2210 * gst/rtsp-server/rtsp-client.c:
2211 client: error out on non-aggregate control
2212 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
2214 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2216 * gst/rtsp-server/rtsp-client.c:
2217 client: rework setup request a little
2218 Cache the media in DESCRIBE based on the longest matching path with the uri
2219 that we can find in the mount points.
2220 Rework the setup request a little to get the media from the session or from
2221 the longest matching path, this way we can derive the control string as
2222 everything after the path instead of hardcoding it.
2223 Find the stream based on the control string and only open a session when all
2226 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2228 * gst/rtsp-server/rtsp-media.c:
2229 * gst/rtsp-server/rtsp-media.h:
2230 media: add method to find a stream by control url
2232 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2234 * gst/rtsp-server/rtsp-stream.c:
2235 * gst/rtsp-server/rtsp-stream.h:
2236 stream: add method to check control url of stream
2238 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2240 * gst/rtsp-server/rtsp-client.c:
2241 * gst/rtsp-server/rtsp-session-media.c:
2242 * gst/rtsp-server/rtsp-session-media.h:
2243 * gst/rtsp-server/rtsp-session.c:
2244 * gst/rtsp-server/rtsp-session.h:
2245 session: use path matching for session media
2246 Use a path string instead of a uri to lookup session media in the sessions. Also
2247 use path matching to find the largest possible path that matches.
2249 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2251 * gst/rtsp-server/rtsp-client.c:
2252 * gst/rtsp-server/rtsp-mount-points.c:
2253 * gst/rtsp-server/rtsp-mount-points.h:
2254 * tests/check/gst/mountpoints.c:
2255 mount-points: remove useless vmethod
2256 Making lookups in the mount points should not be done with a URL, if there is a
2257 mapping to be done from URL to mount points, we'll need to do it somewhere
2260 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2262 * gst/rtsp-server/rtsp-mount-points.c:
2263 * gst/rtsp-server/rtsp-mount-points.h:
2264 * tests/check/gst/mountpoints.c:
2265 mount-points: improve mount point searching
2266 Use a GSequence to keep track of the mount points.
2267 Match a URL to the longest matching registered mount point. This should be the
2268 URL to perform aggreagate control and the remainder is the stream specific
2270 Add some unit tests for this.
2272 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
2274 * gst/rtsp-server/Makefile.am:
2275 rtsp-server: Allow building of static library
2277 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2279 * tests/check/gst/mediafactory.c:
2280 tests: fix compilation
2282 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2284 * gst/rtsp-server/rtsp-sdp.c:
2285 sdp: get control string from stream
2286 Use the control string as configured in the stream.
2288 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2290 * gst/rtsp-server/rtsp-stream.c:
2291 * gst/rtsp-server/rtsp-stream.h:
2292 stream: add methods and property to set control string
2294 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2296 * gst/rtsp-server/rtsp-client.c:
2298 Rename variables for clarity
2299 Keep media in state when we can
2301 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2303 * gst/rtsp-server/rtsp-client.c:
2304 * gst/rtsp-server/rtsp-stream.c:
2305 * gst/rtsp-server/rtsp-stream.h:
2306 stream: add more support for IPv6
2307 Rename _get_address to _get_multicast_address in GstRTSPStream to
2308 make it clear that this function only deals with multicast.
2309 Make it possible to have both an IPv4 and IPv6 multicast address on
2310 a stream. Give the client an IPv4 or IPv6 address depending on the
2311 address it used to connect to the server.
2312 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
2314 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2316 * gst/rtsp-server/rtsp-client.c:
2319 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2321 * gst/rtsp-server/rtsp-stream.c:
2322 stream: handle failed port allocation
2323 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
2324 can't allocate any family at all. Also keep track of what port families we
2326 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
2328 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2330 * gst/rtsp-server/rtsp-stream.c:
2331 stream: improve docs
2333 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2335 * gst/rtsp-server/rtsp-stream-transport.c:
2336 stream-transport: remove old if 0 block
2338 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
2340 * tests/check/gst/client.c:
2342 gst_rtsp_client_get_uri() has been removed
2343 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
2345 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2347 * gst/rtsp-server/rtsp-client.c:
2348 * gst/rtsp-server/rtsp-client.h:
2349 client: add method to filter managed sessions
2350 Add a method to filter the sessions managed by this client connection.
2351 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
2353 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2355 * gst/rtsp-server/rtsp-client.c:
2356 * gst/rtsp-server/rtsp-client.h:
2357 client: remove _get_uri() method
2358 Remove the get_uri() method on the client. A client has no uri, the uri
2359 property is an internal property to manage the last cached media for
2362 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2364 * gst/rtsp-server/rtsp-media-factory.h:
2365 media-factory: fix typo
2367 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
2369 * gst/rtsp-server/rtsp-media.c:
2370 rtsp-media: Do not leak the query in default_query_stop
2371 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
2373 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2375 * gst/rtsp-server/rtsp-media.c:
2376 media: don't unlock when conversion fails
2377 Don't unlock the state lock when conversion fails because it was not locked.
2379 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2381 * gst/rtsp-server/rtsp-media.c:
2382 * gst/rtsp-server/rtsp-media.h:
2383 Add query_position and query_stop vmethods to rtsp-media
2385 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2387 * gst/rtsp-server/rtsp-media.c:
2388 Fix typo in property install for rtsp-media's time-provider
2390 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2392 * gst/rtsp-server/rtsp-client.c:
2393 * gst/rtsp-server/rtsp-client.h:
2394 client: clean some variables
2395 Clean some variables and add some guards to _send_request()
2397 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2399 * gst/rtsp-server/rtsp-client.c:
2400 * gst/rtsp-server/rtsp-client.h:
2401 Add gst_rtsp_client_send_request API
2402 This makes it possible to send arbitrary messages to a client, such as
2403 SET_PARAMETER or GET_PARAMETER
2405 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2407 * gst/rtsp-server/rtsp-media.c:
2408 * gst/rtsp-server/rtsp-media.h:
2409 media: add _get_element() method
2410 Add method to get the element used when creating the media.
2411 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
2413 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2415 * gst/rtsp-server/rtsp-media.c:
2418 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
2420 * gst/rtsp-server/rtsp-stream.c:
2421 * gst/rtsp-server/rtsp-stream.h:
2422 stream: allow access to the rtp session
2423 https://bugzilla.gnome.org/show_bug.cgi?id=703004
2425 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
2427 * gst/rtsp-server/rtsp-stream.c:
2428 * gst/rtsp-server/rtsp-stream.h:
2429 dscp qos support in gst-rtsp-stream
2430 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
2432 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2434 * tests/check/gst/rtspserver.c:
2436 Actually do what the comment says. Also keep the old code around, not sure what
2437 should happen when you get a 454 from a TEARDOWN, does it close the connection?
2438 it currently doesn't.
2440 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2442 * gst/rtsp-server/rtsp-client.c:
2443 client: also watch newly created session
2444 When we newly created a session, start watching it immediately instead of
2445 on the next request.
2447 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
2449 * tests/check/gst/client.c:
2450 tests: add unit test for new-session
2451 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
2453 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2455 * gst/rtsp-server/rtsp-client.c:
2456 client: emit new-session when new session is created
2457 Only emit new-session when we created a new session for a client, not when a
2458 client picked up a previous session.
2459 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
2461 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
2463 * gst/rtsp-server/rtsp-client.c:
2464 client: handle asterisk as path in requests
2465 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
2467 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2469 * gst/rtsp-server/rtsp-media.c:
2470 media: handle segment query format mismatch
2471 It's possible that the segment query returns with a different format than what
2472 we asked for, handle this case also.
2474 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
2476 * gst/rtsp-server/rtsp-media.c:
2477 media: use segment stop in collect_media_stats
2478 Use segment stop instead of duration as range end point.
2479 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
2481 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
2483 * gst/rtsp-server/rtsp-media.c:
2484 * tests/check/gst/media.c:
2485 rtsp-media: Do not leak the element in take_pipeline
2486 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
2488 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
2490 * gst/rtsp-server/rtsp-client.c:
2491 * gst/rtsp-server/rtsp-client.h:
2492 rtsp-client: Make configure_client_transport virtual
2493 This patch makes configure_client_transport virtual. The functionality is
2494 needed to handle some weird clients sending multicast transport settings as url
2496 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
2498 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
2500 * gst/rtsp-server/rtsp-client.c:
2501 * gst/rtsp-server/rtsp-client.h:
2502 rtsp-client: Make param_set and param_get virtual
2503 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
2505 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
2507 * gst/rtsp-server/rtsp-client.c:
2508 * gst/rtsp-server/rtsp-media.c:
2509 * gst/rtsp-server/rtsp-media.h:
2510 media: convert_range replaces get_range_times
2511 get_range_times worked for handling UTC ranges for seeks, but we also
2512 need to convert back from NPT to the requested unit in
2513 get_range_string. convert_range is now used for both.
2514 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
2516 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2518 * gst/rtsp-server/rtsp-client.c:
2519 * gst/rtsp-server/rtsp-sdp.c:
2520 * gst/rtsp-server/rtsp-sdp.h:
2521 sdp: cleanup sdp info
2522 We don't need to pass the proto, we can more easily check a boolean.
2523 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
2525 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
2527 * gst/rtsp-server/rtsp-sdp.c:
2528 use 0.0.0.0 or :: for c= line instead of server address
2530 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
2532 * gst/rtsp-server/rtsp-client.c:
2533 use local address, not remote, in SDP
2534 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
2536 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2539 Automatic update of common submodule
2540 From 098c0d7 to 01a7a46
2542 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
2544 * gst/rtsp-server/rtsp-media.c:
2545 * gst/rtsp-server/rtsp-media.h:
2546 media: possibility to override range time conversion
2547 Make it possible to override the conversion from GstRTSPTimeRange to
2548 GstClockTimes, that is done before seeking on the media
2549 pipeline. Overriding can be useful for UTC ranges, where the default
2550 conversion gives nanoseconds since 1900.
2551 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
2553 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
2555 * gst/rtsp-server/rtsp-server.c:
2556 * gst/rtsp-server/rtsp-server.h:
2557 rtsp-server: Expose the use_client_settings API
2558 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
2560 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
2562 * gst/rtsp-server/rtsp-client.c:
2563 * gst/rtsp-server/rtsp-stream.c:
2564 * gst/rtsp-server/rtsp-stream.h:
2565 rtspstream: handle both ipv4 and ipv6 clients
2566 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
2568 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2570 * gst/rtsp-server/rtsp-sdp.c:
2571 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
2572 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
2573 We already have a way to place extra attributes in the SDP by using a string
2574 property with prefix x- or a- in the caps.
2576 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2578 * gst/rtsp-server/rtsp-sdp.c:
2579 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
2580 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
2581 We already have a way to place extra attributes in the SDP, just make a string
2582 property in the payloader with a- or x- prefix.
2584 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2586 * gst/rtsp-server/rtsp-sdp.c:
2587 rtsp: place a- and x- properties as attributes
2588 application/x-rtp has properties with a- and x- prefixes that should be
2589 placed as attributes in the SDP for the media instead of being added to the
2592 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2594 * examples/Makefile.am:
2595 * examples/test-video.c:
2596 example: add TLS example
2598 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2600 * gst/rtsp-server/rtsp-server.c:
2601 * gst/rtsp-server/rtsp-server.h:
2602 server: add support for TLS
2603 Add methods to set and get a TLS certificate.
2604 Add vmethod to configure a new connection. By default, configure the TLS
2605 certificate in a new connection if needed.
2607 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2609 * gst/rtsp-server/rtsp-server.c:
2610 * gst/rtsp-server/rtsp-server.h:
2611 server: remove accept_client vmethod
2612 This vmethod is not very useful so remove it.
2614 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2616 * gst/rtsp-server/rtsp-server.c:
2617 server: don't crash on NULL GError
2619 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
2621 * gst/rtsp-server/rtsp-session-pool.c:
2622 rtsp-session-pool: corrected session timeout detection
2623 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
2625 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2627 * gst/rtsp-server/rtsp-client.c:
2628 client: improve debug
2630 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2632 * gst/rtsp-server/rtsp-client.c:
2633 * gst/rtsp-server/rtsp-client.h:
2634 * gst/rtsp-server/rtsp-server.c:
2635 server: refactor connection setup
2636 Let the server accept the socket connection and construct a GstRTSPConnection
2637 from it. Remove the code from the client and let the client only deal with
2638 a fully configure GstRTSPConnection object.
2639 We will need this later when the server will configure the connection for
2642 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2644 * gst/rtsp-server/rtsp-stream.c:
2645 stream: keep the transport object alive
2646 Keep the transport object alive while we have it as qdata on the
2649 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
2651 * gst/rtsp-server/rtsp-client.c:
2652 * gst/rtsp-server/rtsp-server.c:
2653 rtsp-server: Do not crash on nmapping of server
2654 * generate error when gst_rtsp_connection_accept fails
2655 * do not stop accepting incoming connections because
2656 accepting a client fails
2657 https://bugzilla.gnome.org/show_bug.cgi?id=701072
2659 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
2661 * gst/rtsp-server/rtsp-client.c:
2662 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
2663 https://bugzilla.gnome.org/show_bug.cgi?id=700953
2665 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
2667 * gst/rtsp-server/rtsp-sdp.c:
2668 rtsp-sdp: Parse framerate caps field and set SDP attribute
2669 The SDP attribute and its format is described in RFC4566.
2670 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
2672 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
2674 * gst/rtsp-server/rtsp-sdp.c:
2675 rtsp-sdp: Parse width/height from caps and set SDP attribute
2676 The SDP attribute and its format is described in RFC6064.
2677 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
2679 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
2681 * gst/rtsp-server/rtsp-sdp.c:
2682 * tests/check/gst/client.c:
2683 rtsp-sdp: add bandwidth line
2684 https://bugzilla.gnome.org/show_bug.cgi?id=699220
2686 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2689 Automatic update of common submodule
2690 From 5edcd85 to 098c0d7
2692 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
2694 * tests/check/gst/media.c:
2695 tests: add dynamic payloader prepare/unprepare check
2697 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2699 * gst/rtsp-server/rtsp-media.c:
2700 media: release lock when removing fakesink
2702 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2704 * gst/rtsp-server/rtsp-stream.c:
2705 stream: set elements to NULL before removing
2706 When removing a stream, set the elements to NULL first. This avoids
2707 element-is-not-in-NULL-state errors when we dispose the elements.
2709 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
2712 Automatic update of common submodule
2713 From 3cb3d3c to 5edcd85
2715 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2717 * gst/rtsp-server/rtsp-media.c:
2718 * gst/rtsp-server/rtsp-media.h:
2719 media: listen to pad-removed signals
2720 Listen to the pad-removed signal and remove the stream associated with the
2722 Add signal to be notified of the removed pad.
2723 Remove the fakesink in unprepare()
2724 Fix signatures of the signal methods
2726 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2728 * examples/test-sdp.c:
2729 tests: add example of reusable pipelines
2731 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
2733 * gst/rtsp-server/rtsp-stream.c:
2734 * gst/rtsp-server/rtsp-stream.h:
2735 stream: add method to get the srcpad
2737 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
2739 * tests/check/gst/media.c:
2740 check: add media prepare/unprepare test
2741 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2743 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
2745 * gst/rtsp-server/rtsp-media.c:
2746 media: disconnect from signal handlers in unprepare()
2747 We connected to the pad-added and no-more-pads signals in prepare() so
2748 we need to disconnect from them in unprepare().
2749 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2751 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
2753 * gst/rtsp-server/rtsp-media.c:
2754 media: don't free streams array
2755 Don't free the streams array in the unprepare() method, they were not
2757 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2759 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
2761 * gst/rtsp-server/rtsp-media.c:
2762 media: don't unref the pipeline in unprepare
2763 Unprepare() should undo what prepare() does. Because the pipeline is
2764 not created in prepare(), we should not unref it in unprepare()
2766 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
2768 * gst/rtsp-server/rtsp-stream.c:
2769 stream: clear session and caps for reuse
2770 Set the session and caps to NULL after unref otherwise we might unref
2772 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2774 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
2776 * gst/rtsp-server/rtsp-client.c:
2777 client: send out teardown signal before tearing down
2778 The advantage is that in the signal handler you get direct access to
2779 information about what streams are about to get torn down (in the
2780 GstRTSPClientState).
2781 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
2783 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
2785 * gst/rtsp-server/rtsp-client.c:
2786 * gst/rtsp-server/rtsp-client.h:
2787 client: expose connection
2788 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
2790 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
2793 Automatic update of common submodule
2794 From aed87ae to 3cb3d3c
2796 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2798 * gst/rtsp-server/rtsp-media.c:
2799 * gst/rtsp-server/rtsp-media.h:
2800 * gst/rtsp-server/rtsp-session-media.c:
2801 * gst/rtsp-server/rtsp-session-media.h:
2802 media: add method to get the base_time of the pipeline
2803 Together with a shared clock, this base-time could eventually be sent to
2804 the client so that it can reconstruct the exact running-time of the clock
2807 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2809 * gst/rtsp-server/Makefile.am:
2810 * gst/rtsp-server/rtsp-media.c:
2811 * gst/rtsp-server/rtsp-media.h:
2812 * gst/rtsp-server/rtsp-sdp.c:
2813 media: add GstNetTimeProvider support
2814 Add a property to let the media provide a GstNetTimeProvider for its clock.
2815 Make methods to get the clock and nettimeprovider
2816 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
2817 provider and also the current time of the clock. This should make it possible
2818 for (GStreamer) clients to slave their clock to the server clock.
2820 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
2823 Automatic update of common submodule
2824 From 04c7a1e to aed87ae
2826 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2828 * gst/rtsp-server/rtsp-media.c:
2829 media: wait for buffering to complete
2830 Wait for buffering to complete before changing the state to the target state.
2832 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2834 * gst/rtsp-server/rtsp-media.c:
2835 media: small cleanup
2837 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
2839 * tests/check/gst/rtspserver.c:
2840 tests: remove extra unref in test_setup_non_existing_stream
2841 The unref is not needed anymore, teardown runs without it.
2842 https://bugzilla.gnome.org/show_bug.cgi?id=696542
2844 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
2846 * tests/check/gst/rtspserver.c:
2847 tests: GSocketService cleanup in test_bind_already_in_use
2848 Use g_socket_service_stop so the rtspserver test stops listening for
2849 incoming connections in test_bind_already_in_use.
2850 https://bugzilla.gnome.org/show_bug.cgi?id=696541
2852 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
2854 * gst/rtsp-server/rtsp-media-factory.c:
2855 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
2856 Instead use a GWeakRef which is safe to use
2857 This is a known GLib bug, see:
2858 https://bugzilla.gnome.org/show_bug.cgi?id=667145
2860 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
2862 * gst/rtsp-server/rtsp-client.c:
2863 * gst/rtsp-server/rtsp-media.c:
2864 * gst/rtsp-server/rtsp-media.h:
2865 * gst/rtsp-server/rtsp-sdp.c:
2866 * tests/check/gst/media.c:
2867 * tests/check/gst/rtspserver.c:
2868 rtsp-media/client: Reply to PLAY request with same type of Range
2869 Remember the type of Range from the PLAY request and use the same type for
2872 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
2874 * gst/rtsp-server/rtsp-client.c:
2875 * gst/rtsp-server/rtsp-client.h:
2876 * tests/check/gst/client.c:
2877 rtsp-client: expose uri
2879 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
2881 * tests/check/gst/mediafactory.c:
2882 tests: Hold ref while creating second media
2883 To test if the media aren't shared, make sure we keep the first one while creating a second
2884 otherwise the same memory address may be reused.
2886 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
2889 configure: remove out-of-date comment
2891 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
2894 .gitignore: ignore more build files
2896 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
2898 * tests/check/Makefile.am:
2899 tests: use right _LIBS variable for gst-plugins-base libs
2901 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2903 * tests/check/Makefile.am:
2904 check: add librtp to libs
2906 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
2908 * tests/check/gst/rtspserver.c:
2909 tests: Add test to check selecting a port the server will send from
2911 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
2913 * tests/check/gst/rtspserver.c:
2914 tests: Make sure packets are actually received
2916 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
2918 * gst/rtsp-server/rtsp-stream.c:
2919 stream: Select unicast address from pool if appropriate
2921 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
2923 * gst/rtsp-server/rtsp-stream.c:
2924 stream: Properties are always there in Gst 1.0
2926 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
2928 * tests/check/gst/addresspool.c:
2929 tests: Add tests for unicast addresses in pool
2931 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
2933 * gst/rtsp-server/rtsp-address-pool.c:
2934 * tests/check/gst/addresspool.c:
2935 address-pool: Verify that multicast addresses are used for multicast and vice-versa
2937 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
2939 * docs/libs/gst-rtsp-server-sections.txt:
2940 * gst/rtsp-server/rtsp-address-pool.c:
2941 * gst/rtsp-server/rtsp-address-pool.h:
2942 * gst/rtsp-server/rtsp-stream.c:
2943 * tests/check/gst/addresspool.c:
2944 address-pool: Add unicast addresses
2946 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
2949 * gst/rtsp-server/rtsp-server.c:
2950 * tests/check/gst/rtspserver.c:
2951 rtsp-server: Limit the number of threads per server instance
2952 If we exceed the maximum, just round robin the clients over the existing
2955 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
2957 * gst/rtsp-server/rtsp-server.c:
2958 rtsp-server: No need to store the GMainContext in the client context
2960 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
2962 * tests/check/gst/rtspserver.c:
2963 tests: Add test for client disconnection
2965 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
2967 * tests/check/gst/rtspserver.c:
2968 tests: Test client and session timeouts with multiple threads
2970 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
2972 * gst/rtsp-server/rtsp-address-pool.c:
2973 * gst/rtsp-server/rtsp-auth.c:
2974 * gst/rtsp-server/rtsp-client.c:
2975 * gst/rtsp-server/rtsp-media-factory-uri.c:
2976 * gst/rtsp-server/rtsp-media-factory.c:
2977 * gst/rtsp-server/rtsp-media.c:
2978 * gst/rtsp-server/rtsp-mount-points.c:
2979 * gst/rtsp-server/rtsp-server.c:
2980 * gst/rtsp-server/rtsp-session-media.c:
2981 * gst/rtsp-server/rtsp-session-pool.c:
2982 * gst/rtsp-server/rtsp-session.c:
2983 Document locking and its order
2985 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
2987 * tests/check/gst/rtspserver.c:
2988 tests: Test that slow DESCRIBE don't block other clients
2990 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
2992 * tests/check/gst/client.c:
2993 tests: Add tests for client-requested multicast address
2995 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
2997 * docs/libs/gst-rtsp-server-sections.txt:
2998 docs: Put the various functions in the right sections
3000 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
3002 * docs/libs/gst-rtsp-server-docs.sgml:
3003 * docs/libs/gst-rtsp-server-sections.txt:
3004 * gst/rtsp-server/rtsp-address-pool.c:
3005 * gst/rtsp-server/rtsp-address-pool.h:
3006 docs: Generate docs for GstRTSPAddressPool
3008 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
3010 * gst/rtsp-server/rtsp-client.c:
3011 * gst/rtsp-server/rtsp-stream.c:
3012 * gst/rtsp-server/rtsp-stream.h:
3013 client: Check client provided addresses against the address pool
3015 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
3017 * gst/rtsp-server/rtsp-address-pool.c:
3018 * gst/rtsp-server/rtsp-address-pool.h:
3019 * tests/check/gst/addresspool.c:
3020 address-pool: Add API to request a specific address from the pool
3021 Also add relevant unit tests.
3023 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
3025 * tests/check/gst/mediafactory.c:
3026 tests: Check the passing around of a RTSPAddressPool
3027 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
3028 way down to the stream.
3030 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
3032 * tests/check/gst/addresspool.c:
3033 tests: Add more tests for the address pool
3035 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
3037 * gst/rtsp-server/rtsp-address-pool.c:
3038 address-pool: Fix off by one error
3039 When splitting a port range, the port after a skip is not part of range.
3041 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
3044 Automatic update of common submodule
3045 From 2de221c to 04c7a1e
3047 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
3050 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
3051 AM_CONFIG_HEADER was removed in automake 1.13
3052 https://bugzilla.gnome.org/show_bug.cgi?id=693368
3054 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
3057 Automatic update of common submodule
3058 From a942293 to 2de221c
3060 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3062 * gst/rtsp-server/rtsp-client.c:
3063 client: make sure the watch exists while sending data
3064 Protect the send_func with a lock. This allows us to wait for sending
3065 to complete before changing the send_func and user_data. We add an
3066 extra ref to the watch to make sure that it remains valid during
3068 When closing the connection, set the send_func to NULL
3069 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
3071 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3073 * tests/check/Makefile.am:
3074 tests: use GST_*_1_0 environment variables everywhere
3075 The _1_0 suffixed environment variables override the
3076 non-suffixed ones, so if we're in an environment that
3077 sets the _1_0 suffixed ones, such as jhbuild, we need
3078 to set those to make sure ours actually always get
3081 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3084 Automatic update of common submodule
3085 From acb04d9 to a942293
3087 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3089 * gst/rtsp-server/rtsp-client.c:
3090 rtsp-client: set the client backlog
3091 Set the client backlog to a reasonable default
3093 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
3095 * gst/rtsp-server/rtsp-media.c:
3096 rtsp-media: Make the element a constructor parameter
3097 https://bugzilla.gnome.org/show_bug.cgi?id=689594
3099 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3101 * docs/libs/Makefile.am:
3102 docs: Link with gcov library when gcov is enabled
3103 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
3105 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3107 * gst/rtsp-server/rtsp-media.c:
3108 media: match prepare with unprepare
3109 Really unprepare when there were an equal amount of prepare calls.
3111 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3113 * gst/rtsp-server/rtsp-media.c:
3114 media: media has to be unprepared in finalize
3115 Because unprepare takes away the last ref on the media.
3117 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3119 * gst/rtsp-server/rtsp-client.c:
3120 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
3121 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
3122 We can't use the refcount to trigger unprepare because it is the unprepare call
3123 that removes the last refcount after all messages are consumed. What we should
3124 probably do is make a prepared refcount and only unprepare when the refcount
3127 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3129 * gst/rtsp-server/rtsp-media.c:
3130 media: let the source unref the last media ref
3131 the last ref to the media is held by the source so we don't need to add more ref
3132 and unrefs, we simply destroy the media when the source is gone.
3134 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3136 * gst/rtsp-server/rtsp-media.c:
3137 media: improve debug
3139 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3141 * gst/rtsp-server/rtsp-media.c:
3143 Make sure we are in the right state when collecting the position and duration.
3144 Only make ourselves PREPARED when we were previously PREPARING.
3146 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3148 * gst/rtsp-server/rtsp-media.c:
3149 media: use g_object_ref/unref for GObjects
3151 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
3153 * gst/rtsp-server/rtsp-client.c:
3154 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
3155 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
3156 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
3157 isn't being used anymore.
3159 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
3161 * gst/rtsp-server/rtsp-media.c:
3162 Fix compiler warning
3164 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
3166 * gst/rtsp-server/rtsp-media-factory-uri.c:
3167 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
3169 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3171 * gst/rtsp-server/rtsp-session-media.h:
3174 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3176 * gst/rtsp-server/rtsp-media.c:
3177 * tests/check/gst/media.c:
3178 media: avoid element leak
3180 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3182 * gst/rtsp-server/rtsp-media.c:
3183 media: require an element in media constructor
3185 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3187 * gst/rtsp-server/rtsp-client.c:
3188 Revert "client: TEARDOWN brings that state to Init again"
3189 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
3190 The object is already disposed, there is no point in setting the state.
3192 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3194 * gst/rtsp-server/rtsp-client.c:
3195 client: TEARDOWN brings that state to Init again
3197 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3199 * docs/libs/gst-rtsp-server-sections.txt:
3200 * examples/test-auth.c:
3201 * gst/rtsp-server/rtsp-auth.c:
3202 * gst/rtsp-server/rtsp-auth.h:
3203 * gst/rtsp-server/rtsp-client.c:
3204 * gst/rtsp-server/rtsp-client.h:
3205 * gst/rtsp-server/rtsp-media-factory-uri.c:
3206 * gst/rtsp-server/rtsp-media-factory-uri.h:
3207 * gst/rtsp-server/rtsp-media-factory.c:
3208 * gst/rtsp-server/rtsp-media-factory.h:
3209 * gst/rtsp-server/rtsp-media.c:
3210 * gst/rtsp-server/rtsp-media.h:
3211 * gst/rtsp-server/rtsp-mount-points.c:
3212 * gst/rtsp-server/rtsp-mount-points.h:
3213 * gst/rtsp-server/rtsp-sdp.c:
3214 * gst/rtsp-server/rtsp-server.c:
3215 * gst/rtsp-server/rtsp-server.h:
3216 * gst/rtsp-server/rtsp-session-media.c:
3217 * gst/rtsp-server/rtsp-session-media.h:
3218 * gst/rtsp-server/rtsp-session-pool.c:
3219 * gst/rtsp-server/rtsp-session-pool.h:
3220 * gst/rtsp-server/rtsp-session.c:
3221 * gst/rtsp-server/rtsp-session.h:
3222 * gst/rtsp-server/rtsp-stream-transport.c:
3223 * gst/rtsp-server/rtsp-stream-transport.h:
3224 * gst/rtsp-server/rtsp-stream.c:
3225 * gst/rtsp-server/rtsp-stream.h:
3226 * tests/check/gst/media.c:
3227 rtsp: make object details private
3228 Make all object details private
3229 Add methods to access private bits
3231 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3233 * tests/check/Makefile.am:
3234 * tests/check/gst/media.c:
3235 tests: add media tests
3237 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3239 * gst/rtsp-server/rtsp-media.c:
3240 media: check if prepared for some methods
3241 Check that the media object is prepared before doing seek and getting the
3242 current position etc.
3243 Add some g_return checks.
3245 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3247 * tests/check/Makefile.am:
3248 * tests/check/gst/mediafactory.c:
3249 tests: add mediafactory test
3251 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3253 * gst/rtsp-server/rtsp-stream.c:
3254 stream: improve debug
3256 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3258 * gst/rtsp-server/rtsp-media.c:
3259 * gst/rtsp-server/rtsp-media.h:
3260 media: unref pipeline in finalize to avoid leaking it
3262 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3264 * gst/rtsp-server/rtsp-media-factory-uri.c:
3265 * gst/rtsp-server/rtsp-media.c:
3266 rtsp: use gst_object_unref on GstObjects
3268 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3270 * gst/rtsp-server/rtsp-media-factory.c:
3271 media-factory: require an url
3273 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3275 * examples/test-uri.c:
3276 examples: fix include
3278 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3280 * gst/rtsp-server/rtsp-server.h:
3281 server: remove unused include
3283 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3285 * tests/check/Makefile.am:
3286 * tests/check/gst/mountpoints.c:
3287 tests: add test for mountpoints
3289 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3291 * gst/rtsp-server/rtsp-client.c:
3292 client: fix factory leak
3293 Keep the factory in the state object only for authorization checks and make
3294 sure we unref it on failure. Also don't keep invalid objects in the state
3297 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3299 * gst/rtsp-server/rtsp-mount-points.c:
3300 mounts: add g_return_if guards
3302 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3304 * tests/check/gst/client.c:
3305 tests: add more tests
3307 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3309 * gst/rtsp-server/rtsp-client.c:
3310 client: improve debug
3312 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3314 * gst/rtsp-server/rtsp-client.c:
3315 client: improve debug and fix leaks
3316 Cleanup the uri and session when there is a bad request.
3318 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3323 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3325 * tests/check/gst/client.c:
3326 test: add test for session in options request
3328 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3330 * gst/rtsp-server/rtsp-client.c:
3331 client: use 454 when session can't be found
3332 We should use 454 when a session can't be found because there was no session
3333 pool configured in the server. This is not a server configuration problem
3334 because the server on which the request is done might not be the same one that
3335 will keep the sessions for us and so it does not need to support sessions.
3337 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3339 * gst/rtsp-server/rtsp-client.c:
3340 client: only free connection when there is one
3341 It's possible that the client doesn't have a connection when we try to free it.
3343 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3345 * tests/check/Makefile.am:
3346 * tests/check/gst/client.c:
3347 tests: add unit test for the client object
3349 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3351 * gst/rtsp-server/rtsp-client.c:
3352 client: small cleanup
3354 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3356 * gst/rtsp-server/rtsp-client.h:
3357 client: remove unused include
3359 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3361 * gst/rtsp-server/rtsp-client.c:
3362 client: fix compilation
3364 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3366 * gst/rtsp-server/rtsp-client.c:
3367 client: call destroy without the lock
3369 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3371 * gst/rtsp-server/rtsp-client.c:
3372 * gst/rtsp-server/rtsp-client.h:
3373 client: make the client usable without a socket
3374 Make a method to let the client handle a message and a callback when the client
3375 wants us to send a response message back. This makes it possible to also use the
3376 client object without the sockets, which should make it easier to test.
3378 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3380 * gst/rtsp-server/rtsp-client.c:
3381 * gst/rtsp-server/rtsp-client.h:
3382 client: small cleanup
3384 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3386 * docs/libs/gst-rtsp-server-sections.txt:
3387 * gst/rtsp-server/rtsp-client.c:
3388 * gst/rtsp-server/rtsp-client.h:
3389 * gst/rtsp-server/rtsp-server.c:
3390 client: remove reference to server
3391 We don't need to keep a ref to the server
3393 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3395 * gst/rtsp-server/rtsp-client.c:
3396 * gst/rtsp-server/rtsp-client.h:
3398 Also add some g_return_if()
3400 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3402 * gst/rtsp-server/rtsp-client.c:
3403 client: log more errors
3405 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3407 * gst/rtsp-server/rtsp-client.c:
3408 client: fix compilation
3410 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3412 * gst/rtsp-server/rtsp-client.c:
3413 * gst/rtsp-server/rtsp-client.h:
3414 client: add generic close-after-send support
3415 Add a property to send_response() to close the connection after the response has
3416 been sent to the client.
3418 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3421 * docs/libs/gst-rtsp-server-docs.sgml:
3422 * docs/libs/gst-rtsp-server-sections.txt:
3423 * docs/libs/gst-rtsp-server.types:
3424 * examples/test-auth.c:
3425 * examples/test-launch.c:
3426 * examples/test-mp4.c:
3427 * examples/test-multicast.c:
3428 * examples/test-multicast2.c:
3429 * examples/test-ogg.c:
3430 * examples/test-readme.c:
3431 * examples/test-sdp.c:
3432 * examples/test-uri.c:
3433 * examples/test-video.c:
3434 * gst/rtsp-server/Makefile.am:
3435 * gst/rtsp-server/rtsp-auth.h:
3436 * gst/rtsp-server/rtsp-client.c:
3437 * gst/rtsp-server/rtsp-client.h:
3438 * gst/rtsp-server/rtsp-media-mapping.c:
3439 * gst/rtsp-server/rtsp-media-mapping.h:
3440 * gst/rtsp-server/rtsp-mount-points.c:
3441 * gst/rtsp-server/rtsp-mount-points.h:
3442 * gst/rtsp-server/rtsp-server.c:
3443 * gst/rtsp-server/rtsp-server.h:
3444 * gst/rtsp-server/rtsp-session-media.c:
3445 * gst/rtsp-server/rtsp-session-pool.c:
3446 * gst/rtsp-server/rtsp-session-pool.h:
3447 * tests/check/gst/rtspserver.c:
3448 MediaMapping -> MountPoints
3449 Describes better what the object manages.
3451 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3454 configure: bump required version of -base
3456 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3458 * gst/rtsp-server/rtsp-media.c:
3461 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3463 * gst/rtsp-server/rtsp-media.c:
3464 * gst/rtsp-server/rtsp-media.h:
3465 media: support more Range formats
3466 Use the new -base methods to convert the Range string into a seek start and stop
3469 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3471 * examples/test-launch.c:
3472 examples: fix whitespace
3474 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3476 * examples/test-auth.c:
3477 test-auth: add example of how to remove sessions
3478 Add an example of the session filter api.
3480 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3482 * examples/test-uri.c:
3483 test-uri: remove mapping example
3485 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3487 * examples/test-uri.c:
3488 test-uri: fix callback signature
3490 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3492 * gst/rtsp-server/rtsp-media-factory.c:
3493 factory: keep ref to factory while media active
3494 While the media from a factory is alive, keep a ref to the factory.
3495 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
3497 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3499 * gst/rtsp-server/rtsp-media-factory-uri.c:
3500 factory-uri: add some debug
3502 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3504 * gst/rtsp-server/rtsp-stream.c:
3505 stream: set udp sources to PLAYING
3506 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
3507 so that it doesn't cause our pipeline to produce ASYNC-DONE.
3509 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3511 * gst/rtsp-server/rtsp-media-factory-uri.c:
3512 factory-uri: take ref to factory
3513 Take a ref to the factory that we place in our list.
3515 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3517 * tests/Makefile.am:
3518 * tests/test-reuse.c:
3519 test: add test for server reuse
3520 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
3522 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
3524 * gst/rtsp-server/rtsp-server.c:
3525 server: start and stop multiple times
3526 Stop listening on the RTSP port when the GSource is removed, so clients
3527 can't connect and the server can be started again.
3528 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
3530 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3532 * gst/rtsp-server/rtsp-server.c:
3533 server: fix small leak
3535 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3537 * gst/rtsp-server/rtsp-media.c:
3538 media: unref source in finish_unprepare
3539 The source is created in prepare, unref it in finish_unprepare.
3540 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
3542 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
3544 * gst/rtsp-server/rtsp-client.c:
3545 * gst/rtsp-server/rtsp-media.c:
3546 rtsp-media: remove bus watch before finalizing
3547 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
3548 * An extra media ref is added for the bus watch. This extra ref is unreffed by
3549 the GDestroyNotify function.
3550 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
3551 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
3552 gst_rtsp_media_unprepare before unreffing the media.
3553 This way, the bus watch will be removed before the media is finalized.
3554 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
3556 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
3558 * gst/rtsp-server/rtsp-client.c:
3559 * gst/rtsp-server/rtsp-client.h:
3560 client: wait until the TEARDOWN response is sent to close the connection
3561 Responses can be sent async so we need to wait until the TEARDOWN response has
3562 been written before we close the connection to the client. This avoids the risk
3563 of writing/polling closed sockets.
3564 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
3566 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
3568 * gst/rtsp-server/rtsp-stream.c:
3569 rtsp-stream: plug socket leak
3570 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
3572 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
3575 Automatic update of common submodule
3576 From 6bb6951 to a72faea
3578 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
3580 * gst/rtsp-server/rtsp-media-factory-uri.c:
3581 rtsp-server: don't use deprecated API
3583 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
3585 * gst/rtsp-server/rtsp-client.c:
3586 rtsp-client: fix unused-but-set-variable compiler warning
3587 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
3589 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3592 * docs/libs/gst-rtsp-server-sections.txt:
3593 * gst/rtsp-server/rtsp-client.c:
3596 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3598 * examples/Makefile.am:
3599 * examples/test-multicast2.c:
3600 examples: add another multicast example
3601 Add an example for how to configure separate multicast ranges for each media
3604 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3606 * examples/test-multicast.c:
3609 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3611 * gst/rtsp-server/rtsp-client.c:
3612 * gst/rtsp-server/rtsp-media.c:
3613 * gst/rtsp-server/rtsp-session-media.c:
3614 * gst/rtsp-server/rtsp-session-media.h:
3615 * gst/rtsp-server/rtsp-stream-transport.c:
3616 * gst/rtsp-server/rtsp-stream-transport.h:
3617 stream: use the address managed by the stream
3618 Use the address managed by the stream for multicast. This allows us to have 1
3619 multicast address for each stream.
3620 Because the address is now managed by the stream we don't have to pass it around
3622 Set the address pool on the streams.
3624 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3626 * gst/rtsp-server/rtsp-client.c:
3627 * gst/rtsp-server/rtsp-media.c:
3628 * gst/rtsp-server/rtsp-stream.c:
3631 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3633 * gst/rtsp-server/rtsp-media.c:
3634 * gst/rtsp-server/rtsp-media.h:
3635 media: add signal for new streams
3636 This allows applications to listen for new streams and configure properties on
3637 them, like the address pool.
3639 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3641 * gst/rtsp-server/rtsp-media.c:
3642 media: configure address pool in new streams
3644 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3646 * gst/rtsp-server/rtsp-stream.c:
3647 * gst/rtsp-server/rtsp-stream.h:
3648 stream: add methods to deal with address pool
3649 Add methods to get and set the address pool for the stream
3650 Add method to allocate and get the multicast addresses for this stream.
3652 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3654 * docs/libs/gst-rtsp-server-sections.txt:
3655 * gst/rtsp-server/rtsp-media.c:
3656 * gst/rtsp-server/rtsp-media.h:
3657 media: remove MTU property
3658 It is a stream property
3660 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3662 * gst/rtsp-server/rtsp-client.c:
3663 client: set blocksize only on stream
3664 Set the blocksize only on the current stream.
3666 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3668 * gst/rtsp-server/rtsp-stream.c:
3669 stream: share src and sink sockets
3670 the allocated socket is in the used-socket property, not socket.
3672 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3674 * gst/rtsp-server/rtsp-address-pool.c:
3675 * gst/rtsp-server/rtsp-address-pool.h:
3676 * gst/rtsp-server/rtsp-client.c:
3677 * gst/rtsp-server/rtsp-session-media.c:
3678 * gst/rtsp-server/rtsp-session-media.h:
3679 * gst/rtsp-server/rtsp-stream-transport.c:
3680 * gst/rtsp-server/rtsp-stream-transport.h:
3681 * tests/check/gst/addresspool.c:
3682 rtsp: make address-pool return an address object
3683 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
3684 store more info in the structure and allows us to more easily return the address
3685 to the right pool when no longer needed.
3686 Pass the address to the StreamTransport so that we can return it to the pool
3687 when the stream transport is freed or changed.
3689 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3691 * examples/Makefile.am:
3692 * examples/test-multicast.c:
3693 examples: add multicast example
3694 Show how to set up the multicast address pool so that media can be
3695 server with multicast.
3697 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3699 * gst/rtsp-server/rtsp-client.c:
3700 * gst/rtsp-server/rtsp-media-factory.c:
3701 * gst/rtsp-server/rtsp-media-factory.h:
3702 * gst/rtsp-server/rtsp-media.c:
3703 * gst/rtsp-server/rtsp-media.h:
3704 rtsp: use AddressPool
3705 Remove the multicast_group property.
3706 Use the configured addresspool to allocate multicast addresses.
3708 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3710 * gst/rtsp-server/rtsp-address-pool.c:
3711 * gst/rtsp-server/rtsp-address-pool.h:
3712 address-pool: add clear method
3714 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3716 * gst/rtsp-server/rtsp-address-pool.c:
3717 address-pool: small cleanups
3719 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3721 * tests/check/Makefile.am:
3722 * tests/check/gst/addresspool.c:
3723 tests: add addresspool unit test
3725 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3727 * gst/rtsp-server/Makefile.am:
3728 * gst/rtsp-server/rtsp-address-pool.c:
3729 * gst/rtsp-server/rtsp-address-pool.h:
3730 address-pool: add object to manage multicast addresses
3731 Make an object that can manage a rage of multicast addresses and ports.
3733 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3735 * gst/rtsp-server/rtsp-server.c:
3736 server: set default max-threads property
3738 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3740 * gst/rtsp-server/rtsp-media.c:
3741 media: wait for concurrent _prepare
3742 If a prepare is busy, wait for the result.
3744 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3746 * gst/rtsp-server/rtsp-media.c:
3747 media: add lock around message handler
3748 We don't want to dispatch messages while we are still processing the result of
3751 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3753 * gst/rtsp-server/rtsp-media.c:
3754 * gst/rtsp-server/rtsp-media.h:
3755 media: add lock to protect state changes
3757 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3759 * gst/rtsp-server/rtsp-stream.c:
3760 * gst/rtsp-server/rtsp-stream.h:
3763 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3765 * gst/rtsp-server/rtsp-stream-transport.c:
3766 * gst/rtsp-server/rtsp-stream-transport.h:
3767 * gst/rtsp-server/rtsp-stream.c:
3768 stream-transport: add keep-alive method
3770 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3772 * gst/rtsp-server/rtsp-stream-transport.c:
3773 * gst/rtsp-server/rtsp-stream-transport.h:
3774 * gst/rtsp-server/rtsp-stream.c:
3775 stream-transport: add method to handle RTP/RTCP
3776 Call new methods instead of poking into the structures directly.
3778 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3780 * gst/rtsp-server/rtsp-session-media.c:
3781 * gst/rtsp-server/rtsp-session-media.h:
3782 session-media: add locking
3784 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3786 * gst/rtsp-server/rtsp-session.c:
3787 * gst/rtsp-server/rtsp-session.h:
3788 session: add locking
3790 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3792 * gst/rtsp-server/rtsp-server.c:
3793 server: free old socket
3795 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3797 * gst/rtsp-server/rtsp-media-mapping.c:
3798 * gst/rtsp-server/rtsp-media-mapping.h:
3799 mapping: add locking
3801 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3803 * gst/rtsp-server/rtsp-media-factory.c:
3804 media-factory: add locking
3806 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3808 * gst/rtsp-server/rtsp-auth.c:
3809 * gst/rtsp-server/rtsp-auth.h:
3812 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3814 * gst/rtsp-server/rtsp-server.c:
3815 * gst/rtsp-server/rtsp-server.h:
3816 server: add max-thread property
3818 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3820 * gst/rtsp-server/rtsp-server.c:
3821 * gst/rtsp-server/rtsp-server.h:
3822 server: use a threadpool for the mainloops
3824 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3826 * gst/rtsp-server/rtsp-client.c:
3827 * gst/rtsp-server/rtsp-client.h:
3828 client: rename method
3829 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
3830 don't really create the client from the socket, we use the socket for the
3833 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3835 * gst/rtsp-server/rtsp-client.c:
3836 * gst/rtsp-server/rtsp-client.h:
3837 * gst/rtsp-server/rtsp-server.c:
3838 server: rework maincontext handling in clients
3839 Make a separate method to attach a client to a MainContext.
3840 Let the server decide in what GMainContext the client will operate and give this
3841 context to the client in attach. Then the server can later decide to use a
3842 separate thread for each client or just use the mainthread.
3844 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3846 * gst/rtsp-server/rtsp-client.c:
3847 * gst/rtsp-server/rtsp-session.c:
3848 * gst/rtsp-server/rtsp-session.h:
3849 session: move session header code in session object
3851 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
3855 * examples/test-auth.c:
3856 * examples/test-launch.c:
3857 * examples/test-mp4.c:
3858 * examples/test-ogg.c:
3859 * examples/test-readme.c:
3860 * examples/test-sdp.c:
3861 * examples/test-uri.c:
3862 * examples/test-video.c:
3863 * gst/rtsp-server/rtsp-auth.c:
3864 * gst/rtsp-server/rtsp-auth.h:
3865 * gst/rtsp-server/rtsp-client.c:
3866 * gst/rtsp-server/rtsp-client.h:
3867 * gst/rtsp-server/rtsp-media-factory-uri.c:
3868 * gst/rtsp-server/rtsp-media-factory-uri.h:
3869 * gst/rtsp-server/rtsp-media-factory.c:
3870 * gst/rtsp-server/rtsp-media-factory.h:
3871 * gst/rtsp-server/rtsp-media-mapping.c:
3872 * gst/rtsp-server/rtsp-media-mapping.h:
3873 * gst/rtsp-server/rtsp-media.c:
3874 * gst/rtsp-server/rtsp-media.h:
3875 * gst/rtsp-server/rtsp-params.c:
3876 * gst/rtsp-server/rtsp-params.h:
3877 * gst/rtsp-server/rtsp-sdp.c:
3878 * gst/rtsp-server/rtsp-sdp.h:
3879 * gst/rtsp-server/rtsp-server.c:
3880 * gst/rtsp-server/rtsp-server.h:
3881 * gst/rtsp-server/rtsp-session-media.c:
3882 * gst/rtsp-server/rtsp-session-media.h:
3883 * gst/rtsp-server/rtsp-session-pool.c:
3884 * gst/rtsp-server/rtsp-session-pool.h:
3885 * gst/rtsp-server/rtsp-session.c:
3886 * gst/rtsp-server/rtsp-session.h:
3887 * gst/rtsp-server/rtsp-stream-transport.c:
3888 * gst/rtsp-server/rtsp-stream-transport.h:
3889 * gst/rtsp-server/rtsp-stream.c:
3890 * gst/rtsp-server/rtsp-stream.h:
3891 * tests/check/gst/rtspserver.c:
3892 * tests/test-cleanup.c:
3895 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
3897 * gst/rtsp-server/rtsp-media.c:
3898 * gst/rtsp-server/rtsp-session-media.c:
3899 * gst/rtsp-server/rtsp-session.c:
3900 rtsp-server: added annotations to indicate type of ownership transfer of return values
3901 https://bugzilla.gnome.org/show_bug.cgi?id=680777
3903 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
3906 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
3908 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
3911 * bindings/Makefile.am:
3912 * bindings/vala/Makefile.am:
3913 * bindings/vala/gst-rtsp-server-0.10.deps:
3914 * bindings/vala/gst-rtsp-server-0.10.vapi:
3915 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
3916 * bindings/vala/packages/gst-rtsp-server-0.10.files:
3917 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
3918 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
3919 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
3921 bindings: remove vala bindings
3922 They'll be reunited with the other GStreamer bindings
3923 https://bugzilla.gnome.org/show_bug.cgi?id=680777
3925 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3927 * gst/rtsp-server/rtsp-client.c:
3928 * gst/rtsp-server/rtsp-session-media.c:
3929 * gst/rtsp-server/rtsp-session-media.h:
3930 * gst/rtsp-server/rtsp-stream-transport.c:
3931 * gst/rtsp-server/rtsp-stream-transport.h:
3932 rtsp: only create transport when needed
3933 Only create the StreamTransport when configured.
3935 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3937 * gst/rtsp-server/rtsp-client.c:
3938 client: small cleanup
3940 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3942 * gst/rtsp-server/rtsp-client.c:
3943 * gst/rtsp-server/rtsp-client.h:
3944 * gst/rtsp-server/rtsp-stream-transport.c:
3945 * gst/rtsp-server/rtsp-stream-transport.h:
3946 rtsp: refactor configuration of transport
3947 Move the configuration of the transport to a place where it makes
3950 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3952 * gst/rtsp-server/rtsp-client.c:
3953 client: refactor transport parsing
3955 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3957 * gst/rtsp-server/rtsp-client.c:
3958 client: refuse to change the MTU on shared media
3959 If we change the MTU of chared media, it changes for all clients.
3960 We don't want to set the MTU to something large for clients that
3963 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3965 * examples/test-mp4.c:
3966 * gst/rtsp-server/rtsp-media.c:
3967 small fixes to docs and debug
3969 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3971 * gst/rtsp-server/rtsp-stream.c:
3972 stream: transports must already have been removed
3974 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3976 * gst/rtsp-server/rtsp-media.c:
3977 * gst/rtsp-server/rtsp-stream.c:
3978 * gst/rtsp-server/rtsp-stream.h:
3979 stream: improve join and leave of the pipeline
3981 Do the cleanup properly
3984 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3986 * gst/rtsp-server/rtsp-media.c:
3987 media: move unprepare below default implementation
3988 Makes it easier to find the default implementation
3990 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3992 * gst/rtsp-server/rtsp-media.c:
3993 media: signal unprepared when we actually finish
3995 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3997 * gst/rtsp-server/rtsp-media.c:
3998 media: no need to unlock, unprepare does that when needed
4000 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4002 * docs/libs/gst-rtsp-server-sections.txt:
4003 * gst/rtsp-server/rtsp-media-factory.h:
4004 * gst/rtsp-server/rtsp-media-mapping.c:
4005 * gst/rtsp-server/rtsp-media.h:
4006 * gst/rtsp-server/rtsp-params.c:
4007 * gst/rtsp-server/rtsp-server.c:
4008 * gst/rtsp-server/rtsp-session-pool.h:
4009 * gst/rtsp-server/rtsp-session.c:
4010 * gst/rtsp-server/rtsp-session.h:
4011 * gst/rtsp-server/rtsp-stream-transport.h:
4012 * gst/rtsp-server/rtsp-stream.h:
4015 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4017 * gst/rtsp-server/rtsp-client.c:
4018 * gst/rtsp-server/rtsp-media-mapping.h:
4019 * gst/rtsp-server/rtsp-media.c:
4020 * gst/rtsp-server/rtsp-media.h:
4021 * gst/rtsp-server/rtsp-server.h:
4022 * gst/rtsp-server/rtsp-stream.c:
4023 * gst/rtsp-server/rtsp-stream.h:
4024 rtsp: fix MTU setting
4025 Fix setting of the MTU. There is no need for a vmethod.
4027 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4032 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4035 configure: bump version number after refactoring
4037 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4039 * gst/rtsp-server/Makefile.am:
4040 * gst/rtsp-server/rtsp-client.c:
4041 * gst/rtsp-server/rtsp-client.h:
4042 * gst/rtsp-server/rtsp-media-factory-uri.c:
4043 * gst/rtsp-server/rtsp-media-factory.c:
4044 * gst/rtsp-server/rtsp-media-factory.h:
4045 * gst/rtsp-server/rtsp-media.c:
4046 * gst/rtsp-server/rtsp-media.h:
4047 * gst/rtsp-server/rtsp-sdp.c:
4048 * gst/rtsp-server/rtsp-session-media.c:
4049 * gst/rtsp-server/rtsp-session-media.h:
4050 * gst/rtsp-server/rtsp-session.c:
4051 * gst/rtsp-server/rtsp-session.h:
4052 * gst/rtsp-server/rtsp-stream-transport.c:
4053 * gst/rtsp-server/rtsp-stream-transport.h:
4054 * gst/rtsp-server/rtsp-stream.c:
4055 * gst/rtsp-server/rtsp-stream.h:
4056 rtsp: massive refactoring
4057 Make GObjects from the remaining simple structures.
4058 Remove GstRTSPSessionStream, it's not needed.
4059 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
4060 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
4061 a GstRTSPStream should be transported to a client.
4062 Rename GstRTSPMediaFactory::get_element -> create_element because that
4063 more accurately describes what it does.
4064 Make nice methods instead of poking in the structures.
4065 Move some methods inside the relevant object source code.
4066 Use GPtrArray to store objects instead of plain arrays, it is more
4067 natural and allows us to more easily clean up.
4068 Move the allocation of udp ports to the Stream object. The Stream object
4069 contains the elements needed to stream the media to a client.
4070 Improve the prepare and unprepare methods. Unprepare should now undo
4071 everything prepare did. Improve also async unprepare when doing EOS on
4072 shutdown. Make sure we always unprepare correctly.
4074 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
4076 * gst/rtsp-server/rtsp-client.c:
4077 rtsp-client: Unref server address clients connected to
4078 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
4080 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
4082 * gst/rtsp-server/rtsp-server.c:
4083 rtsp-server: don't ref server socket if it is NULL
4084 Fixes test_bind_already_in_use unit test again after commit 6a497440.
4085 https://bugzilla.gnome.org/show_bug.cgi?id=686644
4087 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
4089 * tests/check/Makefile.am:
4090 tests: Add libgio link dependency
4091 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
4093 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
4095 * gst/rtsp-server/rtsp-media-mapping.c:
4096 * gst/rtsp-server/rtsp-media-mapping.h:
4097 rtsp-media-mapping: rename find_media vfunc to find_factory
4098 The virtual method and class method should have the same name
4099 so it is correctly represented in GIR file
4100 https://bugzilla.gnome.org/show_bug.cgi?id=680777
4102 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
4104 * gst/rtsp-server/rtsp-auth.c:
4105 * gst/rtsp-server/rtsp-client.c:
4106 * gst/rtsp-server/rtsp-media-factory-uri.c:
4107 * gst/rtsp-server/rtsp-media-factory.c:
4108 * gst/rtsp-server/rtsp-media-mapping.c:
4109 * gst/rtsp-server/rtsp-media.c:
4110 * gst/rtsp-server/rtsp-server.c:
4111 * gst/rtsp-server/rtsp-session-pool.c:
4112 * gst/rtsp-server/rtsp-session.c:
4113 rtsp-server: fixed comments and GIR annotations
4114 https://bugzilla.gnome.org/show_bug.cgi?id=680777
4116 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
4118 * gst/rtsp-server/rtsp-media-mapping.c:
4119 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
4121 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
4123 * gst/rtsp-server/rtsp-server.c:
4124 rtsp-server: allow binding on port 0 (binds on a random port)
4126 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
4128 * gst/rtsp-server/rtsp-server.c:
4129 * gst/rtsp-server/rtsp-server.h:
4130 rtsp-server: add bound-port property
4131 bound-port can be used to retrieve the port number when the server is bound on
4132 port 0, which binds on a random port.
4134 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
4136 * gst/rtsp-server/rtsp-media-factory.c:
4137 * gst/rtsp-server/rtsp-media-factory.h:
4138 rtsp-media-factory: make ::get_element overridable by GI bindings
4139 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
4140 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
4141 as the invoker for ::get_element(), making it overridable by GI generated
4144 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
4146 * gst/rtsp-server/rtsp-media-factory-uri.c:
4147 rtsp-media-factory-uri: don't autoplug parsers in a loop
4148 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
4151 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
4153 * gst/rtsp-server/Makefile.am:
4154 Explicitly link against gio. Fix link error on mac.
4156 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
4158 * gst/rtsp-server/rtsp-session.c:
4159 session: add ttl to the transport header in SETUP
4160 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
4162 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
4164 * gst/rtsp-server/rtsp-client.c:
4165 * gst/rtsp-server/rtsp-client.h:
4166 * gst/rtsp-server/rtsp-media.c:
4167 client: Use client transport settings for multicast if allowed.
4168 This patch makes it possible for the client to send transport settings for
4169 multicast (destination && ttl). Client settings must be explicitly allowed or
4170 the server will use its own settings.
4171 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
4173 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
4176 Automatic update of common submodule
4177 From 6c0b52c to 6bb6951
4179 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
4181 * gst/rtsp-server/rtsp-client.c:
4182 rtsp-client: do not destroy the rtsp watch
4183 Don't destroy the client watch while dispatching. The rtsp watch is
4184 automatically destroyed after the rtsp watch function closed() has
4186 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
4188 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
4191 Automatic update of common submodule
4192 From 4f962f7 to 6c0b52c
4194 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
4196 * gst/rtsp-server/rtsp-media.c:
4197 media: fix check for seekability
4199 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4201 * gst/rtsp-server/rtsp-client.c:
4202 client: use more GIO
4203 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
4205 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4207 * gst/rtsp-server/rtsp-server.c:
4208 server: remove obsolete includes
4210 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4212 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
4213 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
4214 be available in "on_new_ssrc". The transports are added in
4215 gst_rtsp_media_set_state when going to PLAYING state. However,
4216 "on_new_ssrc" might be called before this happens.
4217 https://bugzilla.gnome.org/show_bug.cgi?id=683304
4219 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4221 * gst/rtsp-server/rtsp-client.c:
4222 * gst/rtsp-server/rtsp-client.h:
4223 rtsp-client: add signals for rtsp requests (fixes #683287)
4225 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4227 * gst/rtsp-server/rtsp-client.c:
4228 * gst/rtsp-server/rtsp-client.h:
4229 add new-session signal to rtsp-client (fixes #683058)
4231 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
4234 Automatic update of common submodule
4235 From 668acee to 4f962f7
4237 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
4239 * gst/rtsp-server/rtsp-server.c:
4240 * tests/check/gst/rtspserver.c:
4241 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
4242 Do not assume that *error is set in g_socket_address_enumerator_next.
4243 Added test_bind_already_in_use unit-test.
4244 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
4246 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
4249 Automatic update of common submodule
4250 From 94ccf4c to 668acee
4252 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
4254 * gst/rtsp-server/rtsp-client.c:
4255 * gst/rtsp-server/rtsp-client.h:
4256 rtsp-client: make create_sdp virtual method
4257 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
4259 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4262 Automatic update of common submodule
4263 From 98e386f to 94ccf4c
4265 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4267 * gst/rtsp-server/rtsp-client.c:
4270 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
4272 * gst/rtsp-server/rtsp-client.c:
4273 * gst/rtsp-server/rtsp-client.h:
4274 * gst/rtsp-server/rtsp-server.c:
4275 * gst/rtsp-server/rtsp-server.h:
4276 rtsp-server: use an existing socket to establish HTTP tunnel
4277 Make it possible to transfer a socket from an HTTP server to be used as
4278 an RTSP over HTTP tunnel.
4280 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
4282 * gst/rtsp-server/rtsp-client.c:
4283 * gst/rtsp-server/rtsp-media.c:
4284 * gst/rtsp-server/rtsp-media.h:
4285 rtsp: Handle the blocksize parameter
4286 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
4288 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
4290 * tests/check/Makefile.am:
4291 * tests/check/gst/rtspserver.c:
4292 Have unit test get header from source dir, not installed dir
4293 This makes compilation of unit tests work in a build directory other
4294 than the source directory.
4295 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
4297 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
4299 * gst/rtsp-server/rtsp-media.c:
4300 rtsp-media: update for gst_element_make_from_uri() changes
4302 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
4305 * tests/Makefile.am:
4306 * tests/check/Makefile.am:
4307 * tests/check/gst/rtspserver.c:
4309 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
4311 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
4313 * gst/rtsp-server/rtsp-media.c:
4314 rtsp-media: don't collect media stats when going to NULL
4315 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
4317 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4319 * gst/rtsp-server/rtsp-client.c:
4320 client: don't leak transports
4322 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
4324 * gst/rtsp-server/rtsp-client.c:
4325 rtsp-client: free transport on no_stream in SETUP handler
4327 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
4329 * gst/rtsp-server/rtsp-client.c:
4330 rtsp-client: changed session media iteration
4331 In client_unlink_session: now don't iterate in session->medias
4332 list where items are removed by gst_rtsp_session_release_media.
4333 Instead, repeatedly remove the first item.
4335 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
4337 * gst/rtsp-server/rtsp-client.c:
4338 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
4339 GstRTSPSessionMedia is not a GObject type. When the
4340 GstRTSPSession is freed, it will free the media.
4342 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
4344 * gst/rtsp-server/rtsp-media-factory.c:
4345 factory: plug pad leak in collect_streams
4346 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
4347 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
4348 will take one reference, and the other reference will otherwise
4351 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
4354 configure: suppress some warnings when debug is disabled
4355 Warnings about unused variables should be suppressed if core has the
4356 debug system disabled.
4357 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
4359 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4361 * docs/libs/Makefile.am:
4362 docs: fix build in uninstalled setup
4363 Include gst-plugins-base libs properly.
4365 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
4367 * docs/libs/gst-rtsp-server.types:
4368 docs: include headers defining rtsp-server object types
4369 Fixes compiler warnings during docs build.
4370 https://bugzilla.gnome.org/show_bug.cgi?id=676824
4372 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
4375 configure: Add warning flags for compiler when configuring
4376 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
4378 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4381 Automatic update of common submodule
4382 From 03a0e57 to 98e386f
4384 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4387 Automatic update of common submodule
4388 From 1fab359 to 03a0e57
4390 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
4392 * gst/rtsp-server/rtsp-client.c:
4393 client: fix GSocketAddress leak in gst_rtsp_client_accept
4394 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
4396 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4399 Automatic update of common submodule
4400 From f1b5a96 to 1fab359
4402 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4405 Automatic update of common submodule
4406 From 92b7266 to f1b5a96
4408 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4411 Automatic update of common submodule
4412 From ec1c4a8 to 92b7266
4414 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4417 Automatic update of common submodule
4418 From 3429ba6 to ec1c4a8
4420 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
4422 * gst/rtsp-server/rtsp-auth.c:
4423 * gst/rtsp-server/rtsp-client.c:
4424 * gst/rtsp-server/rtsp-media-factory-uri.c:
4425 * gst/rtsp-server/rtsp-server.c:
4426 rtsp: fix compiler warnings
4427 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
4429 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4432 Automatic update of common submodule
4433 From dc70203 to 3429ba6
4435 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4437 * gst/rtsp-server/rtsp-client.c:
4438 * gst/rtsp-server/rtsp-media-factory.c:
4439 * gst/rtsp-server/rtsp-media-factory.h:
4440 * gst/rtsp-server/rtsp-media.c:
4441 * gst/rtsp-server/rtsp-media.h:
4442 * gst/rtsp-server/rtsp-server.c:
4443 * gst/rtsp-server/rtsp-server.h:
4444 * gst/rtsp-server/rtsp-session-pool.c:
4445 * gst/rtsp-server/rtsp-session-pool.h:
4446 rtsp-server: port to new thread API
4448 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4451 Automatic update of common submodule
4452 From 6db25be to dc70203
4454 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4456 * gst/rtsp-server/rtsp-auth.c:
4457 * gst/rtsp-server/rtsp-auth.h:
4458 * gst/rtsp-server/rtsp-client.c:
4459 rtsp-server: Fix compilation and compiler warnings
4461 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4465 * gst/rtsp-server/Makefile.am:
4466 configure: Modernize autotools setup a bit
4467 Also we now only create tar.bz2 and tar.xz tarballs.
4469 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4472 Automatic update of common submodule
4473 From 464fe15 to 6db25be
4475 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4478 Automatic update of common submodule
4479 From 7fda524 to 464fe15
4481 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4484 * docs/libs/Makefile.am:
4485 * docs/version.entities.in:
4487 * gst/rtsp-server/Makefile.am:
4488 * pkgconfig/Makefile.am:
4489 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4490 * pkgconfig/gstreamer-rtsp-server.pc.in:
4491 * tests/Makefile.am:
4492 rtsp-server: Update versioning
4494 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4496 Merge remote-tracking branch 'origin/0.10'
4498 gst/rtsp-server/rtsp-session-pool.c
4500 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4502 * gst/rtsp-server/rtsp-session-pool.c:
4503 rtsp-server: Don't use deprecated GLib API
4505 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4507 Replace master with 0.11
4509 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4511 Merge branch 'master' into 0.11
4513 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4515 Merge branch 'master' into 0.11
4517 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
4520 A couple minor typo fixes
4522 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4524 * gst/rtsp-server/rtsp-media.c:
4525 media: fix state of the appqueue
4527 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4529 * gst/rtsp-server/rtsp-media-factory-uri.c:
4530 factory: use videoconvert
4532 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4534 * gst/rtsp-server/rtsp-media-factory-uri.c:
4535 factory: change to new style caps
4537 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4539 * gst/rtsp-server/rtsp-client.c:
4540 * gst/rtsp-server/rtsp-client.h:
4541 * gst/rtsp-server/rtsp-media-factory-uri.c:
4542 * gst/rtsp-server/rtsp-media.c:
4543 * gst/rtsp-server/rtsp-server.c:
4544 * gst/rtsp-server/rtsp-server.h:
4545 * gst/rtsp-server/rtsp-session-pool.c:
4546 rtsp-server: port to GIO
4549 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4552 configure: fix build
4554 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4557 docs: fix for gst_rtsp_server_set_port() -> _set_service()
4558 https://bugzilla.gnome.org/show_bug.cgi?id=666548
4560 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4563 * examples/Makefile.am:
4564 First rule of gst-rtsp-server club: don't talk about gst-phonon
4566 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4569 * pkgconfig/Makefile.am:
4570 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
4571 * pkgconfig/gst-rtsp-server.pc.in:
4572 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4573 * pkgconfig/gstreamer-rtsp-server.pc.in:
4574 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
4575 For consistency with all other modules.
4577 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4579 * gst/rtsp-server/rtsp-client.c:
4580 rtsp-client: update for new map API
4582 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4585 * bindings/Makefile.am:
4586 * bindings/python/Makefile.am:
4587 * bindings/python/arg-types.py:
4588 * bindings/python/codegen/Makefile.am:
4589 * bindings/python/codegen/__init__.py:
4590 * bindings/python/codegen/argtypes.py:
4591 * bindings/python/codegen/code-coverage.py:
4592 * bindings/python/codegen/codegen.py:
4593 * bindings/python/codegen/definitions.py:
4594 * bindings/python/codegen/defsparser.py:
4595 * bindings/python/codegen/docextract.py:
4596 * bindings/python/codegen/docgen.py:
4597 * bindings/python/codegen/fileprefix.override:
4598 * bindings/python/codegen/fileprefixmodule.c:
4599 * bindings/python/codegen/h2def.py:
4600 * bindings/python/codegen/mergedefs.py:
4601 * bindings/python/codegen/mkskel.py:
4602 * bindings/python/codegen/override.py:
4603 * bindings/python/codegen/reversewrapper.py:
4604 * bindings/python/codegen/scmexpr.py:
4605 * bindings/python/rtspserver-types.defs:
4606 * bindings/python/rtspserver.defs:
4607 * bindings/python/rtspserver.override:
4608 * bindings/python/rtspservermodule.c:
4609 * bindings/python/test.py:
4611 python: remove pygst-based python bindings
4612 pygi is the future, apparently.
4614 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
4617 Automatic update of common submodule
4618 From c463bc0 to 7fda524
4620 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4623 Automatic update of common submodule
4624 From 2a59016 to c463bc0
4626 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4629 Automatic update of common submodule
4630 From 0807187 to 2a59016
4632 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4635 Automatic update of common submodule
4636 From 11f0cd5 to 0807187
4638 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4640 * examples/test-auth.c:
4641 example: update for new caps
4643 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4645 * examples/test-video.c:
4646 * gst/rtsp-server/rtsp-client.c:
4647 * gst/rtsp-server/rtsp-media-factory-uri.c:
4648 * gst/rtsp-server/rtsp-media.c:
4649 * gst/rtsp-server/rtsp-media.h:
4650 * gst/rtsp-server/rtsp-session.c:
4651 * gst/rtsp-server/rtsp-session.h:
4652 rtsp-server: port some more to 0.11
4654 Remove bufferlist stuff
4656 Add queue before appsink now that preroll-queue-len is gone.
4657 Update for request pad changes.
4659 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4661 Merge branch 'master' into 0.11
4663 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
4665 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
4666 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
4667 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
4669 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
4671 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
4672 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
4673 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
4675 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4677 Merge branch 'master' into 0.11
4679 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4681 * gst/rtsp-server/rtsp-media.c:
4682 * gst/rtsp-server/rtsp-media.h:
4683 media: add a seekable boolean
4684 Maintain the seekable state with a new variable instead of reusing the
4687 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
4689 * gst/rtsp-server/rtsp-media.c:
4690 Disallow seek in live media
4692 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4694 Merge branch 'master' into 0.11
4696 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
4698 * gst/rtsp-server/rtsp-server.c:
4699 #ifdef statements for windows socket creation were missing
4701 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
4704 Automatic update of common submodule
4705 From a39eb83 to 11f0cd5
4707 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
4710 Automatic update of common submodule
4711 From 605cd9a to a39eb83
4713 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4715 Merge branch 'master' into 0.11
4717 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4719 * gst/rtsp-server/rtsp-client.c:
4720 client: use method to access property
4722 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4724 * gst/rtsp-server/rtsp-media-factory.c:
4725 * gst/rtsp-server/rtsp-media-factory.h:
4726 media-factory: add protocols property
4727 Add a property to configure the allowed protocols in the media created from the
4730 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4732 * gst/rtsp-server/rtsp-media-factory.c:
4733 * gst/rtsp-server/rtsp-media-factory.h:
4734 media-factory: add media-configure signal
4735 Add signal to allow the application to configure the media after it was created
4738 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4740 * gst/rtsp-server/rtsp-client.c:
4741 client: use method to access property
4743 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4745 * gst/rtsp-server/rtsp-media-factory.c:
4746 * gst/rtsp-server/rtsp-media-factory.h:
4747 media-factory: add protocols property
4748 Add a property to configure the allowed protocols in the media created from the
4751 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4753 * gst/rtsp-server/rtsp-media-factory.c:
4754 * gst/rtsp-server/rtsp-media-factory.h:
4755 media-factory: add media-configure signal
4756 Add signal to allow the application to configure the media after it was created
4759 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4761 Merge branch 'master' into 0.11
4763 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4765 * gst/rtsp-server/rtsp-client.c:
4766 client: use media multicast group
4768 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4770 * gst/rtsp-server/rtsp-media-factory.h:
4771 * gst/rtsp-server/rtsp-server.h:
4772 * gst/rtsp-server/rtsp-session-pool.h:
4773 * gst/rtsp-server/rtsp-session.h:
4776 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
4778 * gst/rtsp-server/rtsp-client.c:
4779 * gst/rtsp-server/rtsp-sdp.h:
4780 sdp: copy and free the server ip address
4781 Copy and free the server ip address to make memory management easier later.
4783 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4785 * gst/rtsp-server/rtsp-media-factory.c:
4786 media-factory: configure multicast in media
4788 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4790 * gst/rtsp-server/rtsp-media.c:
4791 * gst/rtsp-server/rtsp-media.h:
4792 media: add property for multicast group
4793 Add a property to configure the multicast group in the media.
4794 Based on patches from Marc Leeman and Robert Krakora.
4796 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4798 * gst/rtsp-server/rtsp-media-factory.c:
4799 * gst/rtsp-server/rtsp-media-factory.h:
4800 media-factory: add property for multicast group
4801 Add a property to configure the multicast group in the media factory.
4802 Based on patches from Marc Leeman and Robert Krakora.
4804 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4806 * gst/rtsp-server/rtsp-client.c:
4807 client: do configuration of transport in one place
4808 Move the configuration of the transport destination address to where we also
4809 configure the other bits.
4811 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4813 * gst/rtsp-server/rtsp-client.c:
4814 client: use media multicast group
4816 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4818 * gst/rtsp-server/rtsp-media-factory.h:
4819 * gst/rtsp-server/rtsp-server.h:
4820 * gst/rtsp-server/rtsp-session-pool.h:
4821 * gst/rtsp-server/rtsp-session.h:
4824 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
4826 * gst/rtsp-server/rtsp-client.c:
4827 * gst/rtsp-server/rtsp-sdp.h:
4828 sdp: copy and free the server ip address
4829 Copy and free the server ip address to make memory management easier later.
4831 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4833 * gst/rtsp-server/rtsp-media-factory.c:
4834 media-factory: configure multicast in media
4836 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4838 * gst/rtsp-server/rtsp-media.c:
4839 * gst/rtsp-server/rtsp-media.h:
4840 media: add property for multicast group
4841 Add a property to configure the multicast group in the media.
4842 Based on patches from Marc Leeman and Robert Krakora.
4844 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4846 * gst/rtsp-server/rtsp-media-factory.c:
4847 * gst/rtsp-server/rtsp-media-factory.h:
4848 media-factory: add property for multicast group
4849 Add a property to configure the multicast group in the media factory.
4850 Based on patches from Marc Leeman and Robert Krakora.
4852 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4854 * gst/rtsp-server/rtsp-client.c:
4855 client: do configuration of transport in one place
4856 Move the configuration of the transport destination address to where we also
4857 configure the other bits.
4859 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4861 Merge branch 'master' into 0.11
4863 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
4865 * gst/rtsp-server/rtsp-client.c:
4866 client: destroy pipeline on client disconnect with no prior TEARDOWN.
4867 The problem occurs when the client abruptly closes the connection without
4868 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
4869 server is where the pipeline gets torn down. Since this handler is not called,
4870 the pipeline remains and is up and running. Subsequent clients get their own
4871 pipelines and if the do not issue TEARDOWNs then those pipelines will also
4872 remain up and running. This is a resource leak.
4874 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4876 Merge branch 'master' into 0.11
4878 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
4880 * gst/rtsp-server/rtsp-media-factory.c:
4881 * gst/rtsp-server/rtsp-media-factory.h:
4882 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
4883 For example, it can be used to retrieve source elements like appsrc, in a more
4884 convenient way than subclassing get_element.
4886 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4888 Merge branch 'master' into 0.11
4890 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
4892 * gst/rtsp-server/rtsp-server.c:
4893 rtsp-server: hold on to reference while using object
4895 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4897 * gst/rtsp-server/rtsp-media.c:
4900 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4903 configure: use unstable api
4905 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
4907 * gst/rtsp-server/rtsp-client.c:
4908 client: fix reference counting
4910 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
4912 * gst/rtsp-server/rtsp-client.c:
4913 * gst/rtsp-server/rtsp-media.c:
4914 fix compiler warnings about unused variables
4916 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
4918 * examples/test-launch.c:
4919 * examples/test-readme.c:
4920 * examples/test-uri.c:
4921 * examples/test-video.c:
4922 examples: tell rtsp uri when ready
4924 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
4927 Automatic update of common submodule
4928 From 69b981f to 605cd9a
4930 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4932 * gst/rtsp-server/rtsp-client.c:
4933 client: update for buffer API change
4935 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4937 * gst/rtsp-server/Makefile.am:
4938 Makefile.am: 0.10 => @GST_MAJORMINOR@
4940 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4942 * gst/rtsp-server/rtsp-media-factory-uri.c:
4943 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
4945 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4947 * gst/rtsp-server/.gitignore:
4948 .gitignore: 0.10 => 0.11
4950 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4952 * gst/rtsp-server/Makefile.am:
4953 Makefile.am: 0.10 => @GST_MAJORMINOR@
4955 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4957 Merge branch 'master' into 0.11
4959 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
4962 Automatic update of common submodule
4963 From 9e5bbd5 to 69b981f
4965 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
4968 Automatic update of common submodule
4969 From fd35073 to 9e5bbd5
4971 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
4974 Automatic update of common submodule
4975 From 46dfcea to fd35073
4977 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4979 * gst/rtsp-server/rtsp-media-factory-uri.c:
4980 * gst/rtsp-server/rtsp-media.c:
4981 media: port to new caps API
4983 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4985 Merge branch 'master' into 0.11
4987 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
4989 * bindings/vala/gst-rtsp-server-0.10.vapi:
4990 Updated Vala bindings.
4991 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
4993 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
4995 * gst/rtsp-server/rtsp-server.c:
4996 * gst/rtsp-server/rtsp-server.h:
4997 Add a signal for newly connected clients.
4998 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
5000 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
5002 * bindings/python/rtspserver.override:
5003 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
5005 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5007 * gst/rtsp-server/Makefile.am:
5008 * gst/rtsp-server/rtsp-client.c:
5009 * gst/rtsp-server/rtsp-funnel.c:
5010 * gst/rtsp-server/rtsp-funnel.h:
5011 * gst/rtsp-server/rtsp-media.c:
5012 rtsp-server: port to 0.11
5014 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5019 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5021 Merge branch 'master' into 0.11
5026 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5029 Automatic update of common submodule
5030 From c3cafe1 to 46dfcea
5032 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
5034 * bindings/python/Makefile.am:
5035 * bindings/python/rtspserver.defs:
5036 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
5038 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
5040 * bindings/python/arg-types.py:
5041 python bindings: add GstRTSPUrlParam
5042 Needed to implement MediaFactory virtual proxies
5044 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
5046 * bindings/python/arg-types.py:
5047 python bindings: fix returning GstRTSPUrl types
5049 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
5051 * bindings/python/arg-types.py:
5052 python bindings: add arg type for GstRTSPUrl
5054 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
5056 * bindings/python/rtspserver.defs:
5057 python bindings: fix the definition of MediaFactory.collect_stream
5059 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
5062 Automatic update of common submodule
5063 From 1ccbe09 to c3cafe1
5065 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5068 Automatic update of common submodule
5069 From 193b717 to 1ccbe09
5071 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
5074 Automatic update of common submodule
5075 From b77e2bf to 193b717
5077 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5080 build: Include lcov.mak to allow test coverage report generation
5082 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5085 Automatic update of common submodule
5086 From d8814b6 to b77e2bf
5088 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5091 Automatic update of common submodule
5092 From 6aaa286 to d8814b6
5094 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
5097 Automatic update of common submodule
5098 From 6aec6b9 to 6aaa286
5100 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
5103 autogen: wingo signed comment
5105 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
5107 * gst/rtsp-server/rtsp-session-pool.c:
5108 session: use full charset for RTSP session ID
5109 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
5110 session ID more difficult.
5111 https://bugzilla.gnome.org/show_bug.cgi?id=643812
5113 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5115 * gst/rtsp-server/Makefile.am:
5116 rtsp-server: Don't install the funnel header
5118 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
5121 Automatic update of common submodule
5122 From 1de7f6a to 6aec6b9
5124 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5127 configure: require core/base 0.10.31
5128 Needed at least for gst_plugin_feature_rank_compare_func().
5130 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
5133 Automatic update of common submodule
5134 From f94d739 to 1de7f6a
5136 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5138 * gst/rtsp-server/rtsp-media.c:
5139 media: remove more unused code
5141 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5143 * gst/rtsp-server/rtsp-media.c:
5144 * gst/rtsp-server/rtsp-media.h:
5145 media: remove duplicate filtering
5146 Remove the duplicate filtering code now that we have a released -good version.
5147 Give a warning instead.
5149 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5151 * gst/rtsp-server/rtsp-media-factory.c:
5152 * gst/rtsp-server/rtsp-media.c:
5153 media: fix default buffer size
5155 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5157 * gst/rtsp-server/rtsp-media-factory.c:
5158 * gst/rtsp-server/rtsp-media-factory.h:
5159 media-factory: add property to configure the buffer-size
5160 Add a property to configure the kernel UDP buffer size.
5162 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5164 * gst/rtsp-server/rtsp-media.c:
5165 * gst/rtsp-server/rtsp-media.h:
5166 media: add property to configure kernel buffer sizes
5167 Add a property to configure the kernel UDP buffer size.
5169 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5172 configure: set PYGOBJECT_REQ before using it
5173 https://bugzilla.gnome.org/show_bug.cgi?id=640641
5175 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5178 docs: recursive into sub-directories on 'make upload'
5180 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5182 * docs/libs/gst-rtsp-server-docs.sgml:
5183 * docs/version.entities.in:
5184 docs: mention full version these docs are for, not just major-minor
5186 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5191 === release 0.10.8 ===
5193 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5198 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5200 * gst/rtsp-server/rtsp-server.c:
5201 rtsp-server: clarify docs a little
5203 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5205 * gst/rtsp-server/rtsp-media.c:
5206 media: init debug category before starting thread
5208 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5210 * gst/rtsp-server/rtsp-auth.c:
5211 auth: add realm to make it more spec compliant
5213 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5215 * gst/rtsp-server/rtsp-server.c:
5216 * gst/rtsp-server/rtsp-server.h:
5219 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5221 * examples/test-video.c:
5222 example: improve example docs a little
5224 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5226 * gst/rtsp-server/rtsp-server.c:
5227 server: ensure the watch has a ref to the server
5229 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5231 * gst/rtsp-server/rtsp-server.c:
5232 server: simpify channel function
5234 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5236 * gst/rtsp-server/rtsp-server.c:
5237 * gst/rtsp-server/rtsp-server.h:
5238 server: simplify management of channel and source
5239 We don't need to keep around the channel and source objects. Let the mainloop
5240 and the source manage the source and channel respectively.
5242 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5248 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5251 * tests/Makefile.am:
5252 * tests/test-cleanup.c:
5253 tests: add tests directory and cleanup test
5255 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5257 * gst/rtsp-server/rtsp-media-factory-uri.c:
5258 * gst/rtsp-server/rtsp-media-factory.c:
5259 * gst/rtsp-server/rtsp-media-mapping.c:
5260 * gst/rtsp-server/rtsp-media.c:
5261 * gst/rtsp-server/rtsp-session-pool.c:
5262 * gst/rtsp-server/rtsp-session.c:
5263 server: improve debugging in various objects
5265 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5267 * gst/rtsp-server/rtsp-server.c:
5268 server: chain up to the parent finalize
5270 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
5272 * bindings/python/rtspserver-types.defs:
5273 * bindings/python/rtspserver.defs:
5274 * bindings/python/rtspserver.override:
5275 * bindings/python/test.py:
5276 gst-rtsp-server: update python bindings
5278 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5280 * gst/rtsp-server/rtsp-client.c:
5281 client: use the response from the clientstate
5282 Create the response object only once and store in the client state.
5283 Make all methods use the state response,
5285 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5287 * gst/rtsp-server/rtsp-server.c:
5288 server: use signal to keep track of clients
5289 Keep track of all the clients that the server creates and remove them when they
5290 fire the 'closed' signal.
5292 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5294 * gst/rtsp-server/rtsp-client.c:
5295 * gst/rtsp-server/rtsp-client.h:
5296 client: emit signal when closing
5298 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5300 * examples/.gitignore:
5301 * examples/Makefile.am:
5302 * examples/test-auth.c:
5303 * examples/test-video.c:
5304 * gst/rtsp-server/rtsp-auth.c:
5305 * gst/rtsp-server/rtsp-auth.h:
5306 * gst/rtsp-server/rtsp-client.c:
5307 * gst/rtsp-server/rtsp-media-factory.c:
5308 * gst/rtsp-server/rtsp-media.c:
5309 * gst/rtsp-server/rtsp-media.h:
5310 * gst/rtsp-server/rtsp-session-pool.h:
5311 * gst/rtsp-server/rtsp-session.h:
5312 media: enable per factory authorisations
5313 Allow for adding a GstRTSPAuth on the factory and media level and check
5314 permissions when accessing the factory.
5315 Add hints to the auth methods for future more fine grained authorisation.
5316 Add example application for per factory authentication.
5318 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5320 * gst/rtsp-server/rtsp-auth.c:
5321 * gst/rtsp-server/rtsp-auth.h:
5322 * gst/rtsp-server/rtsp-client.c:
5323 * gst/rtsp-server/rtsp-client.h:
5324 * gst/rtsp-server/rtsp-params.c:
5325 * gst/rtsp-server/rtsp-params.h:
5326 rtsp-server: Pass ClientState structure arround
5327 Pass the collected information for the ongoing request in a GstRTSPClientState
5328 structure that we can then pass around to simplify the method arguments. This
5329 will also be handy when we implement logging functionality.
5331 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5333 * gst/rtsp-server/rtsp-media-factory.c:
5334 * gst/rtsp-server/rtsp-media-factory.h:
5335 media-factory: add methods to configure authorisation
5337 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5339 * gst/rtsp-server/rtsp-client.c:
5340 client: unref auth in finalize
5342 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5344 * gst/rtsp-server/rtsp-server.c:
5345 server: unref auth in finalize
5347 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5349 * docs/libs/gst-rtsp-server-docs.sgml:
5350 * docs/libs/gst-rtsp-server-sections.txt:
5351 * docs/libs/gst-rtsp-server.types:
5354 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5356 * gst/rtsp-server/rtsp-server.c:
5357 * gst/rtsp-server/rtsp-server.h:
5358 server: separate create and accept
5359 Create separate create and accept methods so that subclasses can create custom
5361 Configure the server in the client object and prepare for keeping track of
5364 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5366 * gst/rtsp-server/rtsp-client.c:
5367 * gst/rtsp-server/rtsp-client.h:
5368 client: add support for setting the server.
5369 Add support for keeping a ref to the server that started this client
5372 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5374 * gst/rtsp-server/rtsp-auth.c:
5375 auth: fix memleak and add some docs
5376 Fix a memleak of the basic auth token.
5377 Add docs for the helper function
5379 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5381 * gst/rtsp-server/rtsp-auth.c:
5382 * gst/rtsp-server/rtsp-auth.h:
5383 * gst/rtsp-server/rtsp-client.c:
5384 client: delegate setup of auth to the manager
5385 Delegate the configuration of the authentication tokens to the manager object
5388 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5390 * examples/test-video.c:
5391 * gst/rtsp-server/Makefile.am:
5392 * gst/rtsp-server/rtsp-auth.c:
5393 * gst/rtsp-server/rtsp-auth.h:
5394 * gst/rtsp-server/rtsp-client.c:
5395 * gst/rtsp-server/rtsp-client.h:
5396 * gst/rtsp-server/rtsp-server.c:
5397 * gst/rtsp-server/rtsp-server.h:
5398 auth: add authentication object
5399 Add an object that can check the authorization of requests.
5400 Implement basic authentication.
5401 Add example authentication to test-video
5403 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5405 * gst/rtsp-server/rtsp-server.c:
5406 * gst/rtsp-server/rtsp-server.h:
5407 server: move includes back
5408 the includes are needed for sockaddr_in.
5410 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5412 * gst/rtsp-server/rtsp-client.c:
5413 * gst/rtsp-server/rtsp-client.h:
5414 * gst/rtsp-server/rtsp-server.c:
5415 * gst/rtsp-server/rtsp-server.h:
5416 rtsp: move network includes where they are needed
5418 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
5420 * gst/rtsp-server/rtsp-media.h:
5421 rtsp-media.h: Minor corrections in comments.
5424 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
5427 Automatic update of common submodule
5428 From e572c87 to f94d739
5430 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5434 * docs/libs/.gitignore:
5435 * examples/.gitignore:
5436 * gst/rtsp-server/.gitignore:
5439 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5441 * docs/libs/Makefile.am:
5442 docs: We don't build ps/pdf for API reference docs
5444 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5447 Automatic update of common submodule
5448 From ccbaa85 to e572c87
5450 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5453 Automatic update of common submodule
5454 From 46445ad to ccbaa85
5456 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5458 * gst/rtsp-server/Makefile.am:
5459 * gst/rtsp-server/fs-funnel.c:
5460 * gst/rtsp-server/fs-funnel.h:
5461 * gst/rtsp-server/rtsp-funnel.c:
5462 * gst/rtsp-server/rtsp-funnel.h:
5463 * gst/rtsp-server/rtsp-media.c:
5464 funnel: rename fsfunnel to rtspfunnel
5465 Rename the funnel to avoid conflicts with the farsight one.
5467 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5469 * gst/rtsp-server/Makefile.am:
5470 * gst/rtsp-server/fs-funnel.c:
5471 * gst/rtsp-server/fs-funnel.h:
5472 * gst/rtsp-server/rtsp-media.c:
5473 rtsp-media: add and use fsfunnel
5474 Add a copy of fsfunnel to the build because input-selector removed the (broken)
5475 select-all property that we need.
5477 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5479 * gst/rtsp-server/Makefile.am:
5480 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
5481 Use PKG_CONFIG_PATH specified at configure time (if any) as well
5482 for the g-ir-compiler, rather than just assuming the env var has
5485 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5492 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
5494 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5497 * gst/rtsp-server/Makefile.am:
5498 gobject-introspection: fix g-i build for uninstalled setup
5499 Requires gst-plugins-base git (> 0.10.31.2).
5501 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5503 * examples/test-uri.c:
5504 examples: add some more options and comments
5506 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5508 * gst/rtsp-server/rtsp-media-factory-uri.c:
5509 factory-uri: use right property type
5511 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5513 * gst/rtsp-server/rtsp-media-factory-uri.c:
5514 factory-uri: attempt to configure buffer-lists
5515 Attempt to configure buffer lists in the payloader for improved performance.
5517 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5519 * gst/rtsp-server/rtsp-media.c:
5520 media: attempt to configure bigger UDP buffers
5521 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
5522 send buffers with high bitrate streams.
5524 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
5526 * gst/rtsp-server/rtsp-client.c:
5527 client: use the socket length from getsockname
5528 Use the length returned by getsockname to perform the getnameinfo call because
5529 the size can depend on the socket type and platform.
5532 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5534 * docs/libs/gst-rtsp-server-docs.sgml:
5535 * docs/libs/gst-rtsp-server-sections.txt:
5536 docs: add uri factory to the docs
5538 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5540 * gst/rtsp-server/rtsp-client.c:
5541 * gst/rtsp-server/rtsp-media.h:
5544 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5546 * gst/rtsp-server/rtsp-client.c:
5547 * gst/rtsp-server/rtsp-media.c:
5548 * gst/rtsp-server/rtsp-media.h:
5549 * gst/rtsp-server/rtsp-session.c:
5550 * gst/rtsp-server/rtsp-session.h:
5551 rtsp-server: add support for buffer lists
5552 Add support for sending bufferlists received from appsink.
5555 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5557 * gst/rtsp-server/rtsp-client.c:
5558 * gst/rtsp-server/rtsp-media.c:
5559 * gst/rtsp-server/rtsp-media.h:
5560 * gst/rtsp-server/rtsp-sdp.c:
5561 media: make method to retrieve the play range
5562 Make a method to retrieve the playback range so that we can conditionally create
5563 a different range for the SDP and the PLAY requests.
5565 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5567 * gst/rtsp-server/rtsp-media.c:
5568 * gst/rtsp-server/rtsp-media.h:
5569 media: add signal to notify of state changes
5571 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5573 * gst/rtsp-server/rtsp-client.h:
5574 client: cleanup headers
5576 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5578 * gst/rtsp-server/rtsp-client.c:
5581 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5583 * gst/rtsp-server/rtsp-media-factory-uri.c:
5584 * gst/rtsp-server/rtsp-media-factory-uri.h:
5585 factory-uri: add support for gstpay
5586 Add an option to prefer gstpay over decoder + raw payloader.
5588 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5590 * gst/rtsp-server/rtsp-media-factory-uri.c:
5591 * gst/rtsp-server/rtsp-media-factory-uri.h:
5592 factory-uri: rework the autoplugger.
5593 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
5596 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5598 * gst/rtsp-server/rtsp-media-factory-uri.c:
5599 factory-uri: use better factory filter
5600 Make better payloader filter based on autoplug rank and RTP use case.
5602 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5605 Automatic update of common submodule
5606 From 169462a to 46445ad
5608 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5610 * gst/rtsp-server/rtsp-server.c:
5611 server: set SO_REUSEADDR before bind
5612 Set the SO_REUSEADDR _before_ bind() to make it actually work.
5614 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5616 * gst/rtsp-server/rtsp-media.c:
5617 * gst/rtsp-server/rtsp-media.h:
5618 media: emit prepared signal when prepared
5619 Make a 'prepared' signal and emit it when we successfully prepared the element.
5620 This signal can be used to configure the media object after it has been prepared
5623 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
5626 Automatic update of common submodule
5627 From 011bcc8 to 169462a
5629 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
5631 python an optional dependency
5632 * configure.ac: Move up valgrind and g-i checks. Make the python
5633 dependency optional, as it was before.
5635 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5637 Merge branch 'master' into 0.11
5642 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5644 * gst/rtsp-server/rtsp-media.c:
5645 media: update range when active clients changed
5646 When we changed the number of active clients, update the current range
5647 information because we want the second client connecting to a shared resource
5648 continue from where the stream currently.
5650 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5652 * gst/rtsp-server/rtsp-media-factory-uri.c:
5653 * gst/rtsp-server/rtsp-media-factory-uri.h:
5654 factory-uri: add colorspace and fix pt
5655 Rework the way we pass data to the autoplugger.
5656 When we have raw caps, plug a converter element to make pluggin to raw
5657 payloaders more successful.
5658 Make sure all dynamically plugged payloaders have a unique payload types.
5660 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5662 * examples/Makefile.am:
5663 * examples/test-uri.c:
5664 example: add example of the uri factory
5666 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5668 * gst/rtsp-server/Makefile.am:
5669 * gst/rtsp-server/rtsp-media-factory-uri.c:
5670 * gst/rtsp-server/rtsp-media-factory-uri.h:
5671 * gst/rtsp-server/rtsp-server.h:
5672 factory-uri: add a factory to stream any URI
5673 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
5676 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5678 * gst/rtsp-server/rtsp-media.c:
5679 * gst/rtsp-server/rtsp-media.h:
5680 media: ignore spurious ASYNC_DONE messages
5681 When we are dynamically adding pads, the addition of the udpsrc elements will
5682 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
5683 the real ASYNC_DONE when everything is prerolled.
5685 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5687 * gst/rtsp-server/rtsp-media-factory.c:
5688 * gst/rtsp-server/rtsp-media-factory.h:
5689 media-factory: make lock macro
5691 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
5693 * gst/rtsp-server/rtsp-client.c:
5694 rtsp-server: Remove unused variable and dead assignment
5696 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
5698 * examples/test-launch.c:
5699 * examples/test-mp4.c:
5700 * examples/test-ogg.c:
5701 * examples/test-readme.c:
5702 * examples/test-sdp.c:
5703 * examples/test-video.c:
5704 examples: Run gst-indent
5706 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
5708 * gst/rtsp-server/rtsp-client.c:
5709 * gst/rtsp-server/rtsp-media-factory.c:
5710 * gst/rtsp-server/rtsp-media-mapping.c:
5711 * gst/rtsp-server/rtsp-media.c:
5712 * gst/rtsp-server/rtsp-params.c:
5713 * gst/rtsp-server/rtsp-sdp.c:
5714 * gst/rtsp-server/rtsp-server.c:
5715 * gst/rtsp-server/rtsp-session-pool.c:
5716 * gst/rtsp-server/rtsp-session.c:
5717 rtsp-server: Run gst-indent
5718 Since it wasn't using the upstream common previously, there was no
5719 indentation check before commiting.
5721 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
5723 * gst/rtsp-server/rtsp-media-mapping.h:
5724 * gst/rtsp-server/rtsp-media.c:
5725 * gst/rtsp-server/rtsp-media.h:
5726 * gst/rtsp-server/rtsp-sdp.c:
5727 * gst/rtsp-server/rtsp-session-pool.h:
5728 * gst/rtsp-server/rtsp-session.c:
5729 * gst/rtsp-server/rtsp-session.h:
5730 rtsp-server: Some more doc fixups
5732 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5735 Makefile: Add cruft-cleaning support
5737 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5742 * docs/libs/Makefile.am:
5743 * docs/libs/gst-rtsp-server-docs.sgml:
5744 * docs/libs/gst-rtsp-server-sections.txt:
5745 * docs/libs/gst-rtsp-server.types:
5746 * docs/version.entities.in:
5747 docs: Add gtk-doc build system
5749 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5751 * gst/rtsp-server/Makefile.am:
5752 Makefile.am: Use standard GIR make behaviour
5754 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5758 autogen/configure: Bring more in sync to standard gst module behaviour
5760 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5762 * gst/rtsp-server/rtsp-media.c:
5763 media: warn and fail when gstrtpbin is not found
5765 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5768 configure: open 0.11 branch
5770 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
5774 Add common submodule
5776 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
5779 * common/Makefile.am:
5780 * common/c-to-xml.py:
5782 * common/coverage/coverage-report-entry.pl:
5783 * common/coverage/coverage-report.pl:
5784 * common/coverage/coverage-report.xsl:
5785 * common/coverage/lcov.mak:
5786 * common/gettext.patch:
5787 * common/glib-gen.mak:
5788 * common/gst-autogen.sh:
5789 * common/gst-xmlinspect.py:
5791 * common/gstdoc-scangobj:
5792 * common/gtk-doc-plugins.mak:
5793 * common/gtk-doc.mak:
5794 * common/m4/.gitignore:
5795 * common/m4/Makefile.am:
5797 * common/m4/as-ac-expand.m4:
5798 * common/m4/as-auto-alt.m4:
5799 * common/m4/as-compiler-flag.m4:
5800 * common/m4/as-compiler.m4:
5801 * common/m4/as-docbook.m4:
5802 * common/m4/as-libtool-tags.m4:
5803 * common/m4/as-libtool.m4:
5804 * common/m4/as-python.m4:
5805 * common/m4/as-scrub-include.m4:
5806 * common/m4/as-version.m4:
5807 * common/m4/ax_create_stdint_h.m4:
5808 * common/m4/check.m4:
5809 * common/m4/glib-gettext.m4:
5810 * common/m4/gst-arch.m4:
5811 * common/m4/gst-args.m4:
5812 * common/m4/gst-check.m4:
5813 * common/m4/gst-debuginfo.m4:
5814 * common/m4/gst-default.m4:
5815 * common/m4/gst-doc.m4:
5816 * common/m4/gst-error.m4:
5817 * common/m4/gst-feature.m4:
5818 * common/m4/gst-function.m4:
5819 * common/m4/gst-gettext.m4:
5820 * common/m4/gst-glib2.m4:
5821 * common/m4/gst-libxml2.m4:
5822 * common/m4/gst-plugindir.m4:
5823 * common/m4/gst-valgrind.m4:
5824 * common/m4/gtk-doc.m4:
5825 * common/m4/introspection.m4:
5827 * common/mangle-tmpl.py:
5828 * common/plugins.xsl:
5830 * common/release.mak:
5831 * common/scangobj-merge.py:
5832 * common/upload.mak:
5833 common: Remove static version
5835 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
5837 * common/m4/introspection.m4:
5838 Update introspection.m4 to match usage
5840 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5844 Remove old stuff from the README
5846 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5851 === release 0.10.7 ===
5853 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5858 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5860 * examples/test-ogg.c:
5861 test-ogg: remove parsers
5862 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
5863 buffers with timestamps. Using the parsers also seems to break things.
5865 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
5867 * bindings/vala/gst-rtsp-server-0.10.vapi:
5868 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
5869 Updated Vala bindings
5871 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
5873 * common/m4/introspection.m4:
5875 * gst/rtsp-server/Makefile.am:
5876 Added initial gobject-introspection support
5878 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5880 * gst/rtsp-server/rtsp-media-factory.c:
5881 media-factory: don't use host for shared hash key
5882 When we generate the key to share made between connections, don't include the
5883 host used to connect so that we can share media even if between clients that
5884 connected with localhost and ones with the ip address.
5886 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5888 * bindings/vala/Makefile.am:
5889 build: fix distcheck
5891 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5893 * bindings/vala/gst-rtsp-server-0.10.vapi:
5894 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
5895 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
5896 Update Vala bindings
5898 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5900 * bindings/vala/Makefile.am:
5902 Fix configure checks and installation location for Vala bindings
5905 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5910 === release 0.10.6 ===
5912 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5915 configure: release 0.10.6
5917 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5919 * gst/rtsp-server/rtsp-media.c:
5920 media: help the compiler a little
5922 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5924 * gst/rtsp-server/rtsp-media.c:
5925 * gst/rtsp-server/rtsp-media.h:
5926 * gst/rtsp-server/rtsp-session.c:
5927 media: cleanup media transport before freeing
5928 Cleanup the media transport data before freeing. In particular, remove the qdata
5929 from the rtpsource object.
5931 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5933 * gst/rtsp-server/rtsp-media-factory.c:
5934 * gst/rtsp-server/rtsp-media-factory.h:
5935 * gst/rtsp-server/rtsp-media.c:
5936 * gst/rtsp-server/rtsp-media.h:
5937 media-factory: add eos-shutdown property
5938 Add an eos-shutdown property that will send an EOS to the pipeline before
5939 shutting it down. This allows for nice cleanup in case of a muxer.
5942 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5944 * gst/rtsp-server/rtsp-media.c:
5945 * gst/rtsp-server/rtsp-media.h:
5946 media: use multiudpsink send-duplicates when we can
5947 If we have a new enough multiudpsink with the send-duplicates property, use this
5948 instead of doing our own filtering. Our custom filtering code should eventually
5949 be removed when we can depend on a released -good.
5951 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5953 * gst/rtsp-server/rtsp-media.c:
5954 media: don't leak destinations
5955 Refactor and cleanup the destinations array when the stream is destroyed.
5957 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5959 * gst/rtsp-server/rtsp-media.c:
5960 * gst/rtsp-server/rtsp-media.h:
5961 media: don't add udp addresses multiple times
5962 Keep track of the udp addresses we added to udpsink and never add the same udp
5963 destination twice. This avoids duplicate packets when using multicast.
5965 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5967 * gst/rtsp-server/rtsp-server.c:
5968 server: disable use of SO_LINGER
5969 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
5970 server close()s the connection.
5972 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5974 * gst/rtsp-server/rtsp-server.c:
5975 server: use 5 second linger period in SO_LINGER
5976 Wait 5 seconds before clearing the send buffers and reseting the connection with
5977 the client when we do a close. This should be enough time to get the message to
5981 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
5983 * gst/rtsp-server/rtsp-server.c:
5984 server: use SO_LINGER
5985 SO_LINGER on the socket will make sure that any pending data on the socket is
5986 flushed ASAP and that the socket connection is reset. This makes sure that the
5987 socket can be reused immediately.
5990 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5993 README: add blurb about shared media factories
5995 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
5997 * gst/rtsp-server/rtsp-media.c:
5998 Add stdlib.h for atoi()
6000 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6002 * bindings/python/Makefile.am:
6003 * bindings/vala/Makefile.am:
6004 build: distcheck fixes
6005 Fix 'make distcheck', somewhat (it still fails because it tries to
6006 install files into /usr/share/vala/vapi/ irrespective of the
6009 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6012 configure: bump core/base requirements to released version
6013 Makes things less confusing for people.
6015 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6018 configure: fail if GStreamer core/base requirements are not met
6020 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6022 * gst/rtsp-server/rtsp-client.c:
6023 client: improve client cleanups
6024 Make sure the session does not timeout when using TCP. We need to do this
6025 because quicktime player does not send RTCP for some reason in tunneled
6027 Refactor some cleanup code.
6030 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6032 * gst/rtsp-server/rtsp-session.c:
6033 * gst/rtsp-server/rtsp-session.h:
6034 session: add support for prevent session timeouts
6035 Add an atomix counter to prevent session timeouts when we are, for example,
6038 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6040 * gst/rtsp-server/rtsp-client.c:
6041 client: fix unlink on session timeouts
6042 When our session times out, make sure we unlink all streams in this
6044 Remove the tunnelid when closing the connection.
6046 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6048 * gst/rtsp-server/rtsp-session.c:
6049 session: small cleanups
6051 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6053 * gst/rtsp-server/rtsp-client.c:
6054 client: handle lost_tunnel callbacks
6055 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
6056 hashtable so that we can reuse it for when the client reopens the POST
6058 Close the connection after a TEARDOWN.
6059 Make sure or watchid is cleared when the watch is removed.
6062 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6064 * gst/rtsp-server/rtsp-client.c:
6065 * gst/rtsp-server/rtsp-media.c:
6066 * gst/rtsp-server/rtsp-sdp.c:
6067 rtsp-server: add more support for multicast
6069 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6072 * gst/rtsp-server/rtsp-media.c:
6073 * gst/rtsp-server/rtsp-media.h:
6074 media: allow configuration of allowed lower transport
6076 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6078 * gst/rtsp-server/rtsp-client.h:
6079 * gst/rtsp-server/rtsp-media.c:
6080 * gst/rtsp-server/rtsp-media.h:
6081 * gst/rtsp-server/rtsp-sdp.c:
6082 * gst/rtsp-server/rtsp-sdp.h:
6083 * gst/rtsp-server/rtsp-server.c:
6084 rtsp: keep track of server ip and ipv6
6085 Keep track of how the client connected to the server and setup the udp ports
6086 with the same protocol.
6087 Copy the server ip address in the SDP so that clients can send RTCP back to
6090 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6092 * gst/rtsp-server/rtsp-session.c:
6095 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6097 * gst/rtsp-server/rtsp-client.c:
6098 client: use right size for malloc
6100 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6102 * gst/rtsp-server/rtsp-server.c:
6103 server: comment ipv6 server listening address
6105 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6107 * gst/rtsp-server/rtsp-media.c:
6108 media: allow for ipv6 sockets
6110 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6112 * gst/rtsp-server/rtsp-server.c:
6113 * gst/rtsp-server/rtsp-server.h:
6114 server: rework server part
6115 Allow setting a bind address, make sure we can deal with ipv6.
6116 Remove the port property and change with the service property.
6118 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6120 * gst/rtsp-server/rtsp-media.h:
6121 media: update comments a little
6123 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6125 * gst/rtsp-server/rtsp-client.c:
6126 client: make content-base better
6127 Use the URI formatting functions to make a content-base. Also make sure that
6128 there is a trailing / at the end.
6130 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6132 * gst/rtsp-server/rtsp-client.c:
6133 client: guard against invalid paths
6135 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6137 * examples/test-video.c:
6138 test: catch server bind errors
6140 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
6142 * gst/rtsp-server/rtsp-media.c:
6143 rtspmedia: emit "unprepared" if _prepare fails.
6144 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
6145 media object is removed from its factory's cache.
6147 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6149 * gst/rtsp-server/rtsp-media.c:
6150 media: collect media position when seek completes
6152 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
6154 * gst/rtsp-server/rtsp-client.c:
6155 client: call unlink_streams in client finalize
6158 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6160 * gst/rtsp-server/rtsp-media.c:
6161 media: limit the time to wait to something huge
6162 Avoid waiting forever but limit the timeout to 20 seconds.
6164 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6166 * gst/rtsp-server/rtsp-sdp.c:
6167 sdp: reindent and check for prepared status
6169 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6171 * gst/rtsp-server/rtsp-media.c:
6172 * gst/rtsp-server/rtsp-media.h:
6173 * gst/rtsp-server/rtsp-session.c:
6174 media: avoid doing _get_state() for state changes
6175 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
6176 until the media is prerolled or in error. This avoids doing a blocking call of
6177 gst_element_get_state() that can cause lockups when there is an error.
6180 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6182 * gst/rtsp-server/rtsp-media.c:
6185 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6187 * gst/rtsp-server/rtsp-media-factory.c:
6188 media-factory: better error handling
6189 Improve the error handling a bit.
6191 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6193 * gst/rtsp-server/rtsp-client.c:
6194 client: rework transport parsing
6195 Rework the transport parsing code so that we can ignore transports we don't
6196 support instead of just picking the first one we can parse.
6197 Configure a (for now hardcoded) destination for multicast transports.
6199 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6201 * gst/rtsp-server/rtsp-media.c:
6202 media: set multicast sink parameters
6203 Disable loop and automatic multicast join on the udpsink elements.
6204 Add some more debug info.
6205 Reset some state variables in the right place.
6206 Use the right port numbers for multicast.
6208 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6210 * gst/rtsp-server/rtsp-session.c:
6211 session: handle transport setup correctly
6212 Handle UDP, MCAST and TCP transport negotiation more correctly.
6213 Store the server session SSRC in the transport.
6215 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6217 * gst/rtsp-server/rtsp-client.c:
6218 rtsp-client: implement error_full
6219 Implement error_full to avoid some segfaults when the rtspconnection calls it.
6222 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6225 * gst/rtsp-server/rtsp-client.c:
6226 * gst/rtsp-server/rtsp-server.c:
6227 docs: update docs and comments
6229 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
6231 * gst/rtsp-server/rtsp-sdp.c:
6232 sdp: make server work better when behind a proxy
6234 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6236 * gst/rtsp-server/rtsp-client.c:
6237 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
6239 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6241 * gst/rtsp-server/rtsp-client.c:
6242 * gst/rtsp-server/rtsp-media-factory.c:
6243 * gst/rtsp-server/rtsp-media-mapping.c:
6244 * gst/rtsp-server/rtsp-media.c:
6245 * gst/rtsp-server/rtsp-server.c:
6246 * gst/rtsp-server/rtsp-session-pool.c:
6247 * gst/rtsp-server/rtsp-session.c:
6248 Use GStreamer's debugging subsystem
6250 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6252 * gst/rtsp-server/rtsp-media-factory.c:
6253 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
6255 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6260 === release 0.10.5 ===
6262 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6267 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6270 configure: bump required versions
6272 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
6274 * gst/rtsp-server/rtsp-client.c:
6275 client: call weak-unref on client->sessions from finalize
6278 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6280 * gst/rtsp-server/rtsp-media.c:
6281 media: Fixed crasher where caps got unref'ed too often
6283 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6286 * pkgconfig/.gitignore:
6287 * pkgconfig/Makefile.am:
6288 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
6289 Added pkg-config file to use gst-rtsp-server uninstalled
6291 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6293 * gst/rtsp-server/rtsp-media.c:
6294 media: add some docs
6296 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
6298 * gst/rtsp-server/rtsp-client.c:
6299 rtsp: Use gst_rtsp_watch_send_message().
6300 Use gst_rtsp_watch_send_message() since the old API which used
6301 gst_rtsp_watch_queue_message() has been deprecated.
6303 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6308 === release 0.10.4 ===
6310 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6315 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6317 * gst/rtsp-server/rtsp-client.c:
6318 * gst/rtsp-server/rtsp-session.c:
6319 * gst/rtsp-server/rtsp-session.h:
6320 rtsp: allocate channels in TCP mode
6321 When the client does not provide us with channels in TCP mode, allocate channels
6324 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6326 * gst/rtsp-server/rtsp-client.c:
6327 client: don't crash when tunnelid is missing
6328 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
6329 don't crash but return an error response to the client.
6332 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6334 * bindings/vala/gst-rtsp-server-0.10.vapi:
6335 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6336 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6337 bindings: update vala bindings with new method
6339 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6341 * gst/rtsp-server/rtsp-session-pool.c:
6342 * gst/rtsp-server/rtsp-session-pool.h:
6343 sessionpool: add function to filter sessions
6344 Add generic function to retrieve/remove sessions.
6346 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6349 configure: bump core/base requirements to release
6351 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6353 * gst/rtsp-server/rtsp-media.c:
6354 media: fix indentation
6356 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6358 * gst/rtsp-server/rtsp-media.c:
6359 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
6361 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6363 * gst/rtsp-server/rtsp-media.c:
6364 set state and remove elements of media in for loop
6366 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
6368 * bindings/vala/gst-rtsp-server-0.10.vapi:
6369 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6370 Added gst_rtsp_media_remove_elements function to Vala bindings
6372 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
6374 * gst/rtsp-server/rtsp-media.c:
6375 * gst/rtsp-server/rtsp-media.h:
6376 Added gst_rtsp_media_remove_elements function
6378 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
6380 * gst/rtsp-server/rtsp-media.c:
6381 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
6383 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6385 * bindings/vala/gst-rtsp-server-0.10.vapi:
6386 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6387 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6388 Updated Vala bindings
6390 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6392 * gst/rtsp-server/rtsp-media.c:
6393 * gst/rtsp-server/rtsp-media.h:
6394 Added vmethod unprepare to GstRTSPMedia
6395 The default implementation sets the state of the pipeline to GST_STATE_NULL
6397 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6399 * gst/rtsp-server/rtsp-media-factory.c:
6400 * gst/rtsp-server/rtsp-media-factory.h:
6401 Made collect_streams function public
6403 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6405 * gst/rtsp-server/rtsp-media-factory.c:
6406 * gst/rtsp-server/rtsp-media-factory.h:
6407 * gst/rtsp-server/rtsp-media.c:
6408 Added vmethod create_pipeline to GstRTSPMediaFactory
6409 The pipeline is created in this method and the GstRTSPMedia's element is added to it
6411 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6413 * gst/rtsp-server/rtsp-client.c:
6414 client: use g_source_destroy()
6415 We need to use g_source_destroy() because we might have added the source to a
6416 different main context than the default one.
6418 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6420 * gst/rtsp-server/Makefile.am:
6421 * gst/rtsp-server/rtsp-client.c:
6422 * gst/rtsp-server/rtsp-params.c:
6423 * gst/rtsp-server/rtsp-params.h:
6424 rtsp: prepare for handling GET/SET_PARAMETER
6425 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
6427 Fix return codes of handlers.
6429 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6431 * gst/rtsp-server/rtsp-media.c:
6432 media: don't leak session pads
6434 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6436 * gst/rtsp-server/rtsp-media.c:
6437 media: clean up the messages a bit
6439 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6441 * gst/rtsp-server/rtsp-sdp.c:
6442 sdp: warn and skip streams without media
6444 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6446 * bindings/vala/gst-rtsp-server-0.10.vapi:
6447 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6448 vala: Fixed typo in header file of RTSPMediaStream
6450 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6452 * gst/rtsp-server/rtsp-media.c:
6455 Make dumping RTCP stats configurable
6457 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6459 * gst/rtsp-server/rtsp-media.c:
6460 media: be less verbose and leak less
6462 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6464 * gst/rtsp-server/rtsp-media.c:
6465 media: don't leak the destination address
6467 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6469 * gst/rtsp-server/rtsp-client.c:
6470 * gst/rtsp-server/rtsp-media.c:
6471 * gst/rtsp-server/rtsp-media.h:
6472 * gst/rtsp-server/rtsp-session.c:
6473 * gst/rtsp-server/rtsp-session.h:
6474 rtsp: use RTCP to keep the session alive
6475 Use the RTCP rtcp-from stats field to find the associated session and use this
6476 to keep the session alive.
6478 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6480 * gst/rtsp-server/rtsp-session.c:
6481 session: add 5sec to the real session timeout
6482 Allow the session to live 5sec longer before really timing out. This should give
6483 clients some extra time to keep the session active.
6485 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6487 * gst/rtsp-server/rtsp-client.c:
6488 client: replay OK to GET/SET_PARAMETER
6489 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
6490 so that we return OK for those requests.
6492 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6494 * gst/rtsp-server/rtsp-media.c:
6495 * gst/rtsp-server/rtsp-media.h:
6496 media: keep track of active transports
6497 Keep track of which transport is active to avoid closing the connection too
6499 Remove the destination transport also when going to NULL.
6500 Print some stats about the SDES and other RTCP messages we receive from the
6503 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6505 * examples/.gitignore:
6506 * examples/Makefile.am:
6507 * examples/test-sdp.c:
6508 example: add SDP relay example
6510 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6512 * gst/rtsp-server/rtsp-media.c:
6513 media: also count active TCP connections
6515 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6517 * gst/rtsp-server/rtsp-media-factory.c:
6518 * gst/rtsp-server/rtsp-media.c:
6519 * gst/rtsp-server/rtsp-media.h:
6520 rtsp: add support for dynamic elements
6521 Add support for dynamic elements.
6522 Don't set live pipelines back to paused.
6524 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6526 * gst/rtsp-server/rtsp-sdp.c:
6527 sdp: don't add encoding name when absent in caps
6529 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6531 * gst/rtsp-server/rtsp-client.c:
6532 client: warn when we can't do RTP-Info
6534 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6536 * gst/rtsp-server/rtsp-media-factory.c:
6537 factory: factor out the stream construction
6539 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6541 * gst/rtsp-server/rtsp-client.c:
6542 client: only add RTP-Info when we have the info
6543 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
6546 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6551 === release 0.10.3 ===
6553 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6557 - Fixes a bug where it put the wrong verion in pkgconfig
6558 - Link RTP and RTCP sources
6560 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6562 * gst/rtsp-server/rtsp-media.c:
6563 * gst/rtsp-server/rtsp-media.h:
6564 media: link the RTP udpsrc to the session manager
6565 Link the RTP udpsrc and the appsrc to the session manager so that they don't
6566 shut down when the client sends a packet to open firewalls.
6568 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6570 * pkgconfig/gst-rtsp-server.pc.in:
6571 Don't use hard-coded version number in pkg-config file
6573 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6578 === release 0.10.2 ===
6580 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6585 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6588 * common/m4/.gitignore:
6589 * examples/.gitignore:
6590 * pkgconfig/.gitignore:
6591 add some .gitignore files
6593 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6595 * gst/rtsp-server/rtsp-media.c:
6596 media: seek to key frames
6598 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6600 * gst/rtsp-server/rtsp-media.c:
6601 media: emit the unprepared signal by id
6602 Emit the unprepared signal by id instead of name and set the media as
6605 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6607 * gst/rtsp-server/rtsp-media.c:
6608 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
6610 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6612 * gst/rtsp-server/rtsp-server.c:
6613 Added finalize function to GstRTPSPServer to unref session pool and media mapping
6615 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6617 * bindings/vala/gst-rtsp-server-0.10.vapi:
6618 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6619 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6620 Updated vala bindings
6622 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6624 * gst/rtsp-server/Makefile.am:
6625 * gst/rtsp-server/rtsp-client.c:
6626 * gst/rtsp-server/rtsp-media.c:
6627 server: use appsink and appsrc with the API
6628 Use the appsink/appsrc API instead of the signals for higher
6631 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6633 * examples/test-ogg.c:
6634 tests: set the payload type correctly
6636 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6638 * gst/rtsp-server/rtsp-media-factory.c:
6639 factory: connect to the unprepare signal
6640 Connect to the unprepare signal for non-reusable media so that we can remove
6641 them from the cache.
6643 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6645 * gst/rtsp-server/rtsp-media.c:
6646 * gst/rtsp-server/rtsp-media.h:
6647 media: add signal to notify of unprepare
6649 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6651 * gst/rtsp-server/rtsp-media.c:
6652 * gst/rtsp-server/rtsp-media.h:
6653 media: more work on making the media shared
6654 Add a reusable flag to medias, indicating that they can be reused after a state
6658 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6660 * examples/test-readme.c:
6661 examples: mark the example as shared for testing
6663 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6665 * gst/rtsp-server/rtsp-media.c:
6666 * gst/rtsp-server/rtsp-media.h:
6667 client: support shared media
6668 Always perform the state actions even if the target state of the pipeline is
6669 already correct, we still want to add/remove the transports when we are dealing
6671 Keep a counter of the number of active transports for a media so that we can use
6672 this to perform a state change when needed.
6673 Perform a state change of the pipeline only when the first transport was added
6674 or when there are no active transports.
6676 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6678 * gst/rtsp-server/rtsp-client.c:
6679 client: fix refcounting crasher
6680 Don't need to remove the weak refs in the finalize methods, they are already
6681 removed in the dispose.
6682 Don't register the callback with a DestroyNofity.
6684 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6686 * gst/rtsp-server/rtsp-client.c:
6687 Fix rtsp client refcount management in TCP mode.
6688 Don't unref a client ref we never had. Fixes an unref
6689 of an already-free client object after a client
6690 teardown request for me.
6692 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6694 * gst/rtsp-server/rtsp-session.c:
6695 docs: fix typo in API docs
6697 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6699 * gst/rtsp-server/rtsp-media.c:
6701 Keep the udp sources in playing even if we go to paused. unlock the sources when
6703 Add some more debug info.
6704 Only seek when we need to.
6705 Keep track of the position when we go to paused.
6707 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6709 * gst/rtsp-server/rtsp-client.c:
6710 * gst/rtsp-server/rtsp-media.c:
6711 * gst/rtsp-server/rtsp-media.h:
6712 Add beginnings of seeking.
6713 Parse the Range header and perform a seek on the pipeline for the requested
6714 position. It's disabled currently until I figure out what's going wrong.
6716 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6718 * gst/rtsp-server/rtsp-client.c:
6719 allow pause requests for now.
6722 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6724 * gst/rtsp-server/rtsp-client.c:
6725 Remove weak ref on the session in teardown
6726 We need to remove our weakref from the session when we do a teardown because
6727 else we close the TCP connection prematurely.
6729 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6731 * gst/rtsp-server/rtsp-client.c:
6732 * gst/rtsp-server/rtsp-client.h:
6733 * gst/rtsp-server/rtsp-session-pool.c:
6734 Do some more session cleanup
6735 Make session timeout kill the TCP connection that currently watches the
6737 Remove the client timeout property.
6739 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6741 * gst/rtsp-server/rtsp-client.c:
6742 * gst/rtsp-server/rtsp-client.h:
6743 * gst/rtsp-server/rtsp-media.c:
6744 * gst/rtsp-server/rtsp-media.h:
6745 * gst/rtsp-server/rtsp-server.c:
6746 * gst/rtsp-server/rtsp-session.c:
6747 * gst/rtsp-server/rtsp-session.h:
6749 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
6752 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6754 * examples/Makefile.am:
6755 * examples/test-launch.c:
6756 Add example server that takes launch lines
6757 Add an example server that streams any -launch line.
6759 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6761 * examples/test-readme.c:
6762 * gst/rtsp-server/rtsp-client.c:
6763 * gst/rtsp-server/rtsp-media.c:
6764 * gst/rtsp-server/rtsp-media.h:
6765 Add support for live streams
6766 Add support for live streams and ranges
6767 Start on handling TCP data transfer.
6769 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6771 * gst/rtsp-server/rtsp-media.c:
6772 Free the pipeline before other things
6775 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6777 * gst/rtsp-server/rtsp-client.c:
6778 Only free the pending tunnel if there is one
6781 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6783 * gst/rtsp-server/rtsp-client.c:
6784 * gst/rtsp-server/rtsp-client.h:
6785 * gst/rtsp-server/rtsp-media.c:
6786 rtsp-server: Add support for tunneling
6787 Add support for tunneling over HTTP.
6788 Use new connection methods to retrieve the url.
6789 Dispatch messages based on the message type instead of blindly
6790 assuming it's always a request.
6791 Keep track of the watch id so that we can remove it later.
6792 Set the media pipeline to NULL before unreffing the pipeline.
6794 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6796 * gst/rtsp-server/rtsp-client.c:
6797 * gst/rtsp-server/rtsp-client.h:
6798 Fix for channel -> watch rename in gstreamer
6799 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
6801 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6803 * gst/rtsp-server/rtsp-client.c:
6804 * gst/rtsp-server/rtsp-client.h:
6806 Use the async RTSP channels instead of spawning a new thread for each client.
6807 If a sessionid is specified in a request, fail if we don't have the session.
6809 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6811 * gst/rtsp-server/rtsp-media.c:
6812 Add better debug info
6813 Add some better debug info.
6815 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6817 * examples/test-video.c:
6819 Add support for session timeouts in the example.
6821 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6823 * gst/rtsp-server/rtsp-session-pool.c:
6824 * gst/rtsp-server/rtsp-session-pool.h:
6825 Pass GTimeVal around for performance reasons
6826 Get the current time only once and pass it around so that sessions don't have to
6827 get the current time anymore.
6828 Add experimental support for a GSource that dispatches when the session needs to
6831 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6833 * gst/rtsp-server/rtsp-session.c:
6834 * gst/rtsp-server/rtsp-session.h:
6835 Add better support for session timeouts
6836 Add a method to request the number of milliseconds when a session will timeout.
6838 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6840 * gst/rtsp-server/rtsp-media.c:
6841 * gst/rtsp-server/rtsp-media.h:
6842 Add suport for RTP manager monitoring
6843 Add the first stage in monitoring the rtp manager.
6844 Make sure we don't update the state to something we don't want.
6846 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6848 * gst/rtsp-server/rtsp-client.c:
6849 Add support for session keepalive
6850 Get and update the session timeout for all requests. get the session as early as
6853 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6855 * gst/rtsp-server/rtsp-media-factory.h:
6856 * gst/rtsp-server/rtsp-media.c:
6857 * gst/rtsp-server/rtsp-media.h:
6858 Handle media bus messages
6859 Handle media bus messages in a custom mainloop and dispatch them to the
6860 RTSPMedia objects. Let the default implementation handle some common messages.
6862 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6864 * gst/rtsp-server/rtsp-client.c:
6865 * gst/rtsp-server/rtsp-session-pool.c:
6866 * gst/rtsp-server/rtsp-session.c:
6867 Some more session timeout handling
6868 Move the session header setting code to a central place so that we always add
6869 the timeout parameter too.
6870 Handle timeouts by running the session cleanup code.
6871 Stop media before cleaning up.
6873 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6875 * gst/rtsp-server/rtsp-client.c:
6876 * gst/rtsp-server/rtsp-client.h:
6877 Add timeout property
6878 Add a timeout property ot the client and make the other properties into GObject
6881 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6883 * gst/rtsp-server/rtsp-session-pool.c:
6884 Use getters and setters in property code
6885 Use the getters and setters for the timeout property instead of locking
6888 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6890 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
6892 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6894 * gst/rtsp-server/rtsp-session-pool.c:
6895 * gst/rtsp-server/rtsp-session-pool.h:
6896 * gst/rtsp-server/rtsp-session.c:
6897 * gst/rtsp-server/rtsp-session.h:
6898 Add more timeout stuff
6899 Add method to check if a session is expired.
6900 Add method to perform cleanup on a session pool.
6902 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6904 * gst/rtsp-server/rtsp-client.c:
6905 * gst/rtsp-server/rtsp-session-pool.c:
6906 * gst/rtsp-server/rtsp-session-pool.h:
6907 * gst/rtsp-server/rtsp-session.c:
6908 * gst/rtsp-server/rtsp-session.h:
6909 Add beginnings of session timeouts and limits
6910 Add the timeout value to the Session header for unusual timeout values.
6911 Allow us to configure a limit to the amount of active sessions in a pool. Set a
6912 limit on the amount of retry we do after a sessionid collision.
6913 Add properties to the sessionid and the timeout of a session. Keep track of
6914 creation time and last access time for sessions.
6916 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6918 * gst/rtsp-server/rtsp-client.c:
6919 * gst/rtsp-server/rtsp-media.c:
6920 * gst/rtsp-server/rtsp-media.h:
6921 * gst/rtsp-server/rtsp-sdp.c:
6922 * gst/rtsp-server/rtsp-session-pool.c:
6923 * gst/rtsp-server/rtsp-session.c:
6924 * gst/rtsp-server/rtsp-session.h:
6925 Cleanup of sessions and more
6926 Fix the refcounting of media and sessions in the client. Properly clean up the
6927 session data when the client performs a teardown.
6928 Add Server header to responses.
6929 Allow for multiple uri setups in one session.
6930 Add Range header to the PLAY response and add the range attribute to the SDP
6932 Fix the session pool remove method, it used the wrong key in the hashtable. Also
6933 give the ownership of the sessionid to the session object.
6935 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6937 * gst/rtsp-server/rtsp-server.c:
6938 * gst/rtsp-server/rtsp-server.h:
6940 Rename the 'server_port' variable to simply 'port'.
6942 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6945 * gst/rtsp-server/rtsp-client.c:
6946 * gst/rtsp-server/rtsp-media.c:
6947 * gst/rtsp-server/rtsp-media.h:
6948 * gst/rtsp-server/rtsp-session.c:
6949 * gst/rtsp-server/rtsp-session.h:
6950 Rework the way we handle transports for streams
6951 Make the media accept an array of transports for the streams that we have
6952 configured for the play/pause requests.
6953 Implement server states for a client and its media.
6954 Require 0.10.22.1 (git HEAD) of gstreamer.
6956 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6958 * gst/rtsp-server/rtsp-client.c:
6959 * gst/rtsp-server/rtsp-media-factory.c:
6960 Drop const from functions dealing with urls
6961 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
6962 have the right const in them.
6964 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6966 * gst/rtsp-server/rtsp-client.c:
6967 * gst/rtsp-server/rtsp-media.c:
6968 * gst/rtsp-server/rtsp-sdp.c:
6972 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6974 * gst/rtsp-server/rtsp-client.c:
6975 * gst/rtsp-server/rtsp-media-factory.c:
6976 * gst/rtsp-server/rtsp-media.c:
6977 * gst/rtsp-server/rtsp-media.h:
6979 Don't keep a reference to the GstRTSPMedia in the stream.
6980 Free more things when freeing the GstRTSPMedia.
6982 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6985 * gst/rtsp-server/rtsp-media-factory.c:
6986 * gst/rtsp-server/rtsp-media-factory.h:
6987 * gst/rtsp-server/rtsp-media.c:
6988 * gst/rtsp-server/rtsp-media.h:
6989 * gst/rtsp-server/rtsp-server.c:
6990 * gst/rtsp-server/rtsp-server.h:
6991 More docs and small cleanups
6992 Add some more docs and update the README
6993 Cleanup some method names.
6994 Remove an unneeded idx field in the GstRTSPMediaStream
6996 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6999 * examples/Makefile.am:
7000 * examples/test-readme.c:
7001 Add a README and more example code
7002 Add a README file that contains a small introduction on how to use the server
7003 along with the example code explained in the readme.
7005 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7007 * gst/rtsp-server/rtsp-media.c:
7008 * gst/rtsp-server/rtsp-server.c:
7009 Fix some leaks and change default port
7010 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
7011 we finished the initial preroll. If we keep them locked, setting the pipeline to
7012 NULL will not stop and clean up the sources correctly.
7013 Change the default RTSP port to 8554 aka the official alternative RTSP port.
7015 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7017 * gst/rtsp-server/rtsp-session.c:
7018 * gst/rtsp-server/rtsp-session.h:
7019 Cleanups to the session object
7020 Remove some unneeded variables in the session state of a stream such as the
7021 owner media and the server transport.
7022 Get the configuration of a media stream in a session based on the media_stream
7023 in the original object instead of our cached index.
7024 Free more data in the finalize method.
7026 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7028 * gst/rtsp-server/rtsp-client.c:
7029 * gst/rtsp-server/rtsp-client.h:
7030 Cleanups and reuse media from DESCRIBE
7031 Handle thread create errors.
7032 Rename some internal methods to better match what they actually do.
7033 Handle misconfiguration of session_pool and media_mapping gracefully.
7034 Cache the DESCRIBE media and uri in the client connection and reuse them when
7035 we receive a SETUP request in the same connection for the same uri.
7036 Cleanup the client connection object.
7038 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7040 * gst/rtsp-server/rtsp-media-factory.c:
7041 * gst/rtsp-server/rtsp-media-factory.h:
7042 * gst/rtsp-server/rtsp-media.c:
7043 * gst/rtsp-server/rtsp-media.h:
7044 Add shared properties to media and factory
7045 Add the shared property to media.
7046 Implement some simple caching in the factory depending on if the media is shared
7049 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7051 * gst/rtsp-server/rtsp-client.c:
7052 Add a little comment
7053 Add some comment about the content-base header.
7055 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7057 * examples/Makefile.am:
7059 * examples/test-mp4.c:
7060 * examples/test-ogg.c:
7061 * examples/test-video.c:
7062 * gst/rtsp-server/Makefile.am:
7063 * gst/rtsp-server/rtsp-client.c:
7064 * gst/rtsp-server/rtsp-client.h:
7065 * gst/rtsp-server/rtsp-media-factory.c:
7066 * gst/rtsp-server/rtsp-media-factory.h:
7067 * gst/rtsp-server/rtsp-media.c:
7068 * gst/rtsp-server/rtsp-media.h:
7069 * gst/rtsp-server/rtsp-sdp.c:
7070 * gst/rtsp-server/rtsp-sdp.h:
7071 * gst/rtsp-server/rtsp-server.c:
7072 * gst/rtsp-server/rtsp-server.h:
7073 * gst/rtsp-server/rtsp-session.c:
7074 * gst/rtsp-server/rtsp-session.h:
7075 Reorganize things, prepare for media sharing
7076 Added various other test server examples
7077 Move the SDP message generation to a separate helper.
7078 Refactor common code for finding the session.
7079 Add content-base for realplayer compatibility
7080 Clean up request uris before processing for better vlc compatibility.
7081 Move prerolling and pipeline construction to the RTSPMedia object.
7082 Use multiudpsink for future pipeline reuse.
7084 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7090 === release 0.10.1 ===
7092 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7098 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7100 * bindings/vala/Makefile.am:
7102 Add more directories and files to the dist.
7104 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7106 * bindings/python/Makefile.am:
7107 * bindings/python/rtspserver.override:
7108 Fixed compile error of python bindings
7110 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7112 * bindings/vala/gst-rtsp-server-0.10.vapi:
7113 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7114 Marked values as nullable accordingly
7116 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7118 * bindings/vala/gst-rtsp-server-0.10.vapi:
7119 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
7120 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7121 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7122 Updated Vala bindings
7124 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7126 * gst/rtsp-server/rtsp-client.c:
7127 * gst/rtsp-server/rtsp-media-mapping.c:
7128 * gst/rtsp-server/rtsp-media-mapping.h:
7129 * gst/rtsp-server/rtsp-media.h:
7130 * gst/rtsp-server/rtsp-session-pool.h:
7131 Cleanups and doc updates
7132 Add some more documentation and do some minor cleanups here and there.
7134 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7136 * gst/rtsp-server/rtsp-client.c:
7137 * gst/rtsp-server/rtsp-media-factory.c:
7138 * gst/rtsp-server/rtsp-media-factory.h:
7139 * gst/rtsp-server/rtsp-media.c:
7140 * gst/rtsp-server/rtsp-media.h:
7141 * gst/rtsp-server/rtsp-session.c:
7142 * gst/rtsp-server/rtsp-session.h:
7144 Rename GstRTSPMediaBin to GstRTSPMedia
7145 Parse the request url into a GstRTSPUri object and pass this object to the
7146 various handlers and methods that require the uri.
7148 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7152 Add some more docs and remove some old code from the example.
7154 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7156 * gst/rtsp-server/rtsp-client.c:
7157 Handle state change failures better
7158 Handle state change failures better when changing the state of the pipeline to
7161 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7163 * gst/rtsp-server/rtsp-media-factory.c:
7164 * gst/rtsp-server/rtsp-media-factory.h:
7165 Make element creation more extendible
7166 Add get_element vmethod to the default MediaFactory so that subclasses can just
7167 override that method and still use the default logic for making a MediaBin from
7170 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7173 * gst/rtsp-server/Makefile.am:
7174 * gst/rtsp-server/rtsp-client.c:
7175 * gst/rtsp-server/rtsp-client.h:
7176 * gst/rtsp-server/rtsp-media-factory.c:
7177 * gst/rtsp-server/rtsp-media-factory.h:
7178 * gst/rtsp-server/rtsp-media-mapping.c:
7179 * gst/rtsp-server/rtsp-media-mapping.h:
7180 * gst/rtsp-server/rtsp-media.c:
7181 * gst/rtsp-server/rtsp-media.h:
7182 * gst/rtsp-server/rtsp-server.c:
7183 * gst/rtsp-server/rtsp-server.h:
7184 * gst/rtsp-server/rtsp-session.c:
7185 * gst/rtsp-server/rtsp-session.h:
7186 Make the server handle arbitrary pipelines
7187 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
7188 The GstMediaBin object has a handle to a bin with elements and to a list of
7189 GstMediaStream objects that this bin produces.
7190 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
7191 with methods to register and remove those mappings.
7192 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
7193 used by the server instance.
7194 Modify the example application so that it shows how to create custom pipelines
7195 attached to a specific mount point.
7196 Various misc cleanps.
7198 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7200 * gst/rtsp-server/rtsp-server.c:
7201 * gst/rtsp-server/rtsp-server.h:
7202 Allow setting a custom media factory for a server
7204 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7206 * gst/rtsp-server/rtsp-client.c:
7207 * gst/rtsp-server/rtsp-client.h:
7208 Allow setting a custom media factory for a client.
7210 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7212 * gst/rtsp-server/Makefile.am:
7213 Add Makefile entry for the media factory
7215 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7217 * gst/rtsp-server/rtsp-media-factory.c:
7218 * gst/rtsp-server/rtsp-media-factory.h:
7219 Add media factory to map urls to media pipeline objects.
7221 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7223 * gst/rtsp-server/rtsp-media.c:
7224 * gst/rtsp-server/rtsp-media.h:
7225 Add comments. Remove unused field
7227 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7229 * gst/rtsp-server/rtsp-session-pool.c:
7230 * gst/rtsp-server/rtsp-session-pool.h:
7231 Allow custom session pools to override the session id allocation algorithms Add some comments.
7233 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7235 * gst/rtsp-server/rtsp-session.h:
7238 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7240 * gst/rtsp-server/rtsp-client.c:
7241 * gst/rtsp-server/rtsp-client.h:
7242 Move the connection code in one place Add some comments
7244 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7246 * gst/rtsp-server/rtsp-server.c:
7247 * gst/rtsp-server/rtsp-server.h:
7248 Make vmethod to create and accept new clients. Add some docs.
7250 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7252 * gst/rtsp-server/rtsp-server.c:
7253 * gst/rtsp-server/rtsp-server.h:
7254 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
7256 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7258 * gst/rtsp-server/rtsp-client.c:
7259 * gst/rtsp-server/rtsp-client.h:
7260 Name the parameters more appropriately.
7262 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7264 * gst/rtsp-server/rtsp-session-pool.c:
7265 Do some more cleanup of the session pool.
7267 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7269 * gst/rtsp-server/Makefile.am:
7270 * gst/rtsp-server/rtsp-client.c:
7271 Check if return value of gst_rtsp_session_get_media is not NULL
7273 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7275 * gst/rtsp-server/Makefile.am:
7276 Install rtsp-session and rtsp-session-pool headers
7278 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7283 * bindings/python/Makefile.am:
7284 * bindings/python/arg-types.py:
7285 * bindings/python/codegen/Makefile.am:
7286 * bindings/python/codegen/__init__.py:
7287 * bindings/python/codegen/argtypes.py:
7288 * bindings/python/codegen/code-coverage.py:
7289 * bindings/python/codegen/codegen.py:
7290 * bindings/python/codegen/definitions.py:
7291 * bindings/python/codegen/defsparser.py:
7292 * bindings/python/codegen/docextract.py:
7293 * bindings/python/codegen/docgen.py:
7294 * bindings/python/codegen/fileprefix.override:
7295 * bindings/python/codegen/fileprefixmodule.c:
7296 * bindings/python/codegen/h2def.py:
7297 * bindings/python/codegen/mergedefs.py:
7298 * bindings/python/codegen/mkskel.py:
7299 * bindings/python/codegen/override.py:
7300 * bindings/python/codegen/reversewrapper.py:
7301 * bindings/python/codegen/scmexpr.py:
7302 * bindings/python/rtspserver-types.defs:
7303 * bindings/python/rtspserver.defs:
7304 * bindings/python/rtspserver.override:
7305 * bindings/python/rtspservermodule.c:
7307 Add python bindings.
7309 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7311 * bindings/Makefile.am:
7313 Don't go into python dir when requirements for python bindings are missing
7315 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7317 * bindings/Makefile.am:
7318 * bindings/vala/Makefile.am:
7320 Install Vala bindings if vala is available
7322 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7324 * bindings/vala/gst-rtsp-server-0.10.deps:
7325 * bindings/vala/gst-rtsp-server-0.10.vapi:
7326 * bindings/vala/gst-rtsp-server.vapi:
7327 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
7328 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
7329 * bindings/vala/packages/gst-rtsp-server-0.10.files:
7330 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7331 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7332 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
7333 * bindings/vala/packages/gst-rtsp-server.deps:
7334 * bindings/vala/packages/gst-rtsp-server.excludes:
7335 * bindings/vala/packages/gst-rtsp-server.files:
7336 * bindings/vala/packages/gst-rtsp-server.gi:
7337 * bindings/vala/packages/gst-rtsp-server.metadata:
7338 * bindings/vala/packages/gst-rtsp-server.namespace:
7339 Regenerated Vala bindings
7341 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7343 * bindings/vala/gst-rtsp-server.vapi:
7344 * bindings/vala/packages/gst-rtsp-server.metadata:
7345 Fixed typo in included headers for vala bindings
7347 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7351 * pkgconfig/Makefile.am:
7352 * pkgconfig/gst-rtsp-server.pc.in:
7353 Added pkgconfig file
7355 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
7357 * bindings/vala/gst-rtsp-server.vapi:
7358 * bindings/vala/packages/gst-rtsp-server.excludes:
7359 * bindings/vala/packages/gst-rtsp-server.gi:
7360 * bindings/vala/packages/gst-rtsp-server.metadata:
7361 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
7363 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
7365 * bindings/vala/gst-rtsp-server.vapi:
7366 * bindings/vala/packages/gst-rtsp-server.deps:
7367 * bindings/vala/packages/gst-rtsp-server.files:
7368 * bindings/vala/packages/gst-rtsp-server.gi:
7369 * bindings/vala/packages/gst-rtsp-server.metadata:
7370 * bindings/vala/packages/gst-rtsp-server.namespace:
7373 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
7375 * gst/rtsp-server/rtsp-session.c:
7376 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
7378 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
7380 * examples/Makefile.am:
7381 * gst/rtsp-server/Makefile.am:
7382 Put GStreamer version in library name
7384 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7386 * examples/Makefile.am:
7387 * gst/rtsp-server/Makefile.am:
7388 Fix some issues to pass distcheck
7390 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7392 * gst/rtsp-server/rtsp-server.c:
7393 Added port property to GstRTSPServer class.
7395 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7400 * examples/Makefile.am:
7403 * gst/rtsp-server/Makefile.am:
7404 * gst/rtsp-server/rtsp-client.c:
7405 * gst/rtsp-server/rtsp-client.h:
7406 * gst/rtsp-server/rtsp-media.c:
7407 * gst/rtsp-server/rtsp-media.h:
7408 * gst/rtsp-server/rtsp-server.c:
7409 * gst/rtsp-server/rtsp-server.h:
7410 * gst/rtsp-server/rtsp-session-pool.c:
7411 * gst/rtsp-server/rtsp-session-pool.h:
7412 * gst/rtsp-server/rtsp-session.c:
7413 * gst/rtsp-server/rtsp-session.h:
7416 * src/rtsp-client.c:
7417 * src/rtsp-client.h:
7420 * src/rtsp-server.c:
7421 * src/rtsp-server.h:
7422 * src/rtsp-session-pool.c:
7423 * src/rtsp-session-pool.h:
7424 * src/rtsp-session.c:
7425 * src/rtsp-session.h:
7426 Split in library and example program
7428 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
7430 * src/rtsp-client.h:
7431 Removed obsolete variable
7433 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
7435 * src/rtsp-client.c:
7436 * src/rtsp-client.h:
7437 Removed pipeline variable GstRTSPClient, because it's only used in one function
7439 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7442 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
7444 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
7446 * src/rtsp-session.c:
7447 Initialize some more vars.
7449 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
7451 * src/rtsp-session.c:
7452 Initialize variable to avoid compiler warning.
7454 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
7457 Add a reasonable generic .gitignore