3 2014-06-22 Sebastian Dröge <slomo@coaxion.net>
8 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
10 * gst/rtsp-server/rtsp-client.c:
11 * gst/rtsp-server/rtsp-sdp.c:
12 * gst/rtsp-server/rtsp-sdp.h:
13 mikey: add different key length parameters
14 Add encryption and authentication key length parameters to MIKEY. For
15 the encoders, the key lengths are obtained from the cipher and auth
16 algorithms set in the caps. For the decoders, they are obtained while
17 parsing the key management from the client.
18 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
20 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
22 * tests/check/gst/stream.c:
23 stream tests: Make sure we get right multicast address from stream
24 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
26 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
28 * gst/rtsp-server/rtsp-client.c:
29 client: ref the context until rtsp watch is alive
30 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
32 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
34 * gst/rtsp-server/rtsp-client.c:
35 client: Destroy the rtsp watch after connection close
37 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
39 * gst/rtsp-server/rtsp-media.c:
40 media: fix confusing comment
42 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
44 * gst/rtsp-server/rtsp-session.c:
45 rtsp-session: Timeout in header.
46 Adding the possbilty to always have timout in header.
47 This is configurabe with setting "timeout-always-visible".
48 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
50 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
57 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
64 * gst-rtsp-server.doap:
67 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
70 Automatic update of common submodule
71 From 211fa5f to 1f5d3c3
73 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
75 * gst/rtsp-server/rtsp-client.c:
76 client: store TCP ports in transport
77 Store the TCP ports in the transport when we are doing RTSP over TCP.
78 This way, we can easily get to the ports from the transport.
79 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
81 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
83 * gst/rtsp-server/rtsp-stream.c:
84 stream: add signals for new RTP/RTCP encoders
85 New signals to allow the user to configure the dynamically created
87 https://bugzilla.gnome.org/show_bug.cgi?id=730228
89 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
91 * gst/rtsp-server/rtsp-media.c:
92 * gst/rtsp-server/rtsp-media.h:
93 media: Make suspend()/unsuspend() virtual
94 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
96 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
98 * gst/rtsp-server/rtsp-client.c:
99 client: fix send-message signal marshaller
100 Use generic marshalling for the send-message signal. It has
101 two POINTER arguments, not just one.
102 https://bugzilla.gnome.org/show_bug.cgi?id=729900
104 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
106 * tests/check/gst/media.c:
107 tests: add and remove pads only once
108 In this test we simulate a dynamic pad by watching the caps event.
109 Because of renegotiation in the base payloader now, this caps is sent
110 multiple times but we can only deal with 1 invocation, use a variable to
111 only 'add and remove' the pad once.
113 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
115 * tests/check/gst/rtspserver.c:
116 tests: add unit test for correct handling of Require headers
117 https://bugzilla.gnome.org/show_bug.cgi?id=729426
119 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
121 * gst/rtsp-server/rtsp-client.c:
122 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
123 Servers must handle Require headers and must report a failure
124 if they don't handle any of the Required options, see RFC 2326,
125 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
126 https://bugzilla.gnome.org/show_bug.cgi?id=729426
128 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
133 === release 1.3.1 ===
135 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
141 * gst-rtsp-server.doap:
144 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
147 Automatic update of common submodule
148 From bcb1518 to 211fa5f
150 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
155 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
157 * tests/check/gst/sessionmedia.c:
158 tests: fix memory leak in sessionmedia unit test
160 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
162 * gst/rtsp-server/rtsp-client.c:
163 client: emit a signal before sending a message
164 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
166 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
168 * gst/rtsp-server/rtsp-client.c:
169 client: pass context to send_message
170 Pass the current context to send_message, we will need it later.
172 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
174 * gst/rtsp-server/rtsp-client.c:
175 client: fix typo in comment
177 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
179 * gst/rtsp-server/rtsp-media.c:
180 media: Do not stop thread twice if default_prepare() fails
182 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
184 * gst/rtsp-server/rtsp-client.c:
185 client: set the watch to flushing before going to NULL
186 First set the watch to flushing so that we unblock any current and
187 future attempt to send data on the watch, Then set the pipeline to
189 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
191 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
193 * gst/rtsp-server/rtsp-session-pool.c:
194 * tests/check/gst/sessionpool.c:
195 rtsp-session-pool: Fixes annotation
196 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
197 in the sessionpool test.
198 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
200 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
202 * gst/rtsp-server/rtsp-media.c:
203 * gst/rtsp-server/rtsp-media.h:
204 media: make media_prepare virtual
205 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
207 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
209 * gst/rtsp-server/rtsp-media.c:
210 * tests/check/gst/media.c:
211 media: stop the thread in more error cases
213 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
215 * gst/rtsp-server/rtsp-media.c:
216 * tests/check/gst/media.c:
217 media: allow NULL as the thread
218 Use the default context whan passing a NULL thread.
220 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
222 * gst/rtsp-server/rtsp-client.c:
223 rtsp-client: indent cleanup
224 Coverity was moaning about unreachable code, and I think it was just
225 confused by { being before the label. We'll see if it pops up again.
228 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
230 * gst/rtsp-server/rtsp-client.c:
231 * gst/rtsp-server/rtsp-media.c:
232 client: Add drop-backlog property
233 When we have too many messages queued for a client (currently hardcoded
234 to 100) we overflow and drop the messages. Add a drop-backlog property
235 to control this behaviour. Setting this property to FALSE will retry
236 to send the messages to the client by waiting for more room in the
238 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
240 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
242 * gst/rtsp-server/rtsp-client.c:
243 client: support for POST before GET when setting up a tunnel
245 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
247 * gst/rtsp-server/rtsp-client.c:
248 client: remove watch of the second client after http tunnel setup
249 The second client will be freed after the HTTP tunnel has been set up.
250 Make sure it's RTSP watch is never dispatched again.
251 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
253 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
255 * gst/rtsp-server/rtsp-media.c:
256 * tests/check/gst/media.c:
257 media: Make media_prepare() fail if port allocation fails
258 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
260 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
262 * tests/check/gst/media.c:
263 media test: cleanup the thread pool in tests
265 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
267 * gst/rtsp-server/rtsp-media.c:
268 * tests/check/gst/media.c:
269 rtsp-media: Unblock blocked streams in unprepare
270 The streams will be blocked when a live media is prepared.
271 The streams should be unblocked in gst_rtsp_media_unprepare.
272 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
274 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
276 * gst/rtsp-server/rtsp-media.c:
277 media: release the state lock when going to NULL
278 Set our state to UNPREPARING and release the state-lock before
279 setting the pipeline to the NULL state. This way, any pad-added
280 callback will be able to take the state-lock and check that we are now
281 unpreparing instead of deadlocking.
282 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
284 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
286 * gst/rtsp-server/rtsp-media.c:
287 media: protect status with lock
288 Make sure we only update the status with the lock.
290 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
292 * gst/rtsp-server/rtsp-client.c:
293 * gst/rtsp-server/rtsp-sdp.c:
294 rtsp: update for MIKEY API changes
296 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
298 * gst/rtsp-server/rtsp-client.c:
299 client: parse the mikey response from the client
300 Parse the mikey response from the client and update the policy for
303 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
305 * gst/rtsp-server/rtsp-stream.c:
306 * gst/rtsp-server/rtsp-stream.h:
307 stream: add method to set crypto info
308 Make a method to configure the crypto information of a stream.
309 Set udpsrc in READY instead of PAUSED so that we can configure caps
312 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
314 * gst/rtsp-server/rtsp-client.c:
315 client: cleanup error paths
317 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
319 * gst/rtsp-server/rtsp-media.c:
322 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
324 * examples/test-video.c:
325 test: enable SRTP only on RTSPS
326 We only want to enable SRTP when doing rtsp over TLS so that we can
327 exchange the keys in a secure way.
329 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
331 * examples/test-video.c:
332 test: print an error on failure
334 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
337 * examples/test-video.c:
338 * gst/rtsp-server/rtsp-sdp.c:
339 * gst/rtsp-server/rtsp-stream.c:
340 * tests/check/Makefile.am:
341 stream: add SRTP support
342 Install srtp encoder and decoder elements in rtpbin
345 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
347 * tests/check/Makefile.am:
348 * tests/check/gst/sessionpool.c:
349 tests: Add unit tests for sessionpool
350 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
352 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
354 * tests/check/gst/threadpool.c:
355 tests: Improve code coverage of rtsp-threadpool tests
356 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
358 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
360 * tests/check/gst/sessionmedia.c:
361 tests: Improve code coverage for rtsp-session-media
362 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
364 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
366 gobject-introspection: Add annotations to support language bindings
367 In addition a few cosmetic changes:
368 * Adjust the order of arguments
369 * Fix typo: occured -> occurred
370 * Fix indentation after Return:-clauses
371 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
373 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
375 * gst/rtsp-server/rtsp-stream.c:
376 rtsp-stream: Don't mix IPv4 and IPv6 addresses
377 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
379 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
381 * gst/rtsp-server/rtsp-stream.c:
382 stream: take caps after the session manager
383 Take the caps for the SDP after they leave the rtpbin so that we can
384 also get the properties added by rtpbin elements.
386 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
388 * gst/rtsp-server/rtsp-stream.c:
389 stream: release lock while pushing out packets
390 Keep a cache of the transports and use this to iterate the transport
391 while pushing packets. This allows us to release the lock early.
392 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
394 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
396 * gst/rtsp-server/rtsp-client.c:
397 * gst/rtsp-server/rtsp-client.h:
398 rtsp-client: vmethod for modifying tunnel GET response
399 Add a vmethod tunnel_http_response where the response to the HTTP GET
400 for tunneled connections can be modified.
401 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
403 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
405 * gst/rtsp-server/rtsp-sdp.c:
406 sdp: make 1 media line per profile
407 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
408 line in the SDP for each profile. The client is then supposed to pick
409 one of the profiles in the SETUP request. Because the m= lines have the
410 same pt, the client also knows that only 1 option is possible.
412 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
414 * gst/rtsp-server/rtsp-media-factory.c:
415 * gst/rtsp-server/rtsp-media-factory.h:
416 * gst/rtsp-server/rtsp-media.c:
417 factory: add profile property and pass to media and streams
419 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
421 * examples/test-multicast.c:
422 * gst/rtsp-server/rtsp-sdp.c:
423 sdp: pass multicast connection for multicast-only stream
424 Pass the multicast address of the stream in the connection info in the
425 SDP so that clients try a multicast connection first.
426 Only allow multicast connections in the test-multicast example. Also
427 increase the TTL a little.
429 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
432 .gitignore: Ignore gcov intermediate files
433 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
435 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
437 * gst/rtsp-server/rtsp-stream.c:
438 stream: release some locks in error cases
440 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
442 docs: Enable and fix gtk-doc warnings
443 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
444 * addresspool/mediafactory: Add missing annotation colon
445 * stream: Annotate return value
446 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
448 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
451 Automatic update of common submodule
452 From fe1672e to bcb1518
454 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
457 Automatic update of common submodule
458 From 1a07da9 to fe1672e
460 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
462 * examples/Makefile.am:
463 examples: use LDADD for libs instead of LDFLAGS
465 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
468 configure: make sure releases are in .doap file
470 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
472 * examples/test-cgroups.c:
473 examples: test-cgroups: don't put code with side effects into g_assert()
474 The g_assert() might get compiled out with the right
475 compiler/preprocessor flags.
477 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
479 * examples/.gitignore:
480 examples: add cgroup test binary to .gitignore
482 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
484 * examples/test-cgroups.c:
485 examples: fix cgroup test build
486 Fixes build failure caused by compiler warning:
487 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
489 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
492 .gitignore: ignore temp files created in the course of 'make check'
494 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
496 * gst/rtsp-server/rtsp-media.c:
497 rtsp-media: don't loose frames handling new PLAY request
498 If client supplied a range check if the range specifies the start point.
499 If not, then do an accurate seek to the current position. If a start
500 point was specified do do a key unit seek to make sure the streaming
501 starts with decodeable frames.
502 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
504 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
506 * gst/rtsp-server/rtsp-media.c:
507 Revert "media: only flush when setting a new start position"
508 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
509 We need to do the flush in all cases, demuxer block currently for
512 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
514 * gst/rtsp-server/rtsp-media.c:
515 media: only flush when setting a new start position
516 Only flush the pipeline when we change the start position with
518 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
520 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
522 * gst/rtsp-server/rtsp-stream.c:
523 stream: set ttl-mc before adding the socket
524 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
525 never be set on socket.
526 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
528 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
530 * gst/rtsp-server/rtsp-media.c:
531 media: stop thread if media is already prepared
532 in gst_rtsp_media_prepare() the thread is not used if media is already
533 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
535 https://bugzilla.gnome.org/show_bug.cgi?id=724182
537 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
540 build: Ship gst-rtsp-server.doap file
542 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
544 * tests/check/gst/rtspserver.c:
545 tests: Fix another compiler warning with gcc
547 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
549 * gst/rtsp-server/rtsp-client.c:
550 * gst/rtsp-server/rtsp-mount-points.c:
551 * gst/rtsp-server/rtsp-stream.c:
552 * tests/check/gst/client.c:
553 rtsp-server: Fix lots of compiler warnings with clang
555 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
558 * gst-rtsp-server.doap:
560 configure: Synchronise with the configure scripts of the other modules
562 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
565 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
567 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
569 * gst/rtsp-server/rtsp-media.c:
570 * gst/rtsp-server/rtsp-stream.c:
571 Revert "rtsp-server: support build against last stable release"
572 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
573 Let us require 1.2.3 now, which is going to be released in a few
576 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
578 * gst/rtsp-server/rtsp-session-media.c:
579 * gst/rtsp-server/rtsp-stream-transport.c:
580 session: improve RTP-Info
581 Ignore streams that can't generate RTP-Info instead of failing.
582 Don't return the empty string when all streams are unconfigured but
583 return NULL so that we don't generate and empty RTP-Info header.
584 Improve docs a little.
586 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
588 * gst/rtsp-server/rtsp-session-media.c:
589 Don't free rtpinfo GString when it is NULL
590 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
592 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
594 * gst/rtsp-server/rtsp-media.c:
595 media: only set keyframe flag when modifying start
596 Only set the keyframe flag when we modify the start position. The
597 keyframe flag should probably be ignored when no change is requested but
598 until we can claim this is all documented properly and all demuxer
599 implement this, avoid setting the flag.
600 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
602 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
604 * gst/rtsp-server/rtsp-thread-pool.c:
605 thread-pool: Unref source after mainloop has quit to avoid races in GLib
606 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
608 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
610 * gst/rtsp-server/rtsp-stream.c:
611 stream: handle NULL seqnum and rtptime arguments
613 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
615 * gst/rtsp-server/rtsp-thread-pool.c:
616 * tests/check/gst/threadpool.c:
617 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
618 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
620 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
622 * gst/rtsp-server/rtsp-stream.c:
623 stream: add fallback for missing stats property
624 Use a fallback when the payloader does not have a stats property
625 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
627 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
630 Automatic update of common submodule
631 From f7bc1c3 to 1a07da9
633 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
635 * gst/rtsp-server/rtsp-stream.c:
636 stream: don't leak stats structure
637 Don't leak the stats structure and deal with NULL stats.
639 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
641 * gst/rtsp-server/rtsp-stream.c:
642 stream: Get rtpinfo properties atomically from payloader
643 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
645 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
647 * gst/rtsp-server/rtsp-media.c:
648 media: refactor state change functions and signals
649 Make functions to set the target state and the pipeline state and emit
650 the signals from those functions.
652 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
654 * gst/rtsp-server/rtsp-media.c:
655 * gst/rtsp-server/rtsp-media.h:
656 media: add signal to notify of pending state changes
658 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
660 * gst/rtsp-server/rtsp-media.c:
661 * gst/rtsp-server/rtsp-stream.c:
662 rtsp-server: support build against last stable release
663 Until 1.2.3 is out with the new get_type function and we
666 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
668 * gst/rtsp-server/rtsp-stream.c:
669 stream: fix compilation
671 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
673 * gst/rtsp-server/rtsp-media.c:
674 * gst/rtsp-server/rtsp-media.h:
675 * gst/rtsp-server/rtsp-stream.c:
676 * gst/rtsp-server/rtsp-stream.h:
677 stream: add property to configure profiles
679 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
681 * gst/rtsp-server/rtsp-client.c:
682 client: let stream check supported transport
683 Delegate the check if a transport is allowed to the stream.
684 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
686 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
688 * gst/rtsp-server/rtsp-stream.c:
689 * gst/rtsp-server/rtsp-stream.h:
690 stream: add method to check supported transport
691 Add a method to check if a transport is supported
693 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
696 configure.ac: Only check for gstreamer-check, not check
697 We include check in gstreamer-check since quite some time now.
699 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
701 * gst/rtsp-server/rtsp-session-media.c:
702 * gst/rtsp-server/rtsp-stream-transport.c:
703 * gst/rtsp-server/rtsp-stream.c:
704 * gst/rtsp-server/rtsp-stream.h:
705 stream: return clock-rate from get_rtpinfo
706 And use it to correct the rtptime to the requested start-time.
707 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
709 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
711 * gst/rtsp-server/rtsp-session-media.c:
712 * gst/rtsp-server/rtsp-stream-transport.c:
713 * gst/rtsp-server/rtsp-stream-transport.h:
714 session-media: calculate start-time
716 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
718 * gst/rtsp-server/rtsp-stream-transport.c:
719 * gst/rtsp-server/rtsp-stream.c:
720 * gst/rtsp-server/rtsp-stream.h:
721 stream: also return the running-time
722 Return the running-time in the rtpinfo as well.
724 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
726 * gst/rtsp-server/rtsp-client.c:
727 * gst/rtsp-server/rtsp-session-media.c:
728 * gst/rtsp-server/rtsp-session-media.h:
729 * gst/rtsp-server/rtsp-stream-transport.c:
730 * gst/rtsp-server/rtsp-stream-transport.h:
731 session-media: let the session-media make the RTPInfo
732 Add method to create the RTPInfo for a stream-transport.
733 Add method to create the RTPInfo for all stream-transports in a
735 Use the session-media RTPInfo code in client. This allows us to refactor
736 another method to link the TCP callbacks.
738 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
740 mount-points: sort sequence before g_sequence_lookup
741 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
742 sort sequence if dirty, otherwise lookup will fail.
743 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
745 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
748 configure: rename package from gst-rtsp to gst-rtsp-server
749 To match git module name and avoid confusion with the
750 rtsp lib in gst-plugins-base and rtsp plugin in -good.
752 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
755 configure: bump core/base/good requirement to 1.2.0
756 Bump to released stable version and make implicit
757 requirements explicit.
759 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
764 Fix broken gettext setup which is not used anyway
766 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
769 Automatic update of common submodule
770 From dbedaa0 to d48bed3
772 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
774 * gst/rtsp-server/rtsp-client.c:
775 * gst/rtsp-server/rtsp-media.c:
776 * gst/rtsp-server/rtsp-media.h:
777 media: add setup_sdp vmethod
778 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
779 gst_rtsp_media_setup_sdp.
780 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
782 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
784 * gst/rtsp-server/rtsp-stream.c:
785 rtsp-stream: Check return value of sscanf
786 streamid is only valid if sscanf matched something.
788 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
790 * gst/rtsp-server/rtsp-client.c:
791 rtsp-client: Fix iteration
792 Wouldn't even enter the code block otherwise (i++ was used as the check
793 and not the postfix).
795 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
797 * gst/rtsp-server/rtsp-client.c:
798 * gst/rtsp-server/rtsp-client.h:
799 client: add vmethod to configure media and streams
800 Implement a vmethod that can be used to configure the media and the
801 streams based on the current context. Handle the blocksize handling in
803 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
805 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
808 Make git ignore more unit test binaries
810 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
812 * gst/rtsp-server/rtsp-address-pool.h:
813 * gst/rtsp-server/rtsp-auth.h:
814 * gst/rtsp-server/rtsp-client.h:
815 * gst/rtsp-server/rtsp-context.h:
816 * gst/rtsp-server/rtsp-media-factory-uri.h:
817 * gst/rtsp-server/rtsp-media-factory.h:
818 * gst/rtsp-server/rtsp-media.h:
819 * gst/rtsp-server/rtsp-mount-points.h:
820 * gst/rtsp-server/rtsp-server.h:
821 * gst/rtsp-server/rtsp-session-media.h:
822 * gst/rtsp-server/rtsp-session-pool.h:
823 * gst/rtsp-server/rtsp-session.h:
824 * gst/rtsp-server/rtsp-stream-transport.h:
825 * gst/rtsp-server/rtsp-stream.h:
826 * gst/rtsp-server/rtsp-thread-pool.h:
827 * gst/rtsp-server/rtsp-token.h:
828 rtsp-server: add padding to many public structures
829 Not mini objects though, since they are not subclassable
830 anyway, nor kept on the stack or inlined in a structure.
832 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
834 media: add new create_rtpbin vmethod
835 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
836 https://bugzilla.gnome.org/show_bug.cgi?id=719734
838 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
840 * tests/check/gst/media.c:
841 tests: fix memory leak, free test's thread pool
842 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
844 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
846 * gst/rtsp-server/rtsp-stream-transport.c:
847 stream-transport: free url in finalize
849 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
851 * gst/rtsp-server/rtsp-media.c:
852 media: also do state change in suspended state
854 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
856 * gst/rtsp-server/rtsp-client.c:
857 * gst/rtsp-server/rtsp-media.c:
858 media: also handle prepare and range in suspended state
859 When we are suspended, we are already prepared.
860 We can get the range in the suspended state.
862 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
864 * tests/check/Makefile.am:
865 * tests/check/gst/sessionmedia.c:
866 check: add test for uri in setup
867 Added unit tests for the new functionality in GstRTSPStreamTransport.
868 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
870 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
872 * gst/rtsp-server/rtsp-client.c:
873 client: store setup uri and use in PLAY response
874 Store the uri used when doing the setup and use that in the PLAY
876 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
878 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
880 * gst/rtsp-server/rtsp-stream-transport.c:
881 * gst/rtsp-server/rtsp-stream-transport.h:
882 stream-transport: add method to get/set url
884 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
886 * gst/rtsp-server/rtsp-client.c:
887 client: suspend after SDP and unsuspend before PLAYING
888 Based on patches by Ognyan Tonchev <ognyan@axis.com>
889 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
891 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
893 * gst/rtsp-server/rtsp-media-factory.c:
894 * gst/rtsp-server/rtsp-media-factory.h:
895 * gst/rtsp-server/rtsp-media.c:
896 * gst/rtsp-server/rtsp-media.h:
897 * gst/rtsp-server/rtsp-session-media.c:
898 * gst/rtsp-server/rtsp-session.c:
899 * tests/check/gst/media.c:
900 * tests/check/gst/mediafactory.c:
901 media: add suspend modes
902 Add support for different suspend modes. The stream is suspended right after
903 producing the SDP and after PAUSE. Different suspend modes are available that
904 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
905 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
906 state and RESET will bring the pipeline to the NULL state.
907 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
908 this means that the pipeline needs to be prerolled again.
909 Base on patches by Ognyan Tonchev <ognyan@axis.com>
910 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
912 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
914 * gst/rtsp-server/rtsp-media.c:
915 media: start live streams in blocked state
916 Start live streams in the blocked state and make them preroll using the
917 messages. This ensure that no data is played by the sink until we explicitly
918 unblock the stream right before going to PLAYING.
919 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
921 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
923 * gst/rtsp-server/rtsp-media.c:
924 media: refactor starting and waiting for preroll
925 Based on patches from Ognyan Tonchev <ognyan@axis.com>
926 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
928 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
930 * gst/rtsp-server/rtsp-stream.c:
931 * gst/rtsp-server/rtsp-stream.h:
932 stream: add API to block streams
933 Add an API to block on the streams and make it post a message.
934 Based on patch by Ognyan Tonchev <ognyan@axis.com>
935 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
937 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
939 * docs/libs/Makefile.am:
940 docs: Specify the override file
941 Even if it's empty (for now) it avoids make distcheck complaining
943 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
945 * gst/rtsp-server/rtsp-media.c:
946 media: move default implementations to where they are used
948 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
950 * gst/rtsp-server/rtsp-media.c:
951 media: take the right lock in gst_rtsp_media_set_pipeline_state()
952 We need to take the state_lock when calling this method.
954 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
956 * gst/rtsp-server/rtsp-media.c:
957 media: handle add-added on non-bins too
958 Handle dynamic payloaders that are not bins, as used in the unit-test.
960 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
962 * gst/rtsp-server/rtsp-media-factory.c:
963 * gst/rtsp-server/rtsp-media-factory.h:
964 * gst/rtsp-server/rtsp-media.c:
965 rtsp-media/-factory: Fix request pad name comments
966 These must be escaped for gtk-doc to parse the comments without warnings.
968 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
970 rtsp-media: remove transports if media is in error status
971 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
972 trying to change to GST_STATE_NULL and media is in error status, we
973 remove all transports.
974 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
976 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
978 * gst/rtsp-server/rtsp-media.c:
979 rtsp-media: use element metadata to find payloader
980 Use the element metadata to find the payloader instead of checking
982 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
984 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
986 rtsp-stream: add getter for payload type
987 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
988 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
989 element and create the stream with this one instead of the dynpay%d
991 https://bugzilla.gnome.org/show_bug.cgi?id=712396
993 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
995 * gst/rtsp-server/rtsp-client.c:
996 * gst/rtsp-server/rtsp-context.h:
997 * gst/rtsp-server/rtsp-media.c:
998 * gst/rtsp-server/rtsp-mount-points.c:
999 * gst/rtsp-server/rtsp-server.c:
1000 * gst/rtsp-server/rtsp-token.c:
1001 rtsp-*: Refer to NULL as a constant in comments
1003 https://bugzilla.gnome.org/show_bug.cgi?id=714988
1005 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1007 rtsp-*: Fix type name typos in comments
1008 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
1009 * rtsp-auth: Refer to part of constant name as text
1010 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
1011 * rtsp-session-media: Fix GstRTSPSessionMedia typo
1012 * rtsp-stream: Fix typo when refering to GstBin
1013 https://bugzilla.gnome.org/show_bug.cgi?id=714988
1015 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1018 * docs/libs/gst-rtsp-server-docs.sgml:
1019 * docs/libs/gst-rtsp-server-sections.txt:
1020 docs: Improve documentation
1021 * Include annotation-glossary to quiet gtk-doc
1022 * Rename remaining ClientState -> Context
1023 * Rename object hierarchy file
1024 * Remove stale chapter references
1025 * Add missing function and object references
1026 * Include missing GstRTSPAddressPoolResult
1027 https://bugzilla.gnome.org/show_bug.cgi?id=714988
1029 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
1031 * gst/rtsp-server/rtsp-client.c:
1032 * gst/rtsp-server/rtsp-server.c:
1033 * gst/rtsp-server/rtsp-session-pool.c:
1034 * gst/rtsp-server/rtsp-session.c:
1035 * gst/rtsp-server/rtsp-stream.c:
1036 rtsp-server: sprinkle some allow-none annotations for g-i
1038 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
1040 * gst/rtsp-server/rtsp-stream.c:
1041 * gst/rtsp-server/rtsp-stream.h:
1042 stream: add method to filter transports
1043 Add a method to safely iterate and collect the stream transports
1044 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
1046 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
1048 * gst/rtsp-server/rtsp-client.c:
1049 * gst/rtsp-server/rtsp-server.c:
1050 * gst/rtsp-server/rtsp-session-pool.c:
1051 * gst/rtsp-server/rtsp-session.c:
1052 rtsp: allow NULL func in filters
1053 Passing a null function make the filters return a list of
1056 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
1058 * gst/rtsp-server/rtsp-address-pool.c:
1059 * tests/check/gst/addresspool.c:
1060 address-pool: fix address increment
1061 Use a guint instead of guint8 to increment the address. It's still not
1062 completely correct because a guint might not be able to hold the complete
1063 address range, but that's an enhacement for later.
1064 Add unit test to test improved behaviour.
1065 https://bugzilla.gnome.org/show_bug.cgi?id=708237
1067 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
1069 * gst/rtsp-server/rtsp-client.c:
1070 * tests/check/gst/client.c:
1071 client: allow absolute path in requests
1072 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
1074 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
1076 * gst/rtsp-server/rtsp-client.c:
1077 * gst/rtsp-server/rtsp-client.h:
1078 client: make make_path_from_uri a vmethod
1080 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
1082 * docs/libs/gst-rtsp-server-sections.txt:
1083 * gst/rtsp-server/rtsp-stream.c:
1084 * gst/rtsp-server/rtsp-stream.h:
1085 * tests/check/Makefile.am:
1086 * tests/check/gst/stream.c:
1087 stream: Add functions to get rtp and rtcp sockets
1088 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
1090 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
1092 * gst/rtsp-server/rtsp-context.c:
1093 * gst/rtsp-server/rtsp-context.h:
1094 context: defing a GType for the context
1095 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
1097 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
1099 * gst/rtsp-server/Makefile.am:
1100 * gst/rtsp-server/rtsp-auth.c:
1101 * gst/rtsp-server/rtsp-context.c:
1102 * gst/rtsp-server/rtsp-media.c:
1103 * gst/rtsp-server/rtsp-mount-points.c:
1104 * gst/rtsp-server/rtsp-server.h:
1105 * gst/rtsp-server/rtsp-session-media.c:
1106 * gst/rtsp-server/rtsp-session.c:
1107 * gst/rtsp-server/rtsp-stream.c:
1108 Fixed several GIR warnings
1110 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
1112 * gst/rtsp-server/rtsp-auth.c:
1115 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1117 * tests/check/Makefile.am:
1118 * tests/check/gst/token.c:
1119 tests: Add unit tests for token
1120 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
1122 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1124 * gst/rtsp-server/rtsp-token.c:
1125 token: Validate args for gst_rtsp_token_is_allowed
1126 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
1128 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1130 * gst/rtsp-server/rtsp-token.c:
1131 token: Fix bug when creating empty token
1132 We always want to have a valid GstStructure in the token.
1133 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
1135 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
1137 * gst/rtsp-server/rtsp-thread-pool.c:
1138 thread-pool: avoid race in shutdown
1139 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
1140 don't actually stop the mainloop ever. Solve this race by adding an idle source
1141 to the mainloop that calls the _quit. This way we immediately exit the mainloop
1142 if quit was called before we started it.
1144 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1146 * tests/check/Makefile.am:
1147 * tests/check/gst/permissions.c:
1148 tests: Add unit tests for permissions
1149 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
1151 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1153 * tests/check/gst/mediafactory.c:
1154 tests: Test mediafactory permissions
1155 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
1157 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1159 * gst/rtsp-server/rtsp-permissions.c:
1160 permissions: Fix refcounting when adding/removing roles
1161 Previously a role that was removed was unreffed twice, and when
1162 replacing an existing role the replaced role was freed while still being
1163 referenced. Both bugs are now fixed.
1164 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
1166 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1168 * tests/check/gst/media.c:
1169 * tests/check/gst/mediafactory.c:
1170 * tests/check/gst/rtspserver.c:
1171 tests: Check gst_rtsp_url_parse return value
1172 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
1174 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
1177 Automatic update of common submodule
1178 From 865aa20 to dbedaa0
1180 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
1182 * gst/rtsp-server/rtsp-server.c:
1183 rtsp-server: Fix socket leak
1184 https://bugzilla.gnome.org/show_bug.cgi?id=710088
1186 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
1188 * gst/rtsp-server/rtsp-session-pool.c:
1189 rtsp-session-pool: Make sure session IDs are properly URI-escaped
1190 https://bugzilla.gnome.org/show_bug.cgi?id=643812
1192 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
1194 * examples/.gitignore:
1195 * examples/test-video.c:
1196 examples: fix compilation when WITH_AUTH is defined
1197 https://bugzilla.gnome.org/show_bug.cgi?id=710228
1199 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
1202 gitignore: Add new test binary
1204 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
1206 * tests/check/Makefile.am:
1207 * tests/check/gst/threadpool.c:
1208 thread-pool: Add unit test for the thread pools
1209 https://bugzilla.gnome.org/show_bug.cgi?id=710228
1211 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
1213 * gst/rtsp-server/rtsp-thread-pool.c:
1214 thread-pool: Fix thread leak when reusing threads
1215 https://bugzilla.gnome.org/show_bug.cgi?id=709730
1217 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
1219 * gst/rtsp-server/rtsp-server.c:
1220 * tests/check/gst/rtspserver.c:
1221 tests: fixed racy behavior in rtspserver tests
1222 https://bugzilla.gnome.org/show_bug.cgi?id=710078
1224 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1226 * tests/check/gst/addresspool.c:
1227 tests: Improve address pool unit tests
1228 Add a range with mixed IPV4 and IPV6 addresses to pool.
1229 Get an IPV4 address from an IPV6-only pool.
1230 Get an IPV6 address from an IPV4-only pool.
1231 Reserve a IPV6 address from an IPV4-only pool.
1232 Check for unicast addresses in multicast-only pool.
1233 Check for unicast addresses in uni-/multicast-mixed pool.
1234 https://bugzilla.gnome.org/show_bug.cgi?id=710128
1236 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1238 * gst/rtsp-server/rtsp-client.c:
1239 client: append query string in PAUSE/PLAY/TEARDOWN as well
1241 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
1243 * gst/rtsp-server/rtsp-client.c:
1244 client: Add query to control path
1245 If the SETUP url contains a query it must be appended to the control
1246 path so that it matches any already created stream in the media. The
1247 query will also be appended to the session media path.
1249 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1251 * gst/rtsp-server/rtsp-media.c:
1252 rtsp-media: remove old line
1254 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
1256 * gst/rtsp-server/rtsp-stream.c:
1257 stream: Correct control comparison
1258 https://bugzilla.gnome.org/show_bug.cgi?id=709176
1260 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1262 * gst/rtsp-server/rtsp-media.c:
1263 media: Check dynamically if the pipeline supports seeking
1264 We should not depend on whether or not the pipeline state change
1265 returned NO_PREROLL or not. A media could dynamically change its
1266 element and switch from seekable to non seekable so it's best to test
1267 the seekable nature of the pipeline dynamically when we try to do a seek.
1269 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1271 * gst/rtsp-server/rtsp-media.c:
1272 media: Return FALSE if seeking is not supported
1274 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1276 * gst/rtsp-server/rtsp-media.c:
1277 rtsp-media: don't seek accurate by default
1278 Accurate seeking is perhaps a little overkill in the most common situation and
1279 causes some formats (mp3) over slow media to seek extremely slowly.
1281 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
1283 * tests/check/gst/rtspserver.c:
1284 tests: fix unit test
1285 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
1287 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
1289 * gst/rtsp-server/rtsp-client.c:
1290 client: Reply 400 if media cannot be constructed
1291 Reply 400 Bad Request instead of 503 Service Unavailable if media
1292 cannot be constructed in SETUP.
1293 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
1295 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
1297 * gst/rtsp-server/rtsp-client.c:
1298 client: Send setup reply once only
1299 If find_media() failed in handle_setup_request() two replies was sent.
1300 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
1302 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
1305 Automatic update of common submodule
1306 From 6b03ba7 to 865aa20
1308 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
1310 * gst/rtsp-server/rtsp-server.c:
1311 server: Emit client-connected signal earlier
1312 Emit client-connected before the client ref is given to a GSource,
1313 otherwise client-connected can be emitted after the client object has
1316 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
1318 * gst/rtsp-server/rtsp-address-pool.c:
1319 * gst/rtsp-server/rtsp-address-pool.h:
1320 * gst/rtsp-server/rtsp-stream.c:
1321 * tests/check/gst/addresspool.c:
1322 addresspool: return reason of failure
1323 Let gst_rtsp_address_pool_reserve_address() return the reason why
1324 the address could not be reserved.
1325 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
1327 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
1330 autogen.sh: Sync behaviour with other GStreamer modules
1331 Allows building from outside of tree amongst other things
1333 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
1336 Automatic update of common submodule
1337 From b613661 to 6b03ba7
1339 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
1342 Automatic update of common submodule
1343 From 74a6857 to b613661
1345 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
1348 Automatic update of common submodule
1349 From 01a7a46 to 74a6857
1351 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
1353 * gst/rtsp-server/rtsp-client.c:
1354 client: Do not read beyond end of path string
1355 If the setup was done without a control url, make sure we don't try to read the
1356 non-existing control string and crash.
1358 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1360 * gst/rtsp-server/rtsp-client.c:
1361 client: Fix RTPInfo header
1362 Refactor the method to make the content_base.
1363 Use the content-base and the control url to construct the RTPInfo
1366 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1368 * gst/rtsp-server/rtsp-client.c:
1369 client: map url to path only in describe
1370 Only map the request url to a path in the DESCRIBE method. The SDP then
1371 contains the base and control urls that should be used to SETUP/PAUSE/
1372 PLAY/TEARDOWN the media.
1374 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1376 * gst/rtsp-server/rtsp-client.c:
1377 Revert "client: map URL to path in requests"
1378 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
1379 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
1380 contains the base and control urls which are used in the SETUP, PLAY,
1381 PAUSE and TEARDOWN requests.
1383 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1385 * gst/rtsp-server/rtsp-client.c:
1386 client: map URL to path in requests
1388 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1390 * gst/rtsp-server/rtsp-client.c:
1391 * gst/rtsp-server/rtsp-mount-points.c:
1392 * gst/rtsp-server/rtsp-mount-points.h:
1393 mount-points: make vmethod to make path from uri
1394 Make a vmethod to transform an url into a path. The path is then used to lookup
1395 the factory. This makes it possible to also use other bits of the url, such as
1396 the query parameters, to locate the factory.
1398 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
1400 * gst/rtsp-server/rtsp-thread-pool.c:
1401 * gst/rtsp-server/rtsp-thread-pool.h:
1402 thread-pool: Add cleanup to wait for the threadpool to finish
1403 Also fix race condition if two threads are asking for the first
1404 thread from the thread pool at once. This would case two internal
1405 GThreadPools to be created.
1406 https://bugzilla.gnome.org/show_bug.cgi?id=707753
1408 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
1410 * gst/rtsp-server/rtsp-client.c:
1411 * tests/check/gst/client.c:
1412 client: free threadpool
1413 https://bugzilla.gnome.org/show_bug.cgi?id=707638
1415 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
1417 * tests/check/gst/mountpoints.c:
1418 mountpoints tests: unref matched factories
1419 https://bugzilla.gnome.org/show_bug.cgi?id=707638
1421 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
1423 * tests/check/gst/media.c:
1424 media tests: unref thread pool and caps
1425 https://bugzilla.gnome.org/show_bug.cgi?id=707638
1427 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
1429 * gst/rtsp-server/rtsp-auth.c:
1430 * gst/rtsp-server/rtsp-media-factory.c:
1431 * gst/rtsp-server/rtsp-media.c:
1432 auth, media, media-factory: unref permissions
1433 https://bugzilla.gnome.org/show_bug.cgi?id=707638
1435 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1437 * examples/Makefile.am:
1438 Makefile: add rule for appsrc example
1440 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1442 * examples/test-appsrc.c:
1443 tests: add appsrc example
1444 Add an example on how to use appsrc to feed the server pipeline with data.
1446 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
1448 * gst/rtsp-server/rtsp-client.c:
1449 rtsp-client: remove query part from content-base string
1450 Make sure that after the control url has been resolved, it's
1451 not a part of the query-string.
1452 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
1454 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1456 * gst/rtsp-server/rtsp-client.c:
1457 client: don't check url in response
1458 There is no url or method in the response to check
1460 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1462 * gst/rtsp-server/rtsp-client.c:
1463 * gst/rtsp-server/rtsp-client.h:
1464 Add handle-response signal for when we receive a GET_PARAMETER response
1466 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1468 * gst/rtsp-server/rtsp-server.c:
1469 Fix gst_rtsp_server_client_filter, using wrong variable type
1471 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
1473 * gst/rtsp-server/rtsp-media-factory-uri.c:
1474 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
1475 For AAC we need to check for framed=true instead of parsed=true.
1476 https://bugzilla.gnome.org/show_bug.cgi?id=701384
1478 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1480 * gst/rtsp-server/rtsp-stream.c:
1481 stream: optimize pipeline for protocols
1482 When TCP is not an allowed protocol for the stream, avoid creating the
1483 appsrc/appsink/queue and tee elements.
1485 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1487 * gst/rtsp-server/rtsp-media.c:
1488 media: set protocols on streams
1490 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1492 * gst/rtsp-server/rtsp-client.c:
1493 client: use protocols supported by stream
1495 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1497 * gst/rtsp-server/rtsp-media-factory.c:
1498 * gst/rtsp-server/rtsp-media.c:
1499 * gst/rtsp-server/rtsp-stream.c:
1500 media-factory: allow all protocols
1502 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1504 * gst/rtsp-server/rtsp-media.c:
1505 media: configure protocols in new streams
1507 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1509 * gst/rtsp-server/rtsp-stream.c:
1510 * gst/rtsp-server/rtsp-stream.h:
1511 stream: add protocols property
1513 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1515 * gst/rtsp-server/rtsp-media.c:
1516 rtsp-media: send state in "new-state" signal
1517 https://bugzilla.gnome.org/show_bug.cgi?id=705110
1519 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
1522 build: add subdir-objects to AM_INIT_AUTOMAKE
1523 Fixes warnings with automake 1.14
1524 https://bugzilla.gnome.org/show_bug.cgi?id=705350
1526 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1528 * docs/libs/gst-rtsp-server-sections.txt:
1529 * gst/rtsp-server/rtsp-client.c:
1530 * gst/rtsp-server/rtsp-server.c:
1531 * gst/rtsp-server/rtsp-server.h:
1532 server: add method to iterate clients of server
1534 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1536 * gst/rtsp-server/rtsp-media.c:
1537 * gst/rtsp-server/rtsp-media.h:
1538 Add vmethod for rtsp-media subclass to access rtpbin
1540 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1542 * gst/rtsp-server/rtsp-client.h:
1543 small documentation fix
1545 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1547 * gst/rtsp-server/rtsp-client.c:
1548 Do not take range header if range is invalid
1550 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1552 * docs/libs/gst-rtsp-server-sections.txt:
1553 * gst/rtsp-server/rtsp-media.c:
1554 media: add docs for new method
1556 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1558 * gst/rtsp-server/rtsp-media.c:
1559 * gst/rtsp-server/rtsp-media.h:
1560 Add API to rtsp-media set the pipeline's state
1562 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1564 * gst/rtsp-server/rtsp-media.c:
1565 Update current position/duration when gst_rtsp_media_get_range_string is called
1567 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1569 * examples/test-cgroups.c:
1570 tests: add some more docs
1572 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1574 * examples/test-cgroups.c:
1575 * gst/rtsp-server/Makefile.am:
1576 * gst/rtsp-server/rtsp-auth.c:
1577 * gst/rtsp-server/rtsp-auth.h:
1578 * gst/rtsp-server/rtsp-client.c:
1579 * gst/rtsp-server/rtsp-client.h:
1580 * gst/rtsp-server/rtsp-context.c:
1581 * gst/rtsp-server/rtsp-context.h:
1582 * gst/rtsp-server/rtsp-params.c:
1583 * gst/rtsp-server/rtsp-params.h:
1584 * gst/rtsp-server/rtsp-server.c:
1585 * gst/rtsp-server/rtsp-thread-pool.c:
1586 * gst/rtsp-server/rtsp-thread-pool.h:
1587 * tests/check/gst/client.c:
1588 ClientState -> Context
1589 Rename the clientstate to context and put the code in a separate file.
1591 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1593 * examples/test-auth.c:
1594 * gst/rtsp-server/rtsp-auth.c:
1595 * gst/rtsp-server/rtsp-auth.h:
1596 auth: add support for default token
1597 The default token is used when the user is not authenticated and can be used to
1598 give minimal permissions.
1600 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1602 * examples/test-auth.c:
1603 * gst/rtsp-server/rtsp-auth.c:
1604 auth: use defines when possible
1606 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1608 * gst/rtsp-server/rtsp-address-pool.c:
1609 address-pool: improve docs
1611 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1613 * gst/rtsp-server/rtsp-permissions.c:
1614 permissions: add the role to the copy
1616 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
1618 * gst/rtsp-server/rtsp-permissions.c:
1619 permissions: Also copy the roles
1621 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
1623 * gst/rtsp-server/rtsp-permissions.c:
1624 permissions: Make it build
1626 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1628 * gst/rtsp-server/rtsp-address-pool.h:
1631 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1633 * docs/libs/gst-rtsp-server-sections.txt:
1634 * gst/rtsp-server/rtsp-auth.c:
1635 * gst/rtsp-server/rtsp-auth.h:
1636 * gst/rtsp-server/rtsp-media.c:
1637 * gst/rtsp-server/rtsp-session-media.c:
1638 * gst/rtsp-server/rtsp-stream-transport.c:
1639 * gst/rtsp-server/rtsp-stream-transport.h:
1640 * gst/rtsp-server/rtsp-stream.c:
1641 * tests/check/gst/client.c:
1644 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1646 * docs/libs/gst-rtsp-server-sections.txt:
1647 * gst/rtsp-server/rtsp-address-pool.c:
1648 * gst/rtsp-server/rtsp-address-pool.h:
1649 * tests/check/gst/addresspool.c:
1650 * tests/check/gst/rtspserver.c:
1651 address-pool: cleanups
1652 Remove redundant method, improve docs.
1654 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1656 * docs/libs/gst-rtsp-server-sections.txt:
1657 * gst/rtsp-server/rtsp-auth.h:
1658 * gst/rtsp-server/rtsp-permissions.c:
1659 * gst/rtsp-server/rtsp-permissions.h:
1660 * gst/rtsp-server/rtsp-token.c:
1663 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1665 * gst/rtsp-server/rtsp-permissions.c:
1666 permissions: implement _remove_role
1668 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1670 * gst/rtsp-server/rtsp-permissions.c:
1671 permissions: update docs
1673 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1675 * tests/check/gst/client.c:
1676 tests: simplify tests
1677 Client settings are now disabled by default so we don't need an auth
1678 module to disable them.
1680 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1682 * gst/rtsp-server/rtsp-auth.c:
1683 auth: add default authorizations
1684 When no auth module is specified, use our table of defaults to look up the
1685 default value of the check instead of always allowing everything. This was
1686 we can disallow client settings by default.
1688 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1691 README: update readme
1693 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1695 * gst/rtsp-server/rtsp-thread-pool.c:
1696 * gst/rtsp-server/rtsp-thread-pool.h:
1697 thread-pool: add more docs
1699 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1701 * gst/rtsp-server/rtsp-thread-pool.c:
1702 * gst/rtsp-server/rtsp-thread-pool.h:
1703 thread-pool: fix race in thread reuse
1704 If we try to reuse a thread right after we made it stop, we end up using a
1705 stopped thread. Catch this case and only reuse threads that are not stopping.
1707 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1709 * gst/rtsp-server/rtsp-server.c:
1710 server: add small debug
1712 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1714 * tests/check/gst/client.c:
1716 Add some permissions to media so we can use the auth and enable
1719 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1721 * gst/rtsp-server/rtsp-client.c:
1722 client: support pushed context in handle_request
1723 If we already have a pushed state, reuse it and add our own things. This makes
1724 it easier to write tests.
1726 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1728 * gst/rtsp-server/rtsp-auth.c:
1729 auth: don't auth on methods
1730 Don't authorize on methods anymore but on the resources that we
1731 try to access, this is more flexible.
1732 Move the authorization checks to where they are needed and let the
1733 check return the response on error.
1735 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1737 * gst/rtsp-server/rtsp-mount-points.c:
1738 mount-points: add some debug
1740 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1742 * tests/check/gst/client.c:
1743 tests: almost fix test
1745 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1747 * gst/rtsp-server/rtsp-auth.c:
1748 * gst/rtsp-server/rtsp-auth.h:
1749 * gst/rtsp-server/rtsp-client.c:
1750 * gst/rtsp-server/rtsp-client.h:
1751 * gst/rtsp-server/rtsp-server.c:
1752 * gst/rtsp-server/rtsp-server.h:
1753 auth: let the auth module check client_settings
1754 Let the auth module decide if client settings are allowed for the
1757 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1759 * gst/rtsp-server/rtsp-token.c:
1760 * gst/rtsp-server/rtsp-token.h:
1761 token: add method to check boolean permission
1763 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1765 * examples/test-auth.c:
1766 * examples/test-cgroups.c:
1767 * gst/rtsp-server/rtsp-token.c:
1768 * gst/rtsp-server/rtsp-token.h:
1769 token: simplify token constructor
1770 Use variable arguments to make easier API.
1772 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1774 * examples/test-auth.c:
1775 * examples/test-cgroups.c:
1776 * gst/rtsp-server/rtsp-media-factory.c:
1777 * gst/rtsp-server/rtsp-media-factory.h:
1778 media-factory: add convenience API for factory
1780 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1782 * examples/test-auth.c:
1783 * examples/test-cgroups.c:
1784 * gst/rtsp-server/rtsp-permissions.c:
1785 * gst/rtsp-server/rtsp-permissions.h:
1786 permissions: simplify API a little
1787 Avoid passing GstStructure in the add_role method, use varargs instead
1788 to construct the structure behind the scenes. We can then also use the
1789 structure name as the role and simplify some more logic.
1791 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1793 * gst/rtsp-server/rtsp-auth.c:
1796 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1798 * gst/rtsp-server/rtsp-auth.c:
1799 * gst/rtsp-server/rtsp-auth.h:
1800 * gst/rtsp-server/rtsp-client.c:
1801 auth: handle unauthorized response
1802 Move handling of the unauthorized response to the auth module, it can add
1803 the appropriate headers to request authorization for the required method
1804 much better than the client.
1806 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1808 * gst/rtsp-server/rtsp-client.c:
1809 * gst/rtsp-server/rtsp-client.h:
1810 client: allow for sending any message, not only requests
1811 Change the _send_request() method to _send_message() so that we
1812 can both send requests and replies.
1814 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1816 * docs/libs/gst-rtsp-server-sections.txt:
1817 * gst/rtsp-server/rtsp-server.h:
1820 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1822 * examples/test-video.c:
1823 * gst/rtsp-server/rtsp-auth.c:
1824 * gst/rtsp-server/rtsp-auth.h:
1825 * gst/rtsp-server/rtsp-server.c:
1826 * gst/rtsp-server/rtsp-server.h:
1827 auth: move TLS handling to auth module
1828 Remove the TLS settings on the server and move it to the auth module because
1829 that is where security related bits go.
1831 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1833 * gst/rtsp-server/rtsp-client.c:
1834 * gst/rtsp-server/rtsp-client.h:
1835 client: add state push/pop
1837 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1839 * gst/rtsp-server/rtsp-client.c:
1840 * gst/rtsp-server/rtsp-client.h:
1841 client: add connection to state
1843 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1845 * gst/rtsp-server/rtsp-mount-points.c:
1846 mount-points: fix debug
1848 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1850 * tests/check/gst/media.c:
1851 tests: fix media test
1853 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1855 * gst/rtsp-server/rtsp-thread-pool.c:
1856 thread-pool: we don't require a state
1858 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1860 * gst/rtsp-server/rtsp-server.c:
1861 server: let context ref the server
1862 So that we don't risk losing the server object early anc crash.
1864 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1866 * tests/check/gst/client.c:
1867 tests: fix client test
1869 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1872 * docs/libs/gst-rtsp-server-docs.sgml:
1873 * docs/libs/gst-rtsp-server-sections.txt:
1874 * gst/rtsp-server/rtsp-address-pool.c:
1875 * gst/rtsp-server/rtsp-auth.c:
1876 * gst/rtsp-server/rtsp-client.c:
1877 * gst/rtsp-server/rtsp-client.h:
1878 * gst/rtsp-server/rtsp-media-factory-uri.c:
1879 * gst/rtsp-server/rtsp-media-factory.c:
1880 * gst/rtsp-server/rtsp-media-factory.h:
1881 * gst/rtsp-server/rtsp-media.c:
1882 * gst/rtsp-server/rtsp-mount-points.c:
1883 * gst/rtsp-server/rtsp-params.c:
1884 * gst/rtsp-server/rtsp-permissions.c:
1885 * gst/rtsp-server/rtsp-sdp.c:
1886 * gst/rtsp-server/rtsp-server.c:
1887 * gst/rtsp-server/rtsp-server.h:
1888 * gst/rtsp-server/rtsp-session-media.c:
1889 * gst/rtsp-server/rtsp-session-pool.c:
1890 * gst/rtsp-server/rtsp-session.c:
1891 * gst/rtsp-server/rtsp-stream-transport.c:
1892 * gst/rtsp-server/rtsp-stream.c:
1893 * gst/rtsp-server/rtsp-thread-pool.c:
1894 * gst/rtsp-server/rtsp-token.c:
1897 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1899 * gst/rtsp-server/rtsp-session-pool.c:
1900 * gst/rtsp-server/rtsp-session-pool.h:
1901 session-pool: make vmethod to create a session
1902 Make a vmethod to create a sessions so that subclasses can create
1903 custom session objects
1905 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1907 * gst/rtsp-server/rtsp-auth.c:
1908 * gst/rtsp-server/rtsp-media-factory.h:
1909 * gst/rtsp-server/rtsp-media.h:
1910 * gst/rtsp-server/rtsp-mount-points.h:
1911 * gst/rtsp-server/rtsp-session-pool.h:
1912 * gst/rtsp-server/rtsp-stream.h:
1915 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1917 * docs/libs/gst-rtsp-server-docs.sgml:
1918 * docs/libs/gst-rtsp-server-sections.txt:
1919 * gst/rtsp-server/rtsp-address-pool.c:
1920 * gst/rtsp-server/rtsp-address-pool.h:
1921 * gst/rtsp-server/rtsp-auth.c:
1922 * gst/rtsp-server/rtsp-client.h:
1923 * gst/rtsp-server/rtsp-media-factory.h:
1924 * gst/rtsp-server/rtsp-media.c:
1925 * gst/rtsp-server/rtsp-media.h:
1926 * gst/rtsp-server/rtsp-permissions.c:
1927 * gst/rtsp-server/rtsp-permissions.h:
1928 * gst/rtsp-server/rtsp-server.h:
1929 * gst/rtsp-server/rtsp-session-media.c:
1930 * gst/rtsp-server/rtsp-session-media.h:
1931 * gst/rtsp-server/rtsp-session-pool.h:
1932 * gst/rtsp-server/rtsp-session.h:
1933 * gst/rtsp-server/rtsp-stream-transport.h:
1934 * gst/rtsp-server/rtsp-stream.c:
1935 * gst/rtsp-server/rtsp-thread-pool.h:
1938 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1941 * examples/Makefile.am:
1942 configure: compile cgroup example conditionally
1943 Only compile the cgroup example when we have libcgroup
1945 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1948 * examples/Makefile.am:
1949 * examples/test-cgroups.c:
1950 examples: add cgroups example
1952 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1954 * tests/check/gst/rtspserver.c:
1955 tests: fix compilation
1957 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1959 * gst/rtsp-server/rtsp-thread-pool.c:
1960 thread-pool: fix vmethod invocation
1962 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1964 * gst/rtsp-server/rtsp-thread-pool.c:
1965 * gst/rtsp-server/rtsp-thread-pool.h:
1966 thread-pool: store thread type in thread
1968 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1970 * gst/rtsp-server/rtsp-client.c:
1971 client: pass thread from pool to media _prepare
1972 Get a thread from the configured threadpool and pass it to the prepare method of
1975 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1977 * gst/rtsp-server/rtsp-media.c:
1978 * gst/rtsp-server/rtsp-media.h:
1979 media: Accept a thread in _prepare
1980 Remove out own threadpool handling and use the provided thread and
1981 maincontext for the bus messages and the state changes.
1983 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1985 * gst/rtsp-server/rtsp-server.c:
1986 server: configure client thread pool
1988 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1990 * gst/rtsp-server/rtsp-client.c:
1991 * gst/rtsp-server/rtsp-client.h:
1992 client: add method to configure thread pool
1994 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1996 * gst/rtsp-server/rtsp-client.h:
1997 * gst/rtsp-server/rtsp-server.c:
1998 * gst/rtsp-server/rtsp-server.h:
1999 server: use thread pool
2000 Use the thread pool instead of doing our own thing.
2002 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2004 * gst/rtsp-server/Makefile.am:
2005 * gst/rtsp-server/rtsp-thread-pool.c:
2006 * gst/rtsp-server/rtsp-thread-pool.h:
2007 thread-pool: add object to manage threads
2008 Add an object to manage the client and media threads.
2010 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2012 * gst/rtsp-server/rtsp-auth.c:
2013 auth: debug authorization check
2015 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2017 * gst/rtsp-server/rtsp-media.c:
2018 media: start media pipeline in context
2019 Start the media pipeline in the provided context (or our default one
2020 when NULL). This makes sure that we run the bus thread in this context and that
2021 all media threads are children of this context.
2023 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2025 * gst/rtsp-server/rtsp-media-factory.c:
2026 factory: pass permissions to media by default
2028 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2030 * examples/test-auth.c:
2031 test: add permissions to auth test
2032 Ass some permissions to the media factory in the test.
2034 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2036 * gst/rtsp-server/rtsp-auth.c:
2037 * gst/rtsp-server/rtsp-auth.h:
2038 * gst/rtsp-server/rtsp-client.c:
2039 auth: simplify auth checks
2040 Remove client from methods, it's now in the state
2041 Perform the check specified by the string, use the information from the
2042 thread local context.
2044 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2046 * gst/rtsp-server/rtsp-client.c:
2047 * gst/rtsp-server/rtsp-client.h:
2048 client: add state to current thread
2049 Add the client to the ClientState object.
2050 Place the ClientState on the current thread.
2052 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2054 * gst/rtsp-server/rtsp-media-factory.c:
2055 * gst/rtsp-server/rtsp-media-factory.h:
2056 * gst/rtsp-server/rtsp-media.c:
2057 * gst/rtsp-server/rtsp-media.h:
2058 media: make it possible to set permissions
2059 Make it possible to set permissions on media and media factory objects
2061 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2063 * gst/rtsp-server/Makefile.am:
2064 * gst/rtsp-server/rtsp-permissions.c:
2065 * gst/rtsp-server/rtsp-permissions.h:
2066 permissions: add permissions object
2067 Add a mini object to store permissions based on a role.
2069 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2071 * examples/test-auth.c:
2072 * gst/rtsp-server/rtsp-auth.c:
2073 * gst/rtsp-server/rtsp-auth.h:
2074 * gst/rtsp-server/rtsp-client.c:
2075 auth: add auth checks
2076 Add an enum with auth checks and implement the checks in the auth object.
2077 Perform the checks from the client.
2079 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2081 * examples/test-auth.c:
2082 * gst/rtsp-server/rtsp-auth.c:
2083 * gst/rtsp-server/rtsp-auth.h:
2084 * gst/rtsp-server/rtsp-client.h:
2085 auth: use the token after authentication
2086 After we authenticated a user, keep the Token around in the state.
2088 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2090 * gst/rtsp-server/rtsp-client.c:
2091 * gst/rtsp-server/rtsp-media.c:
2092 * gst/rtsp-server/rtsp-media.h:
2093 * tests/check/gst/media.c:
2094 media: add optional context for bus messages
2095 Add an optional mainloop to _prepare that will handle the bus messages instead
2096 of always using the shared mainloop.
2098 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2100 * gst/rtsp-server/Makefile.am:
2101 * gst/rtsp-server/rtsp-token.c:
2102 * gst/rtsp-server/rtsp-token.h:
2103 token: add authorization token
2104 Add a simply miniobject that contains the authorizations. The object contains a
2105 GstStructure that hold all authorization fields. When a user is authenticated,
2106 the auth module will create a Token for the user. The token is then used to
2107 check what operations the user is allowed to do and various other configuration
2110 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2112 * examples/test-auth.c:
2113 * gst/rtsp-server/rtsp-auth.c:
2114 * gst/rtsp-server/rtsp-auth.h:
2115 * gst/rtsp-server/rtsp-client.c:
2116 * gst/rtsp-server/rtsp-client.h:
2117 * gst/rtsp-server/rtsp-media-factory.c:
2118 * gst/rtsp-server/rtsp-media-factory.h:
2119 * gst/rtsp-server/rtsp-media.c:
2120 * gst/rtsp-server/rtsp-media.h:
2121 auth: remove auth from media and factory
2122 Remove the auth object from media and factory. We want to have the RTSPClient
2123 authenticate and authorize resources, there is no need to place another auth
2124 manager on the media/factory.
2126 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2128 * examples/test-auth.c:
2129 * gst/rtsp-server/rtsp-auth.c:
2130 * gst/rtsp-server/rtsp-auth.h:
2131 * gst/rtsp-server/rtsp-client.h:
2132 auth: add support for multiple basic auth tokens
2133 Make it possible to add multiple basic authorisation tokens to one authorization
2134 object. Associate with each token an authorization group that will define what
2135 capabilities are allowed.
2137 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2139 * gst/rtsp-server/rtsp-client.c:
2140 client: error out on non-aggregate control
2141 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
2143 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2145 * gst/rtsp-server/rtsp-client.c:
2146 client: rework setup request a little
2147 Cache the media in DESCRIBE based on the longest matching path with the uri
2148 that we can find in the mount points.
2149 Rework the setup request a little to get the media from the session or from
2150 the longest matching path, this way we can derive the control string as
2151 everything after the path instead of hardcoding it.
2152 Find the stream based on the control string and only open a session when all
2155 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2157 * gst/rtsp-server/rtsp-media.c:
2158 * gst/rtsp-server/rtsp-media.h:
2159 media: add method to find a stream by control url
2161 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2163 * gst/rtsp-server/rtsp-stream.c:
2164 * gst/rtsp-server/rtsp-stream.h:
2165 stream: add method to check control url of stream
2167 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2169 * gst/rtsp-server/rtsp-client.c:
2170 * gst/rtsp-server/rtsp-session-media.c:
2171 * gst/rtsp-server/rtsp-session-media.h:
2172 * gst/rtsp-server/rtsp-session.c:
2173 * gst/rtsp-server/rtsp-session.h:
2174 session: use path matching for session media
2175 Use a path string instead of a uri to lookup session media in the sessions. Also
2176 use path matching to find the largest possible path that matches.
2178 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2180 * gst/rtsp-server/rtsp-client.c:
2181 * gst/rtsp-server/rtsp-mount-points.c:
2182 * gst/rtsp-server/rtsp-mount-points.h:
2183 * tests/check/gst/mountpoints.c:
2184 mount-points: remove useless vmethod
2185 Making lookups in the mount points should not be done with a URL, if there is a
2186 mapping to be done from URL to mount points, we'll need to do it somewhere
2189 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2191 * gst/rtsp-server/rtsp-mount-points.c:
2192 * gst/rtsp-server/rtsp-mount-points.h:
2193 * tests/check/gst/mountpoints.c:
2194 mount-points: improve mount point searching
2195 Use a GSequence to keep track of the mount points.
2196 Match a URL to the longest matching registered mount point. This should be the
2197 URL to perform aggreagate control and the remainder is the stream specific
2199 Add some unit tests for this.
2201 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
2203 * gst/rtsp-server/Makefile.am:
2204 rtsp-server: Allow building of static library
2206 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2208 * tests/check/gst/mediafactory.c:
2209 tests: fix compilation
2211 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2213 * gst/rtsp-server/rtsp-sdp.c:
2214 sdp: get control string from stream
2215 Use the control string as configured in the stream.
2217 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2219 * gst/rtsp-server/rtsp-stream.c:
2220 * gst/rtsp-server/rtsp-stream.h:
2221 stream: add methods and property to set control string
2223 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2225 * gst/rtsp-server/rtsp-client.c:
2227 Rename variables for clarity
2228 Keep media in state when we can
2230 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2232 * gst/rtsp-server/rtsp-client.c:
2233 * gst/rtsp-server/rtsp-stream.c:
2234 * gst/rtsp-server/rtsp-stream.h:
2235 stream: add more support for IPv6
2236 Rename _get_address to _get_multicast_address in GstRTSPStream to
2237 make it clear that this function only deals with multicast.
2238 Make it possible to have both an IPv4 and IPv6 multicast address on
2239 a stream. Give the client an IPv4 or IPv6 address depending on the
2240 address it used to connect to the server.
2241 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
2243 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2245 * gst/rtsp-server/rtsp-client.c:
2248 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2250 * gst/rtsp-server/rtsp-stream.c:
2251 stream: handle failed port allocation
2252 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
2253 can't allocate any family at all. Also keep track of what port families we
2255 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
2257 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2259 * gst/rtsp-server/rtsp-stream.c:
2260 stream: improve docs
2262 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2264 * gst/rtsp-server/rtsp-stream-transport.c:
2265 stream-transport: remove old if 0 block
2267 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
2269 * tests/check/gst/client.c:
2271 gst_rtsp_client_get_uri() has been removed
2272 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
2274 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2276 * gst/rtsp-server/rtsp-client.c:
2277 * gst/rtsp-server/rtsp-client.h:
2278 client: add method to filter managed sessions
2279 Add a method to filter the sessions managed by this client connection.
2280 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
2282 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2284 * gst/rtsp-server/rtsp-client.c:
2285 * gst/rtsp-server/rtsp-client.h:
2286 client: remove _get_uri() method
2287 Remove the get_uri() method on the client. A client has no uri, the uri
2288 property is an internal property to manage the last cached media for
2291 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2293 * gst/rtsp-server/rtsp-media-factory.h:
2294 media-factory: fix typo
2296 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
2298 * gst/rtsp-server/rtsp-media.c:
2299 rtsp-media: Do not leak the query in default_query_stop
2300 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
2302 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2304 * gst/rtsp-server/rtsp-media.c:
2305 media: don't unlock when conversion fails
2306 Don't unlock the state lock when conversion fails because it was not locked.
2308 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2310 * gst/rtsp-server/rtsp-media.c:
2311 * gst/rtsp-server/rtsp-media.h:
2312 Add query_position and query_stop vmethods to rtsp-media
2314 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2316 * gst/rtsp-server/rtsp-media.c:
2317 Fix typo in property install for rtsp-media's time-provider
2319 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2321 * gst/rtsp-server/rtsp-client.c:
2322 * gst/rtsp-server/rtsp-client.h:
2323 client: clean some variables
2324 Clean some variables and add some guards to _send_request()
2326 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2328 * gst/rtsp-server/rtsp-client.c:
2329 * gst/rtsp-server/rtsp-client.h:
2330 Add gst_rtsp_client_send_request API
2331 This makes it possible to send arbitrary messages to a client, such as
2332 SET_PARAMETER or GET_PARAMETER
2334 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2336 * gst/rtsp-server/rtsp-media.c:
2337 * gst/rtsp-server/rtsp-media.h:
2338 media: add _get_element() method
2339 Add method to get the element used when creating the media.
2340 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
2342 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2344 * gst/rtsp-server/rtsp-media.c:
2347 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
2349 * gst/rtsp-server/rtsp-stream.c:
2350 * gst/rtsp-server/rtsp-stream.h:
2351 stream: allow access to the rtp session
2352 https://bugzilla.gnome.org/show_bug.cgi?id=703004
2354 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
2356 * gst/rtsp-server/rtsp-stream.c:
2357 * gst/rtsp-server/rtsp-stream.h:
2358 dscp qos support in gst-rtsp-stream
2359 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
2361 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2363 * tests/check/gst/rtspserver.c:
2365 Actually do what the comment says. Also keep the old code around, not sure what
2366 should happen when you get a 454 from a TEARDOWN, does it close the connection?
2367 it currently doesn't.
2369 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2371 * gst/rtsp-server/rtsp-client.c:
2372 client: also watch newly created session
2373 When we newly created a session, start watching it immediately instead of
2374 on the next request.
2376 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
2378 * tests/check/gst/client.c:
2379 tests: add unit test for new-session
2380 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
2382 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2384 * gst/rtsp-server/rtsp-client.c:
2385 client: emit new-session when new session is created
2386 Only emit new-session when we created a new session for a client, not when a
2387 client picked up a previous session.
2388 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
2390 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
2392 * gst/rtsp-server/rtsp-client.c:
2393 client: handle asterisk as path in requests
2394 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
2396 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2398 * gst/rtsp-server/rtsp-media.c:
2399 media: handle segment query format mismatch
2400 It's possible that the segment query returns with a different format than what
2401 we asked for, handle this case also.
2403 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
2405 * gst/rtsp-server/rtsp-media.c:
2406 media: use segment stop in collect_media_stats
2407 Use segment stop instead of duration as range end point.
2408 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
2410 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
2412 * gst/rtsp-server/rtsp-media.c:
2413 * tests/check/gst/media.c:
2414 rtsp-media: Do not leak the element in take_pipeline
2415 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
2417 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
2419 * gst/rtsp-server/rtsp-client.c:
2420 * gst/rtsp-server/rtsp-client.h:
2421 rtsp-client: Make configure_client_transport virtual
2422 This patch makes configure_client_transport virtual. The functionality is
2423 needed to handle some weird clients sending multicast transport settings as url
2425 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
2427 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
2429 * gst/rtsp-server/rtsp-client.c:
2430 * gst/rtsp-server/rtsp-client.h:
2431 rtsp-client: Make param_set and param_get virtual
2432 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
2434 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
2436 * gst/rtsp-server/rtsp-client.c:
2437 * gst/rtsp-server/rtsp-media.c:
2438 * gst/rtsp-server/rtsp-media.h:
2439 media: convert_range replaces get_range_times
2440 get_range_times worked for handling UTC ranges for seeks, but we also
2441 need to convert back from NPT to the requested unit in
2442 get_range_string. convert_range is now used for both.
2443 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
2445 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2447 * gst/rtsp-server/rtsp-client.c:
2448 * gst/rtsp-server/rtsp-sdp.c:
2449 * gst/rtsp-server/rtsp-sdp.h:
2450 sdp: cleanup sdp info
2451 We don't need to pass the proto, we can more easily check a boolean.
2452 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
2454 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
2456 * gst/rtsp-server/rtsp-sdp.c:
2457 use 0.0.0.0 or :: for c= line instead of server address
2459 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
2461 * gst/rtsp-server/rtsp-client.c:
2462 use local address, not remote, in SDP
2463 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
2465 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2468 Automatic update of common submodule
2469 From 098c0d7 to 01a7a46
2471 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
2473 * gst/rtsp-server/rtsp-media.c:
2474 * gst/rtsp-server/rtsp-media.h:
2475 media: possibility to override range time conversion
2476 Make it possible to override the conversion from GstRTSPTimeRange to
2477 GstClockTimes, that is done before seeking on the media
2478 pipeline. Overriding can be useful for UTC ranges, where the default
2479 conversion gives nanoseconds since 1900.
2480 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
2482 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
2484 * gst/rtsp-server/rtsp-server.c:
2485 * gst/rtsp-server/rtsp-server.h:
2486 rtsp-server: Expose the use_client_settings API
2487 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
2489 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
2491 * gst/rtsp-server/rtsp-client.c:
2492 * gst/rtsp-server/rtsp-stream.c:
2493 * gst/rtsp-server/rtsp-stream.h:
2494 rtspstream: handle both ipv4 and ipv6 clients
2495 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
2497 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2499 * gst/rtsp-server/rtsp-sdp.c:
2500 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
2501 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
2502 We already have a way to place extra attributes in the SDP by using a string
2503 property with prefix x- or a- in the caps.
2505 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2507 * gst/rtsp-server/rtsp-sdp.c:
2508 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
2509 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
2510 We already have a way to place extra attributes in the SDP, just make a string
2511 property in the payloader with a- or x- prefix.
2513 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2515 * gst/rtsp-server/rtsp-sdp.c:
2516 rtsp: place a- and x- properties as attributes
2517 application/x-rtp has properties with a- and x- prefixes that should be
2518 placed as attributes in the SDP for the media instead of being added to the
2521 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2523 * examples/Makefile.am:
2524 * examples/test-video.c:
2525 example: add TLS example
2527 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2529 * gst/rtsp-server/rtsp-server.c:
2530 * gst/rtsp-server/rtsp-server.h:
2531 server: add support for TLS
2532 Add methods to set and get a TLS certificate.
2533 Add vmethod to configure a new connection. By default, configure the TLS
2534 certificate in a new connection if needed.
2536 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2538 * gst/rtsp-server/rtsp-server.c:
2539 * gst/rtsp-server/rtsp-server.h:
2540 server: remove accept_client vmethod
2541 This vmethod is not very useful so remove it.
2543 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2545 * gst/rtsp-server/rtsp-server.c:
2546 server: don't crash on NULL GError
2548 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
2550 * gst/rtsp-server/rtsp-session-pool.c:
2551 rtsp-session-pool: corrected session timeout detection
2552 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
2554 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2556 * gst/rtsp-server/rtsp-client.c:
2557 client: improve debug
2559 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2561 * gst/rtsp-server/rtsp-client.c:
2562 * gst/rtsp-server/rtsp-client.h:
2563 * gst/rtsp-server/rtsp-server.c:
2564 server: refactor connection setup
2565 Let the server accept the socket connection and construct a GstRTSPConnection
2566 from it. Remove the code from the client and let the client only deal with
2567 a fully configure GstRTSPConnection object.
2568 We will need this later when the server will configure the connection for
2571 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2573 * gst/rtsp-server/rtsp-stream.c:
2574 stream: keep the transport object alive
2575 Keep the transport object alive while we have it as qdata on the
2578 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
2580 * gst/rtsp-server/rtsp-client.c:
2581 * gst/rtsp-server/rtsp-server.c:
2582 rtsp-server: Do not crash on nmapping of server
2583 * generate error when gst_rtsp_connection_accept fails
2584 * do not stop accepting incoming connections because
2585 accepting a client fails
2586 https://bugzilla.gnome.org/show_bug.cgi?id=701072
2588 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
2590 * gst/rtsp-server/rtsp-client.c:
2591 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
2592 https://bugzilla.gnome.org/show_bug.cgi?id=700953
2594 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
2596 * gst/rtsp-server/rtsp-sdp.c:
2597 rtsp-sdp: Parse framerate caps field and set SDP attribute
2598 The SDP attribute and its format is described in RFC4566.
2599 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
2601 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
2603 * gst/rtsp-server/rtsp-sdp.c:
2604 rtsp-sdp: Parse width/height from caps and set SDP attribute
2605 The SDP attribute and its format is described in RFC6064.
2606 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
2608 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
2610 * gst/rtsp-server/rtsp-sdp.c:
2611 * tests/check/gst/client.c:
2612 rtsp-sdp: add bandwidth line
2613 https://bugzilla.gnome.org/show_bug.cgi?id=699220
2615 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2618 Automatic update of common submodule
2619 From 5edcd85 to 098c0d7
2621 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
2623 * tests/check/gst/media.c:
2624 tests: add dynamic payloader prepare/unprepare check
2626 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2628 * gst/rtsp-server/rtsp-media.c:
2629 media: release lock when removing fakesink
2631 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2633 * gst/rtsp-server/rtsp-stream.c:
2634 stream: set elements to NULL before removing
2635 When removing a stream, set the elements to NULL first. This avoids
2636 element-is-not-in-NULL-state errors when we dispose the elements.
2638 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
2641 Automatic update of common submodule
2642 From 3cb3d3c to 5edcd85
2644 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2646 * gst/rtsp-server/rtsp-media.c:
2647 * gst/rtsp-server/rtsp-media.h:
2648 media: listen to pad-removed signals
2649 Listen to the pad-removed signal and remove the stream associated with the
2651 Add signal to be notified of the removed pad.
2652 Remove the fakesink in unprepare()
2653 Fix signatures of the signal methods
2655 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2657 * examples/test-sdp.c:
2658 tests: add example of reusable pipelines
2660 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
2662 * gst/rtsp-server/rtsp-stream.c:
2663 * gst/rtsp-server/rtsp-stream.h:
2664 stream: add method to get the srcpad
2666 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
2668 * tests/check/gst/media.c:
2669 check: add media prepare/unprepare test
2670 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2672 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
2674 * gst/rtsp-server/rtsp-media.c:
2675 media: disconnect from signal handlers in unprepare()
2676 We connected to the pad-added and no-more-pads signals in prepare() so
2677 we need to disconnect from them in unprepare().
2678 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2680 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
2682 * gst/rtsp-server/rtsp-media.c:
2683 media: don't free streams array
2684 Don't free the streams array in the unprepare() method, they were not
2686 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2688 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
2690 * gst/rtsp-server/rtsp-media.c:
2691 media: don't unref the pipeline in unprepare
2692 Unprepare() should undo what prepare() does. Because the pipeline is
2693 not created in prepare(), we should not unref it in unprepare()
2695 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
2697 * gst/rtsp-server/rtsp-stream.c:
2698 stream: clear session and caps for reuse
2699 Set the session and caps to NULL after unref otherwise we might unref
2701 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2703 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
2705 * gst/rtsp-server/rtsp-client.c:
2706 client: send out teardown signal before tearing down
2707 The advantage is that in the signal handler you get direct access to
2708 information about what streams are about to get torn down (in the
2709 GstRTSPClientState).
2710 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
2712 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
2714 * gst/rtsp-server/rtsp-client.c:
2715 * gst/rtsp-server/rtsp-client.h:
2716 client: expose connection
2717 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
2719 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
2722 Automatic update of common submodule
2723 From aed87ae to 3cb3d3c
2725 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2727 * gst/rtsp-server/rtsp-media.c:
2728 * gst/rtsp-server/rtsp-media.h:
2729 * gst/rtsp-server/rtsp-session-media.c:
2730 * gst/rtsp-server/rtsp-session-media.h:
2731 media: add method to get the base_time of the pipeline
2732 Together with a shared clock, this base-time could eventually be sent to
2733 the client so that it can reconstruct the exact running-time of the clock
2736 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2738 * gst/rtsp-server/Makefile.am:
2739 * gst/rtsp-server/rtsp-media.c:
2740 * gst/rtsp-server/rtsp-media.h:
2741 * gst/rtsp-server/rtsp-sdp.c:
2742 media: add GstNetTimeProvider support
2743 Add a property to let the media provide a GstNetTimeProvider for its clock.
2744 Make methods to get the clock and nettimeprovider
2745 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
2746 provider and also the current time of the clock. This should make it possible
2747 for (GStreamer) clients to slave their clock to the server clock.
2749 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
2752 Automatic update of common submodule
2753 From 04c7a1e to aed87ae
2755 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2757 * gst/rtsp-server/rtsp-media.c:
2758 media: wait for buffering to complete
2759 Wait for buffering to complete before changing the state to the target state.
2761 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2763 * gst/rtsp-server/rtsp-media.c:
2764 media: small cleanup
2766 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
2768 * tests/check/gst/rtspserver.c:
2769 tests: remove extra unref in test_setup_non_existing_stream
2770 The unref is not needed anymore, teardown runs without it.
2771 https://bugzilla.gnome.org/show_bug.cgi?id=696542
2773 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
2775 * tests/check/gst/rtspserver.c:
2776 tests: GSocketService cleanup in test_bind_already_in_use
2777 Use g_socket_service_stop so the rtspserver test stops listening for
2778 incoming connections in test_bind_already_in_use.
2779 https://bugzilla.gnome.org/show_bug.cgi?id=696541
2781 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
2783 * gst/rtsp-server/rtsp-media-factory.c:
2784 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
2785 Instead use a GWeakRef which is safe to use
2786 This is a known GLib bug, see:
2787 https://bugzilla.gnome.org/show_bug.cgi?id=667145
2789 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
2791 * gst/rtsp-server/rtsp-client.c:
2792 * gst/rtsp-server/rtsp-media.c:
2793 * gst/rtsp-server/rtsp-media.h:
2794 * gst/rtsp-server/rtsp-sdp.c:
2795 * tests/check/gst/media.c:
2796 * tests/check/gst/rtspserver.c:
2797 rtsp-media/client: Reply to PLAY request with same type of Range
2798 Remember the type of Range from the PLAY request and use the same type for
2801 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
2803 * gst/rtsp-server/rtsp-client.c:
2804 * gst/rtsp-server/rtsp-client.h:
2805 * tests/check/gst/client.c:
2806 rtsp-client: expose uri
2808 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
2810 * tests/check/gst/mediafactory.c:
2811 tests: Hold ref while creating second media
2812 To test if the media aren't shared, make sure we keep the first one while creating a second
2813 otherwise the same memory address may be reused.
2815 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
2818 configure: remove out-of-date comment
2820 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
2823 .gitignore: ignore more build files
2825 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
2827 * tests/check/Makefile.am:
2828 tests: use right _LIBS variable for gst-plugins-base libs
2830 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2832 * tests/check/Makefile.am:
2833 check: add librtp to libs
2835 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
2837 * tests/check/gst/rtspserver.c:
2838 tests: Add test to check selecting a port the server will send from
2840 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
2842 * tests/check/gst/rtspserver.c:
2843 tests: Make sure packets are actually received
2845 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
2847 * gst/rtsp-server/rtsp-stream.c:
2848 stream: Select unicast address from pool if appropriate
2850 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
2852 * gst/rtsp-server/rtsp-stream.c:
2853 stream: Properties are always there in Gst 1.0
2855 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
2857 * tests/check/gst/addresspool.c:
2858 tests: Add tests for unicast addresses in pool
2860 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
2862 * gst/rtsp-server/rtsp-address-pool.c:
2863 * tests/check/gst/addresspool.c:
2864 address-pool: Verify that multicast addresses are used for multicast and vice-versa
2866 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
2868 * docs/libs/gst-rtsp-server-sections.txt:
2869 * gst/rtsp-server/rtsp-address-pool.c:
2870 * gst/rtsp-server/rtsp-address-pool.h:
2871 * gst/rtsp-server/rtsp-stream.c:
2872 * tests/check/gst/addresspool.c:
2873 address-pool: Add unicast addresses
2875 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
2878 * gst/rtsp-server/rtsp-server.c:
2879 * tests/check/gst/rtspserver.c:
2880 rtsp-server: Limit the number of threads per server instance
2881 If we exceed the maximum, just round robin the clients over the existing
2884 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
2886 * gst/rtsp-server/rtsp-server.c:
2887 rtsp-server: No need to store the GMainContext in the client context
2889 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
2891 * tests/check/gst/rtspserver.c:
2892 tests: Add test for client disconnection
2894 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
2896 * tests/check/gst/rtspserver.c:
2897 tests: Test client and session timeouts with multiple threads
2899 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
2901 * gst/rtsp-server/rtsp-address-pool.c:
2902 * gst/rtsp-server/rtsp-auth.c:
2903 * gst/rtsp-server/rtsp-client.c:
2904 * gst/rtsp-server/rtsp-media-factory-uri.c:
2905 * gst/rtsp-server/rtsp-media-factory.c:
2906 * gst/rtsp-server/rtsp-media.c:
2907 * gst/rtsp-server/rtsp-mount-points.c:
2908 * gst/rtsp-server/rtsp-server.c:
2909 * gst/rtsp-server/rtsp-session-media.c:
2910 * gst/rtsp-server/rtsp-session-pool.c:
2911 * gst/rtsp-server/rtsp-session.c:
2912 Document locking and its order
2914 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
2916 * tests/check/gst/rtspserver.c:
2917 tests: Test that slow DESCRIBE don't block other clients
2919 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
2921 * tests/check/gst/client.c:
2922 tests: Add tests for client-requested multicast address
2924 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
2926 * docs/libs/gst-rtsp-server-sections.txt:
2927 docs: Put the various functions in the right sections
2929 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
2931 * docs/libs/gst-rtsp-server-docs.sgml:
2932 * docs/libs/gst-rtsp-server-sections.txt:
2933 * gst/rtsp-server/rtsp-address-pool.c:
2934 * gst/rtsp-server/rtsp-address-pool.h:
2935 docs: Generate docs for GstRTSPAddressPool
2937 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
2939 * gst/rtsp-server/rtsp-client.c:
2940 * gst/rtsp-server/rtsp-stream.c:
2941 * gst/rtsp-server/rtsp-stream.h:
2942 client: Check client provided addresses against the address pool
2944 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
2946 * gst/rtsp-server/rtsp-address-pool.c:
2947 * gst/rtsp-server/rtsp-address-pool.h:
2948 * tests/check/gst/addresspool.c:
2949 address-pool: Add API to request a specific address from the pool
2950 Also add relevant unit tests.
2952 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
2954 * tests/check/gst/mediafactory.c:
2955 tests: Check the passing around of a RTSPAddressPool
2956 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
2957 way down to the stream.
2959 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
2961 * tests/check/gst/addresspool.c:
2962 tests: Add more tests for the address pool
2964 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
2966 * gst/rtsp-server/rtsp-address-pool.c:
2967 address-pool: Fix off by one error
2968 When splitting a port range, the port after a skip is not part of range.
2970 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
2973 Automatic update of common submodule
2974 From 2de221c to 04c7a1e
2976 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
2979 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
2980 AM_CONFIG_HEADER was removed in automake 1.13
2981 https://bugzilla.gnome.org/show_bug.cgi?id=693368
2983 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
2986 Automatic update of common submodule
2987 From a942293 to 2de221c
2989 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2991 * gst/rtsp-server/rtsp-client.c:
2992 client: make sure the watch exists while sending data
2993 Protect the send_func with a lock. This allows us to wait for sending
2994 to complete before changing the send_func and user_data. We add an
2995 extra ref to the watch to make sure that it remains valid during
2997 When closing the connection, set the send_func to NULL
2998 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
3000 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3002 * tests/check/Makefile.am:
3003 tests: use GST_*_1_0 environment variables everywhere
3004 The _1_0 suffixed environment variables override the
3005 non-suffixed ones, so if we're in an environment that
3006 sets the _1_0 suffixed ones, such as jhbuild, we need
3007 to set those to make sure ours actually always get
3010 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3013 Automatic update of common submodule
3014 From acb04d9 to a942293
3016 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3018 * gst/rtsp-server/rtsp-client.c:
3019 rtsp-client: set the client backlog
3020 Set the client backlog to a reasonable default
3022 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
3024 * gst/rtsp-server/rtsp-media.c:
3025 rtsp-media: Make the element a constructor parameter
3026 https://bugzilla.gnome.org/show_bug.cgi?id=689594
3028 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3030 * docs/libs/Makefile.am:
3031 docs: Link with gcov library when gcov is enabled
3032 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
3034 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3036 * gst/rtsp-server/rtsp-media.c:
3037 media: match prepare with unprepare
3038 Really unprepare when there were an equal amount of prepare calls.
3040 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3042 * gst/rtsp-server/rtsp-media.c:
3043 media: media has to be unprepared in finalize
3044 Because unprepare takes away the last ref on the media.
3046 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3048 * gst/rtsp-server/rtsp-client.c:
3049 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
3050 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
3051 We can't use the refcount to trigger unprepare because it is the unprepare call
3052 that removes the last refcount after all messages are consumed. What we should
3053 probably do is make a prepared refcount and only unprepare when the refcount
3056 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3058 * gst/rtsp-server/rtsp-media.c:
3059 media: let the source unref the last media ref
3060 the last ref to the media is held by the source so we don't need to add more ref
3061 and unrefs, we simply destroy the media when the source is gone.
3063 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3065 * gst/rtsp-server/rtsp-media.c:
3066 media: improve debug
3068 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3070 * gst/rtsp-server/rtsp-media.c:
3072 Make sure we are in the right state when collecting the position and duration.
3073 Only make ourselves PREPARED when we were previously PREPARING.
3075 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3077 * gst/rtsp-server/rtsp-media.c:
3078 media: use g_object_ref/unref for GObjects
3080 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
3082 * gst/rtsp-server/rtsp-client.c:
3083 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
3084 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
3085 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
3086 isn't being used anymore.
3088 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
3090 * gst/rtsp-server/rtsp-media.c:
3091 Fix compiler warning
3093 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
3095 * gst/rtsp-server/rtsp-media-factory-uri.c:
3096 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
3098 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3100 * gst/rtsp-server/rtsp-session-media.h:
3103 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3105 * gst/rtsp-server/rtsp-media.c:
3106 * tests/check/gst/media.c:
3107 media: avoid element leak
3109 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3111 * gst/rtsp-server/rtsp-media.c:
3112 media: require an element in media constructor
3114 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3116 * gst/rtsp-server/rtsp-client.c:
3117 Revert "client: TEARDOWN brings that state to Init again"
3118 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
3119 The object is already disposed, there is no point in setting the state.
3121 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3123 * gst/rtsp-server/rtsp-client.c:
3124 client: TEARDOWN brings that state to Init again
3126 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3128 * docs/libs/gst-rtsp-server-sections.txt:
3129 * examples/test-auth.c:
3130 * gst/rtsp-server/rtsp-auth.c:
3131 * gst/rtsp-server/rtsp-auth.h:
3132 * gst/rtsp-server/rtsp-client.c:
3133 * gst/rtsp-server/rtsp-client.h:
3134 * gst/rtsp-server/rtsp-media-factory-uri.c:
3135 * gst/rtsp-server/rtsp-media-factory-uri.h:
3136 * gst/rtsp-server/rtsp-media-factory.c:
3137 * gst/rtsp-server/rtsp-media-factory.h:
3138 * gst/rtsp-server/rtsp-media.c:
3139 * gst/rtsp-server/rtsp-media.h:
3140 * gst/rtsp-server/rtsp-mount-points.c:
3141 * gst/rtsp-server/rtsp-mount-points.h:
3142 * gst/rtsp-server/rtsp-sdp.c:
3143 * gst/rtsp-server/rtsp-server.c:
3144 * gst/rtsp-server/rtsp-server.h:
3145 * gst/rtsp-server/rtsp-session-media.c:
3146 * gst/rtsp-server/rtsp-session-media.h:
3147 * gst/rtsp-server/rtsp-session-pool.c:
3148 * gst/rtsp-server/rtsp-session-pool.h:
3149 * gst/rtsp-server/rtsp-session.c:
3150 * gst/rtsp-server/rtsp-session.h:
3151 * gst/rtsp-server/rtsp-stream-transport.c:
3152 * gst/rtsp-server/rtsp-stream-transport.h:
3153 * gst/rtsp-server/rtsp-stream.c:
3154 * gst/rtsp-server/rtsp-stream.h:
3155 * tests/check/gst/media.c:
3156 rtsp: make object details private
3157 Make all object details private
3158 Add methods to access private bits
3160 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3162 * tests/check/Makefile.am:
3163 * tests/check/gst/media.c:
3164 tests: add media tests
3166 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3168 * gst/rtsp-server/rtsp-media.c:
3169 media: check if prepared for some methods
3170 Check that the media object is prepared before doing seek and getting the
3171 current position etc.
3172 Add some g_return checks.
3174 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3176 * tests/check/Makefile.am:
3177 * tests/check/gst/mediafactory.c:
3178 tests: add mediafactory test
3180 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3182 * gst/rtsp-server/rtsp-stream.c:
3183 stream: improve debug
3185 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3187 * gst/rtsp-server/rtsp-media.c:
3188 * gst/rtsp-server/rtsp-media.h:
3189 media: unref pipeline in finalize to avoid leaking it
3191 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3193 * gst/rtsp-server/rtsp-media-factory-uri.c:
3194 * gst/rtsp-server/rtsp-media.c:
3195 rtsp: use gst_object_unref on GstObjects
3197 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3199 * gst/rtsp-server/rtsp-media-factory.c:
3200 media-factory: require an url
3202 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3204 * examples/test-uri.c:
3205 examples: fix include
3207 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3209 * gst/rtsp-server/rtsp-server.h:
3210 server: remove unused include
3212 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3214 * tests/check/Makefile.am:
3215 * tests/check/gst/mountpoints.c:
3216 tests: add test for mountpoints
3218 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3220 * gst/rtsp-server/rtsp-client.c:
3221 client: fix factory leak
3222 Keep the factory in the state object only for authorization checks and make
3223 sure we unref it on failure. Also don't keep invalid objects in the state
3226 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3228 * gst/rtsp-server/rtsp-mount-points.c:
3229 mounts: add g_return_if guards
3231 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3233 * tests/check/gst/client.c:
3234 tests: add more tests
3236 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3238 * gst/rtsp-server/rtsp-client.c:
3239 client: improve debug
3241 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3243 * gst/rtsp-server/rtsp-client.c:
3244 client: improve debug and fix leaks
3245 Cleanup the uri and session when there is a bad request.
3247 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3252 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3254 * tests/check/gst/client.c:
3255 test: add test for session in options request
3257 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3259 * gst/rtsp-server/rtsp-client.c:
3260 client: use 454 when session can't be found
3261 We should use 454 when a session can't be found because there was no session
3262 pool configured in the server. This is not a server configuration problem
3263 because the server on which the request is done might not be the same one that
3264 will keep the sessions for us and so it does not need to support sessions.
3266 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3268 * gst/rtsp-server/rtsp-client.c:
3269 client: only free connection when there is one
3270 It's possible that the client doesn't have a connection when we try to free it.
3272 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3274 * tests/check/Makefile.am:
3275 * tests/check/gst/client.c:
3276 tests: add unit test for the client object
3278 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3280 * gst/rtsp-server/rtsp-client.c:
3281 client: small cleanup
3283 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3285 * gst/rtsp-server/rtsp-client.h:
3286 client: remove unused include
3288 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3290 * gst/rtsp-server/rtsp-client.c:
3291 client: fix compilation
3293 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3295 * gst/rtsp-server/rtsp-client.c:
3296 client: call destroy without the lock
3298 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3300 * gst/rtsp-server/rtsp-client.c:
3301 * gst/rtsp-server/rtsp-client.h:
3302 client: make the client usable without a socket
3303 Make a method to let the client handle a message and a callback when the client
3304 wants us to send a response message back. This makes it possible to also use the
3305 client object without the sockets, which should make it easier to test.
3307 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3309 * gst/rtsp-server/rtsp-client.c:
3310 * gst/rtsp-server/rtsp-client.h:
3311 client: small cleanup
3313 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3315 * docs/libs/gst-rtsp-server-sections.txt:
3316 * gst/rtsp-server/rtsp-client.c:
3317 * gst/rtsp-server/rtsp-client.h:
3318 * gst/rtsp-server/rtsp-server.c:
3319 client: remove reference to server
3320 We don't need to keep a ref to the server
3322 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3324 * gst/rtsp-server/rtsp-client.c:
3325 * gst/rtsp-server/rtsp-client.h:
3327 Also add some g_return_if()
3329 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3331 * gst/rtsp-server/rtsp-client.c:
3332 client: log more errors
3334 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3336 * gst/rtsp-server/rtsp-client.c:
3337 client: fix compilation
3339 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3341 * gst/rtsp-server/rtsp-client.c:
3342 * gst/rtsp-server/rtsp-client.h:
3343 client: add generic close-after-send support
3344 Add a property to send_response() to close the connection after the response has
3345 been sent to the client.
3347 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3350 * docs/libs/gst-rtsp-server-docs.sgml:
3351 * docs/libs/gst-rtsp-server-sections.txt:
3352 * docs/libs/gst-rtsp-server.types:
3353 * examples/test-auth.c:
3354 * examples/test-launch.c:
3355 * examples/test-mp4.c:
3356 * examples/test-multicast.c:
3357 * examples/test-multicast2.c:
3358 * examples/test-ogg.c:
3359 * examples/test-readme.c:
3360 * examples/test-sdp.c:
3361 * examples/test-uri.c:
3362 * examples/test-video.c:
3363 * gst/rtsp-server/Makefile.am:
3364 * gst/rtsp-server/rtsp-auth.h:
3365 * gst/rtsp-server/rtsp-client.c:
3366 * gst/rtsp-server/rtsp-client.h:
3367 * gst/rtsp-server/rtsp-media-mapping.c:
3368 * gst/rtsp-server/rtsp-media-mapping.h:
3369 * gst/rtsp-server/rtsp-mount-points.c:
3370 * gst/rtsp-server/rtsp-mount-points.h:
3371 * gst/rtsp-server/rtsp-server.c:
3372 * gst/rtsp-server/rtsp-server.h:
3373 * gst/rtsp-server/rtsp-session-media.c:
3374 * gst/rtsp-server/rtsp-session-pool.c:
3375 * gst/rtsp-server/rtsp-session-pool.h:
3376 * tests/check/gst/rtspserver.c:
3377 MediaMapping -> MountPoints
3378 Describes better what the object manages.
3380 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3383 configure: bump required version of -base
3385 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3387 * gst/rtsp-server/rtsp-media.c:
3390 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3392 * gst/rtsp-server/rtsp-media.c:
3393 * gst/rtsp-server/rtsp-media.h:
3394 media: support more Range formats
3395 Use the new -base methods to convert the Range string into a seek start and stop
3398 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3400 * examples/test-launch.c:
3401 examples: fix whitespace
3403 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3405 * examples/test-auth.c:
3406 test-auth: add example of how to remove sessions
3407 Add an example of the session filter api.
3409 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3411 * examples/test-uri.c:
3412 test-uri: remove mapping example
3414 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3416 * examples/test-uri.c:
3417 test-uri: fix callback signature
3419 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3421 * gst/rtsp-server/rtsp-media-factory.c:
3422 factory: keep ref to factory while media active
3423 While the media from a factory is alive, keep a ref to the factory.
3424 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
3426 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3428 * gst/rtsp-server/rtsp-media-factory-uri.c:
3429 factory-uri: add some debug
3431 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3433 * gst/rtsp-server/rtsp-stream.c:
3434 stream: set udp sources to PLAYING
3435 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
3436 so that it doesn't cause our pipeline to produce ASYNC-DONE.
3438 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3440 * gst/rtsp-server/rtsp-media-factory-uri.c:
3441 factory-uri: take ref to factory
3442 Take a ref to the factory that we place in our list.
3444 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3446 * tests/Makefile.am:
3447 * tests/test-reuse.c:
3448 test: add test for server reuse
3449 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
3451 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
3453 * gst/rtsp-server/rtsp-server.c:
3454 server: start and stop multiple times
3455 Stop listening on the RTSP port when the GSource is removed, so clients
3456 can't connect and the server can be started again.
3457 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
3459 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3461 * gst/rtsp-server/rtsp-server.c:
3462 server: fix small leak
3464 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3466 * gst/rtsp-server/rtsp-media.c:
3467 media: unref source in finish_unprepare
3468 The source is created in prepare, unref it in finish_unprepare.
3469 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
3471 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
3473 * gst/rtsp-server/rtsp-client.c:
3474 * gst/rtsp-server/rtsp-media.c:
3475 rtsp-media: remove bus watch before finalizing
3476 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
3477 * An extra media ref is added for the bus watch. This extra ref is unreffed by
3478 the GDestroyNotify function.
3479 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
3480 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
3481 gst_rtsp_media_unprepare before unreffing the media.
3482 This way, the bus watch will be removed before the media is finalized.
3483 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
3485 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
3487 * gst/rtsp-server/rtsp-client.c:
3488 * gst/rtsp-server/rtsp-client.h:
3489 client: wait until the TEARDOWN response is sent to close the connection
3490 Responses can be sent async so we need to wait until the TEARDOWN response has
3491 been written before we close the connection to the client. This avoids the risk
3492 of writing/polling closed sockets.
3493 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
3495 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
3497 * gst/rtsp-server/rtsp-stream.c:
3498 rtsp-stream: plug socket leak
3499 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
3501 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
3504 Automatic update of common submodule
3505 From 6bb6951 to a72faea
3507 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
3509 * gst/rtsp-server/rtsp-media-factory-uri.c:
3510 rtsp-server: don't use deprecated API
3512 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
3514 * gst/rtsp-server/rtsp-client.c:
3515 rtsp-client: fix unused-but-set-variable compiler warning
3516 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
3518 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3521 * docs/libs/gst-rtsp-server-sections.txt:
3522 * gst/rtsp-server/rtsp-client.c:
3525 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3527 * examples/Makefile.am:
3528 * examples/test-multicast2.c:
3529 examples: add another multicast example
3530 Add an example for how to configure separate multicast ranges for each media
3533 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3535 * examples/test-multicast.c:
3538 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3540 * gst/rtsp-server/rtsp-client.c:
3541 * gst/rtsp-server/rtsp-media.c:
3542 * gst/rtsp-server/rtsp-session-media.c:
3543 * gst/rtsp-server/rtsp-session-media.h:
3544 * gst/rtsp-server/rtsp-stream-transport.c:
3545 * gst/rtsp-server/rtsp-stream-transport.h:
3546 stream: use the address managed by the stream
3547 Use the address managed by the stream for multicast. This allows us to have 1
3548 multicast address for each stream.
3549 Because the address is now managed by the stream we don't have to pass it around
3551 Set the address pool on the streams.
3553 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3555 * gst/rtsp-server/rtsp-client.c:
3556 * gst/rtsp-server/rtsp-media.c:
3557 * gst/rtsp-server/rtsp-stream.c:
3560 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3562 * gst/rtsp-server/rtsp-media.c:
3563 * gst/rtsp-server/rtsp-media.h:
3564 media: add signal for new streams
3565 This allows applications to listen for new streams and configure properties on
3566 them, like the address pool.
3568 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3570 * gst/rtsp-server/rtsp-media.c:
3571 media: configure address pool in new streams
3573 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3575 * gst/rtsp-server/rtsp-stream.c:
3576 * gst/rtsp-server/rtsp-stream.h:
3577 stream: add methods to deal with address pool
3578 Add methods to get and set the address pool for the stream
3579 Add method to allocate and get the multicast addresses for this stream.
3581 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3583 * docs/libs/gst-rtsp-server-sections.txt:
3584 * gst/rtsp-server/rtsp-media.c:
3585 * gst/rtsp-server/rtsp-media.h:
3586 media: remove MTU property
3587 It is a stream property
3589 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3591 * gst/rtsp-server/rtsp-client.c:
3592 client: set blocksize only on stream
3593 Set the blocksize only on the current stream.
3595 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3597 * gst/rtsp-server/rtsp-stream.c:
3598 stream: share src and sink sockets
3599 the allocated socket is in the used-socket property, not socket.
3601 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3603 * gst/rtsp-server/rtsp-address-pool.c:
3604 * gst/rtsp-server/rtsp-address-pool.h:
3605 * gst/rtsp-server/rtsp-client.c:
3606 * gst/rtsp-server/rtsp-session-media.c:
3607 * gst/rtsp-server/rtsp-session-media.h:
3608 * gst/rtsp-server/rtsp-stream-transport.c:
3609 * gst/rtsp-server/rtsp-stream-transport.h:
3610 * tests/check/gst/addresspool.c:
3611 rtsp: make address-pool return an address object
3612 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
3613 store more info in the structure and allows us to more easily return the address
3614 to the right pool when no longer needed.
3615 Pass the address to the StreamTransport so that we can return it to the pool
3616 when the stream transport is freed or changed.
3618 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3620 * examples/Makefile.am:
3621 * examples/test-multicast.c:
3622 examples: add multicast example
3623 Show how to set up the multicast address pool so that media can be
3624 server with multicast.
3626 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3628 * gst/rtsp-server/rtsp-client.c:
3629 * gst/rtsp-server/rtsp-media-factory.c:
3630 * gst/rtsp-server/rtsp-media-factory.h:
3631 * gst/rtsp-server/rtsp-media.c:
3632 * gst/rtsp-server/rtsp-media.h:
3633 rtsp: use AddressPool
3634 Remove the multicast_group property.
3635 Use the configured addresspool to allocate multicast addresses.
3637 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3639 * gst/rtsp-server/rtsp-address-pool.c:
3640 * gst/rtsp-server/rtsp-address-pool.h:
3641 address-pool: add clear method
3643 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3645 * gst/rtsp-server/rtsp-address-pool.c:
3646 address-pool: small cleanups
3648 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3650 * tests/check/Makefile.am:
3651 * tests/check/gst/addresspool.c:
3652 tests: add addresspool unit test
3654 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3656 * gst/rtsp-server/Makefile.am:
3657 * gst/rtsp-server/rtsp-address-pool.c:
3658 * gst/rtsp-server/rtsp-address-pool.h:
3659 address-pool: add object to manage multicast addresses
3660 Make an object that can manage a rage of multicast addresses and ports.
3662 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3664 * gst/rtsp-server/rtsp-server.c:
3665 server: set default max-threads property
3667 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3669 * gst/rtsp-server/rtsp-media.c:
3670 media: wait for concurrent _prepare
3671 If a prepare is busy, wait for the result.
3673 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3675 * gst/rtsp-server/rtsp-media.c:
3676 media: add lock around message handler
3677 We don't want to dispatch messages while we are still processing the result of
3680 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3682 * gst/rtsp-server/rtsp-media.c:
3683 * gst/rtsp-server/rtsp-media.h:
3684 media: add lock to protect state changes
3686 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3688 * gst/rtsp-server/rtsp-stream.c:
3689 * gst/rtsp-server/rtsp-stream.h:
3692 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3694 * gst/rtsp-server/rtsp-stream-transport.c:
3695 * gst/rtsp-server/rtsp-stream-transport.h:
3696 * gst/rtsp-server/rtsp-stream.c:
3697 stream-transport: add keep-alive method
3699 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3701 * gst/rtsp-server/rtsp-stream-transport.c:
3702 * gst/rtsp-server/rtsp-stream-transport.h:
3703 * gst/rtsp-server/rtsp-stream.c:
3704 stream-transport: add method to handle RTP/RTCP
3705 Call new methods instead of poking into the structures directly.
3707 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3709 * gst/rtsp-server/rtsp-session-media.c:
3710 * gst/rtsp-server/rtsp-session-media.h:
3711 session-media: add locking
3713 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3715 * gst/rtsp-server/rtsp-session.c:
3716 * gst/rtsp-server/rtsp-session.h:
3717 session: add locking
3719 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3721 * gst/rtsp-server/rtsp-server.c:
3722 server: free old socket
3724 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3726 * gst/rtsp-server/rtsp-media-mapping.c:
3727 * gst/rtsp-server/rtsp-media-mapping.h:
3728 mapping: add locking
3730 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3732 * gst/rtsp-server/rtsp-media-factory.c:
3733 media-factory: add locking
3735 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3737 * gst/rtsp-server/rtsp-auth.c:
3738 * gst/rtsp-server/rtsp-auth.h:
3741 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3743 * gst/rtsp-server/rtsp-server.c:
3744 * gst/rtsp-server/rtsp-server.h:
3745 server: add max-thread property
3747 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3749 * gst/rtsp-server/rtsp-server.c:
3750 * gst/rtsp-server/rtsp-server.h:
3751 server: use a threadpool for the mainloops
3753 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3755 * gst/rtsp-server/rtsp-client.c:
3756 * gst/rtsp-server/rtsp-client.h:
3757 client: rename method
3758 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
3759 don't really create the client from the socket, we use the socket for the
3762 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3764 * gst/rtsp-server/rtsp-client.c:
3765 * gst/rtsp-server/rtsp-client.h:
3766 * gst/rtsp-server/rtsp-server.c:
3767 server: rework maincontext handling in clients
3768 Make a separate method to attach a client to a MainContext.
3769 Let the server decide in what GMainContext the client will operate and give this
3770 context to the client in attach. Then the server can later decide to use a
3771 separate thread for each client or just use the mainthread.
3773 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3775 * gst/rtsp-server/rtsp-client.c:
3776 * gst/rtsp-server/rtsp-session.c:
3777 * gst/rtsp-server/rtsp-session.h:
3778 session: move session header code in session object
3780 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
3784 * examples/test-auth.c:
3785 * examples/test-launch.c:
3786 * examples/test-mp4.c:
3787 * examples/test-ogg.c:
3788 * examples/test-readme.c:
3789 * examples/test-sdp.c:
3790 * examples/test-uri.c:
3791 * examples/test-video.c:
3792 * gst/rtsp-server/rtsp-auth.c:
3793 * gst/rtsp-server/rtsp-auth.h:
3794 * gst/rtsp-server/rtsp-client.c:
3795 * gst/rtsp-server/rtsp-client.h:
3796 * gst/rtsp-server/rtsp-media-factory-uri.c:
3797 * gst/rtsp-server/rtsp-media-factory-uri.h:
3798 * gst/rtsp-server/rtsp-media-factory.c:
3799 * gst/rtsp-server/rtsp-media-factory.h:
3800 * gst/rtsp-server/rtsp-media-mapping.c:
3801 * gst/rtsp-server/rtsp-media-mapping.h:
3802 * gst/rtsp-server/rtsp-media.c:
3803 * gst/rtsp-server/rtsp-media.h:
3804 * gst/rtsp-server/rtsp-params.c:
3805 * gst/rtsp-server/rtsp-params.h:
3806 * gst/rtsp-server/rtsp-sdp.c:
3807 * gst/rtsp-server/rtsp-sdp.h:
3808 * gst/rtsp-server/rtsp-server.c:
3809 * gst/rtsp-server/rtsp-server.h:
3810 * gst/rtsp-server/rtsp-session-media.c:
3811 * gst/rtsp-server/rtsp-session-media.h:
3812 * gst/rtsp-server/rtsp-session-pool.c:
3813 * gst/rtsp-server/rtsp-session-pool.h:
3814 * gst/rtsp-server/rtsp-session.c:
3815 * gst/rtsp-server/rtsp-session.h:
3816 * gst/rtsp-server/rtsp-stream-transport.c:
3817 * gst/rtsp-server/rtsp-stream-transport.h:
3818 * gst/rtsp-server/rtsp-stream.c:
3819 * gst/rtsp-server/rtsp-stream.h:
3820 * tests/check/gst/rtspserver.c:
3821 * tests/test-cleanup.c:
3824 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
3826 * gst/rtsp-server/rtsp-media.c:
3827 * gst/rtsp-server/rtsp-session-media.c:
3828 * gst/rtsp-server/rtsp-session.c:
3829 rtsp-server: added annotations to indicate type of ownership transfer of return values
3830 https://bugzilla.gnome.org/show_bug.cgi?id=680777
3832 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
3835 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
3837 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
3840 * bindings/Makefile.am:
3841 * bindings/vala/Makefile.am:
3842 * bindings/vala/gst-rtsp-server-0.10.deps:
3843 * bindings/vala/gst-rtsp-server-0.10.vapi:
3844 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
3845 * bindings/vala/packages/gst-rtsp-server-0.10.files:
3846 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
3847 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
3848 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
3850 bindings: remove vala bindings
3851 They'll be reunited with the other GStreamer bindings
3852 https://bugzilla.gnome.org/show_bug.cgi?id=680777
3854 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3856 * gst/rtsp-server/rtsp-client.c:
3857 * gst/rtsp-server/rtsp-session-media.c:
3858 * gst/rtsp-server/rtsp-session-media.h:
3859 * gst/rtsp-server/rtsp-stream-transport.c:
3860 * gst/rtsp-server/rtsp-stream-transport.h:
3861 rtsp: only create transport when needed
3862 Only create the StreamTransport when configured.
3864 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3866 * gst/rtsp-server/rtsp-client.c:
3867 client: small cleanup
3869 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3871 * gst/rtsp-server/rtsp-client.c:
3872 * gst/rtsp-server/rtsp-client.h:
3873 * gst/rtsp-server/rtsp-stream-transport.c:
3874 * gst/rtsp-server/rtsp-stream-transport.h:
3875 rtsp: refactor configuration of transport
3876 Move the configuration of the transport to a place where it makes
3879 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3881 * gst/rtsp-server/rtsp-client.c:
3882 client: refactor transport parsing
3884 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3886 * gst/rtsp-server/rtsp-client.c:
3887 client: refuse to change the MTU on shared media
3888 If we change the MTU of chared media, it changes for all clients.
3889 We don't want to set the MTU to something large for clients that
3892 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3894 * examples/test-mp4.c:
3895 * gst/rtsp-server/rtsp-media.c:
3896 small fixes to docs and debug
3898 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3900 * gst/rtsp-server/rtsp-stream.c:
3901 stream: transports must already have been removed
3903 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3905 * gst/rtsp-server/rtsp-media.c:
3906 * gst/rtsp-server/rtsp-stream.c:
3907 * gst/rtsp-server/rtsp-stream.h:
3908 stream: improve join and leave of the pipeline
3910 Do the cleanup properly
3913 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3915 * gst/rtsp-server/rtsp-media.c:
3916 media: move unprepare below default implementation
3917 Makes it easier to find the default implementation
3919 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3921 * gst/rtsp-server/rtsp-media.c:
3922 media: signal unprepared when we actually finish
3924 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3926 * gst/rtsp-server/rtsp-media.c:
3927 media: no need to unlock, unprepare does that when needed
3929 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3931 * docs/libs/gst-rtsp-server-sections.txt:
3932 * gst/rtsp-server/rtsp-media-factory.h:
3933 * gst/rtsp-server/rtsp-media-mapping.c:
3934 * gst/rtsp-server/rtsp-media.h:
3935 * gst/rtsp-server/rtsp-params.c:
3936 * gst/rtsp-server/rtsp-server.c:
3937 * gst/rtsp-server/rtsp-session-pool.h:
3938 * gst/rtsp-server/rtsp-session.c:
3939 * gst/rtsp-server/rtsp-session.h:
3940 * gst/rtsp-server/rtsp-stream-transport.h:
3941 * gst/rtsp-server/rtsp-stream.h:
3944 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3946 * gst/rtsp-server/rtsp-client.c:
3947 * gst/rtsp-server/rtsp-media-mapping.h:
3948 * gst/rtsp-server/rtsp-media.c:
3949 * gst/rtsp-server/rtsp-media.h:
3950 * gst/rtsp-server/rtsp-server.h:
3951 * gst/rtsp-server/rtsp-stream.c:
3952 * gst/rtsp-server/rtsp-stream.h:
3953 rtsp: fix MTU setting
3954 Fix setting of the MTU. There is no need for a vmethod.
3956 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3961 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3964 configure: bump version number after refactoring
3966 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3968 * gst/rtsp-server/Makefile.am:
3969 * gst/rtsp-server/rtsp-client.c:
3970 * gst/rtsp-server/rtsp-client.h:
3971 * gst/rtsp-server/rtsp-media-factory-uri.c:
3972 * gst/rtsp-server/rtsp-media-factory.c:
3973 * gst/rtsp-server/rtsp-media-factory.h:
3974 * gst/rtsp-server/rtsp-media.c:
3975 * gst/rtsp-server/rtsp-media.h:
3976 * gst/rtsp-server/rtsp-sdp.c:
3977 * gst/rtsp-server/rtsp-session-media.c:
3978 * gst/rtsp-server/rtsp-session-media.h:
3979 * gst/rtsp-server/rtsp-session.c:
3980 * gst/rtsp-server/rtsp-session.h:
3981 * gst/rtsp-server/rtsp-stream-transport.c:
3982 * gst/rtsp-server/rtsp-stream-transport.h:
3983 * gst/rtsp-server/rtsp-stream.c:
3984 * gst/rtsp-server/rtsp-stream.h:
3985 rtsp: massive refactoring
3986 Make GObjects from the remaining simple structures.
3987 Remove GstRTSPSessionStream, it's not needed.
3988 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
3989 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
3990 a GstRTSPStream should be transported to a client.
3991 Rename GstRTSPMediaFactory::get_element -> create_element because that
3992 more accurately describes what it does.
3993 Make nice methods instead of poking in the structures.
3994 Move some methods inside the relevant object source code.
3995 Use GPtrArray to store objects instead of plain arrays, it is more
3996 natural and allows us to more easily clean up.
3997 Move the allocation of udp ports to the Stream object. The Stream object
3998 contains the elements needed to stream the media to a client.
3999 Improve the prepare and unprepare methods. Unprepare should now undo
4000 everything prepare did. Improve also async unprepare when doing EOS on
4001 shutdown. Make sure we always unprepare correctly.
4003 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
4005 * gst/rtsp-server/rtsp-client.c:
4006 rtsp-client: Unref server address clients connected to
4007 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
4009 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
4011 * gst/rtsp-server/rtsp-server.c:
4012 rtsp-server: don't ref server socket if it is NULL
4013 Fixes test_bind_already_in_use unit test again after commit 6a497440.
4014 https://bugzilla.gnome.org/show_bug.cgi?id=686644
4016 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
4018 * tests/check/Makefile.am:
4019 tests: Add libgio link dependency
4020 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
4022 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
4024 * gst/rtsp-server/rtsp-media-mapping.c:
4025 * gst/rtsp-server/rtsp-media-mapping.h:
4026 rtsp-media-mapping: rename find_media vfunc to find_factory
4027 The virtual method and class method should have the same name
4028 so it is correctly represented in GIR file
4029 https://bugzilla.gnome.org/show_bug.cgi?id=680777
4031 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
4033 * gst/rtsp-server/rtsp-auth.c:
4034 * gst/rtsp-server/rtsp-client.c:
4035 * gst/rtsp-server/rtsp-media-factory-uri.c:
4036 * gst/rtsp-server/rtsp-media-factory.c:
4037 * gst/rtsp-server/rtsp-media-mapping.c:
4038 * gst/rtsp-server/rtsp-media.c:
4039 * gst/rtsp-server/rtsp-server.c:
4040 * gst/rtsp-server/rtsp-session-pool.c:
4041 * gst/rtsp-server/rtsp-session.c:
4042 rtsp-server: fixed comments and GIR annotations
4043 https://bugzilla.gnome.org/show_bug.cgi?id=680777
4045 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
4047 * gst/rtsp-server/rtsp-media-mapping.c:
4048 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
4050 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
4052 * gst/rtsp-server/rtsp-server.c:
4053 rtsp-server: allow binding on port 0 (binds on a random port)
4055 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
4057 * gst/rtsp-server/rtsp-server.c:
4058 * gst/rtsp-server/rtsp-server.h:
4059 rtsp-server: add bound-port property
4060 bound-port can be used to retrieve the port number when the server is bound on
4061 port 0, which binds on a random port.
4063 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
4065 * gst/rtsp-server/rtsp-media-factory.c:
4066 * gst/rtsp-server/rtsp-media-factory.h:
4067 rtsp-media-factory: make ::get_element overridable by GI bindings
4068 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
4069 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
4070 as the invoker for ::get_element(), making it overridable by GI generated
4073 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
4075 * gst/rtsp-server/rtsp-media-factory-uri.c:
4076 rtsp-media-factory-uri: don't autoplug parsers in a loop
4077 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
4080 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
4082 * gst/rtsp-server/Makefile.am:
4083 Explicitly link against gio. Fix link error on mac.
4085 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
4087 * gst/rtsp-server/rtsp-session.c:
4088 session: add ttl to the transport header in SETUP
4089 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
4091 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
4093 * gst/rtsp-server/rtsp-client.c:
4094 * gst/rtsp-server/rtsp-client.h:
4095 * gst/rtsp-server/rtsp-media.c:
4096 client: Use client transport settings for multicast if allowed.
4097 This patch makes it possible for the client to send transport settings for
4098 multicast (destination && ttl). Client settings must be explicitly allowed or
4099 the server will use its own settings.
4100 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
4102 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
4105 Automatic update of common submodule
4106 From 6c0b52c to 6bb6951
4108 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
4110 * gst/rtsp-server/rtsp-client.c:
4111 rtsp-client: do not destroy the rtsp watch
4112 Don't destroy the client watch while dispatching. The rtsp watch is
4113 automatically destroyed after the rtsp watch function closed() has
4115 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
4117 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
4120 Automatic update of common submodule
4121 From 4f962f7 to 6c0b52c
4123 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
4125 * gst/rtsp-server/rtsp-media.c:
4126 media: fix check for seekability
4128 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4130 * gst/rtsp-server/rtsp-client.c:
4131 client: use more GIO
4132 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
4134 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4136 * gst/rtsp-server/rtsp-server.c:
4137 server: remove obsolete includes
4139 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4141 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
4142 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
4143 be available in "on_new_ssrc". The transports are added in
4144 gst_rtsp_media_set_state when going to PLAYING state. However,
4145 "on_new_ssrc" might be called before this happens.
4146 https://bugzilla.gnome.org/show_bug.cgi?id=683304
4148 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4150 * gst/rtsp-server/rtsp-client.c:
4151 * gst/rtsp-server/rtsp-client.h:
4152 rtsp-client: add signals for rtsp requests (fixes #683287)
4154 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4156 * gst/rtsp-server/rtsp-client.c:
4157 * gst/rtsp-server/rtsp-client.h:
4158 add new-session signal to rtsp-client (fixes #683058)
4160 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
4163 Automatic update of common submodule
4164 From 668acee to 4f962f7
4166 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
4168 * gst/rtsp-server/rtsp-server.c:
4169 * tests/check/gst/rtspserver.c:
4170 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
4171 Do not assume that *error is set in g_socket_address_enumerator_next.
4172 Added test_bind_already_in_use unit-test.
4173 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
4175 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
4178 Automatic update of common submodule
4179 From 94ccf4c to 668acee
4181 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
4183 * gst/rtsp-server/rtsp-client.c:
4184 * gst/rtsp-server/rtsp-client.h:
4185 rtsp-client: make create_sdp virtual method
4186 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
4188 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4191 Automatic update of common submodule
4192 From 98e386f to 94ccf4c
4194 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4196 * gst/rtsp-server/rtsp-client.c:
4199 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
4201 * gst/rtsp-server/rtsp-client.c:
4202 * gst/rtsp-server/rtsp-client.h:
4203 * gst/rtsp-server/rtsp-server.c:
4204 * gst/rtsp-server/rtsp-server.h:
4205 rtsp-server: use an existing socket to establish HTTP tunnel
4206 Make it possible to transfer a socket from an HTTP server to be used as
4207 an RTSP over HTTP tunnel.
4209 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
4211 * gst/rtsp-server/rtsp-client.c:
4212 * gst/rtsp-server/rtsp-media.c:
4213 * gst/rtsp-server/rtsp-media.h:
4214 rtsp: Handle the blocksize parameter
4215 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
4217 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
4219 * tests/check/Makefile.am:
4220 * tests/check/gst/rtspserver.c:
4221 Have unit test get header from source dir, not installed dir
4222 This makes compilation of unit tests work in a build directory other
4223 than the source directory.
4224 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
4226 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
4228 * gst/rtsp-server/rtsp-media.c:
4229 rtsp-media: update for gst_element_make_from_uri() changes
4231 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
4234 * tests/Makefile.am:
4235 * tests/check/Makefile.am:
4236 * tests/check/gst/rtspserver.c:
4238 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
4240 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
4242 * gst/rtsp-server/rtsp-media.c:
4243 rtsp-media: don't collect media stats when going to NULL
4244 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
4246 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4248 * gst/rtsp-server/rtsp-client.c:
4249 client: don't leak transports
4251 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
4253 * gst/rtsp-server/rtsp-client.c:
4254 rtsp-client: free transport on no_stream in SETUP handler
4256 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
4258 * gst/rtsp-server/rtsp-client.c:
4259 rtsp-client: changed session media iteration
4260 In client_unlink_session: now don't iterate in session->medias
4261 list where items are removed by gst_rtsp_session_release_media.
4262 Instead, repeatedly remove the first item.
4264 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
4266 * gst/rtsp-server/rtsp-client.c:
4267 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
4268 GstRTSPSessionMedia is not a GObject type. When the
4269 GstRTSPSession is freed, it will free the media.
4271 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
4273 * gst/rtsp-server/rtsp-media-factory.c:
4274 factory: plug pad leak in collect_streams
4275 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
4276 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
4277 will take one reference, and the other reference will otherwise
4280 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
4283 configure: suppress some warnings when debug is disabled
4284 Warnings about unused variables should be suppressed if core has the
4285 debug system disabled.
4286 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
4288 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4290 * docs/libs/Makefile.am:
4291 docs: fix build in uninstalled setup
4292 Include gst-plugins-base libs properly.
4294 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
4296 * docs/libs/gst-rtsp-server.types:
4297 docs: include headers defining rtsp-server object types
4298 Fixes compiler warnings during docs build.
4299 https://bugzilla.gnome.org/show_bug.cgi?id=676824
4301 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
4304 configure: Add warning flags for compiler when configuring
4305 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
4307 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4310 Automatic update of common submodule
4311 From 03a0e57 to 98e386f
4313 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4316 Automatic update of common submodule
4317 From 1fab359 to 03a0e57
4319 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
4321 * gst/rtsp-server/rtsp-client.c:
4322 client: fix GSocketAddress leak in gst_rtsp_client_accept
4323 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
4325 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4328 Automatic update of common submodule
4329 From f1b5a96 to 1fab359
4331 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4334 Automatic update of common submodule
4335 From 92b7266 to f1b5a96
4337 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4340 Automatic update of common submodule
4341 From ec1c4a8 to 92b7266
4343 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4346 Automatic update of common submodule
4347 From 3429ba6 to ec1c4a8
4349 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
4351 * gst/rtsp-server/rtsp-auth.c:
4352 * gst/rtsp-server/rtsp-client.c:
4353 * gst/rtsp-server/rtsp-media-factory-uri.c:
4354 * gst/rtsp-server/rtsp-server.c:
4355 rtsp: fix compiler warnings
4356 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
4358 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4361 Automatic update of common submodule
4362 From dc70203 to 3429ba6
4364 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4366 * gst/rtsp-server/rtsp-client.c:
4367 * gst/rtsp-server/rtsp-media-factory.c:
4368 * gst/rtsp-server/rtsp-media-factory.h:
4369 * gst/rtsp-server/rtsp-media.c:
4370 * gst/rtsp-server/rtsp-media.h:
4371 * gst/rtsp-server/rtsp-server.c:
4372 * gst/rtsp-server/rtsp-server.h:
4373 * gst/rtsp-server/rtsp-session-pool.c:
4374 * gst/rtsp-server/rtsp-session-pool.h:
4375 rtsp-server: port to new thread API
4377 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4380 Automatic update of common submodule
4381 From 6db25be to dc70203
4383 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4385 * gst/rtsp-server/rtsp-auth.c:
4386 * gst/rtsp-server/rtsp-auth.h:
4387 * gst/rtsp-server/rtsp-client.c:
4388 rtsp-server: Fix compilation and compiler warnings
4390 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4394 * gst/rtsp-server/Makefile.am:
4395 configure: Modernize autotools setup a bit
4396 Also we now only create tar.bz2 and tar.xz tarballs.
4398 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4401 Automatic update of common submodule
4402 From 464fe15 to 6db25be
4404 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4407 Automatic update of common submodule
4408 From 7fda524 to 464fe15
4410 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4413 * docs/libs/Makefile.am:
4414 * docs/version.entities.in:
4416 * gst/rtsp-server/Makefile.am:
4417 * pkgconfig/Makefile.am:
4418 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4419 * pkgconfig/gstreamer-rtsp-server.pc.in:
4420 * tests/Makefile.am:
4421 rtsp-server: Update versioning
4423 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4425 Merge remote-tracking branch 'origin/0.10'
4427 gst/rtsp-server/rtsp-session-pool.c
4429 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4431 * gst/rtsp-server/rtsp-session-pool.c:
4432 rtsp-server: Don't use deprecated GLib API
4434 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4436 Replace master with 0.11
4438 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4440 Merge branch 'master' into 0.11
4442 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4444 Merge branch 'master' into 0.11
4446 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
4449 A couple minor typo fixes
4451 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4453 * gst/rtsp-server/rtsp-media.c:
4454 media: fix state of the appqueue
4456 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4458 * gst/rtsp-server/rtsp-media-factory-uri.c:
4459 factory: use videoconvert
4461 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4463 * gst/rtsp-server/rtsp-media-factory-uri.c:
4464 factory: change to new style caps
4466 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4468 * gst/rtsp-server/rtsp-client.c:
4469 * gst/rtsp-server/rtsp-client.h:
4470 * gst/rtsp-server/rtsp-media-factory-uri.c:
4471 * gst/rtsp-server/rtsp-media.c:
4472 * gst/rtsp-server/rtsp-server.c:
4473 * gst/rtsp-server/rtsp-server.h:
4474 * gst/rtsp-server/rtsp-session-pool.c:
4475 rtsp-server: port to GIO
4478 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4481 configure: fix build
4483 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4486 docs: fix for gst_rtsp_server_set_port() -> _set_service()
4487 https://bugzilla.gnome.org/show_bug.cgi?id=666548
4489 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4492 * examples/Makefile.am:
4493 First rule of gst-rtsp-server club: don't talk about gst-phonon
4495 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4498 * pkgconfig/Makefile.am:
4499 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
4500 * pkgconfig/gst-rtsp-server.pc.in:
4501 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4502 * pkgconfig/gstreamer-rtsp-server.pc.in:
4503 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
4504 For consistency with all other modules.
4506 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4508 * gst/rtsp-server/rtsp-client.c:
4509 rtsp-client: update for new map API
4511 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4514 * bindings/Makefile.am:
4515 * bindings/python/Makefile.am:
4516 * bindings/python/arg-types.py:
4517 * bindings/python/codegen/Makefile.am:
4518 * bindings/python/codegen/__init__.py:
4519 * bindings/python/codegen/argtypes.py:
4520 * bindings/python/codegen/code-coverage.py:
4521 * bindings/python/codegen/codegen.py:
4522 * bindings/python/codegen/definitions.py:
4523 * bindings/python/codegen/defsparser.py:
4524 * bindings/python/codegen/docextract.py:
4525 * bindings/python/codegen/docgen.py:
4526 * bindings/python/codegen/fileprefix.override:
4527 * bindings/python/codegen/fileprefixmodule.c:
4528 * bindings/python/codegen/h2def.py:
4529 * bindings/python/codegen/mergedefs.py:
4530 * bindings/python/codegen/mkskel.py:
4531 * bindings/python/codegen/override.py:
4532 * bindings/python/codegen/reversewrapper.py:
4533 * bindings/python/codegen/scmexpr.py:
4534 * bindings/python/rtspserver-types.defs:
4535 * bindings/python/rtspserver.defs:
4536 * bindings/python/rtspserver.override:
4537 * bindings/python/rtspservermodule.c:
4538 * bindings/python/test.py:
4540 python: remove pygst-based python bindings
4541 pygi is the future, apparently.
4543 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
4546 Automatic update of common submodule
4547 From c463bc0 to 7fda524
4549 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4552 Automatic update of common submodule
4553 From 2a59016 to c463bc0
4555 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4558 Automatic update of common submodule
4559 From 0807187 to 2a59016
4561 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4564 Automatic update of common submodule
4565 From 11f0cd5 to 0807187
4567 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4569 * examples/test-auth.c:
4570 example: update for new caps
4572 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4574 * examples/test-video.c:
4575 * gst/rtsp-server/rtsp-client.c:
4576 * gst/rtsp-server/rtsp-media-factory-uri.c:
4577 * gst/rtsp-server/rtsp-media.c:
4578 * gst/rtsp-server/rtsp-media.h:
4579 * gst/rtsp-server/rtsp-session.c:
4580 * gst/rtsp-server/rtsp-session.h:
4581 rtsp-server: port some more to 0.11
4583 Remove bufferlist stuff
4585 Add queue before appsink now that preroll-queue-len is gone.
4586 Update for request pad changes.
4588 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4590 Merge branch 'master' into 0.11
4592 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
4594 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
4595 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
4596 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
4598 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
4600 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
4601 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
4602 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
4604 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4606 Merge branch 'master' into 0.11
4608 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4610 * gst/rtsp-server/rtsp-media.c:
4611 * gst/rtsp-server/rtsp-media.h:
4612 media: add a seekable boolean
4613 Maintain the seekable state with a new variable instead of reusing the
4616 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
4618 * gst/rtsp-server/rtsp-media.c:
4619 Disallow seek in live media
4621 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4623 Merge branch 'master' into 0.11
4625 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
4627 * gst/rtsp-server/rtsp-server.c:
4628 #ifdef statements for windows socket creation were missing
4630 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
4633 Automatic update of common submodule
4634 From a39eb83 to 11f0cd5
4636 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
4639 Automatic update of common submodule
4640 From 605cd9a to a39eb83
4642 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4644 Merge branch 'master' into 0.11
4646 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4648 * gst/rtsp-server/rtsp-client.c:
4649 client: use method to access property
4651 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4653 * gst/rtsp-server/rtsp-media-factory.c:
4654 * gst/rtsp-server/rtsp-media-factory.h:
4655 media-factory: add protocols property
4656 Add a property to configure the allowed protocols in the media created from the
4659 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4661 * gst/rtsp-server/rtsp-media-factory.c:
4662 * gst/rtsp-server/rtsp-media-factory.h:
4663 media-factory: add media-configure signal
4664 Add signal to allow the application to configure the media after it was created
4667 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4669 * gst/rtsp-server/rtsp-client.c:
4670 client: use method to access property
4672 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4674 * gst/rtsp-server/rtsp-media-factory.c:
4675 * gst/rtsp-server/rtsp-media-factory.h:
4676 media-factory: add protocols property
4677 Add a property to configure the allowed protocols in the media created from the
4680 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4682 * gst/rtsp-server/rtsp-media-factory.c:
4683 * gst/rtsp-server/rtsp-media-factory.h:
4684 media-factory: add media-configure signal
4685 Add signal to allow the application to configure the media after it was created
4688 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4690 Merge branch 'master' into 0.11
4692 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4694 * gst/rtsp-server/rtsp-client.c:
4695 client: use media multicast group
4697 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4699 * gst/rtsp-server/rtsp-media-factory.h:
4700 * gst/rtsp-server/rtsp-server.h:
4701 * gst/rtsp-server/rtsp-session-pool.h:
4702 * gst/rtsp-server/rtsp-session.h:
4705 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
4707 * gst/rtsp-server/rtsp-client.c:
4708 * gst/rtsp-server/rtsp-sdp.h:
4709 sdp: copy and free the server ip address
4710 Copy and free the server ip address to make memory management easier later.
4712 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4714 * gst/rtsp-server/rtsp-media-factory.c:
4715 media-factory: configure multicast in media
4717 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4719 * gst/rtsp-server/rtsp-media.c:
4720 * gst/rtsp-server/rtsp-media.h:
4721 media: add property for multicast group
4722 Add a property to configure the multicast group in the media.
4723 Based on patches from Marc Leeman and Robert Krakora.
4725 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4727 * gst/rtsp-server/rtsp-media-factory.c:
4728 * gst/rtsp-server/rtsp-media-factory.h:
4729 media-factory: add property for multicast group
4730 Add a property to configure the multicast group in the media factory.
4731 Based on patches from Marc Leeman and Robert Krakora.
4733 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4735 * gst/rtsp-server/rtsp-client.c:
4736 client: do configuration of transport in one place
4737 Move the configuration of the transport destination address to where we also
4738 configure the other bits.
4740 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4742 * gst/rtsp-server/rtsp-client.c:
4743 client: use media multicast group
4745 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4747 * gst/rtsp-server/rtsp-media-factory.h:
4748 * gst/rtsp-server/rtsp-server.h:
4749 * gst/rtsp-server/rtsp-session-pool.h:
4750 * gst/rtsp-server/rtsp-session.h:
4753 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
4755 * gst/rtsp-server/rtsp-client.c:
4756 * gst/rtsp-server/rtsp-sdp.h:
4757 sdp: copy and free the server ip address
4758 Copy and free the server ip address to make memory management easier later.
4760 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4762 * gst/rtsp-server/rtsp-media-factory.c:
4763 media-factory: configure multicast in media
4765 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4767 * gst/rtsp-server/rtsp-media.c:
4768 * gst/rtsp-server/rtsp-media.h:
4769 media: add property for multicast group
4770 Add a property to configure the multicast group in the media.
4771 Based on patches from Marc Leeman and Robert Krakora.
4773 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4775 * gst/rtsp-server/rtsp-media-factory.c:
4776 * gst/rtsp-server/rtsp-media-factory.h:
4777 media-factory: add property for multicast group
4778 Add a property to configure the multicast group in the media factory.
4779 Based on patches from Marc Leeman and Robert Krakora.
4781 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4783 * gst/rtsp-server/rtsp-client.c:
4784 client: do configuration of transport in one place
4785 Move the configuration of the transport destination address to where we also
4786 configure the other bits.
4788 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4790 Merge branch 'master' into 0.11
4792 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
4794 * gst/rtsp-server/rtsp-client.c:
4795 client: destroy pipeline on client disconnect with no prior TEARDOWN.
4796 The problem occurs when the client abruptly closes the connection without
4797 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
4798 server is where the pipeline gets torn down. Since this handler is not called,
4799 the pipeline remains and is up and running. Subsequent clients get their own
4800 pipelines and if the do not issue TEARDOWNs then those pipelines will also
4801 remain up and running. This is a resource leak.
4803 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4805 Merge branch 'master' into 0.11
4807 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
4809 * gst/rtsp-server/rtsp-media-factory.c:
4810 * gst/rtsp-server/rtsp-media-factory.h:
4811 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
4812 For example, it can be used to retrieve source elements like appsrc, in a more
4813 convenient way than subclassing get_element.
4815 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4817 Merge branch 'master' into 0.11
4819 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
4821 * gst/rtsp-server/rtsp-server.c:
4822 rtsp-server: hold on to reference while using object
4824 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4826 * gst/rtsp-server/rtsp-media.c:
4829 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4832 configure: use unstable api
4834 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
4836 * gst/rtsp-server/rtsp-client.c:
4837 client: fix reference counting
4839 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
4841 * gst/rtsp-server/rtsp-client.c:
4842 * gst/rtsp-server/rtsp-media.c:
4843 fix compiler warnings about unused variables
4845 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
4847 * examples/test-launch.c:
4848 * examples/test-readme.c:
4849 * examples/test-uri.c:
4850 * examples/test-video.c:
4851 examples: tell rtsp uri when ready
4853 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
4856 Automatic update of common submodule
4857 From 69b981f to 605cd9a
4859 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4861 * gst/rtsp-server/rtsp-client.c:
4862 client: update for buffer API change
4864 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4866 * gst/rtsp-server/Makefile.am:
4867 Makefile.am: 0.10 => @GST_MAJORMINOR@
4869 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4871 * gst/rtsp-server/rtsp-media-factory-uri.c:
4872 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
4874 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4876 * gst/rtsp-server/.gitignore:
4877 .gitignore: 0.10 => 0.11
4879 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4881 * gst/rtsp-server/Makefile.am:
4882 Makefile.am: 0.10 => @GST_MAJORMINOR@
4884 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4886 Merge branch 'master' into 0.11
4888 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
4891 Automatic update of common submodule
4892 From 9e5bbd5 to 69b981f
4894 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
4897 Automatic update of common submodule
4898 From fd35073 to 9e5bbd5
4900 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
4903 Automatic update of common submodule
4904 From 46dfcea to fd35073
4906 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4908 * gst/rtsp-server/rtsp-media-factory-uri.c:
4909 * gst/rtsp-server/rtsp-media.c:
4910 media: port to new caps API
4912 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4914 Merge branch 'master' into 0.11
4916 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
4918 * bindings/vala/gst-rtsp-server-0.10.vapi:
4919 Updated Vala bindings.
4920 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
4922 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
4924 * gst/rtsp-server/rtsp-server.c:
4925 * gst/rtsp-server/rtsp-server.h:
4926 Add a signal for newly connected clients.
4927 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
4929 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
4931 * bindings/python/rtspserver.override:
4932 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
4934 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4936 * gst/rtsp-server/Makefile.am:
4937 * gst/rtsp-server/rtsp-client.c:
4938 * gst/rtsp-server/rtsp-funnel.c:
4939 * gst/rtsp-server/rtsp-funnel.h:
4940 * gst/rtsp-server/rtsp-media.c:
4941 rtsp-server: port to 0.11
4943 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4948 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4950 Merge branch 'master' into 0.11
4955 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4958 Automatic update of common submodule
4959 From c3cafe1 to 46dfcea
4961 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
4963 * bindings/python/Makefile.am:
4964 * bindings/python/rtspserver.defs:
4965 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
4967 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
4969 * bindings/python/arg-types.py:
4970 python bindings: add GstRTSPUrlParam
4971 Needed to implement MediaFactory virtual proxies
4973 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
4975 * bindings/python/arg-types.py:
4976 python bindings: fix returning GstRTSPUrl types
4978 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
4980 * bindings/python/arg-types.py:
4981 python bindings: add arg type for GstRTSPUrl
4983 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
4985 * bindings/python/rtspserver.defs:
4986 python bindings: fix the definition of MediaFactory.collect_stream
4988 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
4991 Automatic update of common submodule
4992 From 1ccbe09 to c3cafe1
4994 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4997 Automatic update of common submodule
4998 From 193b717 to 1ccbe09
5000 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
5003 Automatic update of common submodule
5004 From b77e2bf to 193b717
5006 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5009 build: Include lcov.mak to allow test coverage report generation
5011 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5014 Automatic update of common submodule
5015 From d8814b6 to b77e2bf
5017 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5020 Automatic update of common submodule
5021 From 6aaa286 to d8814b6
5023 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
5026 Automatic update of common submodule
5027 From 6aec6b9 to 6aaa286
5029 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
5032 autogen: wingo signed comment
5034 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
5036 * gst/rtsp-server/rtsp-session-pool.c:
5037 session: use full charset for RTSP session ID
5038 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
5039 session ID more difficult.
5040 https://bugzilla.gnome.org/show_bug.cgi?id=643812
5042 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5044 * gst/rtsp-server/Makefile.am:
5045 rtsp-server: Don't install the funnel header
5047 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
5050 Automatic update of common submodule
5051 From 1de7f6a to 6aec6b9
5053 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5056 configure: require core/base 0.10.31
5057 Needed at least for gst_plugin_feature_rank_compare_func().
5059 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
5062 Automatic update of common submodule
5063 From f94d739 to 1de7f6a
5065 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5067 * gst/rtsp-server/rtsp-media.c:
5068 media: remove more unused code
5070 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5072 * gst/rtsp-server/rtsp-media.c:
5073 * gst/rtsp-server/rtsp-media.h:
5074 media: remove duplicate filtering
5075 Remove the duplicate filtering code now that we have a released -good version.
5076 Give a warning instead.
5078 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5080 * gst/rtsp-server/rtsp-media-factory.c:
5081 * gst/rtsp-server/rtsp-media.c:
5082 media: fix default buffer size
5084 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5086 * gst/rtsp-server/rtsp-media-factory.c:
5087 * gst/rtsp-server/rtsp-media-factory.h:
5088 media-factory: add property to configure the buffer-size
5089 Add a property to configure the kernel UDP buffer size.
5091 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5093 * gst/rtsp-server/rtsp-media.c:
5094 * gst/rtsp-server/rtsp-media.h:
5095 media: add property to configure kernel buffer sizes
5096 Add a property to configure the kernel UDP buffer size.
5098 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5101 configure: set PYGOBJECT_REQ before using it
5102 https://bugzilla.gnome.org/show_bug.cgi?id=640641
5104 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5107 docs: recursive into sub-directories on 'make upload'
5109 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5111 * docs/libs/gst-rtsp-server-docs.sgml:
5112 * docs/version.entities.in:
5113 docs: mention full version these docs are for, not just major-minor
5115 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5120 === release 0.10.8 ===
5122 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5127 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5129 * gst/rtsp-server/rtsp-server.c:
5130 rtsp-server: clarify docs a little
5132 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5134 * gst/rtsp-server/rtsp-media.c:
5135 media: init debug category before starting thread
5137 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5139 * gst/rtsp-server/rtsp-auth.c:
5140 auth: add realm to make it more spec compliant
5142 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5144 * gst/rtsp-server/rtsp-server.c:
5145 * gst/rtsp-server/rtsp-server.h:
5148 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5150 * examples/test-video.c:
5151 example: improve example docs a little
5153 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5155 * gst/rtsp-server/rtsp-server.c:
5156 server: ensure the watch has a ref to the server
5158 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5160 * gst/rtsp-server/rtsp-server.c:
5161 server: simpify channel function
5163 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5165 * gst/rtsp-server/rtsp-server.c:
5166 * gst/rtsp-server/rtsp-server.h:
5167 server: simplify management of channel and source
5168 We don't need to keep around the channel and source objects. Let the mainloop
5169 and the source manage the source and channel respectively.
5171 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5177 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5180 * tests/Makefile.am:
5181 * tests/test-cleanup.c:
5182 tests: add tests directory and cleanup test
5184 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5186 * gst/rtsp-server/rtsp-media-factory-uri.c:
5187 * gst/rtsp-server/rtsp-media-factory.c:
5188 * gst/rtsp-server/rtsp-media-mapping.c:
5189 * gst/rtsp-server/rtsp-media.c:
5190 * gst/rtsp-server/rtsp-session-pool.c:
5191 * gst/rtsp-server/rtsp-session.c:
5192 server: improve debugging in various objects
5194 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5196 * gst/rtsp-server/rtsp-server.c:
5197 server: chain up to the parent finalize
5199 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
5201 * bindings/python/rtspserver-types.defs:
5202 * bindings/python/rtspserver.defs:
5203 * bindings/python/rtspserver.override:
5204 * bindings/python/test.py:
5205 gst-rtsp-server: update python bindings
5207 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5209 * gst/rtsp-server/rtsp-client.c:
5210 client: use the response from the clientstate
5211 Create the response object only once and store in the client state.
5212 Make all methods use the state response,
5214 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5216 * gst/rtsp-server/rtsp-server.c:
5217 server: use signal to keep track of clients
5218 Keep track of all the clients that the server creates and remove them when they
5219 fire the 'closed' signal.
5221 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5223 * gst/rtsp-server/rtsp-client.c:
5224 * gst/rtsp-server/rtsp-client.h:
5225 client: emit signal when closing
5227 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5229 * examples/.gitignore:
5230 * examples/Makefile.am:
5231 * examples/test-auth.c:
5232 * examples/test-video.c:
5233 * gst/rtsp-server/rtsp-auth.c:
5234 * gst/rtsp-server/rtsp-auth.h:
5235 * gst/rtsp-server/rtsp-client.c:
5236 * gst/rtsp-server/rtsp-media-factory.c:
5237 * gst/rtsp-server/rtsp-media.c:
5238 * gst/rtsp-server/rtsp-media.h:
5239 * gst/rtsp-server/rtsp-session-pool.h:
5240 * gst/rtsp-server/rtsp-session.h:
5241 media: enable per factory authorisations
5242 Allow for adding a GstRTSPAuth on the factory and media level and check
5243 permissions when accessing the factory.
5244 Add hints to the auth methods for future more fine grained authorisation.
5245 Add example application for per factory authentication.
5247 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5249 * gst/rtsp-server/rtsp-auth.c:
5250 * gst/rtsp-server/rtsp-auth.h:
5251 * gst/rtsp-server/rtsp-client.c:
5252 * gst/rtsp-server/rtsp-client.h:
5253 * gst/rtsp-server/rtsp-params.c:
5254 * gst/rtsp-server/rtsp-params.h:
5255 rtsp-server: Pass ClientState structure arround
5256 Pass the collected information for the ongoing request in a GstRTSPClientState
5257 structure that we can then pass around to simplify the method arguments. This
5258 will also be handy when we implement logging functionality.
5260 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5262 * gst/rtsp-server/rtsp-media-factory.c:
5263 * gst/rtsp-server/rtsp-media-factory.h:
5264 media-factory: add methods to configure authorisation
5266 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5268 * gst/rtsp-server/rtsp-client.c:
5269 client: unref auth in finalize
5271 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5273 * gst/rtsp-server/rtsp-server.c:
5274 server: unref auth in finalize
5276 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5278 * docs/libs/gst-rtsp-server-docs.sgml:
5279 * docs/libs/gst-rtsp-server-sections.txt:
5280 * docs/libs/gst-rtsp-server.types:
5283 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5285 * gst/rtsp-server/rtsp-server.c:
5286 * gst/rtsp-server/rtsp-server.h:
5287 server: separate create and accept
5288 Create separate create and accept methods so that subclasses can create custom
5290 Configure the server in the client object and prepare for keeping track of
5293 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5295 * gst/rtsp-server/rtsp-client.c:
5296 * gst/rtsp-server/rtsp-client.h:
5297 client: add support for setting the server.
5298 Add support for keeping a ref to the server that started this client
5301 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5303 * gst/rtsp-server/rtsp-auth.c:
5304 auth: fix memleak and add some docs
5305 Fix a memleak of the basic auth token.
5306 Add docs for the helper function
5308 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5310 * gst/rtsp-server/rtsp-auth.c:
5311 * gst/rtsp-server/rtsp-auth.h:
5312 * gst/rtsp-server/rtsp-client.c:
5313 client: delegate setup of auth to the manager
5314 Delegate the configuration of the authentication tokens to the manager object
5317 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5319 * examples/test-video.c:
5320 * gst/rtsp-server/Makefile.am:
5321 * gst/rtsp-server/rtsp-auth.c:
5322 * gst/rtsp-server/rtsp-auth.h:
5323 * gst/rtsp-server/rtsp-client.c:
5324 * gst/rtsp-server/rtsp-client.h:
5325 * gst/rtsp-server/rtsp-server.c:
5326 * gst/rtsp-server/rtsp-server.h:
5327 auth: add authentication object
5328 Add an object that can check the authorization of requests.
5329 Implement basic authentication.
5330 Add example authentication to test-video
5332 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5334 * gst/rtsp-server/rtsp-server.c:
5335 * gst/rtsp-server/rtsp-server.h:
5336 server: move includes back
5337 the includes are needed for sockaddr_in.
5339 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5341 * gst/rtsp-server/rtsp-client.c:
5342 * gst/rtsp-server/rtsp-client.h:
5343 * gst/rtsp-server/rtsp-server.c:
5344 * gst/rtsp-server/rtsp-server.h:
5345 rtsp: move network includes where they are needed
5347 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
5349 * gst/rtsp-server/rtsp-media.h:
5350 rtsp-media.h: Minor corrections in comments.
5353 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
5356 Automatic update of common submodule
5357 From e572c87 to f94d739
5359 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5363 * docs/libs/.gitignore:
5364 * examples/.gitignore:
5365 * gst/rtsp-server/.gitignore:
5368 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5370 * docs/libs/Makefile.am:
5371 docs: We don't build ps/pdf for API reference docs
5373 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5376 Automatic update of common submodule
5377 From ccbaa85 to e572c87
5379 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5382 Automatic update of common submodule
5383 From 46445ad to ccbaa85
5385 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5387 * gst/rtsp-server/Makefile.am:
5388 * gst/rtsp-server/fs-funnel.c:
5389 * gst/rtsp-server/fs-funnel.h:
5390 * gst/rtsp-server/rtsp-funnel.c:
5391 * gst/rtsp-server/rtsp-funnel.h:
5392 * gst/rtsp-server/rtsp-media.c:
5393 funnel: rename fsfunnel to rtspfunnel
5394 Rename the funnel to avoid conflicts with the farsight one.
5396 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5398 * gst/rtsp-server/Makefile.am:
5399 * gst/rtsp-server/fs-funnel.c:
5400 * gst/rtsp-server/fs-funnel.h:
5401 * gst/rtsp-server/rtsp-media.c:
5402 rtsp-media: add and use fsfunnel
5403 Add a copy of fsfunnel to the build because input-selector removed the (broken)
5404 select-all property that we need.
5406 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5408 * gst/rtsp-server/Makefile.am:
5409 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
5410 Use PKG_CONFIG_PATH specified at configure time (if any) as well
5411 for the g-ir-compiler, rather than just assuming the env var has
5414 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5421 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
5423 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5426 * gst/rtsp-server/Makefile.am:
5427 gobject-introspection: fix g-i build for uninstalled setup
5428 Requires gst-plugins-base git (> 0.10.31.2).
5430 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5432 * examples/test-uri.c:
5433 examples: add some more options and comments
5435 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5437 * gst/rtsp-server/rtsp-media-factory-uri.c:
5438 factory-uri: use right property type
5440 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5442 * gst/rtsp-server/rtsp-media-factory-uri.c:
5443 factory-uri: attempt to configure buffer-lists
5444 Attempt to configure buffer lists in the payloader for improved performance.
5446 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5448 * gst/rtsp-server/rtsp-media.c:
5449 media: attempt to configure bigger UDP buffers
5450 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
5451 send buffers with high bitrate streams.
5453 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
5455 * gst/rtsp-server/rtsp-client.c:
5456 client: use the socket length from getsockname
5457 Use the length returned by getsockname to perform the getnameinfo call because
5458 the size can depend on the socket type and platform.
5461 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5463 * docs/libs/gst-rtsp-server-docs.sgml:
5464 * docs/libs/gst-rtsp-server-sections.txt:
5465 docs: add uri factory to the docs
5467 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5469 * gst/rtsp-server/rtsp-client.c:
5470 * gst/rtsp-server/rtsp-media.h:
5473 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5475 * gst/rtsp-server/rtsp-client.c:
5476 * gst/rtsp-server/rtsp-media.c:
5477 * gst/rtsp-server/rtsp-media.h:
5478 * gst/rtsp-server/rtsp-session.c:
5479 * gst/rtsp-server/rtsp-session.h:
5480 rtsp-server: add support for buffer lists
5481 Add support for sending bufferlists received from appsink.
5484 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5486 * gst/rtsp-server/rtsp-client.c:
5487 * gst/rtsp-server/rtsp-media.c:
5488 * gst/rtsp-server/rtsp-media.h:
5489 * gst/rtsp-server/rtsp-sdp.c:
5490 media: make method to retrieve the play range
5491 Make a method to retrieve the playback range so that we can conditionally create
5492 a different range for the SDP and the PLAY requests.
5494 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5496 * gst/rtsp-server/rtsp-media.c:
5497 * gst/rtsp-server/rtsp-media.h:
5498 media: add signal to notify of state changes
5500 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5502 * gst/rtsp-server/rtsp-client.h:
5503 client: cleanup headers
5505 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5507 * gst/rtsp-server/rtsp-client.c:
5510 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5512 * gst/rtsp-server/rtsp-media-factory-uri.c:
5513 * gst/rtsp-server/rtsp-media-factory-uri.h:
5514 factory-uri: add support for gstpay
5515 Add an option to prefer gstpay over decoder + raw payloader.
5517 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5519 * gst/rtsp-server/rtsp-media-factory-uri.c:
5520 * gst/rtsp-server/rtsp-media-factory-uri.h:
5521 factory-uri: rework the autoplugger.
5522 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
5525 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5527 * gst/rtsp-server/rtsp-media-factory-uri.c:
5528 factory-uri: use better factory filter
5529 Make better payloader filter based on autoplug rank and RTP use case.
5531 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5534 Automatic update of common submodule
5535 From 169462a to 46445ad
5537 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5539 * gst/rtsp-server/rtsp-server.c:
5540 server: set SO_REUSEADDR before bind
5541 Set the SO_REUSEADDR _before_ bind() to make it actually work.
5543 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5545 * gst/rtsp-server/rtsp-media.c:
5546 * gst/rtsp-server/rtsp-media.h:
5547 media: emit prepared signal when prepared
5548 Make a 'prepared' signal and emit it when we successfully prepared the element.
5549 This signal can be used to configure the media object after it has been prepared
5552 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
5555 Automatic update of common submodule
5556 From 011bcc8 to 169462a
5558 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
5560 python an optional dependency
5561 * configure.ac: Move up valgrind and g-i checks. Make the python
5562 dependency optional, as it was before.
5564 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5566 Merge branch 'master' into 0.11
5571 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5573 * gst/rtsp-server/rtsp-media.c:
5574 media: update range when active clients changed
5575 When we changed the number of active clients, update the current range
5576 information because we want the second client connecting to a shared resource
5577 continue from where the stream currently.
5579 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5581 * gst/rtsp-server/rtsp-media-factory-uri.c:
5582 * gst/rtsp-server/rtsp-media-factory-uri.h:
5583 factory-uri: add colorspace and fix pt
5584 Rework the way we pass data to the autoplugger.
5585 When we have raw caps, plug a converter element to make pluggin to raw
5586 payloaders more successful.
5587 Make sure all dynamically plugged payloaders have a unique payload types.
5589 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5591 * examples/Makefile.am:
5592 * examples/test-uri.c:
5593 example: add example of the uri factory
5595 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5597 * gst/rtsp-server/Makefile.am:
5598 * gst/rtsp-server/rtsp-media-factory-uri.c:
5599 * gst/rtsp-server/rtsp-media-factory-uri.h:
5600 * gst/rtsp-server/rtsp-server.h:
5601 factory-uri: add a factory to stream any URI
5602 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
5605 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5607 * gst/rtsp-server/rtsp-media.c:
5608 * gst/rtsp-server/rtsp-media.h:
5609 media: ignore spurious ASYNC_DONE messages
5610 When we are dynamically adding pads, the addition of the udpsrc elements will
5611 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
5612 the real ASYNC_DONE when everything is prerolled.
5614 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5616 * gst/rtsp-server/rtsp-media-factory.c:
5617 * gst/rtsp-server/rtsp-media-factory.h:
5618 media-factory: make lock macro
5620 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
5622 * gst/rtsp-server/rtsp-client.c:
5623 rtsp-server: Remove unused variable and dead assignment
5625 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
5627 * examples/test-launch.c:
5628 * examples/test-mp4.c:
5629 * examples/test-ogg.c:
5630 * examples/test-readme.c:
5631 * examples/test-sdp.c:
5632 * examples/test-video.c:
5633 examples: Run gst-indent
5635 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
5637 * gst/rtsp-server/rtsp-client.c:
5638 * gst/rtsp-server/rtsp-media-factory.c:
5639 * gst/rtsp-server/rtsp-media-mapping.c:
5640 * gst/rtsp-server/rtsp-media.c:
5641 * gst/rtsp-server/rtsp-params.c:
5642 * gst/rtsp-server/rtsp-sdp.c:
5643 * gst/rtsp-server/rtsp-server.c:
5644 * gst/rtsp-server/rtsp-session-pool.c:
5645 * gst/rtsp-server/rtsp-session.c:
5646 rtsp-server: Run gst-indent
5647 Since it wasn't using the upstream common previously, there was no
5648 indentation check before commiting.
5650 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
5652 * gst/rtsp-server/rtsp-media-mapping.h:
5653 * gst/rtsp-server/rtsp-media.c:
5654 * gst/rtsp-server/rtsp-media.h:
5655 * gst/rtsp-server/rtsp-sdp.c:
5656 * gst/rtsp-server/rtsp-session-pool.h:
5657 * gst/rtsp-server/rtsp-session.c:
5658 * gst/rtsp-server/rtsp-session.h:
5659 rtsp-server: Some more doc fixups
5661 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5664 Makefile: Add cruft-cleaning support
5666 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5671 * docs/libs/Makefile.am:
5672 * docs/libs/gst-rtsp-server-docs.sgml:
5673 * docs/libs/gst-rtsp-server-sections.txt:
5674 * docs/libs/gst-rtsp-server.types:
5675 * docs/version.entities.in:
5676 docs: Add gtk-doc build system
5678 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5680 * gst/rtsp-server/Makefile.am:
5681 Makefile.am: Use standard GIR make behaviour
5683 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5687 autogen/configure: Bring more in sync to standard gst module behaviour
5689 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5691 * gst/rtsp-server/rtsp-media.c:
5692 media: warn and fail when gstrtpbin is not found
5694 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5697 configure: open 0.11 branch
5699 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
5703 Add common submodule
5705 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
5708 * common/Makefile.am:
5709 * common/c-to-xml.py:
5711 * common/coverage/coverage-report-entry.pl:
5712 * common/coverage/coverage-report.pl:
5713 * common/coverage/coverage-report.xsl:
5714 * common/coverage/lcov.mak:
5715 * common/gettext.patch:
5716 * common/glib-gen.mak:
5717 * common/gst-autogen.sh:
5718 * common/gst-xmlinspect.py:
5720 * common/gstdoc-scangobj:
5721 * common/gtk-doc-plugins.mak:
5722 * common/gtk-doc.mak:
5723 * common/m4/.gitignore:
5724 * common/m4/Makefile.am:
5726 * common/m4/as-ac-expand.m4:
5727 * common/m4/as-auto-alt.m4:
5728 * common/m4/as-compiler-flag.m4:
5729 * common/m4/as-compiler.m4:
5730 * common/m4/as-docbook.m4:
5731 * common/m4/as-libtool-tags.m4:
5732 * common/m4/as-libtool.m4:
5733 * common/m4/as-python.m4:
5734 * common/m4/as-scrub-include.m4:
5735 * common/m4/as-version.m4:
5736 * common/m4/ax_create_stdint_h.m4:
5737 * common/m4/check.m4:
5738 * common/m4/glib-gettext.m4:
5739 * common/m4/gst-arch.m4:
5740 * common/m4/gst-args.m4:
5741 * common/m4/gst-check.m4:
5742 * common/m4/gst-debuginfo.m4:
5743 * common/m4/gst-default.m4:
5744 * common/m4/gst-doc.m4:
5745 * common/m4/gst-error.m4:
5746 * common/m4/gst-feature.m4:
5747 * common/m4/gst-function.m4:
5748 * common/m4/gst-gettext.m4:
5749 * common/m4/gst-glib2.m4:
5750 * common/m4/gst-libxml2.m4:
5751 * common/m4/gst-plugindir.m4:
5752 * common/m4/gst-valgrind.m4:
5753 * common/m4/gtk-doc.m4:
5754 * common/m4/introspection.m4:
5756 * common/mangle-tmpl.py:
5757 * common/plugins.xsl:
5759 * common/release.mak:
5760 * common/scangobj-merge.py:
5761 * common/upload.mak:
5762 common: Remove static version
5764 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
5766 * common/m4/introspection.m4:
5767 Update introspection.m4 to match usage
5769 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5773 Remove old stuff from the README
5775 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5780 === release 0.10.7 ===
5782 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5787 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5789 * examples/test-ogg.c:
5790 test-ogg: remove parsers
5791 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
5792 buffers with timestamps. Using the parsers also seems to break things.
5794 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
5796 * bindings/vala/gst-rtsp-server-0.10.vapi:
5797 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
5798 Updated Vala bindings
5800 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
5802 * common/m4/introspection.m4:
5804 * gst/rtsp-server/Makefile.am:
5805 Added initial gobject-introspection support
5807 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5809 * gst/rtsp-server/rtsp-media-factory.c:
5810 media-factory: don't use host for shared hash key
5811 When we generate the key to share made between connections, don't include the
5812 host used to connect so that we can share media even if between clients that
5813 connected with localhost and ones with the ip address.
5815 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5817 * bindings/vala/Makefile.am:
5818 build: fix distcheck
5820 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5822 * bindings/vala/gst-rtsp-server-0.10.vapi:
5823 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
5824 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
5825 Update Vala bindings
5827 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5829 * bindings/vala/Makefile.am:
5831 Fix configure checks and installation location for Vala bindings
5834 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5839 === release 0.10.6 ===
5841 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5844 configure: release 0.10.6
5846 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5848 * gst/rtsp-server/rtsp-media.c:
5849 media: help the compiler a little
5851 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5853 * gst/rtsp-server/rtsp-media.c:
5854 * gst/rtsp-server/rtsp-media.h:
5855 * gst/rtsp-server/rtsp-session.c:
5856 media: cleanup media transport before freeing
5857 Cleanup the media transport data before freeing. In particular, remove the qdata
5858 from the rtpsource object.
5860 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5862 * gst/rtsp-server/rtsp-media-factory.c:
5863 * gst/rtsp-server/rtsp-media-factory.h:
5864 * gst/rtsp-server/rtsp-media.c:
5865 * gst/rtsp-server/rtsp-media.h:
5866 media-factory: add eos-shutdown property
5867 Add an eos-shutdown property that will send an EOS to the pipeline before
5868 shutting it down. This allows for nice cleanup in case of a muxer.
5871 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5873 * gst/rtsp-server/rtsp-media.c:
5874 * gst/rtsp-server/rtsp-media.h:
5875 media: use multiudpsink send-duplicates when we can
5876 If we have a new enough multiudpsink with the send-duplicates property, use this
5877 instead of doing our own filtering. Our custom filtering code should eventually
5878 be removed when we can depend on a released -good.
5880 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5882 * gst/rtsp-server/rtsp-media.c:
5883 media: don't leak destinations
5884 Refactor and cleanup the destinations array when the stream is destroyed.
5886 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5888 * gst/rtsp-server/rtsp-media.c:
5889 * gst/rtsp-server/rtsp-media.h:
5890 media: don't add udp addresses multiple times
5891 Keep track of the udp addresses we added to udpsink and never add the same udp
5892 destination twice. This avoids duplicate packets when using multicast.
5894 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5896 * gst/rtsp-server/rtsp-server.c:
5897 server: disable use of SO_LINGER
5898 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
5899 server close()s the connection.
5901 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5903 * gst/rtsp-server/rtsp-server.c:
5904 server: use 5 second linger period in SO_LINGER
5905 Wait 5 seconds before clearing the send buffers and reseting the connection with
5906 the client when we do a close. This should be enough time to get the message to
5910 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
5912 * gst/rtsp-server/rtsp-server.c:
5913 server: use SO_LINGER
5914 SO_LINGER on the socket will make sure that any pending data on the socket is
5915 flushed ASAP and that the socket connection is reset. This makes sure that the
5916 socket can be reused immediately.
5919 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5922 README: add blurb about shared media factories
5924 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
5926 * gst/rtsp-server/rtsp-media.c:
5927 Add stdlib.h for atoi()
5929 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5931 * bindings/python/Makefile.am:
5932 * bindings/vala/Makefile.am:
5933 build: distcheck fixes
5934 Fix 'make distcheck', somewhat (it still fails because it tries to
5935 install files into /usr/share/vala/vapi/ irrespective of the
5938 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5941 configure: bump core/base requirements to released version
5942 Makes things less confusing for people.
5944 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5947 configure: fail if GStreamer core/base requirements are not met
5949 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5951 * gst/rtsp-server/rtsp-client.c:
5952 client: improve client cleanups
5953 Make sure the session does not timeout when using TCP. We need to do this
5954 because quicktime player does not send RTCP for some reason in tunneled
5956 Refactor some cleanup code.
5959 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5961 * gst/rtsp-server/rtsp-session.c:
5962 * gst/rtsp-server/rtsp-session.h:
5963 session: add support for prevent session timeouts
5964 Add an atomix counter to prevent session timeouts when we are, for example,
5967 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5969 * gst/rtsp-server/rtsp-client.c:
5970 client: fix unlink on session timeouts
5971 When our session times out, make sure we unlink all streams in this
5973 Remove the tunnelid when closing the connection.
5975 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5977 * gst/rtsp-server/rtsp-session.c:
5978 session: small cleanups
5980 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5982 * gst/rtsp-server/rtsp-client.c:
5983 client: handle lost_tunnel callbacks
5984 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
5985 hashtable so that we can reuse it for when the client reopens the POST
5987 Close the connection after a TEARDOWN.
5988 Make sure or watchid is cleared when the watch is removed.
5991 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5993 * gst/rtsp-server/rtsp-client.c:
5994 * gst/rtsp-server/rtsp-media.c:
5995 * gst/rtsp-server/rtsp-sdp.c:
5996 rtsp-server: add more support for multicast
5998 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6001 * gst/rtsp-server/rtsp-media.c:
6002 * gst/rtsp-server/rtsp-media.h:
6003 media: allow configuration of allowed lower transport
6005 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6007 * gst/rtsp-server/rtsp-client.h:
6008 * gst/rtsp-server/rtsp-media.c:
6009 * gst/rtsp-server/rtsp-media.h:
6010 * gst/rtsp-server/rtsp-sdp.c:
6011 * gst/rtsp-server/rtsp-sdp.h:
6012 * gst/rtsp-server/rtsp-server.c:
6013 rtsp: keep track of server ip and ipv6
6014 Keep track of how the client connected to the server and setup the udp ports
6015 with the same protocol.
6016 Copy the server ip address in the SDP so that clients can send RTCP back to
6019 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6021 * gst/rtsp-server/rtsp-session.c:
6024 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6026 * gst/rtsp-server/rtsp-client.c:
6027 client: use right size for malloc
6029 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6031 * gst/rtsp-server/rtsp-server.c:
6032 server: comment ipv6 server listening address
6034 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6036 * gst/rtsp-server/rtsp-media.c:
6037 media: allow for ipv6 sockets
6039 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6041 * gst/rtsp-server/rtsp-server.c:
6042 * gst/rtsp-server/rtsp-server.h:
6043 server: rework server part
6044 Allow setting a bind address, make sure we can deal with ipv6.
6045 Remove the port property and change with the service property.
6047 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6049 * gst/rtsp-server/rtsp-media.h:
6050 media: update comments a little
6052 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6054 * gst/rtsp-server/rtsp-client.c:
6055 client: make content-base better
6056 Use the URI formatting functions to make a content-base. Also make sure that
6057 there is a trailing / at the end.
6059 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6061 * gst/rtsp-server/rtsp-client.c:
6062 client: guard against invalid paths
6064 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6066 * examples/test-video.c:
6067 test: catch server bind errors
6069 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
6071 * gst/rtsp-server/rtsp-media.c:
6072 rtspmedia: emit "unprepared" if _prepare fails.
6073 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
6074 media object is removed from its factory's cache.
6076 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6078 * gst/rtsp-server/rtsp-media.c:
6079 media: collect media position when seek completes
6081 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
6083 * gst/rtsp-server/rtsp-client.c:
6084 client: call unlink_streams in client finalize
6087 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6089 * gst/rtsp-server/rtsp-media.c:
6090 media: limit the time to wait to something huge
6091 Avoid waiting forever but limit the timeout to 20 seconds.
6093 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6095 * gst/rtsp-server/rtsp-sdp.c:
6096 sdp: reindent and check for prepared status
6098 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6100 * gst/rtsp-server/rtsp-media.c:
6101 * gst/rtsp-server/rtsp-media.h:
6102 * gst/rtsp-server/rtsp-session.c:
6103 media: avoid doing _get_state() for state changes
6104 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
6105 until the media is prerolled or in error. This avoids doing a blocking call of
6106 gst_element_get_state() that can cause lockups when there is an error.
6109 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6111 * gst/rtsp-server/rtsp-media.c:
6114 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6116 * gst/rtsp-server/rtsp-media-factory.c:
6117 media-factory: better error handling
6118 Improve the error handling a bit.
6120 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6122 * gst/rtsp-server/rtsp-client.c:
6123 client: rework transport parsing
6124 Rework the transport parsing code so that we can ignore transports we don't
6125 support instead of just picking the first one we can parse.
6126 Configure a (for now hardcoded) destination for multicast transports.
6128 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6130 * gst/rtsp-server/rtsp-media.c:
6131 media: set multicast sink parameters
6132 Disable loop and automatic multicast join on the udpsink elements.
6133 Add some more debug info.
6134 Reset some state variables in the right place.
6135 Use the right port numbers for multicast.
6137 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6139 * gst/rtsp-server/rtsp-session.c:
6140 session: handle transport setup correctly
6141 Handle UDP, MCAST and TCP transport negotiation more correctly.
6142 Store the server session SSRC in the transport.
6144 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6146 * gst/rtsp-server/rtsp-client.c:
6147 rtsp-client: implement error_full
6148 Implement error_full to avoid some segfaults when the rtspconnection calls it.
6151 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6154 * gst/rtsp-server/rtsp-client.c:
6155 * gst/rtsp-server/rtsp-server.c:
6156 docs: update docs and comments
6158 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
6160 * gst/rtsp-server/rtsp-sdp.c:
6161 sdp: make server work better when behind a proxy
6163 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6165 * gst/rtsp-server/rtsp-client.c:
6166 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
6168 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6170 * gst/rtsp-server/rtsp-client.c:
6171 * gst/rtsp-server/rtsp-media-factory.c:
6172 * gst/rtsp-server/rtsp-media-mapping.c:
6173 * gst/rtsp-server/rtsp-media.c:
6174 * gst/rtsp-server/rtsp-server.c:
6175 * gst/rtsp-server/rtsp-session-pool.c:
6176 * gst/rtsp-server/rtsp-session.c:
6177 Use GStreamer's debugging subsystem
6179 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6181 * gst/rtsp-server/rtsp-media-factory.c:
6182 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
6184 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6189 === release 0.10.5 ===
6191 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6196 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6199 configure: bump required versions
6201 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
6203 * gst/rtsp-server/rtsp-client.c:
6204 client: call weak-unref on client->sessions from finalize
6207 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6209 * gst/rtsp-server/rtsp-media.c:
6210 media: Fixed crasher where caps got unref'ed too often
6212 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6215 * pkgconfig/.gitignore:
6216 * pkgconfig/Makefile.am:
6217 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
6218 Added pkg-config file to use gst-rtsp-server uninstalled
6220 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6222 * gst/rtsp-server/rtsp-media.c:
6223 media: add some docs
6225 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
6227 * gst/rtsp-server/rtsp-client.c:
6228 rtsp: Use gst_rtsp_watch_send_message().
6229 Use gst_rtsp_watch_send_message() since the old API which used
6230 gst_rtsp_watch_queue_message() has been deprecated.
6232 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6237 === release 0.10.4 ===
6239 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6244 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6246 * gst/rtsp-server/rtsp-client.c:
6247 * gst/rtsp-server/rtsp-session.c:
6248 * gst/rtsp-server/rtsp-session.h:
6249 rtsp: allocate channels in TCP mode
6250 When the client does not provide us with channels in TCP mode, allocate channels
6253 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6255 * gst/rtsp-server/rtsp-client.c:
6256 client: don't crash when tunnelid is missing
6257 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
6258 don't crash but return an error response to the client.
6261 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6263 * bindings/vala/gst-rtsp-server-0.10.vapi:
6264 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6265 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6266 bindings: update vala bindings with new method
6268 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6270 * gst/rtsp-server/rtsp-session-pool.c:
6271 * gst/rtsp-server/rtsp-session-pool.h:
6272 sessionpool: add function to filter sessions
6273 Add generic function to retrieve/remove sessions.
6275 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6278 configure: bump core/base requirements to release
6280 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6282 * gst/rtsp-server/rtsp-media.c:
6283 media: fix indentation
6285 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6287 * gst/rtsp-server/rtsp-media.c:
6288 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
6290 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6292 * gst/rtsp-server/rtsp-media.c:
6293 set state and remove elements of media in for loop
6295 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
6297 * bindings/vala/gst-rtsp-server-0.10.vapi:
6298 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6299 Added gst_rtsp_media_remove_elements function to Vala bindings
6301 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
6303 * gst/rtsp-server/rtsp-media.c:
6304 * gst/rtsp-server/rtsp-media.h:
6305 Added gst_rtsp_media_remove_elements function
6307 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
6309 * gst/rtsp-server/rtsp-media.c:
6310 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
6312 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6314 * bindings/vala/gst-rtsp-server-0.10.vapi:
6315 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6316 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6317 Updated Vala bindings
6319 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6321 * gst/rtsp-server/rtsp-media.c:
6322 * gst/rtsp-server/rtsp-media.h:
6323 Added vmethod unprepare to GstRTSPMedia
6324 The default implementation sets the state of the pipeline to GST_STATE_NULL
6326 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6328 * gst/rtsp-server/rtsp-media-factory.c:
6329 * gst/rtsp-server/rtsp-media-factory.h:
6330 Made collect_streams function public
6332 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6334 * gst/rtsp-server/rtsp-media-factory.c:
6335 * gst/rtsp-server/rtsp-media-factory.h:
6336 * gst/rtsp-server/rtsp-media.c:
6337 Added vmethod create_pipeline to GstRTSPMediaFactory
6338 The pipeline is created in this method and the GstRTSPMedia's element is added to it
6340 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6342 * gst/rtsp-server/rtsp-client.c:
6343 client: use g_source_destroy()
6344 We need to use g_source_destroy() because we might have added the source to a
6345 different main context than the default one.
6347 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6349 * gst/rtsp-server/Makefile.am:
6350 * gst/rtsp-server/rtsp-client.c:
6351 * gst/rtsp-server/rtsp-params.c:
6352 * gst/rtsp-server/rtsp-params.h:
6353 rtsp: prepare for handling GET/SET_PARAMETER
6354 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
6356 Fix return codes of handlers.
6358 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6360 * gst/rtsp-server/rtsp-media.c:
6361 media: don't leak session pads
6363 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6365 * gst/rtsp-server/rtsp-media.c:
6366 media: clean up the messages a bit
6368 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6370 * gst/rtsp-server/rtsp-sdp.c:
6371 sdp: warn and skip streams without media
6373 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6375 * bindings/vala/gst-rtsp-server-0.10.vapi:
6376 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6377 vala: Fixed typo in header file of RTSPMediaStream
6379 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6381 * gst/rtsp-server/rtsp-media.c:
6384 Make dumping RTCP stats configurable
6386 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6388 * gst/rtsp-server/rtsp-media.c:
6389 media: be less verbose and leak less
6391 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6393 * gst/rtsp-server/rtsp-media.c:
6394 media: don't leak the destination address
6396 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6398 * gst/rtsp-server/rtsp-client.c:
6399 * gst/rtsp-server/rtsp-media.c:
6400 * gst/rtsp-server/rtsp-media.h:
6401 * gst/rtsp-server/rtsp-session.c:
6402 * gst/rtsp-server/rtsp-session.h:
6403 rtsp: use RTCP to keep the session alive
6404 Use the RTCP rtcp-from stats field to find the associated session and use this
6405 to keep the session alive.
6407 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6409 * gst/rtsp-server/rtsp-session.c:
6410 session: add 5sec to the real session timeout
6411 Allow the session to live 5sec longer before really timing out. This should give
6412 clients some extra time to keep the session active.
6414 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6416 * gst/rtsp-server/rtsp-client.c:
6417 client: replay OK to GET/SET_PARAMETER
6418 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
6419 so that we return OK for those requests.
6421 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6423 * gst/rtsp-server/rtsp-media.c:
6424 * gst/rtsp-server/rtsp-media.h:
6425 media: keep track of active transports
6426 Keep track of which transport is active to avoid closing the connection too
6428 Remove the destination transport also when going to NULL.
6429 Print some stats about the SDES and other RTCP messages we receive from the
6432 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6434 * examples/.gitignore:
6435 * examples/Makefile.am:
6436 * examples/test-sdp.c:
6437 example: add SDP relay example
6439 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6441 * gst/rtsp-server/rtsp-media.c:
6442 media: also count active TCP connections
6444 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6446 * gst/rtsp-server/rtsp-media-factory.c:
6447 * gst/rtsp-server/rtsp-media.c:
6448 * gst/rtsp-server/rtsp-media.h:
6449 rtsp: add support for dynamic elements
6450 Add support for dynamic elements.
6451 Don't set live pipelines back to paused.
6453 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6455 * gst/rtsp-server/rtsp-sdp.c:
6456 sdp: don't add encoding name when absent in caps
6458 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6460 * gst/rtsp-server/rtsp-client.c:
6461 client: warn when we can't do RTP-Info
6463 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6465 * gst/rtsp-server/rtsp-media-factory.c:
6466 factory: factor out the stream construction
6468 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6470 * gst/rtsp-server/rtsp-client.c:
6471 client: only add RTP-Info when we have the info
6472 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
6475 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6480 === release 0.10.3 ===
6482 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6486 - Fixes a bug where it put the wrong verion in pkgconfig
6487 - Link RTP and RTCP sources
6489 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6491 * gst/rtsp-server/rtsp-media.c:
6492 * gst/rtsp-server/rtsp-media.h:
6493 media: link the RTP udpsrc to the session manager
6494 Link the RTP udpsrc and the appsrc to the session manager so that they don't
6495 shut down when the client sends a packet to open firewalls.
6497 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6499 * pkgconfig/gst-rtsp-server.pc.in:
6500 Don't use hard-coded version number in pkg-config file
6502 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6507 === release 0.10.2 ===
6509 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6514 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6517 * common/m4/.gitignore:
6518 * examples/.gitignore:
6519 * pkgconfig/.gitignore:
6520 add some .gitignore files
6522 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6524 * gst/rtsp-server/rtsp-media.c:
6525 media: seek to key frames
6527 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6529 * gst/rtsp-server/rtsp-media.c:
6530 media: emit the unprepared signal by id
6531 Emit the unprepared signal by id instead of name and set the media as
6534 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6536 * gst/rtsp-server/rtsp-media.c:
6537 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
6539 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6541 * gst/rtsp-server/rtsp-server.c:
6542 Added finalize function to GstRTPSPServer to unref session pool and media mapping
6544 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6546 * bindings/vala/gst-rtsp-server-0.10.vapi:
6547 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6548 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6549 Updated vala bindings
6551 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6553 * gst/rtsp-server/Makefile.am:
6554 * gst/rtsp-server/rtsp-client.c:
6555 * gst/rtsp-server/rtsp-media.c:
6556 server: use appsink and appsrc with the API
6557 Use the appsink/appsrc API instead of the signals for higher
6560 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6562 * examples/test-ogg.c:
6563 tests: set the payload type correctly
6565 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6567 * gst/rtsp-server/rtsp-media-factory.c:
6568 factory: connect to the unprepare signal
6569 Connect to the unprepare signal for non-reusable media so that we can remove
6570 them from the cache.
6572 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6574 * gst/rtsp-server/rtsp-media.c:
6575 * gst/rtsp-server/rtsp-media.h:
6576 media: add signal to notify of unprepare
6578 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6580 * gst/rtsp-server/rtsp-media.c:
6581 * gst/rtsp-server/rtsp-media.h:
6582 media: more work on making the media shared
6583 Add a reusable flag to medias, indicating that they can be reused after a state
6587 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6589 * examples/test-readme.c:
6590 examples: mark the example as shared for testing
6592 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6594 * gst/rtsp-server/rtsp-media.c:
6595 * gst/rtsp-server/rtsp-media.h:
6596 client: support shared media
6597 Always perform the state actions even if the target state of the pipeline is
6598 already correct, we still want to add/remove the transports when we are dealing
6600 Keep a counter of the number of active transports for a media so that we can use
6601 this to perform a state change when needed.
6602 Perform a state change of the pipeline only when the first transport was added
6603 or when there are no active transports.
6605 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6607 * gst/rtsp-server/rtsp-client.c:
6608 client: fix refcounting crasher
6609 Don't need to remove the weak refs in the finalize methods, they are already
6610 removed in the dispose.
6611 Don't register the callback with a DestroyNofity.
6613 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6615 * gst/rtsp-server/rtsp-client.c:
6616 Fix rtsp client refcount management in TCP mode.
6617 Don't unref a client ref we never had. Fixes an unref
6618 of an already-free client object after a client
6619 teardown request for me.
6621 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6623 * gst/rtsp-server/rtsp-session.c:
6624 docs: fix typo in API docs
6626 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6628 * gst/rtsp-server/rtsp-media.c:
6630 Keep the udp sources in playing even if we go to paused. unlock the sources when
6632 Add some more debug info.
6633 Only seek when we need to.
6634 Keep track of the position when we go to paused.
6636 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6638 * gst/rtsp-server/rtsp-client.c:
6639 * gst/rtsp-server/rtsp-media.c:
6640 * gst/rtsp-server/rtsp-media.h:
6641 Add beginnings of seeking.
6642 Parse the Range header and perform a seek on the pipeline for the requested
6643 position. It's disabled currently until I figure out what's going wrong.
6645 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6647 * gst/rtsp-server/rtsp-client.c:
6648 allow pause requests for now.
6651 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6653 * gst/rtsp-server/rtsp-client.c:
6654 Remove weak ref on the session in teardown
6655 We need to remove our weakref from the session when we do a teardown because
6656 else we close the TCP connection prematurely.
6658 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6660 * gst/rtsp-server/rtsp-client.c:
6661 * gst/rtsp-server/rtsp-client.h:
6662 * gst/rtsp-server/rtsp-session-pool.c:
6663 Do some more session cleanup
6664 Make session timeout kill the TCP connection that currently watches the
6666 Remove the client timeout property.
6668 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6670 * gst/rtsp-server/rtsp-client.c:
6671 * gst/rtsp-server/rtsp-client.h:
6672 * gst/rtsp-server/rtsp-media.c:
6673 * gst/rtsp-server/rtsp-media.h:
6674 * gst/rtsp-server/rtsp-server.c:
6675 * gst/rtsp-server/rtsp-session.c:
6676 * gst/rtsp-server/rtsp-session.h:
6678 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
6681 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6683 * examples/Makefile.am:
6684 * examples/test-launch.c:
6685 Add example server that takes launch lines
6686 Add an example server that streams any -launch line.
6688 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6690 * examples/test-readme.c:
6691 * gst/rtsp-server/rtsp-client.c:
6692 * gst/rtsp-server/rtsp-media.c:
6693 * gst/rtsp-server/rtsp-media.h:
6694 Add support for live streams
6695 Add support for live streams and ranges
6696 Start on handling TCP data transfer.
6698 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6700 * gst/rtsp-server/rtsp-media.c:
6701 Free the pipeline before other things
6704 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6706 * gst/rtsp-server/rtsp-client.c:
6707 Only free the pending tunnel if there is one
6710 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6712 * gst/rtsp-server/rtsp-client.c:
6713 * gst/rtsp-server/rtsp-client.h:
6714 * gst/rtsp-server/rtsp-media.c:
6715 rtsp-server: Add support for tunneling
6716 Add support for tunneling over HTTP.
6717 Use new connection methods to retrieve the url.
6718 Dispatch messages based on the message type instead of blindly
6719 assuming it's always a request.
6720 Keep track of the watch id so that we can remove it later.
6721 Set the media pipeline to NULL before unreffing the pipeline.
6723 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6725 * gst/rtsp-server/rtsp-client.c:
6726 * gst/rtsp-server/rtsp-client.h:
6727 Fix for channel -> watch rename in gstreamer
6728 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
6730 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6732 * gst/rtsp-server/rtsp-client.c:
6733 * gst/rtsp-server/rtsp-client.h:
6735 Use the async RTSP channels instead of spawning a new thread for each client.
6736 If a sessionid is specified in a request, fail if we don't have the session.
6738 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6740 * gst/rtsp-server/rtsp-media.c:
6741 Add better debug info
6742 Add some better debug info.
6744 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6746 * examples/test-video.c:
6748 Add support for session timeouts in the example.
6750 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6752 * gst/rtsp-server/rtsp-session-pool.c:
6753 * gst/rtsp-server/rtsp-session-pool.h:
6754 Pass GTimeVal around for performance reasons
6755 Get the current time only once and pass it around so that sessions don't have to
6756 get the current time anymore.
6757 Add experimental support for a GSource that dispatches when the session needs to
6760 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6762 * gst/rtsp-server/rtsp-session.c:
6763 * gst/rtsp-server/rtsp-session.h:
6764 Add better support for session timeouts
6765 Add a method to request the number of milliseconds when a session will timeout.
6767 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6769 * gst/rtsp-server/rtsp-media.c:
6770 * gst/rtsp-server/rtsp-media.h:
6771 Add suport for RTP manager monitoring
6772 Add the first stage in monitoring the rtp manager.
6773 Make sure we don't update the state to something we don't want.
6775 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6777 * gst/rtsp-server/rtsp-client.c:
6778 Add support for session keepalive
6779 Get and update the session timeout for all requests. get the session as early as
6782 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6784 * gst/rtsp-server/rtsp-media-factory.h:
6785 * gst/rtsp-server/rtsp-media.c:
6786 * gst/rtsp-server/rtsp-media.h:
6787 Handle media bus messages
6788 Handle media bus messages in a custom mainloop and dispatch them to the
6789 RTSPMedia objects. Let the default implementation handle some common messages.
6791 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6793 * gst/rtsp-server/rtsp-client.c:
6794 * gst/rtsp-server/rtsp-session-pool.c:
6795 * gst/rtsp-server/rtsp-session.c:
6796 Some more session timeout handling
6797 Move the session header setting code to a central place so that we always add
6798 the timeout parameter too.
6799 Handle timeouts by running the session cleanup code.
6800 Stop media before cleaning up.
6802 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6804 * gst/rtsp-server/rtsp-client.c:
6805 * gst/rtsp-server/rtsp-client.h:
6806 Add timeout property
6807 Add a timeout property ot the client and make the other properties into GObject
6810 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6812 * gst/rtsp-server/rtsp-session-pool.c:
6813 Use getters and setters in property code
6814 Use the getters and setters for the timeout property instead of locking
6817 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6819 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
6821 2009-02-04 20:10:39 +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 * gst/rtsp-server/rtsp-session.c:
6826 * gst/rtsp-server/rtsp-session.h:
6827 Add more timeout stuff
6828 Add method to check if a session is expired.
6829 Add method to perform cleanup on a session pool.
6831 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6833 * gst/rtsp-server/rtsp-client.c:
6834 * gst/rtsp-server/rtsp-session-pool.c:
6835 * gst/rtsp-server/rtsp-session-pool.h:
6836 * gst/rtsp-server/rtsp-session.c:
6837 * gst/rtsp-server/rtsp-session.h:
6838 Add beginnings of session timeouts and limits
6839 Add the timeout value to the Session header for unusual timeout values.
6840 Allow us to configure a limit to the amount of active sessions in a pool. Set a
6841 limit on the amount of retry we do after a sessionid collision.
6842 Add properties to the sessionid and the timeout of a session. Keep track of
6843 creation time and last access time for sessions.
6845 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6847 * gst/rtsp-server/rtsp-client.c:
6848 * gst/rtsp-server/rtsp-media.c:
6849 * gst/rtsp-server/rtsp-media.h:
6850 * gst/rtsp-server/rtsp-sdp.c:
6851 * gst/rtsp-server/rtsp-session-pool.c:
6852 * gst/rtsp-server/rtsp-session.c:
6853 * gst/rtsp-server/rtsp-session.h:
6854 Cleanup of sessions and more
6855 Fix the refcounting of media and sessions in the client. Properly clean up the
6856 session data when the client performs a teardown.
6857 Add Server header to responses.
6858 Allow for multiple uri setups in one session.
6859 Add Range header to the PLAY response and add the range attribute to the SDP
6861 Fix the session pool remove method, it used the wrong key in the hashtable. Also
6862 give the ownership of the sessionid to the session object.
6864 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6866 * gst/rtsp-server/rtsp-server.c:
6867 * gst/rtsp-server/rtsp-server.h:
6869 Rename the 'server_port' variable to simply 'port'.
6871 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6874 * gst/rtsp-server/rtsp-client.c:
6875 * gst/rtsp-server/rtsp-media.c:
6876 * gst/rtsp-server/rtsp-media.h:
6877 * gst/rtsp-server/rtsp-session.c:
6878 * gst/rtsp-server/rtsp-session.h:
6879 Rework the way we handle transports for streams
6880 Make the media accept an array of transports for the streams that we have
6881 configured for the play/pause requests.
6882 Implement server states for a client and its media.
6883 Require 0.10.22.1 (git HEAD) of gstreamer.
6885 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6887 * gst/rtsp-server/rtsp-client.c:
6888 * gst/rtsp-server/rtsp-media-factory.c:
6889 Drop const from functions dealing with urls
6890 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
6891 have the right const in them.
6893 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6895 * gst/rtsp-server/rtsp-client.c:
6896 * gst/rtsp-server/rtsp-media.c:
6897 * gst/rtsp-server/rtsp-sdp.c:
6901 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6903 * gst/rtsp-server/rtsp-client.c:
6904 * gst/rtsp-server/rtsp-media-factory.c:
6905 * gst/rtsp-server/rtsp-media.c:
6906 * gst/rtsp-server/rtsp-media.h:
6908 Don't keep a reference to the GstRTSPMedia in the stream.
6909 Free more things when freeing the GstRTSPMedia.
6911 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6914 * gst/rtsp-server/rtsp-media-factory.c:
6915 * gst/rtsp-server/rtsp-media-factory.h:
6916 * gst/rtsp-server/rtsp-media.c:
6917 * gst/rtsp-server/rtsp-media.h:
6918 * gst/rtsp-server/rtsp-server.c:
6919 * gst/rtsp-server/rtsp-server.h:
6920 More docs and small cleanups
6921 Add some more docs and update the README
6922 Cleanup some method names.
6923 Remove an unneeded idx field in the GstRTSPMediaStream
6925 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6928 * examples/Makefile.am:
6929 * examples/test-readme.c:
6930 Add a README and more example code
6931 Add a README file that contains a small introduction on how to use the server
6932 along with the example code explained in the readme.
6934 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6936 * gst/rtsp-server/rtsp-media.c:
6937 * gst/rtsp-server/rtsp-server.c:
6938 Fix some leaks and change default port
6939 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
6940 we finished the initial preroll. If we keep them locked, setting the pipeline to
6941 NULL will not stop and clean up the sources correctly.
6942 Change the default RTSP port to 8554 aka the official alternative RTSP port.
6944 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6946 * gst/rtsp-server/rtsp-session.c:
6947 * gst/rtsp-server/rtsp-session.h:
6948 Cleanups to the session object
6949 Remove some unneeded variables in the session state of a stream such as the
6950 owner media and the server transport.
6951 Get the configuration of a media stream in a session based on the media_stream
6952 in the original object instead of our cached index.
6953 Free more data in the finalize method.
6955 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6957 * gst/rtsp-server/rtsp-client.c:
6958 * gst/rtsp-server/rtsp-client.h:
6959 Cleanups and reuse media from DESCRIBE
6960 Handle thread create errors.
6961 Rename some internal methods to better match what they actually do.
6962 Handle misconfiguration of session_pool and media_mapping gracefully.
6963 Cache the DESCRIBE media and uri in the client connection and reuse them when
6964 we receive a SETUP request in the same connection for the same uri.
6965 Cleanup the client connection object.
6967 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6969 * gst/rtsp-server/rtsp-media-factory.c:
6970 * gst/rtsp-server/rtsp-media-factory.h:
6971 * gst/rtsp-server/rtsp-media.c:
6972 * gst/rtsp-server/rtsp-media.h:
6973 Add shared properties to media and factory
6974 Add the shared property to media.
6975 Implement some simple caching in the factory depending on if the media is shared
6978 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6980 * gst/rtsp-server/rtsp-client.c:
6981 Add a little comment
6982 Add some comment about the content-base header.
6984 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6986 * examples/Makefile.am:
6988 * examples/test-mp4.c:
6989 * examples/test-ogg.c:
6990 * examples/test-video.c:
6991 * gst/rtsp-server/Makefile.am:
6992 * gst/rtsp-server/rtsp-client.c:
6993 * gst/rtsp-server/rtsp-client.h:
6994 * gst/rtsp-server/rtsp-media-factory.c:
6995 * gst/rtsp-server/rtsp-media-factory.h:
6996 * gst/rtsp-server/rtsp-media.c:
6997 * gst/rtsp-server/rtsp-media.h:
6998 * gst/rtsp-server/rtsp-sdp.c:
6999 * gst/rtsp-server/rtsp-sdp.h:
7000 * gst/rtsp-server/rtsp-server.c:
7001 * gst/rtsp-server/rtsp-server.h:
7002 * gst/rtsp-server/rtsp-session.c:
7003 * gst/rtsp-server/rtsp-session.h:
7004 Reorganize things, prepare for media sharing
7005 Added various other test server examples
7006 Move the SDP message generation to a separate helper.
7007 Refactor common code for finding the session.
7008 Add content-base for realplayer compatibility
7009 Clean up request uris before processing for better vlc compatibility.
7010 Move prerolling and pipeline construction to the RTSPMedia object.
7011 Use multiudpsink for future pipeline reuse.
7013 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7019 === release 0.10.1 ===
7021 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7027 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7029 * bindings/vala/Makefile.am:
7031 Add more directories and files to the dist.
7033 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7035 * bindings/python/Makefile.am:
7036 * bindings/python/rtspserver.override:
7037 Fixed compile error of python bindings
7039 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7041 * bindings/vala/gst-rtsp-server-0.10.vapi:
7042 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7043 Marked values as nullable accordingly
7045 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7047 * bindings/vala/gst-rtsp-server-0.10.vapi:
7048 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
7049 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7050 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7051 Updated Vala bindings
7053 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7055 * gst/rtsp-server/rtsp-client.c:
7056 * gst/rtsp-server/rtsp-media-mapping.c:
7057 * gst/rtsp-server/rtsp-media-mapping.h:
7058 * gst/rtsp-server/rtsp-media.h:
7059 * gst/rtsp-server/rtsp-session-pool.h:
7060 Cleanups and doc updates
7061 Add some more documentation and do some minor cleanups here and there.
7063 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7065 * gst/rtsp-server/rtsp-client.c:
7066 * gst/rtsp-server/rtsp-media-factory.c:
7067 * gst/rtsp-server/rtsp-media-factory.h:
7068 * gst/rtsp-server/rtsp-media.c:
7069 * gst/rtsp-server/rtsp-media.h:
7070 * gst/rtsp-server/rtsp-session.c:
7071 * gst/rtsp-server/rtsp-session.h:
7073 Rename GstRTSPMediaBin to GstRTSPMedia
7074 Parse the request url into a GstRTSPUri object and pass this object to the
7075 various handlers and methods that require the uri.
7077 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7081 Add some more docs and remove some old code from the example.
7083 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7085 * gst/rtsp-server/rtsp-client.c:
7086 Handle state change failures better
7087 Handle state change failures better when changing the state of the pipeline to
7090 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7092 * gst/rtsp-server/rtsp-media-factory.c:
7093 * gst/rtsp-server/rtsp-media-factory.h:
7094 Make element creation more extendible
7095 Add get_element vmethod to the default MediaFactory so that subclasses can just
7096 override that method and still use the default logic for making a MediaBin from
7099 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7102 * gst/rtsp-server/Makefile.am:
7103 * gst/rtsp-server/rtsp-client.c:
7104 * gst/rtsp-server/rtsp-client.h:
7105 * gst/rtsp-server/rtsp-media-factory.c:
7106 * gst/rtsp-server/rtsp-media-factory.h:
7107 * gst/rtsp-server/rtsp-media-mapping.c:
7108 * gst/rtsp-server/rtsp-media-mapping.h:
7109 * gst/rtsp-server/rtsp-media.c:
7110 * gst/rtsp-server/rtsp-media.h:
7111 * gst/rtsp-server/rtsp-server.c:
7112 * gst/rtsp-server/rtsp-server.h:
7113 * gst/rtsp-server/rtsp-session.c:
7114 * gst/rtsp-server/rtsp-session.h:
7115 Make the server handle arbitrary pipelines
7116 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
7117 The GstMediaBin object has a handle to a bin with elements and to a list of
7118 GstMediaStream objects that this bin produces.
7119 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
7120 with methods to register and remove those mappings.
7121 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
7122 used by the server instance.
7123 Modify the example application so that it shows how to create custom pipelines
7124 attached to a specific mount point.
7125 Various misc cleanps.
7127 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7129 * gst/rtsp-server/rtsp-server.c:
7130 * gst/rtsp-server/rtsp-server.h:
7131 Allow setting a custom media factory for a server
7133 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7135 * gst/rtsp-server/rtsp-client.c:
7136 * gst/rtsp-server/rtsp-client.h:
7137 Allow setting a custom media factory for a client.
7139 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7141 * gst/rtsp-server/Makefile.am:
7142 Add Makefile entry for the media factory
7144 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7146 * gst/rtsp-server/rtsp-media-factory.c:
7147 * gst/rtsp-server/rtsp-media-factory.h:
7148 Add media factory to map urls to media pipeline objects.
7150 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7152 * gst/rtsp-server/rtsp-media.c:
7153 * gst/rtsp-server/rtsp-media.h:
7154 Add comments. Remove unused field
7156 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7158 * gst/rtsp-server/rtsp-session-pool.c:
7159 * gst/rtsp-server/rtsp-session-pool.h:
7160 Allow custom session pools to override the session id allocation algorithms Add some comments.
7162 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7164 * gst/rtsp-server/rtsp-session.h:
7167 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7169 * gst/rtsp-server/rtsp-client.c:
7170 * gst/rtsp-server/rtsp-client.h:
7171 Move the connection code in one place Add some comments
7173 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7175 * gst/rtsp-server/rtsp-server.c:
7176 * gst/rtsp-server/rtsp-server.h:
7177 Make vmethod to create and accept new clients. Add some docs.
7179 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7181 * gst/rtsp-server/rtsp-server.c:
7182 * gst/rtsp-server/rtsp-server.h:
7183 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
7185 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7187 * gst/rtsp-server/rtsp-client.c:
7188 * gst/rtsp-server/rtsp-client.h:
7189 Name the parameters more appropriately.
7191 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7193 * gst/rtsp-server/rtsp-session-pool.c:
7194 Do some more cleanup of the session pool.
7196 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7198 * gst/rtsp-server/Makefile.am:
7199 * gst/rtsp-server/rtsp-client.c:
7200 Check if return value of gst_rtsp_session_get_media is not NULL
7202 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7204 * gst/rtsp-server/Makefile.am:
7205 Install rtsp-session and rtsp-session-pool headers
7207 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7212 * bindings/python/Makefile.am:
7213 * bindings/python/arg-types.py:
7214 * bindings/python/codegen/Makefile.am:
7215 * bindings/python/codegen/__init__.py:
7216 * bindings/python/codegen/argtypes.py:
7217 * bindings/python/codegen/code-coverage.py:
7218 * bindings/python/codegen/codegen.py:
7219 * bindings/python/codegen/definitions.py:
7220 * bindings/python/codegen/defsparser.py:
7221 * bindings/python/codegen/docextract.py:
7222 * bindings/python/codegen/docgen.py:
7223 * bindings/python/codegen/fileprefix.override:
7224 * bindings/python/codegen/fileprefixmodule.c:
7225 * bindings/python/codegen/h2def.py:
7226 * bindings/python/codegen/mergedefs.py:
7227 * bindings/python/codegen/mkskel.py:
7228 * bindings/python/codegen/override.py:
7229 * bindings/python/codegen/reversewrapper.py:
7230 * bindings/python/codegen/scmexpr.py:
7231 * bindings/python/rtspserver-types.defs:
7232 * bindings/python/rtspserver.defs:
7233 * bindings/python/rtspserver.override:
7234 * bindings/python/rtspservermodule.c:
7236 Add python bindings.
7238 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7240 * bindings/Makefile.am:
7242 Don't go into python dir when requirements for python bindings are missing
7244 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7246 * bindings/Makefile.am:
7247 * bindings/vala/Makefile.am:
7249 Install Vala bindings if vala is available
7251 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7253 * bindings/vala/gst-rtsp-server-0.10.deps:
7254 * bindings/vala/gst-rtsp-server-0.10.vapi:
7255 * bindings/vala/gst-rtsp-server.vapi:
7256 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
7257 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
7258 * bindings/vala/packages/gst-rtsp-server-0.10.files:
7259 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7260 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7261 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
7262 * bindings/vala/packages/gst-rtsp-server.deps:
7263 * bindings/vala/packages/gst-rtsp-server.excludes:
7264 * bindings/vala/packages/gst-rtsp-server.files:
7265 * bindings/vala/packages/gst-rtsp-server.gi:
7266 * bindings/vala/packages/gst-rtsp-server.metadata:
7267 * bindings/vala/packages/gst-rtsp-server.namespace:
7268 Regenerated Vala bindings
7270 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7272 * bindings/vala/gst-rtsp-server.vapi:
7273 * bindings/vala/packages/gst-rtsp-server.metadata:
7274 Fixed typo in included headers for vala bindings
7276 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7280 * pkgconfig/Makefile.am:
7281 * pkgconfig/gst-rtsp-server.pc.in:
7282 Added pkgconfig file
7284 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
7286 * bindings/vala/gst-rtsp-server.vapi:
7287 * bindings/vala/packages/gst-rtsp-server.excludes:
7288 * bindings/vala/packages/gst-rtsp-server.gi:
7289 * bindings/vala/packages/gst-rtsp-server.metadata:
7290 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
7292 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
7294 * bindings/vala/gst-rtsp-server.vapi:
7295 * bindings/vala/packages/gst-rtsp-server.deps:
7296 * bindings/vala/packages/gst-rtsp-server.files:
7297 * bindings/vala/packages/gst-rtsp-server.gi:
7298 * bindings/vala/packages/gst-rtsp-server.metadata:
7299 * bindings/vala/packages/gst-rtsp-server.namespace:
7302 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
7304 * gst/rtsp-server/rtsp-session.c:
7305 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
7307 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
7309 * examples/Makefile.am:
7310 * gst/rtsp-server/Makefile.am:
7311 Put GStreamer version in library name
7313 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7315 * examples/Makefile.am:
7316 * gst/rtsp-server/Makefile.am:
7317 Fix some issues to pass distcheck
7319 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7321 * gst/rtsp-server/rtsp-server.c:
7322 Added port property to GstRTSPServer class.
7324 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7329 * examples/Makefile.am:
7332 * gst/rtsp-server/Makefile.am:
7333 * gst/rtsp-server/rtsp-client.c:
7334 * gst/rtsp-server/rtsp-client.h:
7335 * gst/rtsp-server/rtsp-media.c:
7336 * gst/rtsp-server/rtsp-media.h:
7337 * gst/rtsp-server/rtsp-server.c:
7338 * gst/rtsp-server/rtsp-server.h:
7339 * gst/rtsp-server/rtsp-session-pool.c:
7340 * gst/rtsp-server/rtsp-session-pool.h:
7341 * gst/rtsp-server/rtsp-session.c:
7342 * gst/rtsp-server/rtsp-session.h:
7345 * src/rtsp-client.c:
7346 * src/rtsp-client.h:
7349 * src/rtsp-server.c:
7350 * src/rtsp-server.h:
7351 * src/rtsp-session-pool.c:
7352 * src/rtsp-session-pool.h:
7353 * src/rtsp-session.c:
7354 * src/rtsp-session.h:
7355 Split in library and example program
7357 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
7359 * src/rtsp-client.h:
7360 Removed obsolete variable
7362 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
7364 * src/rtsp-client.c:
7365 * src/rtsp-client.h:
7366 Removed pipeline variable GstRTSPClient, because it's only used in one function
7368 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7371 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
7373 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
7375 * src/rtsp-session.c:
7376 Initialize some more vars.
7378 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
7380 * src/rtsp-session.c:
7381 Initialize variable to avoid compiler warning.
7383 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
7386 Add a reasonable generic .gitignore