3 2016-07-06 Sebastian Dröge <slomo@coaxion.net>
8 2016-06-24 02:02:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
11 configure: Need to add -DGST_STATIC_COMPILATION when building only statically
12 https://bugzilla.gnome.org/show_bug.cgi?id=767463
14 2016-06-21 11:49:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
17 Automatic update of common submodule
18 From ac2f647 to f363b32
20 2016-04-14 22:56:11 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
22 * gst/rtsp-server/rtsp-sdp.c:
23 * gst/rtsp-server/rtsp-sdp.h:
24 * gst/rtsp-server/rtsp-stream.c:
25 * gst/rtsp-server/rtsp-stream.h:
26 sdp: add rollover counters for all sender SSRC
27 We add different crypto sessions in MIKEY, one for each sender
28 SSRC. Currently, all of them will have the same security policy, 0.
29 The rollover counters are obtained from the srtpenc element using the
31 https://bugzilla.gnome.org/show_bug.cgi?id=730539
33 2016-06-07 20:44:42 +0100 Tim-Philipp Müller <tim@centricular.com>
35 * gst/rtsp-server/rtsp-media-factory.h:
36 * gst/rtsp-server/rtsp-server.h:
39 2016-05-25 10:28:43 +0100 Tim-Philipp Müller <tim@centricular.com>
41 * gst/rtsp-server/Makefile.am:
42 g-i: pass compiler env to g-ir-scanner
43 It's what introspection.mak does as well. Should
44 fix spurious build failures on gnome-continuous
45 (caused by g-ir-scanner getting compiler details
46 via python which is broken in some environments
47 so passing the compiler details bypasses that).
49 2016-05-18 16:48:44 +0100 Ian <ian.arkver.dev@gmail.com>
51 * gst/rtsp-server/rtsp-session.c:
52 rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header
53 This works with rtspsrc and live555, but fails with e.g. ffmpeg.
54 https://bugzilla.gnome.org/show_bug.cgi?id=766619
56 2016-03-07 14:48:38 +0100 Edward Hervey <bilboed@bilboed.com>
58 * gst/rtsp-sink/gstrtspclientsink.c:
59 rtspclientsink: Check return value of sscanf
60 And just make sure we always have 0/0 if we have an error
63 2016-04-25 08:55:25 -0400 Jake Foytik <jake.foytik@ipconfigure.com>
65 * gst/rtsp-server/rtsp-stream.c:
66 * tests/check/gst/rtspserver.c:
67 * tests/check/gst/stream.c:
68 rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
69 - Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
70 - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
71 - Create unit test for shared media.
72 https://bugzilla.gnome.org/show_bug.cgi?id=764744
74 2016-04-11 10:55:23 +0300 Sebastian Dröge <sebastian@centricular.com>
76 * gst/rtsp-server/rtsp-stream.c:
77 rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
78 For IPv6 addresses, binding to a multicast group does not work on Linux
79 either. Always bind to ANY and then later join the multicast group.
80 https://bugzilla.gnome.org/show_bug.cgi?id=764679
82 2016-04-14 10:05:02 +0100 Julien Isorce <j.isorce@samsung.com>
85 Automatic update of common submodule
86 From 6f2d209 to ac2f647
88 2016-04-06 10:09:46 +0200 Patricia Muscalu <patricia@axis.com>
90 * gst/rtsp-server/rtsp-thread-pool.c:
91 rtsp-thread-pool: explained why GSource is a part of ThreadImpl
92 Clarified why it is necessary to add source information to
93 GstRTSPThreadImpl. See the reported bug in GLib:
94 https://bugzilla.gnome.org/show_bug.cgi?id=720186
96 https://bugzilla.gnome.org/show_bug.cgi?id=761702
98 2016-04-04 12:58:38 +0300 Sebastian Dröge <sebastian@centricular.com>
100 * examples/Makefile.am:
101 examples: Clean up CFLAGS/LDADD even more
102 The internal .la should come first and is part of LDADD, as is
105 2016-04-04 12:39:39 +0300 Sebastian Dröge <sebastian@centricular.com>
107 * examples/Makefile.am:
108 examples: Clean up CFLAGS/LDADD to link with the correct versions of all libraries
110 2016-04-03 12:06:29 +0300 Sebastian Dröge <sebastian@centricular.com>
112 * gst/rtsp-server/Makefile.am:
113 rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS)
115 2015-12-30 18:39:05 +0200 Sebastian Dröge <sebastian@centricular.com>
117 * gst/rtsp-server/rtsp-client.c:
118 * gst/rtsp-server/rtsp-media-factory.c:
119 * gst/rtsp-server/rtsp-media-factory.h:
120 * gst/rtsp-server/rtsp-media.c:
121 * gst/rtsp-server/rtsp-media.h:
122 * gst/rtsp-server/rtsp-sdp.c:
123 * gst/rtsp-server/rtsp-stream.c:
124 * gst/rtsp-server/rtsp-stream.h:
125 rtsp-server: Implement clock signalling according to RFC7273
126 For NTP and PTP clocks we signal the actual clock that is used and signal
127 the direct media clock offset.
128 For all other clocks we at least signal that it's the local sender clock.
129 This allows receivers to know which clock was used to generate the media and
130 its RTP timestamps. Receivers can then implement network synchronization,
131 either absolute or at least relative by getting the sender clock rate directly
132 via NTP/PTP instead of estimating it from RTP timestamps and packet receive
134 https://bugzilla.gnome.org/show_bug.cgi?id=760005
136 2016-03-02 19:42:58 +0200 Sebastian Dröge <sebastian@centricular.com>
138 * gst/rtsp-sink/gstrtspclientsink.c:
139 rtspclientsink: Add support for setting the multicast interface
140 https://bugzilla.gnome.org/show_bug.cgi?id=763000
142 2016-03-02 19:42:13 +0200 Sebastian Dröge <sebastian@centricular.com>
144 * gst/rtsp-server/rtsp-media-factory.c:
145 * gst/rtsp-server/rtsp-media-factory.h:
146 * gst/rtsp-server/rtsp-media.c:
147 * gst/rtsp-server/rtsp-media.h:
148 * gst/rtsp-server/rtsp-stream.c:
149 * gst/rtsp-server/rtsp-stream.h:
150 rtsp-media: Add support for setting the multicast interface
151 https://bugzilla.gnome.org/show_bug.cgi?id=763000
153 2016-03-07 08:50:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
155 * gst/rtsp-sink/gstrtspclientsink.c:
156 rtspclientsink: use new gst_element_class_add_static_pad_template()
157 https://bugzilla.gnome.org/show_bug.cgi?id=763196
159 2016-03-24 13:33:43 +0200 Sebastian Dröge <sebastian@centricular.com>
164 === release 1.8.0 ===
166 2016-03-24 13:00:35 +0200 Sebastian Dröge <sebastian@centricular.com>
172 * gst-rtsp-server.doap:
175 2016-03-16 23:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
177 * gst/rtsp-server/rtsp-stream.c:
178 rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
179 This would get us NO_PREROLL in the bin again and break seeking.
180 Thanks to Carlos Rafael Giani for helping to debug this!
181 https://bugzilla.gnome.org/show_bug.cgi?id=740509
183 === release 1.7.91 ===
185 2016-03-15 12:26:13 +0200 Sebastian Dröge <sebastian@centricular.com>
191 * gst-rtsp-server.doap:
194 2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
196 * gst/rtsp-server/rtsp-stream.c:
197 rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
198 Without this, RECORD pipelines are broken because
199 a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
200 added later. Previously it was there earlier and due to NO_PREROLL caused the
201 pipeline to preroll immediately
202 b) the udpsrc for the pipeline is added later and never set to PLAYING state,
203 as the corresponding code previously was only for PLAY pipelines.
204 https://bugzilla.gnome.org/show_bug.cgi?id=763281
206 2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
208 * gst/rtsp-server/rtsp-stream.c:
209 rtsp-stream: Fix typo in the docstring
210 gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
212 2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
214 * gst/rtsp-server/rtsp-stream.c:
215 rtsp-stream: Disable multicast loopback for all our sockets
216 On Windows this is a receiver-side setting, on Linux a sender-side setting. As
217 we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
218 loopback setting on the socket... while udpsink does which unfortunately has
219 no effect here on Windows but on Linux.
220 https://bugzilla.gnome.org/show_bug.cgi?id=757488
222 2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
224 * tests/check/gst/stream.c:
225 stream tests: added new tests
226 Test a case when the address pool only contains multicast addresses
227 and the client is requesting unicast udp.
228 Added tests for multicast ports allocation.
229 https://bugzilla.gnome.org/show_bug.cgi?id=757488
231 2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
233 * gst/rtsp-server/rtsp-stream.c:
234 rtsp-stream: Only bind multicast sockets to ANY on Windows
235 On Linux it is still needed to bind to the multicast address
236 to filter out random other packets, while on Windows binding
237 to multicast addresses just fails.
239 2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
241 * gst/rtsp-server/rtsp-stream.c:
242 rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
243 Otherwise we fail to allocate UDP ports if the pool only contains multicast
244 addresses, which is something that used to work before. For unicast addresses
245 if the pool contains none, we just allocate them as if there is no pool at
247 https://bugzilla.gnome.org/show_bug.cgi?id=757488
249 2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
251 * gst/rtsp-server/rtsp-client.c:
252 * gst/rtsp-server/rtsp-stream.c:
253 rtsp-server: Fix indentation
255 2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
257 * gst/rtsp-server/rtsp-stream.c:
258 rtsp-stream: Don't bind the sockets to multicast addresses
259 This works on Linux but fails completely on Windows. You're supposed
260 to bind to ANY and then join the multicast group.
261 https://bugzilla.gnome.org/show_bug.cgi?id=757488
263 === release 1.7.90 ===
265 2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
271 * gst-rtsp-server.doap:
274 2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
277 Automatic update of common submodule
278 From b64f03f to 6f2d209
280 2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
282 * gst/rtsp-sink/gstrtspclientsink.c:
283 * tests/check/gst/rtspclientsink.c:
284 rtspsink: Fix some leaks in rtspclientsink and the unit test.
285 https://bugzilla.gnome.org/show_bug.cgi?id=762525
287 2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
289 * tests/check/gst/media.c:
290 * tests/check/gst/rtspclientsink.c:
291 * tests/check/gst/rtspserver.c:
292 * tests/check/gst/stream.c:
293 tests: unit test fixes
294 Removed port allocation test from the media suite.
295 The port allocation failure is now in the stream suite.
297 Make sure that the media is suspended after the DESCRIBE request
298 before reconfiguring the UDP sinks.
300 In the RECORD case we have to set async property to false
301 for the appsink element in the test in order to make sure
302 that the media pipeline doesn't hang in start_preroll().
303 https://bugzilla.gnome.org/show_bug.cgi?id=757488
305 2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
307 * gst/rtsp-server/rtsp-client.c:
308 * gst/rtsp-server/rtsp-stream.c:
309 * gst/rtsp-server/rtsp-stream.h:
310 rtsp-stream: postpone UDP socket allocation until SETUP
311 Postpone the allocation of the UDP sockets until we know
312 what transport has been chosen by the client.
313 Both unicast and multicast UDP sources are created in one
315 https://bugzilla.gnome.org/show_bug.cgi?id=757488
317 2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
319 * gst/rtsp-server/rtsp-stream.c:
320 rtsp-stream: postpone the creation of the UDP sources
321 Code refactoring: allocate the UDP ports after the sender and
322 the reciver parts have been created.
323 We postpone the creation of the UDP sources until the UDP
324 ports have been allocated.
325 https://bugzilla.gnome.org/show_bug.cgi?id=757488
327 2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
329 * gst/rtsp-server/rtsp-stream.c:
330 rtsp-stream: added function for setting UDP sources to PLAYING state
331 Code refactoring: Introduced a function for setting UDP sources
333 https://bugzilla.gnome.org/show_bug.cgi?id=757488
335 2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
337 * gst/rtsp-server/rtsp-stream.c:
338 rtsp-stream: added function for creating and configuring UDP sources
339 Code refactoring: create and configure UDP sources in a separate function.
340 https://bugzilla.gnome.org/show_bug.cgi?id=757488
342 2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
344 * gst/rtsp-server/rtsp-stream.c:
345 rtsp-stream: added function for RTP/RTCP socket configuration
346 Code refactoring: configure RTP and RTCP sockets for UDP sinks
347 in a separate function.
348 https://bugzilla.gnome.org/show_bug.cgi?id=757488
350 2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
352 * gst/rtsp-server/rtsp-stream.c:
353 rtsp-stream: added function for creating and configuring UDP sinks
354 Code refactoring: create and configure UDP sinks in a separate function.
355 https://bugzilla.gnome.org/show_bug.cgi?id=757488
357 2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
359 * gst/rtsp-server/rtsp-stream.c:
360 rtsp-stream: added helper function for creating the sender/receiver parts
361 Code refactoring: introduced helper function for creating
362 the receiver and the sender parts of the streaming pipeline.
363 https://bugzilla.gnome.org/show_bug.cgi?id=757488
365 2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
370 === release 1.7.2 ===
372 2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
378 * gst-rtsp-server.doap:
381 2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
383 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
384 uninstalled.pc: add support for non libtool build systems
385 Currently the .la path is provided which requires to use libtool as
386 mentioned in the GStreamer manual section-helloworld-compilerun.html.
387 It is fine as long as the application is built using libtool.
388 So currently it is not possible to compile a GStreamer application
389 within gst-uninstalled with CMake or other build system different
391 This patch allows to do the following in gst-uninstalled env:
392 gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
393 gstreamer-rtsp-server-1.0)
394 Previously it required to prepend libtool --mode=link
395 https://bugzilla.gnome.org/show_bug.cgi?id=720778
397 2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
399 * gst/rtsp-sink/gstrtspclientsink.c:
400 rtspclientsink: remove check for impossible condition
401 Goto error label checks stream to see if it needs to be unreferenced before
402 returning, but this goto jumps happens before the stream is ever set, so it
403 will always be NULL in this error label.
406 2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
408 * gst/rtsp-sink/gstrtspclientsink.c:
409 rtspclientsink: clean switch statements
410 Coverity demands for fallthrough statements to be clearly commented,
411 to distinguish from accidental fall throughs. And it also needs all
412 cases to finish with a break, even if the break is never going to be
413 executed like in the case of a continue jump.
417 2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
419 * tests/check/Makefile.am:
420 tests: extend the AM_TESTS_ENVIRONMENT from check.mak
421 To get the CK_DEFAULT_TIMEOUT defined for all tests
422 Also removes a 120 seconds timeout that was set as default
423 explicitly in this module
424 https://bugzilla.gnome.org/show_bug.cgi?id=761472
426 2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
430 Automatic update of common submodule
431 From 86e4663 to b64f03f
433 2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
435 * gst/rtsp-server/rtsp-media.c:
436 rtsp-media: fix state_lock not locked again when preroll fails
437 https://bugzilla.gnome.org/show_bug.cgi?id=761399
439 2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
442 configure: Move plugin specific flags below all the others
443 They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
444 -no-undefined. And -no-undefined is required on Windows to build DLLs.
446 2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
448 * gst/rtsp-sink/gstrtspclientsink.c:
449 rtspclientsink: Simplify slightly using new -base API
450 Use the new Mikey and SDP API in the base plugins libs
451 to simplify some code.
452 https://bugzilla.gnome.org/show_bug.cgi?id=758180
454 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
459 * gst/rtsp-sink/Makefile.am:
460 * gst/rtsp-sink/gstrtspclientsink.c:
461 * gst/rtsp-sink/gstrtspclientsink.h:
462 * gst/rtsp-sink/plugin.c:
463 * tests/check/Makefile.am:
464 * tests/check/gst/rtspclientsink.c:
465 rtspsink: Add rtspclientsink element
466 Add an rtspclientsink element that accepts streams for which
467 there is a registered payloader and sends them to
468 an RTSP server using RECORD.
469 Sending is synchronised to the pipeline clock. Payload-types
470 are automatically selected. The 'new-payloader' signal is fired
471 for custom configuration of payloaders when they are created.
472 Can now stream a movie like this:
474 ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
475 decodebin name=depay1 ! audioconvert ! autoaudiosink )"
477 gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
478 queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
479 https://bugzilla.gnome.org/show_bug.cgi?id=758180
481 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
483 * gst/rtsp-server/rtsp-stream.c:
484 * gst/rtsp-server/rtsp-stream.h:
485 rtsp-stream: Add functions for using rtsp-stream from the client
486 Add a boolean to indicate that the rtsp-stream is running on the
487 'client' side of an RTSP connection, for sending streams via
488 RECORD. In that case, the roles of the client/server ports
489 in transport setup are swapped.
490 https://bugzilla.gnome.org/show_bug.cgi?id=758180
492 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
494 * gst/rtsp-server/rtsp-sdp.c:
495 * gst/rtsp-server/rtsp-sdp.h:
496 rtsp-sdp: Add gst_rtsp_sdp_from_stream()
497 A new function that adds info from a GstRTSPStream into an SDP message.
498 https://bugzilla.gnome.org/show_bug.cgi?id=758180
500 2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
502 * gst/rtsp-server/rtsp-media.c:
503 rtsp-media: Fix mutex beeing unlocked while they should be locked
504 https://bugzilla.gnome.org/show_bug.cgi?id=761226
506 2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
508 * gst/rtsp-server/rtsp-media-factory.c:
509 rtsp-media-factory: add missing break in "clock" property setter
512 2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
514 * gst/rtsp-server/rtsp-stream.c:
515 rtsp-stream: fixed assert during update transport
516 When RTSP server trying update transport during multicast, it throws an
517 assert. The assert is thrown because it is trying to get the parent of
518 an non-existing funnel element.
519 https://bugzilla.gnome.org/show_bug.cgi?id=760150
521 2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
523 * gst/rtsp-server/rtsp-permissions.h:
524 * gst/rtsp-server/rtsp-thread-pool.h:
525 * gst/rtsp-server/rtsp-token.h:
526 docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
527 gtk-doc can handle static inline functions just fine these days,
528 there's no need for this stuff any more.
530 2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
532 * gst/rtsp-server/rtsp-media.c:
533 * gst/rtsp-server/rtsp-sdp.c:
534 sdp: replace duplicated codes to call new base sdp apis
535 https://bugzilla.gnome.org/show_bug.cgi?id=745880
537 2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
539 * examples/test-netclock.c:
540 test-netclock: Use the new API to configure a clock directly
542 2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
544 * gst/rtsp-server/rtsp-media-factory.c:
545 * gst/rtsp-server/rtsp-media-factory.h:
546 * gst/rtsp-server/rtsp-media.c:
547 * gst/rtsp-server/rtsp-media.h:
548 rtsp-media: Add API to directly configure a clock on the media pipelines
550 2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
552 * gst/rtsp-server/rtsp-media.c:
553 rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
555 2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
557 * gst/rtsp-server/rtsp-media-factory.c:
558 rtsp-media-factory: Add FIXME for 2.0
560 2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
562 * gst/rtsp-server/rtsp-stream.c:
563 rtsp-stream: Fix indentation
565 2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
567 * gst/rtsp-server/rtsp-media.c:
568 rtsp-media: Do not prepare media after media times out
569 Deferred calls to start_prepare() can be deferred past the point until
570 which wait_preroll() and by proxy gst_rtsp_media_get_status() is
571 prepared to wait. Previously there was no lock and no check for this
572 situation. This meant that a media could be prepared and unprepared
573 simultaneously by two different threads. Now a lock is in place and a
574 suitable check is done.
575 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
577 2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
579 * gst/rtsp-server/rtsp-client.c:
580 * gst/rtsp-server/rtsp-media-factory.c:
581 * gst/rtsp-server/rtsp-media-factory.h:
582 * gst/rtsp-server/rtsp-media.c:
583 * gst/rtsp-server/rtsp-media.h:
584 rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
585 Without TEARDOWN it might be desireable to keep the media running and continue
586 sending data to the client, even if the RTSP connection itself is
588 Only do this for session medias that have only UDP transports. If there's at
589 least on TCP transport, it will stop working and cause problems when the
590 connection is disconnected.
591 https://bugzilla.gnome.org/show_bug.cgi?id=758999
593 2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
598 === release 1.7.1 ===
600 2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
606 * gst-rtsp-server.doap:
609 2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
612 configure: Make -Bsymbolic check work with clang.
613 Update the -Bsymbolic check with the version glib has. This version
615 https://bugzilla.gnome.org/show_bug.cgi?id=759713
617 2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
619 * gst/rtsp-server/rtsp-session-pool.c:
620 rtsp-session-pool: Avoid dollar sign ($) in session ids
621 Live555 in VLC strips off dollar signs and then gets very confused,
622 we don't loose too much entropy by just skipping it.
624 2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
626 * gst/rtsp-server/rtsp-address-pool.h:
627 * gst/rtsp-server/rtsp-auth.h:
628 * gst/rtsp-server/rtsp-client.h:
629 * gst/rtsp-server/rtsp-media-factory-uri.h:
630 * gst/rtsp-server/rtsp-media-factory.h:
631 * gst/rtsp-server/rtsp-media.h:
632 * gst/rtsp-server/rtsp-mount-points.h:
633 * gst/rtsp-server/rtsp-permissions.h:
634 * gst/rtsp-server/rtsp-server.h:
635 * gst/rtsp-server/rtsp-session-media.h:
636 * gst/rtsp-server/rtsp-session-pool.h:
637 * gst/rtsp-server/rtsp-session.h:
638 * gst/rtsp-server/rtsp-stream-transport.h:
639 * gst/rtsp-server/rtsp-stream.h:
640 * gst/rtsp-server/rtsp-thread-pool.h:
641 * gst/rtsp-server/rtsp-token.h:
642 rtsp-server: Add g_autoptr() support to all types
643 https://bugzilla.gnome.org/show_bug.cgi?id=754464
645 2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
647 * gst/rtsp-server/rtsp-stream.c:
648 rtsp-stream: fixed valgrind error
649 Fixed the valgrind error in unit test. The UDP source created during
650 gst_rtsp_stream_join_bin() was not released while destroying the rtp
652 https://bugzilla.gnome.org/show_bug.cgi?id=759010
654 2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
658 Automatic update of common submodule
659 From b319909 to 86e4663
661 2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
663 * gst/rtsp-server/rtsp-client.c:
664 rtsp-client: suspend media during setup request
665 SETUP request from clients needs to suspend the media to clear the
666 prerolled buffers. Otherwise it will not affect the prerolled buffer
667 and the prerolled buffers will be incorrect (for example block-size
668 from setup request will not affect the prerolled buffer unless the
670 https://bugzilla.gnome.org/show_bug.cgi?id=758268
672 2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
674 * gst/rtsp-server/rtsp-stream.c:
675 rtsp-stream: create stream pipeline based on transport
676 Based on the protocol, create the rtsp stream pipeline. If only TCP or
677 only UDP is set as the transport protocol, it will not add the extra tee
678 or queue element to the pipeline. Both these elements will be added, if
679 it supports both TCP and UDP protocols. This improves the pipeline
680 performance when one protocol is present.
681 https://bugzilla.gnome.org/show_bug.cgi?id=758179
683 2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
685 * gst/rtsp-server/rtsp-stream.c:
686 rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
687 Adding them when not needed will start some logic inside rtpbin that might be
688 problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
689 would start up a rtpjitterbuffer and behave in weird ways.
690 We still set up the UDP sources for RTP receiving for a sender media to be
691 able to receive any packets sent by the client for NAT traversal. They will
692 all go to a fakesink though.
693 Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
694 NO_PREROLL, which will cause deadlocks when seeking the media as it will never
695 receive ASYNC_DONE after a seek.
696 https://bugzilla.gnome.org/show_bug.cgi?id=758319
698 2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
700 * gst/rtsp-server/rtsp-stream.c:
701 rtsp-stream: Disable multicast loopback for the multicast udp sources too
702 On POSIX this setting is for sender sockets, on Windows for receiver sockets.
703 Previously we were only setting this for sender sockets, which caused looped
704 back packets to be received on Windows if a multicast transport was used.
706 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
708 * examples/test-record-auth.c:
709 * examples/test-record.c:
710 examples: Actually use the provided port in the record examples
712 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
714 * examples/test-record-auth.c:
715 test-record-auth: Add the option to build in TLS support
717 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
719 * examples/test-auth.c:
720 test-auth: Use an 'anonymous' user for unauthenticated default
721 There's a comment on one of the resources that 'user' and 'admin'
722 shouldn't even be able to see it, but they can if the default
723 token is 'admin2', since that gives them access anyway.
725 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
727 * examples/.gitignore:
728 * examples/Makefile.am:
729 * examples/test-record-auth.c:
730 Add test-record-auth example
732 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
734 * gst/rtsp-server/rtsp-client.c:
735 * tests/check/gst/client.c:
736 rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
738 2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
740 * gst/rtsp-server/rtsp-server.c:
741 rtsp-server: Change the logic so we don't pop a NULL context
742 When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
743 will sometimes fail. This call is made before any context is pushed
744 resulting in an attempt to pop a NULL context.
745 https://bugzilla.gnome.org/show_bug.cgi?id=757949
747 2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
749 * tests/check/gst/rtspserver.c:
750 rtspserver: Add udp-mcast transport SETUP test
751 Refactor utility functions in the test file so they can handle
752 more than UDP and TCP as lower transport.
753 https://bugzilla.gnome.org/show_bug.cgi?id=756969
755 2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
757 * gst/rtsp-server/rtsp-stream.c:
758 rtsp-stream: Always unref return value of gst_object_get_parent()
759 Fixes a leak of a GstBin in the udp-mcast case.
760 https://bugzilla.gnome.org/show_bug.cgi?id=756968
762 2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
765 Automatic update of common submodule
766 From b99800a to b319909
768 2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
771 Use new GST_ENABLE_EXTRA_CHECKS #define
772 https://bugzilla.gnome.org/show_bug.cgi?id=756870
774 2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
777 Automatic update of common submodule
778 From 6babecd to b99800a
780 2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
783 Update GLib dependency to 2.40.0
785 2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
787 * examples/test-mp4.c:
788 * gst/rtsp-server/rtsp-stream.c:
789 stream: listen to sender ssrc signals
790 https://bugzilla.gnome.org/show_bug.cgi?id=746747
792 2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
795 common: update for new suppression
796 Makes check-valgrind pass with glib 2.46
798 2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
800 * gst/rtsp-server/rtsp-media.c:
801 rtsp-media: Take reference to media that will be prepared
802 default_prepare() takes a transfer-none reference GstRTSPMedia object.
803 Later on a g_idle_source_new() is created and a pointer to the media
804 object is passed as user data. If the media is freed before the idle
805 source is dispatched the media object pointer is invalid, but the idle
806 source callback expects it to still be valid. To fix this a reference to
807 the media object is taken when registering the source callback function
808 and a corresponding release of the reference is done when the souce is
810 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
812 2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
814 * examples/test-launch.c:
815 * examples/test-mp4.c:
816 * examples/test-ogg.c:
817 * examples/test-record.c:
818 * examples/test-uri.c:
819 rtsp-server: Fix memory leaks when context parse fails
820 When g_option_context_parse fails, context and error variables are not getting free'd
821 which results in memory leaks. Free'ing the same.
822 And replacing g_error_free with g_clear_error, which checks if the error being passed
823 is not NULL and sets the variable to NULL on free'ing.
824 https://bugzilla.gnome.org/show_bug.cgi?id=753863
826 2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
831 === release 1.6.0 ===
833 2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
839 * gst-rtsp-server.doap:
842 === release 1.5.91 ===
844 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
850 * gst-rtsp-server.doap:
853 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
855 * docs/libs/gst-rtsp-server-sections.txt:
856 * gst/rtsp-server/rtsp-stream.c:
857 stream: fix docs for recently-added get/set_buffer_size API
858 https://bugzilla.gnome.org/show_bug.cgi?id=749095
860 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
862 * gst/rtsp-server/rtsp-media.c:
863 rtsp-media: Don't crash on encrypted RTX SDP
864 In parse_keymgmt(), don't mutate the input string that's been passed
865 as const, especially since we might need the original value again if
866 the same key info applies to multiple streams (RTX, for example).
867 https://bugzilla.gnome.org/show_bug.cgi?id=754753
869 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
871 * examples/test-mp4.c:
872 test-mp4: Support filenames with spaces in them. Error out on too few arguments
874 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
876 * examples/test-record.c:
877 test-record: Check parameter count and print out help
878 If no launch pipeline was supplied, print out some help
880 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
882 * gst/rtsp-server/rtsp-media.c:
883 * gst/rtsp-server/rtsp-stream.c:
884 * gst/rtsp-server/rtsp-stream.h:
885 rtsp-stream: Implement UDP buffer size setting.
886 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
888 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
889 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
891 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
893 * gst/rtsp-server/rtsp-media.h:
894 rtsp-media: Fix small typo causing gtk-doc to complain
896 === release 1.5.90 ===
898 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
904 * gst-rtsp-server.doap:
907 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
909 * gst/rtsp-server/rtsp-media-factory.c:
910 media-factory: get port number through gst_rtsp_url_get_port
911 https://bugzilla.gnome.org/show_bug.cgi?id=753473
913 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
915 * tests/check/gst/media.c:
916 media-test: Removing unnecessary assertion
917 https://bugzilla.gnome.org/show_bug.cgi?id=753385
919 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
921 * gst/rtsp-server/rtsp-server.c:
922 Document that source keeps a ref on server until it's destroyed
923 https://bugzilla.gnome.org/show_bug.cgi?id=749227
925 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
927 * tests/check/gst/media.c:
928 media-test: Test for multiple dynamic payload
929 https://bugzilla.gnome.org/show_bug.cgi?id=753385
931 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
933 * gst/rtsp-server/rtsp-media.c:
934 media: Only add fakesink once per pipeline
935 The intention is to prevent going PLAYING state before pads are created.
936 If there was mutilple dynamic payload, it would leak few fakesink and
937 actually prevent from ever reaching playing state.
938 https://bugzilla.gnome.org/show_bug.cgi?id=753385
940 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
942 * gst/rtsp-server/rtsp-media.c:
943 Revert "rtsp-media: Only add 1 fakesink per pipeline"
944 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
946 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
948 * gst/rtsp-server/rtsp-media.c:
949 rtsp-media: Only add 1 fakesink per pipeline
950 There should be only one fakesink per pipeline, not per dynpay. This
951 would lead to element naming clash.
953 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
955 * gst/rtsp-server/rtsp-media.c:
956 rtsp-media: assertion error due to wrong condition check
957 In media to caps function, reserved_keys array is being used for variable i,
958 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
959 changed it to variable j
960 https://bugzilla.gnome.org/show_bug.cgi?id=753009
962 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
964 * gst/rtsp-server/rtsp-media.c:
965 rtsp-media: Strip keys from the fmtp that we use internally in our caps
966 Skip keys from the fmtp, which we already use ourselves for the
967 caps. Some software is adding random things like clock-rate into
968 the fmtp, and we would otherwise here set a string-typed clock-rate
969 in the caps... and thus fail to create valid RTP caps
970 https://bugzilla.gnome.org/show_bug.cgi?id=753009
972 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
974 * gst/rtsp-server/rtsp-thread-pool.c:
975 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
976 https://bugzilla.gnome.org/show_bug.cgi?id=752640
978 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
981 Automatic update of common submodule
982 From f74b2df to 9aed1d7
984 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
989 === release 1.5.2 ===
991 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
997 * gst-rtsp-server.doap:
1000 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
1002 * gst/rtsp-server/rtsp-client.c:
1003 * gst/rtsp-server/rtsp-client.h:
1004 * tests/check/gst/client.c:
1005 rtsp-client: allow application to decide what requirements are supported
1006 Add "check-requirements" signal and vfunc to allow application
1007 (and subclasses) to check the requirements.
1008 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
1009 https://bugzilla.gnome.org/show_bug.cgi?id=749417
1011 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1014 Automatic update of common submodule
1015 From 6015d26 to f74b2df
1017 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
1019 * gst/rtsp-server/rtsp-media.c:
1020 rtsp-media: Always use real payloader when creating streams
1021 A bin that contains the real payloader might be used as payloader. In this
1022 case we have to get the real payloader for the various properties it provides.
1023 Example use cases for this are bins that payload some media and then have
1024 additional elements that add metadata or RTP extension headers to the stream.
1025 https://bugzilla.gnome.org/show_bug.cgi?id=750800
1027 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
1029 * examples/test-netclock-client.c:
1030 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
1032 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
1034 * examples/test-netclock-client.c:
1035 * examples/test-netclock.c:
1036 test-netclock: Use new ntp-time-source property on rtpbin
1037 Select the clock time to be used as NTP time source. This allows proper
1038 synchronization between receivers, independent of sharing base times, and just
1039 requires them to use the same clock.
1041 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
1043 * examples/test-netclock-client.c:
1044 * examples/test-netclock.c:
1045 test-netclock: Setting the same base time on sender and receiver is not necessary
1046 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
1048 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1050 * gst/rtsp-server/rtsp-stream.c:
1051 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
1052 https://bugzilla.gnome.org/show_bug.cgi?id=750764
1054 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1056 * docs/libs/gst-rtsp-server.types:
1057 docs: add missing types
1058 https://bugzilla.gnome.org/show_bug.cgi?id=750764
1060 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1062 * docs/libs/gst-rtsp-server-sections.txt:
1063 docs: add missing apis
1064 https://bugzilla.gnome.org/show_bug.cgi?id=750764
1066 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
1068 * examples/test-netclock-client.c:
1069 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
1071 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1073 * docs/libs/gst-rtsp-server-sections.txt:
1074 * gst/rtsp-server/rtsp-auth.c:
1075 * gst/rtsp-server/rtsp-auth.h:
1076 GstRTSPAuth: Add client certificate authentication support
1077 https://bugzilla.gnome.org/show_bug.cgi?id=750471
1079 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
1081 * examples/test-netclock-client.c:
1082 test-netclock-client: Use new GstClock API to wait for clock synchronization
1084 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
1086 * examples/test-netclock-client.c:
1087 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
1088 A mainloop is needed to get glimagesink to display something on OSX, and
1089 the source-setup signal just makes things a little bit easier.
1091 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
1094 Automatic update of common submodule
1095 From d9a3353 to 6015d26
1097 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
1100 Automatic update of common submodule
1101 From d37af32 to d9a3353
1103 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
1106 Automatic update of common submodule
1107 From 21ba2e5 to d37af32
1109 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
1112 Automatic update of common submodule
1113 From c408583 to 21ba2e5
1115 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
1117 * docs/libs/Makefile.am:
1118 docs: remove variables that we define in the snippet from common
1119 This is syncing our Makefile.am with upstream gtkdoc.
1121 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
1124 Automatic update of common submodule
1125 From 44a3517 to c408583
1127 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
1132 === release 1.5.1 ===
1134 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
1140 * gst-rtsp-server.doap:
1143 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
1145 * gst/rtsp-server/rtsp-client.c:
1146 rtsp-client: No flush during Teardown.
1147 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
1148 backlog is empty it can happen that just a part of a message will be
1149 sent and rest is in backlog queue. If then flush during teardown
1150 just a part of message will be sent.This can lead to client miss
1151 teardown response since it expect to get the last part of message.
1152 The flushing during teardown was introduced to fix a deadlock that now
1153 is fixed more generally in handle_request by temporary setting backlog
1155 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
1157 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
1159 * tests/check/Makefile.am:
1160 tests: Use AM_TESTS_ENVIRONMENT
1161 Needed by the new automake test runner and the
1162 current version of the common submodule.
1164 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
1166 * gst/rtsp-server/rtsp-media.h:
1167 * gst/rtsp-server/rtsp-stream.h:
1168 rtsp-server: Use single-include rtsp header to make sure we get all definitions
1170 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
1172 * gst/rtsp-server/rtsp-media.c:
1173 rtsp-media: Mark some more functions static
1175 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
1177 * gst/rtsp-server/rtsp-media.c:
1178 rtsp-media: Only unblock the media in suspend() when actually changing the state
1179 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
1181 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
1183 * examples/test-video-rtx.c:
1184 examples: Use AVPF profile for the RTX example
1186 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
1188 * gst/rtsp-server/rtsp-sdp.c:
1189 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
1191 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1193 * gst/rtsp-server/rtsp-stream.c:
1194 rtsp-stream: get valid clock-rate from last-sample
1195 clock-rate in last-sample's caps is integer, not unsigned.
1196 To get this value properly, variable needs to be type-casted to int.
1197 https://bugzilla.gnome.org/show_bug.cgi?id=747614
1199 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
1203 autogen.sh: only run autopoint if gettext requested in configure.ac
1204 Not just because there happens to be a po directory.
1205 https://bugzilla.gnome.org/show_bug.cgi?id=748058
1207 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
1210 Revert "configure.ac: uncomment gettext version setup"
1211 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
1212 We don't need a gettext setup here and there's no po
1213 directory either, so no reason why autopoint would be
1214 run in the first place.
1215 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
1217 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
1219 * examples/test-multicast.c:
1220 * examples/test-multicast2.c:
1221 * examples/test-sdp.c:
1222 * examples/test-video-rtx.c:
1223 * examples/test-video.c:
1224 * tests/test-cleanup.c:
1225 * tests/test-reuse.c:
1226 Fix timeout function signatures across tests and examples
1228 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
1230 * tests/check/Makefile.am:
1231 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
1232 Make sure the test environment is set up.
1233 https://bugzilla.gnome.org//show_bug.cgi?id=747624
1235 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
1238 configure: bump automake requirement to 1.14 and autoconf to 2.69
1239 This is only required for builds from git, people can still
1240 build tarballs if they only have older autotools.
1241 https://bugzilla.gnome.org//show_bug.cgi?id=747624
1243 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
1246 configure.ac: uncomment gettext version setup
1247 Fixes autogen.sh. It would run autopoint, which would complain
1248 that it could not find the gettext version in configure.ac.
1249 https://bugzilla.gnome.org/show_bug.cgi?id=748058
1251 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1253 * examples/test-video-rtx.c:
1254 test-video-rtx: set exact payload type to PCMA payloader
1255 Setting wrong payload type causes failure to do retransmission through audio stream
1256 https://bugzilla.gnome.org/show_bug.cgi?id=747839
1258 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1260 * gst/rtsp-server/rtsp-media.c:
1261 * gst/rtsp-server/rtsp-stream.c:
1262 * gst/rtsp-server/rtsp-stream.h:
1263 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
1264 Because of duplicated g_signal_connect for request-aux-sender signal,
1265 wrong stream pointer is passed to the signal handler.
1266 Instead of passing each stream, pass stream array and get the relevant stream.
1267 https://bugzilla.gnome.org/show_bug.cgi?id=747839
1269 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
1273 Update autogen.sh to latest version from common
1274 Fixes build after aclocal_check etc. helpers have been removed.
1276 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
1279 Automatic update of common submodule
1280 From bc76a8b to c8fb372
1282 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
1284 * gst/rtsp-server/rtsp-stream.c:
1285 rtsp-stream: Limit the queues to 1 buffer
1286 We only need them to be able to pre-roll, queueing up more data here
1287 is only going to harm latency and memory usage.
1289 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
1291 * gst/rtsp-server/rtsp-stream.c:
1292 rtsp-stream: Update comment and ASCII art to the latest code
1293 We have a queue in front of the udpsink too to prevent the pipeline from
1296 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
1298 * gst/rtsp-server/rtsp-stream.c:
1299 rtsp-media: Properly return first rtptime
1300 Instead we where returning first GstBuffer timestamp. This would result
1301 in clock skew and unwanted behaviour in RTSP playback.
1302 https://bugzilla.gnome.org/show_bug.cgi?id=746479
1304 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
1306 * gst/rtsp-server/rtsp-stream.c:
1307 rtsp-stream: Don't leave buffer mapped
1308 If the seq is NULL, the RTP buffer was left mapped. We should always
1311 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
1316 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
1318 * gst/rtsp-server/rtsp-media-factory.c:
1319 * tests/check/gst/client.c:
1320 Fix double semicolons
1322 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
1324 * gst/rtsp-server/rtsp-stream.c:
1325 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
1326 This gives more accurate values than asking the payloader. There might be
1327 queueing happening between the payloader and the sink.
1328 https://bugzilla.gnome.org/show_bug.cgi?id=745704
1330 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
1332 * gst/rtsp-server/rtsp-media.c:
1333 rtsp-media: Don't seek for PLAY if the position will not change
1334 https://bugzilla.gnome.org/show_bug.cgi?id=745704
1336 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
1338 * gst/rtsp-server/rtsp-media.c:
1339 rtsp-media: Don't include payload type in the caps for framesize
1340 When the sdp media attribute framesize are converted to caps
1341 the <payload> should not be included.
1342 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
1343 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
1345 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
1347 * gst/rtsp-server/rtsp-sdp.c:
1348 rtsp-sdp: add payload type to the sdp framesize attribute
1349 The sdp framesize attribute is desribed in RFC6064. It is specified
1350 for payloading of H263 and has the following form
1351 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
1352 should be added to the caps in a payloader and the <payload type> should
1353 be added by the rtsp-server.
1354 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
1356 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
1358 * examples/test-uri.c:
1359 examples: test-uri: fix tainted variable
1360 Insignificant but this keeps Coverity happy.
1363 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
1365 * examples/.gitignore:
1366 * examples/Makefile.am:
1367 * examples/test-netclock-client.c:
1368 * examples/test-netclock.c:
1369 examples: Add a simple example of network synch for live streams.
1370 An example server and client that works for synchronising live streams
1371 only - as it can't support pause/play.
1373 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
1375 * gst/rtsp-server/rtsp-media-factory.c:
1376 * gst/rtsp-server/rtsp-media-factory.h:
1377 rtsp-media-factory: Add functions to set/get the media gtype
1378 Allow specifying the GType of a GstRtspMedia subclass to create
1379 as a simpler way to get the factory to create a custom
1380 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
1382 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
1384 * gst/rtsp-server/rtsp-media.c:
1385 rtsp-media: fix double unlock in _get_buffer_size()
1386 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
1387 because of double g_mutex_unlock () usage.
1388 https://bugzilla.gnome.org/show_bug.cgi?id=745434
1390 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
1392 * gst/rtsp-server/rtsp-session-pool.c:
1393 * gst/rtsp-server/rtsp-session.c:
1394 * gst/rtsp-server/rtsp-session.h:
1395 rtsp-session: Use monotonic time for RTSP session timeout
1396 Changed RTSP session timeout handling to monotonic time
1397 and deprecating the API for current system time.
1398 This fixes timeouts when the system time changes.
1399 https://bugzilla.gnome.org/show_bug.cgi?id=743346
1401 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
1403 * gst/rtsp-server/rtsp-client.c:
1404 * gst/rtsp-server/rtsp-media.c:
1405 rtsp-client: Only error out in PLAY if seeking actually failed
1406 If the media was just not seekable, we continue from whatever position we are
1407 and let the client decide if that is what is wanted or not.
1408 Only if the actual seek failed, we can't really recover and should error out.
1410 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
1412 * gst/rtsp-server/rtsp-stream.c:
1413 rtsp-stream: Add necessary queues between tee and multiudpsink
1414 https://bugzilla.gnome.org/show_bug.cgi?id=744379
1416 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
1418 * gst/rtsp-server/rtsp-client.c:
1419 * gst/rtsp-server/rtsp-media.c:
1420 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
1421 Instead error out properly the same way as if the SEEKING query already
1424 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
1426 * gst/rtsp-server/rtsp-stream.h:
1427 rtsp-stream: minor code formatting fix
1429 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
1431 * gst/rtsp-server/rtsp-media.c:
1432 rtsp-media: fix logic for collect_streams
1433 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
1434 all streams it knows if it got any, and can check if the transport mode is OK.
1437 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
1439 * gst/rtsp-server/rtsp-media.c:
1440 rtsp-media: Don't set the transport mode based on what elements we find
1441 Just print a warning if the one that was set before disagrees with what
1442 elements we found. It must already be set to something before as this
1443 function is called after we received the SDP from ANNOUNCE in RECORD mode,
1444 and we would reject ANNOUNCE if the RECORD flag was not set.
1446 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
1448 * tests/check/gst/rtspserver.c:
1449 tests: rtspserver: rename shadowed variable
1450 We have two different 'sink' variables here,
1451 rename one of them for clarity.
1453 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
1455 * gst/rtsp-server/rtsp-client.c:
1456 rtsp-client: fix awkward if clause
1458 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
1460 * examples/test-uri.c:
1461 examples: test-uri: improve uri argument handling and accept file names
1462 Print an error if the argument passed is not a URI and can't
1463 be converted into one, or no arguments have been provided.
1465 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
1467 * examples/test-uri.c:
1468 examples: test-uri: don't remove mount point after 10 seconds
1469 It's very irritating when trying to test stuff repeatedly
1470 and serves no real purpose other than showing that it can
1473 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
1475 * examples/.gitignore:
1476 examples: add new test-record to .gitignore
1478 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
1480 * examples/test-record.c:
1481 * gst/rtsp-server/rtsp-client.c:
1482 * gst/rtsp-server/rtsp-media-factory.c:
1483 * gst/rtsp-server/rtsp-media-factory.h:
1484 * gst/rtsp-server/rtsp-media.c:
1485 * gst/rtsp-server/rtsp-media.h:
1486 * tests/check/gst/rtspserver.c:
1487 rtsp-media: Use flags to distinguish between PLAY and RECORD media
1489 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
1491 * examples/test-record.c:
1492 test-record: Set latency for playback-style example to 2s instead of 200ms
1494 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
1496 * tests/check/gst/rtspserver.c:
1497 tests: add some unit tests for ANNOUNCE and RECORD
1498 https://bugzilla.gnome.org/show_bug.cgi?id=743175
1500 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
1502 * gst/rtsp-server/rtsp-client.c:
1503 rtsp-client: fix a couple of leaks in handle_announce
1505 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
1507 * gst/rtsp-server/rtsp-media-factory.c:
1508 * gst/rtsp-server/rtsp-media-factory.h:
1509 * gst/rtsp-server/rtsp-media.c:
1510 * gst/rtsp-server/rtsp-media.h:
1511 rtsp-media: Expose latency setting for setting the rtpbin latency
1513 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
1515 * examples/test-record.c:
1516 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
1518 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
1520 * gst/rtsp-server/rtsp-stream.c:
1521 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
1523 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
1525 * examples/Makefile.am:
1526 * examples/test-record.c:
1527 * gst/rtsp-server/rtsp-client.c:
1528 * gst/rtsp-server/rtsp-client.h:
1529 * gst/rtsp-server/rtsp-media-factory.c:
1530 * gst/rtsp-server/rtsp-media-factory.h:
1531 * gst/rtsp-server/rtsp-media.c:
1532 * gst/rtsp-server/rtsp-media.h:
1533 * gst/rtsp-server/rtsp-session-media.c:
1534 * gst/rtsp-server/rtsp-stream.c:
1535 * gst/rtsp-server/rtsp-stream.h:
1536 Add initial support for RECORD
1537 We currently only support media that is RECORD or PLAY only, not both at once.
1538 https://bugzilla.gnome.org/show_bug.cgi?id=743175
1540 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
1542 * gst/rtsp-server/rtsp-stream.c:
1543 rtsp-stream: RTCP and RTP transport cache cookies seperated
1544 RTCP packets were not sent because the same tr_cache_cookie was used for
1545 both RTP and RTCP. So only one of the tr_cache lists were populated
1546 depending on which one was sent first. If the tr_cache list is not
1547 populated then no packets can be sent. Most often this happened to be
1548 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
1549 resulted in both the tr_cache_lists to be populated regardless of which
1551 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
1553 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
1555 * gst/rtsp-server/rtsp-stream.c:
1556 rtsp-stream: fix false compiler warning
1557 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
1559 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
1561 * gst/rtsp-server/rtsp-client.c:
1562 rtsp-client: log interleaved data received
1564 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
1566 * gst/rtsp-server/rtsp-client.c:
1567 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
1569 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
1571 * gst/rtsp-server/rtsp-client.c:
1572 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
1574 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
1576 * gst/rtsp-server/rtsp-client.c:
1577 rtsp-client: Use a random session ID in the SDP
1578 RFC4566 Section 5.2 says that it should make the username, session id,
1579 nettype, addrtype and unicast address tuple globally unique. Always using
1580 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
1581 Instead let's create a 64 bit random number, which at least brings us
1582 closer to the goal of global uniqueness.
1583 https://tools.ietf.org/html/rfc4566#section-5.2
1585 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
1587 * examples/test-launch.c:
1588 * examples/test-mp4.c:
1589 * examples/test-ogg.c:
1590 * examples/test-uri.c:
1591 examples: Don't call gst_init() and gst_get_option_group()
1592 The latter calls the former at the appropriate time.
1594 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
1596 * gst/rtsp-server/rtsp-client.c:
1597 rtsp-client: Drop trailing \0 of RTSP DATA messages
1598 We add a trailing \0 in GstRTSPConnection to make parsing of
1599 string message bodies easier (e.g. the SDP from DESCRIBE) but
1600 for actual data this means we have to drop it or otherwise
1601 create invalid data.
1603 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
1605 * gst/rtsp-server/rtsp-stream.c:
1606 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
1607 Fixes crash when two threads access handle_new_sample() at the same
1608 time, one for RTP, one for RTCP.
1609 Otherwise, when iterating over the transports cache, it might be modified by
1610 another thread at the same time if the transports cookie has changed.
1611 https://bugzilla.gnome.org/show_bug.cgi?id=742954
1613 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
1615 * gst/rtsp-server/rtsp-stream.c:
1616 rtsp-stream: Set format=TIME on our app sources for TCP
1618 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
1620 * gst/rtsp-server/rtsp-session-pool.c:
1621 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
1622 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
1623 RFC 2326 states that session IDs may consist of alphanumeric as well as
1624 the safe characters $-_.+ -- N.B. the percent character is not allowed.
1625 Previously the session ID was URI-escaped, this meant that any character
1626 which was not alphanumeric or any of the characters +-._~ would be
1627 percent encoded. While the RFC (surprisingly) mentions that linear white
1628 space in session IDs should be URI-escaped, it does not say anything
1629 about other characters. Moreover no white space is allowed in the
1630 session ID. Finally the percent character which is the result of
1631 URI-escaping is not allowed in a session ID.
1632 So there is no reason to do any URI-escaping, and now it is removed.
1633 https://bugzilla.gnome.org/show_bug.cgi?id=742869
1635 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
1638 Automatic update of common submodule
1639 From f2c6b95 to bc76a8b
1641 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
1644 Fix 'make check' from top-level directory
1646 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
1648 * examples/test-launch.c:
1649 * examples/test-mp4.c:
1650 * examples/test-ogg.c:
1651 * examples/test-uri.c:
1652 examples: Add command-line parsing and take a 'port' argument
1653 This allows users to run multiple servers on different ports for testing.
1654 Only done for examples that actually take arguments and hence are capable of
1655 outputting different streams for each instance on each port.
1656 https://bugzilla.gnome.org/show_bug.cgi?id=742115
1658 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
1660 * gst/rtsp-server/rtsp-client.c:
1661 * gst/rtsp-server/rtsp-client.h:
1662 rtsp-client: Add a send_message default signal handler
1663 This allows subclasses to easily hook into the response sending
1664 mechanism without doing everything from a signal, which seems
1665 awkward from subclasses.
1667 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
1670 Automatic update of common submodule
1671 From ef1ffdc to f2c6b95
1673 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1677 configure: add --disable-examples switch
1678 https://bugzilla.gnome.org/show_bug.cgi?id=741678
1680 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
1682 * examples/.gitignore:
1683 * examples/Makefile.am:
1684 * examples/test-video-rtx.c:
1685 examples: add a retransmisison example implementing RFC4588
1686 Currently only SSRC-multiplexed rtx streams are supported
1688 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
1690 * gst/rtsp-server/rtsp-stream.c:
1691 rtsp-stream: Fix some minor memory leaks
1693 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
1695 * gst/rtsp-server/rtsp-media.c:
1696 rtsp-media: Some minor cleanup
1698 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
1700 * gst/rtsp-server/rtsp-stream.c:
1701 rtsp-stream: Fix compiler warnings
1702 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
1703 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1705 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
1706 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1709 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
1711 * docs/libs/gst-rtsp-server-sections.txt:
1712 * gst/rtsp-server/rtsp-media-factory.c:
1713 * gst/rtsp-server/rtsp-media-factory.h:
1714 * gst/rtsp-server/rtsp-media.c:
1715 * gst/rtsp-server/rtsp-media.h:
1716 * gst/rtsp-server/rtsp-sdp.c:
1717 * gst/rtsp-server/rtsp-stream.c:
1718 * gst/rtsp-server/rtsp-stream.h:
1719 media: implement ssrc-multiplexed retransmission support
1720 based off RFC 4588 and the server-rtpaux example in -good
1722 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
1724 * gst/rtsp-server/rtsp-client.c:
1725 * gst/rtsp-server/rtsp-stream-transport.c:
1726 * gst/rtsp-server/rtsp-stream.c:
1727 rtsp: Ref transports in hash table.
1728 Also ref streams for transports.
1729 This solves a crash when reciving a rtcp after teardown but before
1731 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
1733 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
1736 Automatic update of common submodule
1737 From 7bb2bce to ef1ffdc
1739 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
1741 * gst/rtsp-server/rtsp-client.c:
1742 client: refactor cleanup of cached media
1744 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
1746 * tests/check/gst/client.c:
1748 The session leak is now fixed, lets remove those FIXME comments.
1750 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
1752 * tests/check/gst/rtspserver.c:
1753 tests: Test to setup two sessions on one connection
1754 https://bugzilla.gnome.org/show_bug.cgi?id=739112
1756 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
1758 * tests/check/gst/rtspserver.c:
1759 tests: Test setup with tcp transport
1760 https://bugzilla.gnome.org/show_bug.cgi?id=739112
1762 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
1764 * gst/rtsp-server/rtsp-client.c:
1765 client: Configure transport after creating session media
1766 The default implementation of configure_client_transport() in
1767 rtsp-client uses the session media when it chooses channels for
1768 interleaved traffic.
1769 https://bugzilla.gnome.org/show_bug.cgi?id=739112
1771 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
1773 * gst/rtsp-server/rtsp-client.c:
1774 * gst/rtsp-server/rtsp-session-media.c:
1775 client: Stop caching media in client when doing setup
1776 If the media has been managed by a session media, it should not be
1777 cached in the client any longer. The GstRTSPSessionMedia object is now
1778 responsible for unpreparing the GstRTSPMedia object using
1779 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
1781 https://bugzilla.gnome.org/show_bug.cgi?id=739112
1783 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1785 * gst/rtsp-server/rtsp-stream.c:
1786 rtsp-stream: unref srtp decoder when leaving bin
1787 https://bugzilla.gnome.org/show_bug.cgi?id=739481
1789 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1791 * gst/rtsp-server/rtsp-client.c:
1792 rtsp-client: mikey memory leaks
1793 https://bugzilla.gnome.org/show_bug.cgi?id=739383
1795 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
1798 Automatic update of common submodule
1799 From 84d06cd to 7bb2bce
1801 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
1804 Parallelise 'make check-valgrind'
1806 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
1809 Automatic update of common submodule
1810 From a8c8939 to 84d06cd
1812 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
1815 Automatic update of common submodule
1816 From 36388a1 to a8c8939
1818 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
1820 * gst/rtsp-server/rtsp-media.c:
1821 rtsp-media: deactivate media when shutting down from paused
1822 This was only done when going directly from playing.
1823 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
1825 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1827 * gst/rtsp-server/rtsp-client.c:
1828 * gst/rtsp-server/rtsp-context.h:
1829 rtsp-client: add stream transport to context
1830 We add the stream transport to the context so we can get the configured
1831 client stream transport in the setup request signal.
1832 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
1834 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1836 * gst/rtsp-server/rtsp-stream.c:
1837 stream: release lock even not all transports have been removed
1838 We don't want to keep the lock even we return FALSE because not all the
1839 transports have been removed. This could lead into a deadlock.
1840 https://bugzilla.gnome.org/show_bug.cgi?id=737797
1842 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
1844 * gst/rtsp-server/rtsp-sdp.c:
1845 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
1846 These were renamed in GstRTPBasePayload in 1.0
1848 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1850 * gst/rtsp-server/rtsp-client.c:
1851 client: set session media to NULL without the lock
1852 We need to set session medias to NULL without the client lock otherwise
1853 we can end up in a deadlock if another thread is waiting for the lock
1854 and media unprepare is also waiting for that thread to end.
1855 https://bugzilla.gnome.org/show_bug.cgi?id=737690
1857 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
1859 * gst/rtsp-server/rtsp-media.c:
1860 rtsp-media: Set state to UNPREPARING in all cases
1862 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
1864 * gst/rtsp-server/rtsp-media.c:
1865 media: set state to unpreparing when unprepare is initiated
1866 https://bugzilla.gnome.org/show_bug.cgi?id=737675
1868 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
1870 * gst/rtsp-server/rtsp-client.c:
1871 rtsp-client: Remove backlog limit while processings requests
1872 If the backlog limit is kept two cases of deadlocks may be
1873 encountered when streaming over TCP. Without the backlog
1874 limit this deadlocks can not happen, at the expence of
1876 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
1878 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
1880 * gst/rtsp-server/rtsp-client.c:
1881 rtsp-client: do not free main context before rtsp watch
1882 https://bugzilla.gnome.org/show_bug.cgi?id=737110
1884 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
1886 * tests/check/gst/rtspserver.c:
1887 tests: Extend unit test timeout to accomodate for valgrind
1888 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1890 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
1892 * gst/rtsp-server/rtsp-client.c:
1893 * gst/rtsp-server/rtsp-session.c:
1894 * gst/rtsp-server/rtsp-stream-transport.c:
1895 rtsp-*: Treat sending packets to clients as keepalive
1896 As long as gst-rtsp-server can successfully send RTP/RTCP data to
1897 clients then the client must be reading. This change makes the server
1898 timeout the connection if the client stops reading.
1899 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1901 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
1903 * gst/rtsp-server/rtsp-client.c:
1904 rtsp-client: Allow backlog to grow while expiring session
1905 Allow the send backlog in the RTSP watch to grow to unlimited size while
1906 attempting to bring the media pipeline to NULL due to a session
1907 expiring. Without this change the appsink element cannot change state
1908 because it is blocked while rendering data in the new_sample callback.
1909 This callback will block until it has successfully put the data into the
1910 send backlog. There is a chance that the send backlog is full at this
1911 point which means that the callback may block for a long time, possibly
1912 forever. Therefore the media pipeline may also be prevented from
1913 changing state for a long time.
1914 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1916 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
1918 * gst/rtsp-server/rtsp-client.c:
1919 rtsp-client: Make old compilers happy
1920 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
1921 Just in case that guint8 doesn't fit in a pointer. Just in case ...
1923 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
1925 * gst/rtsp-server/rtsp-client.c:
1926 client: raise the backlog limits before pausing
1927 We need to raise the backlog limits before pausing the pipeline or else
1928 the appsink might be blocking in the render method in wait_backlog() and
1929 we would deadlock waiting for paused.
1930 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
1932 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
1934 * gst/rtsp-server/rtsp-client.c:
1935 client: make define for the WATCH_BACKLOG
1936 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
1938 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
1940 * gst/rtsp-server/rtsp-client.c:
1941 client: simplify session transport handling
1942 link/unlink of the transport in a session was done to keep track of all
1943 TCP transports and to send RTP/RTCP data to the streams. We can simplify
1944 that by putting all the TCP transports in a hashtable indexed with the
1946 We also don't need to link/unlink the transports when we pause/resume
1947 the streams. The same effect is already achieved when we pause/play the
1948 media. Indeed, when we pause the media, the transport is removed from
1949 the media and the callbacks will not be called anymore.
1950 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
1952 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
1954 * gst/rtsp-server/rtsp-stream-transport.c:
1955 * gst/rtsp-server/rtsp-stream-transport.h:
1956 stream-transport: make method to handle received data
1957 Make a method to handle the data received on a channel. It sends the
1958 data to the stream of the transport on the RTP or RTCP pads based on
1961 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
1963 * examples/test-mp4.c:
1964 test: add example of dumping RTCP reports
1966 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
1968 * gst/rtsp-server/rtsp-media.c:
1969 * gst/rtsp-server/rtsp-stream.c:
1970 * gst/rtsp-server/rtsp-stream.h:
1971 rtsp-media: Make sure that sequence numbers are monotonic after pause
1972 The sequence number is not monotonic for RTP packets after pause. The
1973 reason is basepayloader generates a randon sequence number when the
1974 pipeline goes from ready to pause. With this fix generation of sequence
1975 number will be monotonic when going from pause to play request.
1976 https://bugzilla.gnome.org/show_bug.cgi?id=736017
1978 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
1980 * gst/rtsp-server/rtsp-client.c:
1981 rtsp-client: Protect saved clients watch with a mutex
1982 Fixes a crash when close() is called while merging clients
1983 in handle_tunnel(). In that case close() would destroy the
1984 watch while it is still being used in handle_tunnel().
1985 https://bugzilla.gnome.org/show_bug.cgi?id=735570
1987 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
1989 * gst/rtsp-server/rtsp-stream.c:
1990 rtsp-stream: Remove the multicast group udp sources when removing from the bin
1992 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
1994 * gst/rtsp-server/rtsp-media.c:
1995 * gst/rtsp-server/rtsp-stream.c:
1996 * gst/rtsp-server/rtsp-stream.h:
1997 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
1998 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
1999 seeking and will always continue counting the time. This leads to
2000 the NPT after a backwards seek to be something completely different
2001 to the actual seek position.
2002 https://bugzilla.gnome.org/show_bug.cgi?id=732644
2004 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
2006 * examples/test-appsrc.c:
2007 examples: fix another reference leak
2008 gst_rtsp_media_get_element() returns a new ref.
2010 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2012 * examples/test-appsrc.c:
2013 examples: unref element after usage
2014 gst_bin_get_by_name_recurse_up() returns an element
2015 reference that must be unreffed after usage.
2016 https://bugzilla.gnome.org/show_bug.cgi?id=734546
2018 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
2020 * gst/rtsp-server/rtsp-media.c:
2021 signals: Fix copy-pasto in target-state signal offset
2023 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
2027 Makefile: Add usage of build-checks step
2028 Allows building checks without running them
2030 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
2032 * gst/rtsp-server/rtsp-stream.c:
2033 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
2034 When a UDP multicast transport is used it is expected that the server listens
2035 for RTP and RTCP packets on the multicast group with the corresponding port.
2036 Without this we will never get RTCP packets from clients in multicast mode.
2037 https://bugzilla.gnome.org/show_bug.cgi?id=732238
2039 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
2044 === release 1.4.0 ===
2046 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
2052 * gst-rtsp-server.doap:
2055 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
2057 * gst/rtsp-server/rtsp-media.h:
2058 media: correct misspelled words in description
2059 https://bugzilla.gnome.org/show_bug.cgi?id=733244
2061 === release 1.3.91 ===
2063 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
2069 * gst-rtsp-server.doap:
2072 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
2074 * docs/libs/gst-rtsp-server-sections.txt:
2077 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
2079 * gst/rtsp-server/rtsp-server.c:
2080 server: implement client REMOVE filter
2082 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
2084 * gst/rtsp-server/rtsp-client.c:
2085 * gst/rtsp-server/rtsp-client.h:
2086 client: expose _close() method
2087 Expose a previously internal close method to close the client
2090 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
2092 * gst/rtsp-server/rtsp-session-pool.c:
2093 session-pool: signal session-removed outside of the lock
2094 Release the lock before emiting the session-removed signal.
2096 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
2098 * gst/rtsp-server/rtsp-client.c:
2099 * gst/rtsp-server/rtsp-server.c:
2100 * gst/rtsp-server/rtsp-session-pool.c:
2101 * gst/rtsp-server/rtsp-session.c:
2102 * gst/rtsp-server/rtsp-stream.c:
2103 filter: Release lock in filter functions
2104 Release the object lock before calling the filter functions. We need to
2105 keep a cookie to detect when the list changed during the filter
2106 callback. We also keep a hashtable to make sure we only call the filter
2107 function once for each object in case of concurrent modification.
2108 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
2110 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
2112 * gst/rtsp-server/rtsp-client.c:
2113 client: check if watch is set in handle_teardown()
2114 The unit tests run without a watch
2116 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
2118 * tests/check/gst/client.c:
2119 client tests: send teardown to cleanup session
2121 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
2123 * tests/check/gst/rtspserver.c:
2124 server tests: send teardown to cleanup session
2126 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
2128 * gst/rtsp-server/rtsp-client.c:
2129 client: keep ref to client for the session removed handler
2130 This extra ref will be dropped when all client sessions have been
2131 removed. A session is removed when a client sends teardown, closes its
2132 endpoint of the TCP connection or the sessions expires.
2133 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
2135 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
2137 * gst/rtsp-server/rtsp-client.c:
2138 * gst/rtsp-server/rtsp-session.c:
2139 * tests/check/gst/client.c:
2140 client: manage media in session as a last step
2141 Once we manage a media in a session, we can't unmanage it anymore
2142 without destroying it. Therefore, first check everything before we
2143 manage the media, otherwise if something is wrong we have no way to
2145 If we created a new session and something went wrong, remove the session
2146 again. Fixes a leak in the unit test.
2148 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
2150 * examples/test-mp4.c:
2151 * examples/test-ogg.c:
2152 examples: print 'stream ready at url' for mp4 and ogg example
2154 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
2156 * gst/rtsp-server/rtsp-client.c:
2157 * gst/rtsp-server/rtsp-sdp.c:
2158 rtsp: fix for MIKEY api change
2160 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
2162 * gst/rtsp-server/rtsp-client.c:
2163 client: free watch context only once
2164 The watch context is freed when the source is destroyed. Avoids
2165 a CRITICAL when we try to unref the context twice.
2167 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
2169 * gst/rtsp-server/rtsp-client.c:
2172 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
2174 * gst/rtsp-server/rtsp-client.c:
2175 client: protect sessions with lock
2176 Protect the list of sessions with the lock.
2177 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
2179 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
2181 * gst/rtsp-server/rtsp-client.c:
2182 Client: keep a ref to the session
2183 Don't just keep a weak ref to the session objects but use a hard ref. We
2184 will be notified when a session is removed from the pool (expired) with
2185 the new session-removed signal.
2186 Don't automatically close the RTSP connection when all the sessions of
2187 a client are removed, a client can continue to operate and it can create
2188 a new session if it wants. If you want to remove the client from the
2189 server, you have to use gst_rtsp_server_client_filter() now.
2190 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
2191 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
2193 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
2195 * gst/rtsp-server/rtsp-session-pool.c:
2196 * gst/rtsp-server/rtsp-session-pool.h:
2197 session-pool: add session-removed signal
2198 Add a signal to be notified when a session is removed from the pool.
2200 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
2202 * gst/rtsp-server/Makefile.am:
2203 * gst/rtsp-server/rtsp-server.h:
2204 Make rtsp-server.h a single-include header, use it for G-I
2205 https://bugzilla.gnome.org/show_bug.cgi?id=732411
2207 === release 1.3.90 ===
2209 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
2215 * gst-rtsp-server.doap:
2218 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
2220 * gst/rtsp-server/rtsp-stream.c:
2221 stream: crypto can be NULL
2223 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
2225 * gst/rtsp-server/rtsp-client.c:
2226 * gst/rtsp-server/rtsp-media.c:
2227 * gst/rtsp-server/rtsp-mount-points.c:
2228 introspection: add missing allow-none annotations
2229 https://bugzilla.gnome.org/show_bug.cgi?id=730952
2231 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
2233 * gst/rtsp-server/rtsp-address-pool.c:
2234 * gst/rtsp-server/rtsp-media.c:
2235 * gst/rtsp-server/rtsp-session-media.c:
2236 * gst/rtsp-server/rtsp-session-pool.c:
2237 * gst/rtsp-server/rtsp-stream-transport.c:
2238 * gst/rtsp-server/rtsp-stream.c:
2239 * gst/rtsp-server/rtsp-token.c:
2240 introspection: add (nullable) annotations to return values
2241 https://bugzilla.gnome.org/show_bug.cgi?id=730952
2243 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
2245 * gst/rtsp-server/rtsp-client.c:
2246 * gst/rtsp-server/rtsp-stream.c:
2247 gi: improve annotations
2248 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
2250 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
2252 * gst/rtsp-server/rtsp-client.c:
2253 * gst/rtsp-server/rtsp-media-factory.c:
2254 * gst/rtsp-server/rtsp-media.c:
2255 * gst/rtsp-server/rtsp-server.c:
2256 signals: use generic marshal function
2257 Use the generic C marshal function.
2258 Use more explicit type instead of G_TYPE_POINTER
2260 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
2262 * gst/rtsp-server/rtsp-context.h:
2263 context: add type macro
2265 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
2267 * gst/rtsp-server/rtsp-client.c:
2268 * gst/rtsp-server/rtsp-sdp.c:
2269 * gst/rtsp-server/rtsp-sdp.h:
2270 sdp: hide key length defines
2271 They don't have a namespace.
2273 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
2278 === release 1.3.3 ===
2280 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
2286 * gst-rtsp-server.doap:
2289 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2291 * gst/rtsp-server/rtsp-client.c:
2292 * gst/rtsp-server/rtsp-sdp.c:
2293 * gst/rtsp-server/rtsp-sdp.h:
2294 mikey: add different key length parameters
2295 Add encryption and authentication key length parameters to MIKEY. For
2296 the encoders, the key lengths are obtained from the cipher and auth
2297 algorithms set in the caps. For the decoders, they are obtained while
2298 parsing the key management from the client.
2299 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
2301 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
2303 * tests/check/gst/stream.c:
2304 stream tests: Make sure we get right multicast address from stream
2305 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
2307 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
2309 * gst/rtsp-server/rtsp-client.c:
2310 client: ref the context until rtsp watch is alive
2311 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
2313 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
2315 * gst/rtsp-server/rtsp-client.c:
2316 client: Destroy the rtsp watch after connection close
2318 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
2320 * gst/rtsp-server/rtsp-media.c:
2321 media: fix confusing comment
2323 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
2325 * gst/rtsp-server/rtsp-session.c:
2326 rtsp-session: Timeout in header.
2327 Adding the possbilty to always have timout in header.
2328 This is configurabe with setting "timeout-always-visible".
2329 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
2331 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
2336 === release 1.3.2 ===
2338 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
2345 * gst-rtsp-server.doap:
2348 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
2351 Automatic update of common submodule
2352 From 211fa5f to 1f5d3c3
2354 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
2356 * gst/rtsp-server/rtsp-client.c:
2357 client: store TCP ports in transport
2358 Store the TCP ports in the transport when we are doing RTSP over TCP.
2359 This way, we can easily get to the ports from the transport.
2360 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
2362 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2364 * gst/rtsp-server/rtsp-stream.c:
2365 stream: add signals for new RTP/RTCP encoders
2366 New signals to allow the user to configure the dynamically created
2368 https://bugzilla.gnome.org/show_bug.cgi?id=730228
2370 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
2372 * gst/rtsp-server/rtsp-media.c:
2373 * gst/rtsp-server/rtsp-media.h:
2374 media: Make suspend()/unsuspend() virtual
2375 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
2377 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2379 * gst/rtsp-server/rtsp-client.c:
2380 client: fix send-message signal marshaller
2381 Use generic marshalling for the send-message signal. It has
2382 two POINTER arguments, not just one.
2383 https://bugzilla.gnome.org/show_bug.cgi?id=729900
2385 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
2387 * tests/check/gst/media.c:
2388 tests: add and remove pads only once
2389 In this test we simulate a dynamic pad by watching the caps event.
2390 Because of renegotiation in the base payloader now, this caps is sent
2391 multiple times but we can only deal with 1 invocation, use a variable to
2392 only 'add and remove' the pad once.
2394 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
2396 * tests/check/gst/rtspserver.c:
2397 tests: add unit test for correct handling of Require headers
2398 https://bugzilla.gnome.org/show_bug.cgi?id=729426
2400 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
2402 * gst/rtsp-server/rtsp-client.c:
2403 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
2404 Servers must handle Require headers and must report a failure
2405 if they don't handle any of the Required options, see RFC 2326,
2406 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
2407 https://bugzilla.gnome.org/show_bug.cgi?id=729426
2409 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
2414 === release 1.3.1 ===
2416 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
2422 * gst-rtsp-server.doap:
2425 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
2428 Automatic update of common submodule
2429 From bcb1518 to 211fa5f
2431 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
2436 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
2438 * tests/check/gst/sessionmedia.c:
2439 tests: fix memory leak in sessionmedia unit test
2441 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
2443 * gst/rtsp-server/rtsp-client.c:
2444 client: emit a signal before sending a message
2445 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
2447 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
2449 * gst/rtsp-server/rtsp-client.c:
2450 client: pass context to send_message
2451 Pass the current context to send_message, we will need it later.
2453 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
2455 * gst/rtsp-server/rtsp-client.c:
2456 client: fix typo in comment
2458 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
2460 * gst/rtsp-server/rtsp-media.c:
2461 media: Do not stop thread twice if default_prepare() fails
2463 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
2465 * gst/rtsp-server/rtsp-client.c:
2466 client: set the watch to flushing before going to NULL
2467 First set the watch to flushing so that we unblock any current and
2468 future attempt to send data on the watch, Then set the pipeline to
2470 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
2472 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
2474 * gst/rtsp-server/rtsp-session-pool.c:
2475 * tests/check/gst/sessionpool.c:
2476 rtsp-session-pool: Fixes annotation
2477 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
2478 in the sessionpool test.
2479 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
2481 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
2483 * gst/rtsp-server/rtsp-media.c:
2484 * gst/rtsp-server/rtsp-media.h:
2485 media: make media_prepare virtual
2486 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
2488 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
2490 * gst/rtsp-server/rtsp-media.c:
2491 * tests/check/gst/media.c:
2492 media: stop the thread in more error cases
2494 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
2496 * gst/rtsp-server/rtsp-media.c:
2497 * tests/check/gst/media.c:
2498 media: allow NULL as the thread
2499 Use the default context whan passing a NULL thread.
2501 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
2503 * gst/rtsp-server/rtsp-client.c:
2504 rtsp-client: indent cleanup
2505 Coverity was moaning about unreachable code, and I think it was just
2506 confused by { being before the label. We'll see if it pops up again.
2509 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
2511 * gst/rtsp-server/rtsp-client.c:
2512 * gst/rtsp-server/rtsp-media.c:
2513 client: Add drop-backlog property
2514 When we have too many messages queued for a client (currently hardcoded
2515 to 100) we overflow and drop the messages. Add a drop-backlog property
2516 to control this behaviour. Setting this property to FALSE will retry
2517 to send the messages to the client by waiting for more room in the
2519 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
2521 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
2523 * gst/rtsp-server/rtsp-client.c:
2524 client: support for POST before GET when setting up a tunnel
2526 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
2528 * gst/rtsp-server/rtsp-client.c:
2529 client: remove watch of the second client after http tunnel setup
2530 The second client will be freed after the HTTP tunnel has been set up.
2531 Make sure it's RTSP watch is never dispatched again.
2532 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
2534 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
2536 * gst/rtsp-server/rtsp-media.c:
2537 * tests/check/gst/media.c:
2538 media: Make media_prepare() fail if port allocation fails
2539 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
2541 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
2543 * tests/check/gst/media.c:
2544 media test: cleanup the thread pool in tests
2546 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
2548 * gst/rtsp-server/rtsp-media.c:
2549 * tests/check/gst/media.c:
2550 rtsp-media: Unblock blocked streams in unprepare
2551 The streams will be blocked when a live media is prepared.
2552 The streams should be unblocked in gst_rtsp_media_unprepare.
2553 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
2555 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
2557 * gst/rtsp-server/rtsp-media.c:
2558 media: release the state lock when going to NULL
2559 Set our state to UNPREPARING and release the state-lock before
2560 setting the pipeline to the NULL state. This way, any pad-added
2561 callback will be able to take the state-lock and check that we are now
2562 unpreparing instead of deadlocking.
2563 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
2565 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
2567 * gst/rtsp-server/rtsp-media.c:
2568 media: protect status with lock
2569 Make sure we only update the status with the lock.
2571 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
2573 * gst/rtsp-server/rtsp-client.c:
2574 * gst/rtsp-server/rtsp-sdp.c:
2575 rtsp: update for MIKEY API changes
2577 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
2579 * gst/rtsp-server/rtsp-client.c:
2580 client: parse the mikey response from the client
2581 Parse the mikey response from the client and update the policy for
2584 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
2586 * gst/rtsp-server/rtsp-stream.c:
2587 * gst/rtsp-server/rtsp-stream.h:
2588 stream: add method to set crypto info
2589 Make a method to configure the crypto information of a stream.
2590 Set udpsrc in READY instead of PAUSED so that we can configure caps
2593 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
2595 * gst/rtsp-server/rtsp-client.c:
2596 client: cleanup error paths
2598 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
2600 * gst/rtsp-server/rtsp-media.c:
2603 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
2605 * examples/test-video.c:
2606 test: enable SRTP only on RTSPS
2607 We only want to enable SRTP when doing rtsp over TLS so that we can
2608 exchange the keys in a secure way.
2610 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
2612 * examples/test-video.c:
2613 test: print an error on failure
2615 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
2618 * examples/test-video.c:
2619 * gst/rtsp-server/rtsp-sdp.c:
2620 * gst/rtsp-server/rtsp-stream.c:
2621 * tests/check/Makefile.am:
2622 stream: add SRTP support
2623 Install srtp encoder and decoder elements in rtpbin
2626 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2628 * tests/check/Makefile.am:
2629 * tests/check/gst/sessionpool.c:
2630 tests: Add unit tests for sessionpool
2631 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
2633 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2635 * tests/check/gst/threadpool.c:
2636 tests: Improve code coverage of rtsp-threadpool tests
2637 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
2639 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2641 * tests/check/gst/sessionmedia.c:
2642 tests: Improve code coverage for rtsp-session-media
2643 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
2645 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2647 gobject-introspection: Add annotations to support language bindings
2648 In addition a few cosmetic changes:
2649 * Adjust the order of arguments
2650 * Fix typo: occured -> occurred
2651 * Fix indentation after Return:-clauses
2652 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
2654 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2656 * gst/rtsp-server/rtsp-stream.c:
2657 rtsp-stream: Don't mix IPv4 and IPv6 addresses
2658 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
2660 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
2662 * gst/rtsp-server/rtsp-stream.c:
2663 stream: take caps after the session manager
2664 Take the caps for the SDP after they leave the rtpbin so that we can
2665 also get the properties added by rtpbin elements.
2667 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
2669 * gst/rtsp-server/rtsp-stream.c:
2670 stream: release lock while pushing out packets
2671 Keep a cache of the transports and use this to iterate the transport
2672 while pushing packets. This allows us to release the lock early.
2673 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
2675 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
2677 * gst/rtsp-server/rtsp-client.c:
2678 * gst/rtsp-server/rtsp-client.h:
2679 rtsp-client: vmethod for modifying tunnel GET response
2680 Add a vmethod tunnel_http_response where the response to the HTTP GET
2681 for tunneled connections can be modified.
2682 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
2684 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
2686 * gst/rtsp-server/rtsp-sdp.c:
2687 sdp: make 1 media line per profile
2688 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
2689 line in the SDP for each profile. The client is then supposed to pick
2690 one of the profiles in the SETUP request. Because the m= lines have the
2691 same pt, the client also knows that only 1 option is possible.
2693 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
2695 * gst/rtsp-server/rtsp-media-factory.c:
2696 * gst/rtsp-server/rtsp-media-factory.h:
2697 * gst/rtsp-server/rtsp-media.c:
2698 factory: add profile property and pass to media and streams
2700 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
2702 * examples/test-multicast.c:
2703 * gst/rtsp-server/rtsp-sdp.c:
2704 sdp: pass multicast connection for multicast-only stream
2705 Pass the multicast address of the stream in the connection info in the
2706 SDP so that clients try a multicast connection first.
2707 Only allow multicast connections in the test-multicast example. Also
2708 increase the TTL a little.
2710 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2713 .gitignore: Ignore gcov intermediate files
2714 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
2716 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
2718 * gst/rtsp-server/rtsp-stream.c:
2719 stream: release some locks in error cases
2721 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2723 docs: Enable and fix gtk-doc warnings
2724 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
2725 * addresspool/mediafactory: Add missing annotation colon
2726 * stream: Annotate return value
2727 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
2729 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
2732 Automatic update of common submodule
2733 From fe1672e to bcb1518
2735 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
2738 Automatic update of common submodule
2739 From 1a07da9 to fe1672e
2741 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
2743 * examples/Makefile.am:
2744 examples: use LDADD for libs instead of LDFLAGS
2746 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
2749 configure: make sure releases are in .doap file
2751 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
2753 * examples/test-cgroups.c:
2754 examples: test-cgroups: don't put code with side effects into g_assert()
2755 The g_assert() might get compiled out with the right
2756 compiler/preprocessor flags.
2758 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
2760 * examples/.gitignore:
2761 examples: add cgroup test binary to .gitignore
2763 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
2765 * examples/test-cgroups.c:
2766 examples: fix cgroup test build
2767 Fixes build failure caused by compiler warning:
2768 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
2770 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
2773 .gitignore: ignore temp files created in the course of 'make check'
2775 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
2777 * gst/rtsp-server/rtsp-media.c:
2778 rtsp-media: don't loose frames handling new PLAY request
2779 If client supplied a range check if the range specifies the start point.
2780 If not, then do an accurate seek to the current position. If a start
2781 point was specified do do a key unit seek to make sure the streaming
2782 starts with decodeable frames.
2783 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
2785 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
2787 * gst/rtsp-server/rtsp-media.c:
2788 Revert "media: only flush when setting a new start position"
2789 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
2790 We need to do the flush in all cases, demuxer block currently for
2793 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
2795 * gst/rtsp-server/rtsp-media.c:
2796 media: only flush when setting a new start position
2797 Only flush the pipeline when we change the start position with
2799 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
2801 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
2803 * gst/rtsp-server/rtsp-stream.c:
2804 stream: set ttl-mc before adding the socket
2805 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
2806 never be set on socket.
2807 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
2809 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
2811 * gst/rtsp-server/rtsp-media.c:
2812 media: stop thread if media is already prepared
2813 in gst_rtsp_media_prepare() the thread is not used if media is already
2814 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
2816 https://bugzilla.gnome.org/show_bug.cgi?id=724182
2818 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
2821 build: Ship gst-rtsp-server.doap file
2823 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
2825 * tests/check/gst/rtspserver.c:
2826 tests: Fix another compiler warning with gcc
2828 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
2830 * gst/rtsp-server/rtsp-client.c:
2831 * gst/rtsp-server/rtsp-mount-points.c:
2832 * gst/rtsp-server/rtsp-stream.c:
2833 * tests/check/gst/client.c:
2834 rtsp-server: Fix lots of compiler warnings with clang
2836 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
2839 * gst-rtsp-server.doap:
2840 * tests/Makefile.am:
2841 configure: Synchronise with the configure scripts of the other modules
2843 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
2846 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
2848 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
2850 * gst/rtsp-server/rtsp-media.c:
2851 * gst/rtsp-server/rtsp-stream.c:
2852 Revert "rtsp-server: support build against last stable release"
2853 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
2854 Let us require 1.2.3 now, which is going to be released in a few
2857 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
2859 * gst/rtsp-server/rtsp-session-media.c:
2860 * gst/rtsp-server/rtsp-stream-transport.c:
2861 session: improve RTP-Info
2862 Ignore streams that can't generate RTP-Info instead of failing.
2863 Don't return the empty string when all streams are unconfigured but
2864 return NULL so that we don't generate and empty RTP-Info header.
2865 Improve docs a little.
2867 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
2869 * gst/rtsp-server/rtsp-session-media.c:
2870 Don't free rtpinfo GString when it is NULL
2871 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
2873 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
2875 * gst/rtsp-server/rtsp-media.c:
2876 media: only set keyframe flag when modifying start
2877 Only set the keyframe flag when we modify the start position. The
2878 keyframe flag should probably be ignored when no change is requested but
2879 until we can claim this is all documented properly and all demuxer
2880 implement this, avoid setting the flag.
2881 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
2883 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
2885 * gst/rtsp-server/rtsp-thread-pool.c:
2886 thread-pool: Unref source after mainloop has quit to avoid races in GLib
2887 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
2889 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
2891 * gst/rtsp-server/rtsp-stream.c:
2892 stream: handle NULL seqnum and rtptime arguments
2894 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
2896 * gst/rtsp-server/rtsp-thread-pool.c:
2897 * tests/check/gst/threadpool.c:
2898 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
2899 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
2901 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
2903 * gst/rtsp-server/rtsp-stream.c:
2904 stream: add fallback for missing stats property
2905 Use a fallback when the payloader does not have a stats property
2906 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
2908 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
2911 Automatic update of common submodule
2912 From f7bc1c3 to 1a07da9
2914 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
2916 * gst/rtsp-server/rtsp-stream.c:
2917 stream: don't leak stats structure
2918 Don't leak the stats structure and deal with NULL stats.
2920 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
2922 * gst/rtsp-server/rtsp-stream.c:
2923 stream: Get rtpinfo properties atomically from payloader
2924 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
2926 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
2928 * gst/rtsp-server/rtsp-media.c:
2929 media: refactor state change functions and signals
2930 Make functions to set the target state and the pipeline state and emit
2931 the signals from those functions.
2933 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
2935 * gst/rtsp-server/rtsp-media.c:
2936 * gst/rtsp-server/rtsp-media.h:
2937 media: add signal to notify of pending state changes
2939 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
2941 * gst/rtsp-server/rtsp-media.c:
2942 * gst/rtsp-server/rtsp-stream.c:
2943 rtsp-server: support build against last stable release
2944 Until 1.2.3 is out with the new get_type function and we
2947 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
2949 * gst/rtsp-server/rtsp-stream.c:
2950 stream: fix compilation
2952 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
2954 * gst/rtsp-server/rtsp-media.c:
2955 * gst/rtsp-server/rtsp-media.h:
2956 * gst/rtsp-server/rtsp-stream.c:
2957 * gst/rtsp-server/rtsp-stream.h:
2958 stream: add property to configure profiles
2960 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
2962 * gst/rtsp-server/rtsp-client.c:
2963 client: let stream check supported transport
2964 Delegate the check if a transport is allowed to the stream.
2965 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
2967 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
2969 * gst/rtsp-server/rtsp-stream.c:
2970 * gst/rtsp-server/rtsp-stream.h:
2971 stream: add method to check supported transport
2972 Add a method to check if a transport is supported
2974 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
2977 configure.ac: Only check for gstreamer-check, not check
2978 We include check in gstreamer-check since quite some time now.
2980 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
2982 * gst/rtsp-server/rtsp-session-media.c:
2983 * gst/rtsp-server/rtsp-stream-transport.c:
2984 * gst/rtsp-server/rtsp-stream.c:
2985 * gst/rtsp-server/rtsp-stream.h:
2986 stream: return clock-rate from get_rtpinfo
2987 And use it to correct the rtptime to the requested start-time.
2988 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
2990 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
2992 * gst/rtsp-server/rtsp-session-media.c:
2993 * gst/rtsp-server/rtsp-stream-transport.c:
2994 * gst/rtsp-server/rtsp-stream-transport.h:
2995 session-media: calculate start-time
2997 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
2999 * gst/rtsp-server/rtsp-stream-transport.c:
3000 * gst/rtsp-server/rtsp-stream.c:
3001 * gst/rtsp-server/rtsp-stream.h:
3002 stream: also return the running-time
3003 Return the running-time in the rtpinfo as well.
3005 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
3007 * gst/rtsp-server/rtsp-client.c:
3008 * gst/rtsp-server/rtsp-session-media.c:
3009 * gst/rtsp-server/rtsp-session-media.h:
3010 * gst/rtsp-server/rtsp-stream-transport.c:
3011 * gst/rtsp-server/rtsp-stream-transport.h:
3012 session-media: let the session-media make the RTPInfo
3013 Add method to create the RTPInfo for a stream-transport.
3014 Add method to create the RTPInfo for all stream-transports in a
3016 Use the session-media RTPInfo code in client. This allows us to refactor
3017 another method to link the TCP callbacks.
3019 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
3021 mount-points: sort sequence before g_sequence_lookup
3022 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
3023 sort sequence if dirty, otherwise lookup will fail.
3024 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
3026 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
3029 configure: rename package from gst-rtsp to gst-rtsp-server
3030 To match git module name and avoid confusion with the
3031 rtsp lib in gst-plugins-base and rtsp plugin in -good.
3033 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
3036 configure: bump core/base/good requirement to 1.2.0
3037 Bump to released stable version and make implicit
3038 requirements explicit.
3040 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
3045 Fix broken gettext setup which is not used anyway
3047 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
3050 Automatic update of common submodule
3051 From dbedaa0 to d48bed3
3053 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
3055 * gst/rtsp-server/rtsp-client.c:
3056 * gst/rtsp-server/rtsp-media.c:
3057 * gst/rtsp-server/rtsp-media.h:
3058 media: add setup_sdp vmethod
3059 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
3060 gst_rtsp_media_setup_sdp.
3061 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
3063 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
3065 * gst/rtsp-server/rtsp-stream.c:
3066 rtsp-stream: Check return value of sscanf
3067 streamid is only valid if sscanf matched something.
3069 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
3071 * gst/rtsp-server/rtsp-client.c:
3072 rtsp-client: Fix iteration
3073 Wouldn't even enter the code block otherwise (i++ was used as the check
3074 and not the postfix).
3076 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
3078 * gst/rtsp-server/rtsp-client.c:
3079 * gst/rtsp-server/rtsp-client.h:
3080 client: add vmethod to configure media and streams
3081 Implement a vmethod that can be used to configure the media and the
3082 streams based on the current context. Handle the blocksize handling in
3083 the default handler.
3084 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
3086 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
3089 Make git ignore more unit test binaries
3091 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
3093 * gst/rtsp-server/rtsp-address-pool.h:
3094 * gst/rtsp-server/rtsp-auth.h:
3095 * gst/rtsp-server/rtsp-client.h:
3096 * gst/rtsp-server/rtsp-context.h:
3097 * gst/rtsp-server/rtsp-media-factory-uri.h:
3098 * gst/rtsp-server/rtsp-media-factory.h:
3099 * gst/rtsp-server/rtsp-media.h:
3100 * gst/rtsp-server/rtsp-mount-points.h:
3101 * gst/rtsp-server/rtsp-server.h:
3102 * gst/rtsp-server/rtsp-session-media.h:
3103 * gst/rtsp-server/rtsp-session-pool.h:
3104 * gst/rtsp-server/rtsp-session.h:
3105 * gst/rtsp-server/rtsp-stream-transport.h:
3106 * gst/rtsp-server/rtsp-stream.h:
3107 * gst/rtsp-server/rtsp-thread-pool.h:
3108 * gst/rtsp-server/rtsp-token.h:
3109 rtsp-server: add padding to many public structures
3110 Not mini objects though, since they are not subclassable
3111 anyway, nor kept on the stack or inlined in a structure.
3113 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
3115 media: add new create_rtpbin vmethod
3116 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
3117 https://bugzilla.gnome.org/show_bug.cgi?id=719734
3119 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
3121 * tests/check/gst/media.c:
3122 tests: fix memory leak, free test's thread pool
3123 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
3125 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
3127 * gst/rtsp-server/rtsp-stream-transport.c:
3128 stream-transport: free url in finalize
3130 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
3132 * gst/rtsp-server/rtsp-media.c:
3133 media: also do state change in suspended state
3135 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
3137 * gst/rtsp-server/rtsp-client.c:
3138 * gst/rtsp-server/rtsp-media.c:
3139 media: also handle prepare and range in suspended state
3140 When we are suspended, we are already prepared.
3141 We can get the range in the suspended state.
3143 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
3145 * tests/check/Makefile.am:
3146 * tests/check/gst/sessionmedia.c:
3147 check: add test for uri in setup
3148 Added unit tests for the new functionality in GstRTSPStreamTransport.
3149 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
3151 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
3153 * gst/rtsp-server/rtsp-client.c:
3154 client: store setup uri and use in PLAY response
3155 Store the uri used when doing the setup and use that in the PLAY
3157 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
3159 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
3161 * gst/rtsp-server/rtsp-stream-transport.c:
3162 * gst/rtsp-server/rtsp-stream-transport.h:
3163 stream-transport: add method to get/set url
3165 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
3167 * gst/rtsp-server/rtsp-client.c:
3168 client: suspend after SDP and unsuspend before PLAYING
3169 Based on patches by Ognyan Tonchev <ognyan@axis.com>
3170 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
3172 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
3174 * gst/rtsp-server/rtsp-media-factory.c:
3175 * gst/rtsp-server/rtsp-media-factory.h:
3176 * gst/rtsp-server/rtsp-media.c:
3177 * gst/rtsp-server/rtsp-media.h:
3178 * gst/rtsp-server/rtsp-session-media.c:
3179 * gst/rtsp-server/rtsp-session.c:
3180 * tests/check/gst/media.c:
3181 * tests/check/gst/mediafactory.c:
3182 media: add suspend modes
3183 Add support for different suspend modes. The stream is suspended right after
3184 producing the SDP and after PAUSE. Different suspend modes are available that
3185 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
3186 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
3187 state and RESET will bring the pipeline to the NULL state.
3188 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
3189 this means that the pipeline needs to be prerolled again.
3190 Base on patches by Ognyan Tonchev <ognyan@axis.com>
3191 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3193 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
3195 * gst/rtsp-server/rtsp-media.c:
3196 media: start live streams in blocked state
3197 Start live streams in the blocked state and make them preroll using the
3198 messages. This ensure that no data is played by the sink until we explicitly
3199 unblock the stream right before going to PLAYING.
3200 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3202 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
3204 * gst/rtsp-server/rtsp-media.c:
3205 media: refactor starting and waiting for preroll
3206 Based on patches from Ognyan Tonchev <ognyan@axis.com>
3207 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3209 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
3211 * gst/rtsp-server/rtsp-stream.c:
3212 * gst/rtsp-server/rtsp-stream.h:
3213 stream: add API to block streams
3214 Add an API to block on the streams and make it post a message.
3215 Based on patch by Ognyan Tonchev <ognyan@axis.com>
3216 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3218 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
3220 * docs/libs/Makefile.am:
3221 docs: Specify the override file
3222 Even if it's empty (for now) it avoids make distcheck complaining
3224 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
3226 * gst/rtsp-server/rtsp-media.c:
3227 media: move default implementations to where they are used
3229 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
3231 * gst/rtsp-server/rtsp-media.c:
3232 media: take the right lock in gst_rtsp_media_set_pipeline_state()
3233 We need to take the state_lock when calling this method.
3235 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
3237 * gst/rtsp-server/rtsp-media.c:
3238 media: handle add-added on non-bins too
3239 Handle dynamic payloaders that are not bins, as used in the unit-test.
3241 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3243 * gst/rtsp-server/rtsp-media-factory.c:
3244 * gst/rtsp-server/rtsp-media-factory.h:
3245 * gst/rtsp-server/rtsp-media.c:
3246 rtsp-media/-factory: Fix request pad name comments
3247 These must be escaped for gtk-doc to parse the comments without warnings.
3249 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
3251 rtsp-media: remove transports if media is in error status
3252 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
3253 trying to change to GST_STATE_NULL and media is in error status, we
3254 remove all transports.
3255 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
3257 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
3259 * gst/rtsp-server/rtsp-media.c:
3260 rtsp-media: use element metadata to find payloader
3261 Use the element metadata to find the payloader instead of checking
3263 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
3265 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
3267 rtsp-stream: add getter for payload type
3268 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
3269 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
3270 element and create the stream with this one instead of the dynpay%d
3272 https://bugzilla.gnome.org/show_bug.cgi?id=712396
3274 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3276 * gst/rtsp-server/rtsp-client.c:
3277 * gst/rtsp-server/rtsp-context.h:
3278 * gst/rtsp-server/rtsp-media.c:
3279 * gst/rtsp-server/rtsp-mount-points.c:
3280 * gst/rtsp-server/rtsp-server.c:
3281 * gst/rtsp-server/rtsp-token.c:
3282 rtsp-*: Refer to NULL as a constant in comments
3284 https://bugzilla.gnome.org/show_bug.cgi?id=714988
3286 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3288 rtsp-*: Fix type name typos in comments
3289 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
3290 * rtsp-auth: Refer to part of constant name as text
3291 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
3292 * rtsp-session-media: Fix GstRTSPSessionMedia typo
3293 * rtsp-stream: Fix typo when refering to GstBin
3294 https://bugzilla.gnome.org/show_bug.cgi?id=714988
3296 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3299 * docs/libs/gst-rtsp-server-docs.sgml:
3300 * docs/libs/gst-rtsp-server-sections.txt:
3301 docs: Improve documentation
3302 * Include annotation-glossary to quiet gtk-doc
3303 * Rename remaining ClientState -> Context
3304 * Rename object hierarchy file
3305 * Remove stale chapter references
3306 * Add missing function and object references
3307 * Include missing GstRTSPAddressPoolResult
3308 https://bugzilla.gnome.org/show_bug.cgi?id=714988
3310 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
3312 * gst/rtsp-server/rtsp-client.c:
3313 * gst/rtsp-server/rtsp-server.c:
3314 * gst/rtsp-server/rtsp-session-pool.c:
3315 * gst/rtsp-server/rtsp-session.c:
3316 * gst/rtsp-server/rtsp-stream.c:
3317 rtsp-server: sprinkle some allow-none annotations for g-i
3319 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
3321 * gst/rtsp-server/rtsp-stream.c:
3322 * gst/rtsp-server/rtsp-stream.h:
3323 stream: add method to filter transports
3324 Add a method to safely iterate and collect the stream transports
3325 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
3327 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
3329 * gst/rtsp-server/rtsp-client.c:
3330 * gst/rtsp-server/rtsp-server.c:
3331 * gst/rtsp-server/rtsp-session-pool.c:
3332 * gst/rtsp-server/rtsp-session.c:
3333 rtsp: allow NULL func in filters
3334 Passing a null function make the filters return a list of
3337 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
3339 * gst/rtsp-server/rtsp-address-pool.c:
3340 * tests/check/gst/addresspool.c:
3341 address-pool: fix address increment
3342 Use a guint instead of guint8 to increment the address. It's still not
3343 completely correct because a guint might not be able to hold the complete
3344 address range, but that's an enhacement for later.
3345 Add unit test to test improved behaviour.
3346 https://bugzilla.gnome.org/show_bug.cgi?id=708237
3348 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
3350 * gst/rtsp-server/rtsp-client.c:
3351 * tests/check/gst/client.c:
3352 client: allow absolute path in requests
3353 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
3355 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
3357 * gst/rtsp-server/rtsp-client.c:
3358 * gst/rtsp-server/rtsp-client.h:
3359 client: make make_path_from_uri a vmethod
3361 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
3363 * docs/libs/gst-rtsp-server-sections.txt:
3364 * gst/rtsp-server/rtsp-stream.c:
3365 * gst/rtsp-server/rtsp-stream.h:
3366 * tests/check/Makefile.am:
3367 * tests/check/gst/stream.c:
3368 stream: Add functions to get rtp and rtcp sockets
3369 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
3371 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
3373 * gst/rtsp-server/rtsp-context.c:
3374 * gst/rtsp-server/rtsp-context.h:
3375 context: defing a GType for the context
3376 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
3378 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
3380 * gst/rtsp-server/Makefile.am:
3381 * gst/rtsp-server/rtsp-auth.c:
3382 * gst/rtsp-server/rtsp-context.c:
3383 * gst/rtsp-server/rtsp-media.c:
3384 * gst/rtsp-server/rtsp-mount-points.c:
3385 * gst/rtsp-server/rtsp-server.h:
3386 * gst/rtsp-server/rtsp-session-media.c:
3387 * gst/rtsp-server/rtsp-session.c:
3388 * gst/rtsp-server/rtsp-stream.c:
3389 Fixed several GIR warnings
3391 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
3393 * gst/rtsp-server/rtsp-auth.c:
3396 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3398 * tests/check/Makefile.am:
3399 * tests/check/gst/token.c:
3400 tests: Add unit tests for token
3401 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
3403 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3405 * gst/rtsp-server/rtsp-token.c:
3406 token: Validate args for gst_rtsp_token_is_allowed
3407 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
3409 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3411 * gst/rtsp-server/rtsp-token.c:
3412 token: Fix bug when creating empty token
3413 We always want to have a valid GstStructure in the token.
3414 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
3416 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
3418 * gst/rtsp-server/rtsp-thread-pool.c:
3419 thread-pool: avoid race in shutdown
3420 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
3421 don't actually stop the mainloop ever. Solve this race by adding an idle source
3422 to the mainloop that calls the _quit. This way we immediately exit the mainloop
3423 if quit was called before we started it.
3425 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3427 * tests/check/Makefile.am:
3428 * tests/check/gst/permissions.c:
3429 tests: Add unit tests for permissions
3430 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
3432 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3434 * tests/check/gst/mediafactory.c:
3435 tests: Test mediafactory permissions
3436 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
3438 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3440 * gst/rtsp-server/rtsp-permissions.c:
3441 permissions: Fix refcounting when adding/removing roles
3442 Previously a role that was removed was unreffed twice, and when
3443 replacing an existing role the replaced role was freed while still being
3444 referenced. Both bugs are now fixed.
3445 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
3447 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3449 * tests/check/gst/media.c:
3450 * tests/check/gst/mediafactory.c:
3451 * tests/check/gst/rtspserver.c:
3452 tests: Check gst_rtsp_url_parse return value
3453 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
3455 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
3458 Automatic update of common submodule
3459 From 865aa20 to dbedaa0
3461 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
3463 * gst/rtsp-server/rtsp-server.c:
3464 rtsp-server: Fix socket leak
3465 https://bugzilla.gnome.org/show_bug.cgi?id=710088
3467 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
3469 * gst/rtsp-server/rtsp-session-pool.c:
3470 rtsp-session-pool: Make sure session IDs are properly URI-escaped
3471 https://bugzilla.gnome.org/show_bug.cgi?id=643812
3473 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
3475 * examples/.gitignore:
3476 * examples/test-video.c:
3477 examples: fix compilation when WITH_AUTH is defined
3478 https://bugzilla.gnome.org/show_bug.cgi?id=710228
3480 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
3483 gitignore: Add new test binary
3485 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
3487 * tests/check/Makefile.am:
3488 * tests/check/gst/threadpool.c:
3489 thread-pool: Add unit test for the thread pools
3490 https://bugzilla.gnome.org/show_bug.cgi?id=710228
3492 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
3494 * gst/rtsp-server/rtsp-thread-pool.c:
3495 thread-pool: Fix thread leak when reusing threads
3496 https://bugzilla.gnome.org/show_bug.cgi?id=709730
3498 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
3500 * gst/rtsp-server/rtsp-server.c:
3501 * tests/check/gst/rtspserver.c:
3502 tests: fixed racy behavior in rtspserver tests
3503 https://bugzilla.gnome.org/show_bug.cgi?id=710078
3505 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3507 * tests/check/gst/addresspool.c:
3508 tests: Improve address pool unit tests
3509 Add a range with mixed IPV4 and IPV6 addresses to pool.
3510 Get an IPV4 address from an IPV6-only pool.
3511 Get an IPV6 address from an IPV4-only pool.
3512 Reserve a IPV6 address from an IPV4-only pool.
3513 Check for unicast addresses in multicast-only pool.
3514 Check for unicast addresses in uni-/multicast-mixed pool.
3515 https://bugzilla.gnome.org/show_bug.cgi?id=710128
3517 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3519 * gst/rtsp-server/rtsp-client.c:
3520 client: append query string in PAUSE/PLAY/TEARDOWN as well
3522 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
3524 * gst/rtsp-server/rtsp-client.c:
3525 client: Add query to control path
3526 If the SETUP url contains a query it must be appended to the control
3527 path so that it matches any already created stream in the media. The
3528 query will also be appended to the session media path.
3530 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3532 * gst/rtsp-server/rtsp-media.c:
3533 rtsp-media: remove old line
3535 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
3537 * gst/rtsp-server/rtsp-stream.c:
3538 stream: Correct control comparison
3539 https://bugzilla.gnome.org/show_bug.cgi?id=709176
3541 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3543 * gst/rtsp-server/rtsp-media.c:
3544 media: Check dynamically if the pipeline supports seeking
3545 We should not depend on whether or not the pipeline state change
3546 returned NO_PREROLL or not. A media could dynamically change its
3547 element and switch from seekable to non seekable so it's best to test
3548 the seekable nature of the pipeline dynamically when we try to do a seek.
3550 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3552 * gst/rtsp-server/rtsp-media.c:
3553 media: Return FALSE if seeking is not supported
3555 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3557 * gst/rtsp-server/rtsp-media.c:
3558 rtsp-media: don't seek accurate by default
3559 Accurate seeking is perhaps a little overkill in the most common situation and
3560 causes some formats (mp3) over slow media to seek extremely slowly.
3562 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
3564 * tests/check/gst/rtspserver.c:
3565 tests: fix unit test
3566 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
3568 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
3570 * gst/rtsp-server/rtsp-client.c:
3571 client: Reply 400 if media cannot be constructed
3572 Reply 400 Bad Request instead of 503 Service Unavailable if media
3573 cannot be constructed in SETUP.
3574 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
3576 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
3578 * gst/rtsp-server/rtsp-client.c:
3579 client: Send setup reply once only
3580 If find_media() failed in handle_setup_request() two replies was sent.
3581 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
3583 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
3586 Automatic update of common submodule
3587 From 6b03ba7 to 865aa20
3589 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
3591 * gst/rtsp-server/rtsp-server.c:
3592 server: Emit client-connected signal earlier
3593 Emit client-connected before the client ref is given to a GSource,
3594 otherwise client-connected can be emitted after the client object has
3597 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
3599 * gst/rtsp-server/rtsp-address-pool.c:
3600 * gst/rtsp-server/rtsp-address-pool.h:
3601 * gst/rtsp-server/rtsp-stream.c:
3602 * tests/check/gst/addresspool.c:
3603 addresspool: return reason of failure
3604 Let gst_rtsp_address_pool_reserve_address() return the reason why
3605 the address could not be reserved.
3606 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
3608 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
3611 autogen.sh: Sync behaviour with other GStreamer modules
3612 Allows building from outside of tree amongst other things
3614 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
3617 Automatic update of common submodule
3618 From b613661 to 6b03ba7
3620 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
3623 Automatic update of common submodule
3624 From 74a6857 to b613661
3626 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
3629 Automatic update of common submodule
3630 From 01a7a46 to 74a6857
3632 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
3634 * gst/rtsp-server/rtsp-client.c:
3635 client: Do not read beyond end of path string
3636 If the setup was done without a control url, make sure we don't try to read the
3637 non-existing control string and crash.
3639 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3641 * gst/rtsp-server/rtsp-client.c:
3642 client: Fix RTPInfo header
3643 Refactor the method to make the content_base.
3644 Use the content-base and the control url to construct the RTPInfo
3647 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3649 * gst/rtsp-server/rtsp-client.c:
3650 client: map url to path only in describe
3651 Only map the request url to a path in the DESCRIBE method. The SDP then
3652 contains the base and control urls that should be used to SETUP/PAUSE/
3653 PLAY/TEARDOWN the media.
3655 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3657 * gst/rtsp-server/rtsp-client.c:
3658 Revert "client: map URL to path in requests"
3659 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
3660 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
3661 contains the base and control urls which are used in the SETUP, PLAY,
3662 PAUSE and TEARDOWN requests.
3664 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3666 * gst/rtsp-server/rtsp-client.c:
3667 client: map URL to path in requests
3669 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3671 * gst/rtsp-server/rtsp-client.c:
3672 * gst/rtsp-server/rtsp-mount-points.c:
3673 * gst/rtsp-server/rtsp-mount-points.h:
3674 mount-points: make vmethod to make path from uri
3675 Make a vmethod to transform an url into a path. The path is then used to lookup
3676 the factory. This makes it possible to also use other bits of the url, such as
3677 the query parameters, to locate the factory.
3679 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
3681 * gst/rtsp-server/rtsp-thread-pool.c:
3682 * gst/rtsp-server/rtsp-thread-pool.h:
3683 thread-pool: Add cleanup to wait for the threadpool to finish
3684 Also fix race condition if two threads are asking for the first
3685 thread from the thread pool at once. This would case two internal
3686 GThreadPools to be created.
3687 https://bugzilla.gnome.org/show_bug.cgi?id=707753
3689 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
3691 * gst/rtsp-server/rtsp-client.c:
3692 * tests/check/gst/client.c:
3693 client: free threadpool
3694 https://bugzilla.gnome.org/show_bug.cgi?id=707638
3696 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
3698 * tests/check/gst/mountpoints.c:
3699 mountpoints tests: unref matched factories
3700 https://bugzilla.gnome.org/show_bug.cgi?id=707638
3702 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
3704 * tests/check/gst/media.c:
3705 media tests: unref thread pool and caps
3706 https://bugzilla.gnome.org/show_bug.cgi?id=707638
3708 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
3710 * gst/rtsp-server/rtsp-auth.c:
3711 * gst/rtsp-server/rtsp-media-factory.c:
3712 * gst/rtsp-server/rtsp-media.c:
3713 auth, media, media-factory: unref permissions
3714 https://bugzilla.gnome.org/show_bug.cgi?id=707638
3716 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3718 * examples/Makefile.am:
3719 Makefile: add rule for appsrc example
3721 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3723 * examples/test-appsrc.c:
3724 tests: add appsrc example
3725 Add an example on how to use appsrc to feed the server pipeline with data.
3727 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
3729 * gst/rtsp-server/rtsp-client.c:
3730 rtsp-client: remove query part from content-base string
3731 Make sure that after the control url has been resolved, it's
3732 not a part of the query-string.
3733 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
3735 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3737 * gst/rtsp-server/rtsp-client.c:
3738 client: don't check url in response
3739 There is no url or method in the response to check
3741 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3743 * gst/rtsp-server/rtsp-client.c:
3744 * gst/rtsp-server/rtsp-client.h:
3745 Add handle-response signal for when we receive a GET_PARAMETER response
3747 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3749 * gst/rtsp-server/rtsp-server.c:
3750 Fix gst_rtsp_server_client_filter, using wrong variable type
3752 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
3754 * gst/rtsp-server/rtsp-media-factory-uri.c:
3755 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
3756 For AAC we need to check for framed=true instead of parsed=true.
3757 https://bugzilla.gnome.org/show_bug.cgi?id=701384
3759 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3761 * gst/rtsp-server/rtsp-stream.c:
3762 stream: optimize pipeline for protocols
3763 When TCP is not an allowed protocol for the stream, avoid creating the
3764 appsrc/appsink/queue and tee elements.
3766 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3768 * gst/rtsp-server/rtsp-media.c:
3769 media: set protocols on streams
3771 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3773 * gst/rtsp-server/rtsp-client.c:
3774 client: use protocols supported by stream
3776 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3778 * gst/rtsp-server/rtsp-media-factory.c:
3779 * gst/rtsp-server/rtsp-media.c:
3780 * gst/rtsp-server/rtsp-stream.c:
3781 media-factory: allow all protocols
3783 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3785 * gst/rtsp-server/rtsp-media.c:
3786 media: configure protocols in new streams
3788 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3790 * gst/rtsp-server/rtsp-stream.c:
3791 * gst/rtsp-server/rtsp-stream.h:
3792 stream: add protocols property
3794 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3796 * gst/rtsp-server/rtsp-media.c:
3797 rtsp-media: send state in "new-state" signal
3798 https://bugzilla.gnome.org/show_bug.cgi?id=705110
3800 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
3803 build: add subdir-objects to AM_INIT_AUTOMAKE
3804 Fixes warnings with automake 1.14
3805 https://bugzilla.gnome.org/show_bug.cgi?id=705350
3807 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3809 * docs/libs/gst-rtsp-server-sections.txt:
3810 * gst/rtsp-server/rtsp-client.c:
3811 * gst/rtsp-server/rtsp-server.c:
3812 * gst/rtsp-server/rtsp-server.h:
3813 server: add method to iterate clients of server
3815 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3817 * gst/rtsp-server/rtsp-media.c:
3818 * gst/rtsp-server/rtsp-media.h:
3819 Add vmethod for rtsp-media subclass to access rtpbin
3821 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3823 * gst/rtsp-server/rtsp-client.h:
3824 small documentation fix
3826 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3828 * gst/rtsp-server/rtsp-client.c:
3829 Do not take range header if range is invalid
3831 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3833 * docs/libs/gst-rtsp-server-sections.txt:
3834 * gst/rtsp-server/rtsp-media.c:
3835 media: add docs for new method
3837 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3839 * gst/rtsp-server/rtsp-media.c:
3840 * gst/rtsp-server/rtsp-media.h:
3841 Add API to rtsp-media set the pipeline's state
3843 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3845 * gst/rtsp-server/rtsp-media.c:
3846 Update current position/duration when gst_rtsp_media_get_range_string is called
3848 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3850 * examples/test-cgroups.c:
3851 tests: add some more docs
3853 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3855 * examples/test-cgroups.c:
3856 * gst/rtsp-server/Makefile.am:
3857 * gst/rtsp-server/rtsp-auth.c:
3858 * gst/rtsp-server/rtsp-auth.h:
3859 * gst/rtsp-server/rtsp-client.c:
3860 * gst/rtsp-server/rtsp-client.h:
3861 * gst/rtsp-server/rtsp-context.c:
3862 * gst/rtsp-server/rtsp-context.h:
3863 * gst/rtsp-server/rtsp-params.c:
3864 * gst/rtsp-server/rtsp-params.h:
3865 * gst/rtsp-server/rtsp-server.c:
3866 * gst/rtsp-server/rtsp-thread-pool.c:
3867 * gst/rtsp-server/rtsp-thread-pool.h:
3868 * tests/check/gst/client.c:
3869 ClientState -> Context
3870 Rename the clientstate to context and put the code in a separate file.
3872 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3874 * examples/test-auth.c:
3875 * gst/rtsp-server/rtsp-auth.c:
3876 * gst/rtsp-server/rtsp-auth.h:
3877 auth: add support for default token
3878 The default token is used when the user is not authenticated and can be used to
3879 give minimal permissions.
3881 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3883 * examples/test-auth.c:
3884 * gst/rtsp-server/rtsp-auth.c:
3885 auth: use defines when possible
3887 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3889 * gst/rtsp-server/rtsp-address-pool.c:
3890 address-pool: improve docs
3892 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3894 * gst/rtsp-server/rtsp-permissions.c:
3895 permissions: add the role to the copy
3897 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
3899 * gst/rtsp-server/rtsp-permissions.c:
3900 permissions: Also copy the roles
3902 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
3904 * gst/rtsp-server/rtsp-permissions.c:
3905 permissions: Make it build
3907 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3909 * gst/rtsp-server/rtsp-address-pool.h:
3912 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3914 * docs/libs/gst-rtsp-server-sections.txt:
3915 * gst/rtsp-server/rtsp-auth.c:
3916 * gst/rtsp-server/rtsp-auth.h:
3917 * gst/rtsp-server/rtsp-media.c:
3918 * gst/rtsp-server/rtsp-session-media.c:
3919 * gst/rtsp-server/rtsp-stream-transport.c:
3920 * gst/rtsp-server/rtsp-stream-transport.h:
3921 * gst/rtsp-server/rtsp-stream.c:
3922 * tests/check/gst/client.c:
3925 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3927 * docs/libs/gst-rtsp-server-sections.txt:
3928 * gst/rtsp-server/rtsp-address-pool.c:
3929 * gst/rtsp-server/rtsp-address-pool.h:
3930 * tests/check/gst/addresspool.c:
3931 * tests/check/gst/rtspserver.c:
3932 address-pool: cleanups
3933 Remove redundant method, improve docs.
3935 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3937 * docs/libs/gst-rtsp-server-sections.txt:
3938 * gst/rtsp-server/rtsp-auth.h:
3939 * gst/rtsp-server/rtsp-permissions.c:
3940 * gst/rtsp-server/rtsp-permissions.h:
3941 * gst/rtsp-server/rtsp-token.c:
3944 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3946 * gst/rtsp-server/rtsp-permissions.c:
3947 permissions: implement _remove_role
3949 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3951 * gst/rtsp-server/rtsp-permissions.c:
3952 permissions: update docs
3954 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3956 * tests/check/gst/client.c:
3957 tests: simplify tests
3958 Client settings are now disabled by default so we don't need an auth
3959 module to disable them.
3961 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3963 * gst/rtsp-server/rtsp-auth.c:
3964 auth: add default authorizations
3965 When no auth module is specified, use our table of defaults to look up the
3966 default value of the check instead of always allowing everything. This was
3967 we can disallow client settings by default.
3969 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3972 README: update readme
3974 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3976 * gst/rtsp-server/rtsp-thread-pool.c:
3977 * gst/rtsp-server/rtsp-thread-pool.h:
3978 thread-pool: add more docs
3980 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3982 * gst/rtsp-server/rtsp-thread-pool.c:
3983 * gst/rtsp-server/rtsp-thread-pool.h:
3984 thread-pool: fix race in thread reuse
3985 If we try to reuse a thread right after we made it stop, we end up using a
3986 stopped thread. Catch this case and only reuse threads that are not stopping.
3988 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3990 * gst/rtsp-server/rtsp-server.c:
3991 server: add small debug
3993 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3995 * tests/check/gst/client.c:
3997 Add some permissions to media so we can use the auth and enable
4000 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4002 * gst/rtsp-server/rtsp-client.c:
4003 client: support pushed context in handle_request
4004 If we already have a pushed state, reuse it and add our own things. This makes
4005 it easier to write tests.
4007 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4009 * gst/rtsp-server/rtsp-auth.c:
4010 auth: don't auth on methods
4011 Don't authorize on methods anymore but on the resources that we
4012 try to access, this is more flexible.
4013 Move the authorization checks to where they are needed and let the
4014 check return the response on error.
4016 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4018 * gst/rtsp-server/rtsp-mount-points.c:
4019 mount-points: add some debug
4021 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4023 * tests/check/gst/client.c:
4024 tests: almost fix test
4026 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4028 * gst/rtsp-server/rtsp-auth.c:
4029 * gst/rtsp-server/rtsp-auth.h:
4030 * gst/rtsp-server/rtsp-client.c:
4031 * gst/rtsp-server/rtsp-client.h:
4032 * gst/rtsp-server/rtsp-server.c:
4033 * gst/rtsp-server/rtsp-server.h:
4034 auth: let the auth module check client_settings
4035 Let the auth module decide if client settings are allowed for the
4038 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4040 * gst/rtsp-server/rtsp-token.c:
4041 * gst/rtsp-server/rtsp-token.h:
4042 token: add method to check boolean permission
4044 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4046 * examples/test-auth.c:
4047 * examples/test-cgroups.c:
4048 * gst/rtsp-server/rtsp-token.c:
4049 * gst/rtsp-server/rtsp-token.h:
4050 token: simplify token constructor
4051 Use variable arguments to make easier API.
4053 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4055 * examples/test-auth.c:
4056 * examples/test-cgroups.c:
4057 * gst/rtsp-server/rtsp-media-factory.c:
4058 * gst/rtsp-server/rtsp-media-factory.h:
4059 media-factory: add convenience API for factory
4061 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4063 * examples/test-auth.c:
4064 * examples/test-cgroups.c:
4065 * gst/rtsp-server/rtsp-permissions.c:
4066 * gst/rtsp-server/rtsp-permissions.h:
4067 permissions: simplify API a little
4068 Avoid passing GstStructure in the add_role method, use varargs instead
4069 to construct the structure behind the scenes. We can then also use the
4070 structure name as the role and simplify some more logic.
4072 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4074 * gst/rtsp-server/rtsp-auth.c:
4077 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4079 * gst/rtsp-server/rtsp-auth.c:
4080 * gst/rtsp-server/rtsp-auth.h:
4081 * gst/rtsp-server/rtsp-client.c:
4082 auth: handle unauthorized response
4083 Move handling of the unauthorized response to the auth module, it can add
4084 the appropriate headers to request authorization for the required method
4085 much better than the client.
4087 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4089 * gst/rtsp-server/rtsp-client.c:
4090 * gst/rtsp-server/rtsp-client.h:
4091 client: allow for sending any message, not only requests
4092 Change the _send_request() method to _send_message() so that we
4093 can both send requests and replies.
4095 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4097 * docs/libs/gst-rtsp-server-sections.txt:
4098 * gst/rtsp-server/rtsp-server.h:
4101 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4103 * examples/test-video.c:
4104 * gst/rtsp-server/rtsp-auth.c:
4105 * gst/rtsp-server/rtsp-auth.h:
4106 * gst/rtsp-server/rtsp-server.c:
4107 * gst/rtsp-server/rtsp-server.h:
4108 auth: move TLS handling to auth module
4109 Remove the TLS settings on the server and move it to the auth module because
4110 that is where security related bits go.
4112 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4114 * gst/rtsp-server/rtsp-client.c:
4115 * gst/rtsp-server/rtsp-client.h:
4116 client: add state push/pop
4118 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4120 * gst/rtsp-server/rtsp-client.c:
4121 * gst/rtsp-server/rtsp-client.h:
4122 client: add connection to state
4124 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4126 * gst/rtsp-server/rtsp-mount-points.c:
4127 mount-points: fix debug
4129 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4131 * tests/check/gst/media.c:
4132 tests: fix media test
4134 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4136 * gst/rtsp-server/rtsp-thread-pool.c:
4137 thread-pool: we don't require a state
4139 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4141 * gst/rtsp-server/rtsp-server.c:
4142 server: let context ref the server
4143 So that we don't risk losing the server object early anc crash.
4145 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4147 * tests/check/gst/client.c:
4148 tests: fix client test
4150 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4153 * docs/libs/gst-rtsp-server-docs.sgml:
4154 * docs/libs/gst-rtsp-server-sections.txt:
4155 * gst/rtsp-server/rtsp-address-pool.c:
4156 * gst/rtsp-server/rtsp-auth.c:
4157 * gst/rtsp-server/rtsp-client.c:
4158 * gst/rtsp-server/rtsp-client.h:
4159 * gst/rtsp-server/rtsp-media-factory-uri.c:
4160 * gst/rtsp-server/rtsp-media-factory.c:
4161 * gst/rtsp-server/rtsp-media-factory.h:
4162 * gst/rtsp-server/rtsp-media.c:
4163 * gst/rtsp-server/rtsp-mount-points.c:
4164 * gst/rtsp-server/rtsp-params.c:
4165 * gst/rtsp-server/rtsp-permissions.c:
4166 * gst/rtsp-server/rtsp-sdp.c:
4167 * gst/rtsp-server/rtsp-server.c:
4168 * gst/rtsp-server/rtsp-server.h:
4169 * gst/rtsp-server/rtsp-session-media.c:
4170 * gst/rtsp-server/rtsp-session-pool.c:
4171 * gst/rtsp-server/rtsp-session.c:
4172 * gst/rtsp-server/rtsp-stream-transport.c:
4173 * gst/rtsp-server/rtsp-stream.c:
4174 * gst/rtsp-server/rtsp-thread-pool.c:
4175 * gst/rtsp-server/rtsp-token.c:
4178 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4180 * gst/rtsp-server/rtsp-session-pool.c:
4181 * gst/rtsp-server/rtsp-session-pool.h:
4182 session-pool: make vmethod to create a session
4183 Make a vmethod to create a sessions so that subclasses can create
4184 custom session objects
4186 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4188 * gst/rtsp-server/rtsp-auth.c:
4189 * gst/rtsp-server/rtsp-media-factory.h:
4190 * gst/rtsp-server/rtsp-media.h:
4191 * gst/rtsp-server/rtsp-mount-points.h:
4192 * gst/rtsp-server/rtsp-session-pool.h:
4193 * gst/rtsp-server/rtsp-stream.h:
4196 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4198 * docs/libs/gst-rtsp-server-docs.sgml:
4199 * docs/libs/gst-rtsp-server-sections.txt:
4200 * gst/rtsp-server/rtsp-address-pool.c:
4201 * gst/rtsp-server/rtsp-address-pool.h:
4202 * gst/rtsp-server/rtsp-auth.c:
4203 * gst/rtsp-server/rtsp-client.h:
4204 * gst/rtsp-server/rtsp-media-factory.h:
4205 * gst/rtsp-server/rtsp-media.c:
4206 * gst/rtsp-server/rtsp-media.h:
4207 * gst/rtsp-server/rtsp-permissions.c:
4208 * gst/rtsp-server/rtsp-permissions.h:
4209 * gst/rtsp-server/rtsp-server.h:
4210 * gst/rtsp-server/rtsp-session-media.c:
4211 * gst/rtsp-server/rtsp-session-media.h:
4212 * gst/rtsp-server/rtsp-session-pool.h:
4213 * gst/rtsp-server/rtsp-session.h:
4214 * gst/rtsp-server/rtsp-stream-transport.h:
4215 * gst/rtsp-server/rtsp-stream.c:
4216 * gst/rtsp-server/rtsp-thread-pool.h:
4219 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4222 * examples/Makefile.am:
4223 configure: compile cgroup example conditionally
4224 Only compile the cgroup example when we have libcgroup
4226 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4229 * examples/Makefile.am:
4230 * examples/test-cgroups.c:
4231 examples: add cgroups example
4233 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4235 * tests/check/gst/rtspserver.c:
4236 tests: fix compilation
4238 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4240 * gst/rtsp-server/rtsp-thread-pool.c:
4241 thread-pool: fix vmethod invocation
4243 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4245 * gst/rtsp-server/rtsp-thread-pool.c:
4246 * gst/rtsp-server/rtsp-thread-pool.h:
4247 thread-pool: store thread type in thread
4249 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4251 * gst/rtsp-server/rtsp-client.c:
4252 client: pass thread from pool to media _prepare
4253 Get a thread from the configured threadpool and pass it to the prepare method of
4256 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4258 * gst/rtsp-server/rtsp-media.c:
4259 * gst/rtsp-server/rtsp-media.h:
4260 media: Accept a thread in _prepare
4261 Remove out own threadpool handling and use the provided thread and
4262 maincontext for the bus messages and the state changes.
4264 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4266 * gst/rtsp-server/rtsp-server.c:
4267 server: configure client thread pool
4269 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4271 * gst/rtsp-server/rtsp-client.c:
4272 * gst/rtsp-server/rtsp-client.h:
4273 client: add method to configure thread pool
4275 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4277 * gst/rtsp-server/rtsp-client.h:
4278 * gst/rtsp-server/rtsp-server.c:
4279 * gst/rtsp-server/rtsp-server.h:
4280 server: use thread pool
4281 Use the thread pool instead of doing our own thing.
4283 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4285 * gst/rtsp-server/Makefile.am:
4286 * gst/rtsp-server/rtsp-thread-pool.c:
4287 * gst/rtsp-server/rtsp-thread-pool.h:
4288 thread-pool: add object to manage threads
4289 Add an object to manage the client and media threads.
4291 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4293 * gst/rtsp-server/rtsp-auth.c:
4294 auth: debug authorization check
4296 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4298 * gst/rtsp-server/rtsp-media.c:
4299 media: start media pipeline in context
4300 Start the media pipeline in the provided context (or our default one
4301 when NULL). This makes sure that we run the bus thread in this context and that
4302 all media threads are children of this context.
4304 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4306 * gst/rtsp-server/rtsp-media-factory.c:
4307 factory: pass permissions to media by default
4309 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4311 * examples/test-auth.c:
4312 test: add permissions to auth test
4313 Ass some permissions to the media factory in the test.
4315 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4317 * gst/rtsp-server/rtsp-auth.c:
4318 * gst/rtsp-server/rtsp-auth.h:
4319 * gst/rtsp-server/rtsp-client.c:
4320 auth: simplify auth checks
4321 Remove client from methods, it's now in the state
4322 Perform the check specified by the string, use the information from the
4323 thread local context.
4325 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4327 * gst/rtsp-server/rtsp-client.c:
4328 * gst/rtsp-server/rtsp-client.h:
4329 client: add state to current thread
4330 Add the client to the ClientState object.
4331 Place the ClientState on the current thread.
4333 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4335 * gst/rtsp-server/rtsp-media-factory.c:
4336 * gst/rtsp-server/rtsp-media-factory.h:
4337 * gst/rtsp-server/rtsp-media.c:
4338 * gst/rtsp-server/rtsp-media.h:
4339 media: make it possible to set permissions
4340 Make it possible to set permissions on media and media factory objects
4342 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4344 * gst/rtsp-server/Makefile.am:
4345 * gst/rtsp-server/rtsp-permissions.c:
4346 * gst/rtsp-server/rtsp-permissions.h:
4347 permissions: add permissions object
4348 Add a mini object to store permissions based on a role.
4350 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4352 * examples/test-auth.c:
4353 * gst/rtsp-server/rtsp-auth.c:
4354 * gst/rtsp-server/rtsp-auth.h:
4355 * gst/rtsp-server/rtsp-client.c:
4356 auth: add auth checks
4357 Add an enum with auth checks and implement the checks in the auth object.
4358 Perform the checks from the client.
4360 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4362 * examples/test-auth.c:
4363 * gst/rtsp-server/rtsp-auth.c:
4364 * gst/rtsp-server/rtsp-auth.h:
4365 * gst/rtsp-server/rtsp-client.h:
4366 auth: use the token after authentication
4367 After we authenticated a user, keep the Token around in the state.
4369 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4371 * gst/rtsp-server/rtsp-client.c:
4372 * gst/rtsp-server/rtsp-media.c:
4373 * gst/rtsp-server/rtsp-media.h:
4374 * tests/check/gst/media.c:
4375 media: add optional context for bus messages
4376 Add an optional mainloop to _prepare that will handle the bus messages instead
4377 of always using the shared mainloop.
4379 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4381 * gst/rtsp-server/Makefile.am:
4382 * gst/rtsp-server/rtsp-token.c:
4383 * gst/rtsp-server/rtsp-token.h:
4384 token: add authorization token
4385 Add a simply miniobject that contains the authorizations. The object contains a
4386 GstStructure that hold all authorization fields. When a user is authenticated,
4387 the auth module will create a Token for the user. The token is then used to
4388 check what operations the user is allowed to do and various other configuration
4391 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4393 * examples/test-auth.c:
4394 * gst/rtsp-server/rtsp-auth.c:
4395 * gst/rtsp-server/rtsp-auth.h:
4396 * gst/rtsp-server/rtsp-client.c:
4397 * gst/rtsp-server/rtsp-client.h:
4398 * gst/rtsp-server/rtsp-media-factory.c:
4399 * gst/rtsp-server/rtsp-media-factory.h:
4400 * gst/rtsp-server/rtsp-media.c:
4401 * gst/rtsp-server/rtsp-media.h:
4402 auth: remove auth from media and factory
4403 Remove the auth object from media and factory. We want to have the RTSPClient
4404 authenticate and authorize resources, there is no need to place another auth
4405 manager on the media/factory.
4407 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4409 * examples/test-auth.c:
4410 * gst/rtsp-server/rtsp-auth.c:
4411 * gst/rtsp-server/rtsp-auth.h:
4412 * gst/rtsp-server/rtsp-client.h:
4413 auth: add support for multiple basic auth tokens
4414 Make it possible to add multiple basic authorisation tokens to one authorization
4415 object. Associate with each token an authorization group that will define what
4416 capabilities are allowed.
4418 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4420 * gst/rtsp-server/rtsp-client.c:
4421 client: error out on non-aggregate control
4422 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
4424 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4426 * gst/rtsp-server/rtsp-client.c:
4427 client: rework setup request a little
4428 Cache the media in DESCRIBE based on the longest matching path with the uri
4429 that we can find in the mount points.
4430 Rework the setup request a little to get the media from the session or from
4431 the longest matching path, this way we can derive the control string as
4432 everything after the path instead of hardcoding it.
4433 Find the stream based on the control string and only open a session when all
4436 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4438 * gst/rtsp-server/rtsp-media.c:
4439 * gst/rtsp-server/rtsp-media.h:
4440 media: add method to find a stream by control url
4442 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4444 * gst/rtsp-server/rtsp-stream.c:
4445 * gst/rtsp-server/rtsp-stream.h:
4446 stream: add method to check control url of stream
4448 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4450 * gst/rtsp-server/rtsp-client.c:
4451 * gst/rtsp-server/rtsp-session-media.c:
4452 * gst/rtsp-server/rtsp-session-media.h:
4453 * gst/rtsp-server/rtsp-session.c:
4454 * gst/rtsp-server/rtsp-session.h:
4455 session: use path matching for session media
4456 Use a path string instead of a uri to lookup session media in the sessions. Also
4457 use path matching to find the largest possible path that matches.
4459 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4461 * gst/rtsp-server/rtsp-client.c:
4462 * gst/rtsp-server/rtsp-mount-points.c:
4463 * gst/rtsp-server/rtsp-mount-points.h:
4464 * tests/check/gst/mountpoints.c:
4465 mount-points: remove useless vmethod
4466 Making lookups in the mount points should not be done with a URL, if there is a
4467 mapping to be done from URL to mount points, we'll need to do it somewhere
4470 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4472 * gst/rtsp-server/rtsp-mount-points.c:
4473 * gst/rtsp-server/rtsp-mount-points.h:
4474 * tests/check/gst/mountpoints.c:
4475 mount-points: improve mount point searching
4476 Use a GSequence to keep track of the mount points.
4477 Match a URL to the longest matching registered mount point. This should be the
4478 URL to perform aggreagate control and the remainder is the stream specific
4480 Add some unit tests for this.
4482 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
4484 * gst/rtsp-server/Makefile.am:
4485 rtsp-server: Allow building of static library
4487 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4489 * tests/check/gst/mediafactory.c:
4490 tests: fix compilation
4492 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4494 * gst/rtsp-server/rtsp-sdp.c:
4495 sdp: get control string from stream
4496 Use the control string as configured in the stream.
4498 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4500 * gst/rtsp-server/rtsp-stream.c:
4501 * gst/rtsp-server/rtsp-stream.h:
4502 stream: add methods and property to set control string
4504 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4506 * gst/rtsp-server/rtsp-client.c:
4508 Rename variables for clarity
4509 Keep media in state when we can
4511 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4513 * gst/rtsp-server/rtsp-client.c:
4514 * gst/rtsp-server/rtsp-stream.c:
4515 * gst/rtsp-server/rtsp-stream.h:
4516 stream: add more support for IPv6
4517 Rename _get_address to _get_multicast_address in GstRTSPStream to
4518 make it clear that this function only deals with multicast.
4519 Make it possible to have both an IPv4 and IPv6 multicast address on
4520 a stream. Give the client an IPv4 or IPv6 address depending on the
4521 address it used to connect to the server.
4522 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
4524 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4526 * gst/rtsp-server/rtsp-client.c:
4529 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4531 * gst/rtsp-server/rtsp-stream.c:
4532 stream: handle failed port allocation
4533 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
4534 can't allocate any family at all. Also keep track of what port families we
4536 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
4538 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4540 * gst/rtsp-server/rtsp-stream.c:
4541 stream: improve docs
4543 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4545 * gst/rtsp-server/rtsp-stream-transport.c:
4546 stream-transport: remove old if 0 block
4548 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
4550 * tests/check/gst/client.c:
4552 gst_rtsp_client_get_uri() has been removed
4553 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
4555 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4557 * gst/rtsp-server/rtsp-client.c:
4558 * gst/rtsp-server/rtsp-client.h:
4559 client: add method to filter managed sessions
4560 Add a method to filter the sessions managed by this client connection.
4561 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
4563 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4565 * gst/rtsp-server/rtsp-client.c:
4566 * gst/rtsp-server/rtsp-client.h:
4567 client: remove _get_uri() method
4568 Remove the get_uri() method on the client. A client has no uri, the uri
4569 property is an internal property to manage the last cached media for
4572 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4574 * gst/rtsp-server/rtsp-media-factory.h:
4575 media-factory: fix typo
4577 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
4579 * gst/rtsp-server/rtsp-media.c:
4580 rtsp-media: Do not leak the query in default_query_stop
4581 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
4583 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4585 * gst/rtsp-server/rtsp-media.c:
4586 media: don't unlock when conversion fails
4587 Don't unlock the state lock when conversion fails because it was not locked.
4589 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4591 * gst/rtsp-server/rtsp-media.c:
4592 * gst/rtsp-server/rtsp-media.h:
4593 Add query_position and query_stop vmethods to rtsp-media
4595 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4597 * gst/rtsp-server/rtsp-media.c:
4598 Fix typo in property install for rtsp-media's time-provider
4600 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4602 * gst/rtsp-server/rtsp-client.c:
4603 * gst/rtsp-server/rtsp-client.h:
4604 client: clean some variables
4605 Clean some variables and add some guards to _send_request()
4607 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4609 * gst/rtsp-server/rtsp-client.c:
4610 * gst/rtsp-server/rtsp-client.h:
4611 Add gst_rtsp_client_send_request API
4612 This makes it possible to send arbitrary messages to a client, such as
4613 SET_PARAMETER or GET_PARAMETER
4615 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4617 * gst/rtsp-server/rtsp-media.c:
4618 * gst/rtsp-server/rtsp-media.h:
4619 media: add _get_element() method
4620 Add method to get the element used when creating the media.
4621 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
4623 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4625 * gst/rtsp-server/rtsp-media.c:
4628 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4630 * gst/rtsp-server/rtsp-stream.c:
4631 * gst/rtsp-server/rtsp-stream.h:
4632 stream: allow access to the rtp session
4633 https://bugzilla.gnome.org/show_bug.cgi?id=703004
4635 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
4637 * gst/rtsp-server/rtsp-stream.c:
4638 * gst/rtsp-server/rtsp-stream.h:
4639 dscp qos support in gst-rtsp-stream
4640 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
4642 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4644 * tests/check/gst/rtspserver.c:
4646 Actually do what the comment says. Also keep the old code around, not sure what
4647 should happen when you get a 454 from a TEARDOWN, does it close the connection?
4648 it currently doesn't.
4650 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4652 * gst/rtsp-server/rtsp-client.c:
4653 client: also watch newly created session
4654 When we newly created a session, start watching it immediately instead of
4655 on the next request.
4657 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
4659 * tests/check/gst/client.c:
4660 tests: add unit test for new-session
4661 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
4663 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4665 * gst/rtsp-server/rtsp-client.c:
4666 client: emit new-session when new session is created
4667 Only emit new-session when we created a new session for a client, not when a
4668 client picked up a previous session.
4669 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
4671 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
4673 * gst/rtsp-server/rtsp-client.c:
4674 client: handle asterisk as path in requests
4675 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
4677 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4679 * gst/rtsp-server/rtsp-media.c:
4680 media: handle segment query format mismatch
4681 It's possible that the segment query returns with a different format than what
4682 we asked for, handle this case also.
4684 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
4686 * gst/rtsp-server/rtsp-media.c:
4687 media: use segment stop in collect_media_stats
4688 Use segment stop instead of duration as range end point.
4689 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
4691 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
4693 * gst/rtsp-server/rtsp-media.c:
4694 * tests/check/gst/media.c:
4695 rtsp-media: Do not leak the element in take_pipeline
4696 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
4698 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
4700 * gst/rtsp-server/rtsp-client.c:
4701 * gst/rtsp-server/rtsp-client.h:
4702 rtsp-client: Make configure_client_transport virtual
4703 This patch makes configure_client_transport virtual. The functionality is
4704 needed to handle some weird clients sending multicast transport settings as url
4706 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
4708 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
4710 * gst/rtsp-server/rtsp-client.c:
4711 * gst/rtsp-server/rtsp-client.h:
4712 rtsp-client: Make param_set and param_get virtual
4713 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
4715 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
4717 * gst/rtsp-server/rtsp-client.c:
4718 * gst/rtsp-server/rtsp-media.c:
4719 * gst/rtsp-server/rtsp-media.h:
4720 media: convert_range replaces get_range_times
4721 get_range_times worked for handling UTC ranges for seeks, but we also
4722 need to convert back from NPT to the requested unit in
4723 get_range_string. convert_range is now used for both.
4724 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
4726 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4728 * gst/rtsp-server/rtsp-client.c:
4729 * gst/rtsp-server/rtsp-sdp.c:
4730 * gst/rtsp-server/rtsp-sdp.h:
4731 sdp: cleanup sdp info
4732 We don't need to pass the proto, we can more easily check a boolean.
4733 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
4735 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
4737 * gst/rtsp-server/rtsp-sdp.c:
4738 use 0.0.0.0 or :: for c= line instead of server address
4740 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
4742 * gst/rtsp-server/rtsp-client.c:
4743 use local address, not remote, in SDP
4744 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
4746 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4749 Automatic update of common submodule
4750 From 098c0d7 to 01a7a46
4752 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
4754 * gst/rtsp-server/rtsp-media.c:
4755 * gst/rtsp-server/rtsp-media.h:
4756 media: possibility to override range time conversion
4757 Make it possible to override the conversion from GstRTSPTimeRange to
4758 GstClockTimes, that is done before seeking on the media
4759 pipeline. Overriding can be useful for UTC ranges, where the default
4760 conversion gives nanoseconds since 1900.
4761 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
4763 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
4765 * gst/rtsp-server/rtsp-server.c:
4766 * gst/rtsp-server/rtsp-server.h:
4767 rtsp-server: Expose the use_client_settings API
4768 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
4770 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
4772 * gst/rtsp-server/rtsp-client.c:
4773 * gst/rtsp-server/rtsp-stream.c:
4774 * gst/rtsp-server/rtsp-stream.h:
4775 rtspstream: handle both ipv4 and ipv6 clients
4776 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
4778 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4780 * gst/rtsp-server/rtsp-sdp.c:
4781 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
4782 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
4783 We already have a way to place extra attributes in the SDP by using a string
4784 property with prefix x- or a- in the caps.
4786 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4788 * gst/rtsp-server/rtsp-sdp.c:
4789 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
4790 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
4791 We already have a way to place extra attributes in the SDP, just make a string
4792 property in the payloader with a- or x- prefix.
4794 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4796 * gst/rtsp-server/rtsp-sdp.c:
4797 rtsp: place a- and x- properties as attributes
4798 application/x-rtp has properties with a- and x- prefixes that should be
4799 placed as attributes in the SDP for the media instead of being added to the
4802 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4804 * examples/Makefile.am:
4805 * examples/test-video.c:
4806 example: add TLS example
4808 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4810 * gst/rtsp-server/rtsp-server.c:
4811 * gst/rtsp-server/rtsp-server.h:
4812 server: add support for TLS
4813 Add methods to set and get a TLS certificate.
4814 Add vmethod to configure a new connection. By default, configure the TLS
4815 certificate in a new connection if needed.
4817 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4819 * gst/rtsp-server/rtsp-server.c:
4820 * gst/rtsp-server/rtsp-server.h:
4821 server: remove accept_client vmethod
4822 This vmethod is not very useful so remove it.
4824 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4826 * gst/rtsp-server/rtsp-server.c:
4827 server: don't crash on NULL GError
4829 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
4831 * gst/rtsp-server/rtsp-session-pool.c:
4832 rtsp-session-pool: corrected session timeout detection
4833 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
4835 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4837 * gst/rtsp-server/rtsp-client.c:
4838 client: improve debug
4840 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4842 * gst/rtsp-server/rtsp-client.c:
4843 * gst/rtsp-server/rtsp-client.h:
4844 * gst/rtsp-server/rtsp-server.c:
4845 server: refactor connection setup
4846 Let the server accept the socket connection and construct a GstRTSPConnection
4847 from it. Remove the code from the client and let the client only deal with
4848 a fully configure GstRTSPConnection object.
4849 We will need this later when the server will configure the connection for
4852 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4854 * gst/rtsp-server/rtsp-stream.c:
4855 stream: keep the transport object alive
4856 Keep the transport object alive while we have it as qdata on the
4859 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
4861 * gst/rtsp-server/rtsp-client.c:
4862 * gst/rtsp-server/rtsp-server.c:
4863 rtsp-server: Do not crash on nmapping of server
4864 * generate error when gst_rtsp_connection_accept fails
4865 * do not stop accepting incoming connections because
4866 accepting a client fails
4867 https://bugzilla.gnome.org/show_bug.cgi?id=701072
4869 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
4871 * gst/rtsp-server/rtsp-client.c:
4872 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
4873 https://bugzilla.gnome.org/show_bug.cgi?id=700953
4875 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
4877 * gst/rtsp-server/rtsp-sdp.c:
4878 rtsp-sdp: Parse framerate caps field and set SDP attribute
4879 The SDP attribute and its format is described in RFC4566.
4880 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
4882 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
4884 * gst/rtsp-server/rtsp-sdp.c:
4885 rtsp-sdp: Parse width/height from caps and set SDP attribute
4886 The SDP attribute and its format is described in RFC6064.
4887 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
4889 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
4891 * gst/rtsp-server/rtsp-sdp.c:
4892 * tests/check/gst/client.c:
4893 rtsp-sdp: add bandwidth line
4894 https://bugzilla.gnome.org/show_bug.cgi?id=699220
4896 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4899 Automatic update of common submodule
4900 From 5edcd85 to 098c0d7
4902 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
4904 * tests/check/gst/media.c:
4905 tests: add dynamic payloader prepare/unprepare check
4907 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4909 * gst/rtsp-server/rtsp-media.c:
4910 media: release lock when removing fakesink
4912 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4914 * gst/rtsp-server/rtsp-stream.c:
4915 stream: set elements to NULL before removing
4916 When removing a stream, set the elements to NULL first. This avoids
4917 element-is-not-in-NULL-state errors when we dispose the elements.
4919 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
4922 Automatic update of common submodule
4923 From 3cb3d3c to 5edcd85
4925 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4927 * gst/rtsp-server/rtsp-media.c:
4928 * gst/rtsp-server/rtsp-media.h:
4929 media: listen to pad-removed signals
4930 Listen to the pad-removed signal and remove the stream associated with the
4932 Add signal to be notified of the removed pad.
4933 Remove the fakesink in unprepare()
4934 Fix signatures of the signal methods
4936 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4938 * examples/test-sdp.c:
4939 tests: add example of reusable pipelines
4941 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
4943 * gst/rtsp-server/rtsp-stream.c:
4944 * gst/rtsp-server/rtsp-stream.h:
4945 stream: add method to get the srcpad
4947 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
4949 * tests/check/gst/media.c:
4950 check: add media prepare/unprepare test
4951 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4953 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
4955 * gst/rtsp-server/rtsp-media.c:
4956 media: disconnect from signal handlers in unprepare()
4957 We connected to the pad-added and no-more-pads signals in prepare() so
4958 we need to disconnect from them in unprepare().
4959 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4961 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
4963 * gst/rtsp-server/rtsp-media.c:
4964 media: don't free streams array
4965 Don't free the streams array in the unprepare() method, they were not
4967 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4969 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
4971 * gst/rtsp-server/rtsp-media.c:
4972 media: don't unref the pipeline in unprepare
4973 Unprepare() should undo what prepare() does. Because the pipeline is
4974 not created in prepare(), we should not unref it in unprepare()
4976 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
4978 * gst/rtsp-server/rtsp-stream.c:
4979 stream: clear session and caps for reuse
4980 Set the session and caps to NULL after unref otherwise we might unref
4982 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4984 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
4986 * gst/rtsp-server/rtsp-client.c:
4987 client: send out teardown signal before tearing down
4988 The advantage is that in the signal handler you get direct access to
4989 information about what streams are about to get torn down (in the
4990 GstRTSPClientState).
4991 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
4993 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
4995 * gst/rtsp-server/rtsp-client.c:
4996 * gst/rtsp-server/rtsp-client.h:
4997 client: expose connection
4998 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
5000 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
5003 Automatic update of common submodule
5004 From aed87ae to 3cb3d3c
5006 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5008 * gst/rtsp-server/rtsp-media.c:
5009 * gst/rtsp-server/rtsp-media.h:
5010 * gst/rtsp-server/rtsp-session-media.c:
5011 * gst/rtsp-server/rtsp-session-media.h:
5012 media: add method to get the base_time of the pipeline
5013 Together with a shared clock, this base-time could eventually be sent to
5014 the client so that it can reconstruct the exact running-time of the clock
5017 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5019 * gst/rtsp-server/Makefile.am:
5020 * gst/rtsp-server/rtsp-media.c:
5021 * gst/rtsp-server/rtsp-media.h:
5022 * gst/rtsp-server/rtsp-sdp.c:
5023 media: add GstNetTimeProvider support
5024 Add a property to let the media provide a GstNetTimeProvider for its clock.
5025 Make methods to get the clock and nettimeprovider
5026 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
5027 provider and also the current time of the clock. This should make it possible
5028 for (GStreamer) clients to slave their clock to the server clock.
5030 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
5033 Automatic update of common submodule
5034 From 04c7a1e to aed87ae
5036 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5038 * gst/rtsp-server/rtsp-media.c:
5039 media: wait for buffering to complete
5040 Wait for buffering to complete before changing the state to the target state.
5042 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5044 * gst/rtsp-server/rtsp-media.c:
5045 media: small cleanup
5047 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
5049 * tests/check/gst/rtspserver.c:
5050 tests: remove extra unref in test_setup_non_existing_stream
5051 The unref is not needed anymore, teardown runs without it.
5052 https://bugzilla.gnome.org/show_bug.cgi?id=696542
5054 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
5056 * tests/check/gst/rtspserver.c:
5057 tests: GSocketService cleanup in test_bind_already_in_use
5058 Use g_socket_service_stop so the rtspserver test stops listening for
5059 incoming connections in test_bind_already_in_use.
5060 https://bugzilla.gnome.org/show_bug.cgi?id=696541
5062 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
5064 * gst/rtsp-server/rtsp-media-factory.c:
5065 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
5066 Instead use a GWeakRef which is safe to use
5067 This is a known GLib bug, see:
5068 https://bugzilla.gnome.org/show_bug.cgi?id=667145
5070 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
5072 * gst/rtsp-server/rtsp-client.c:
5073 * gst/rtsp-server/rtsp-media.c:
5074 * gst/rtsp-server/rtsp-media.h:
5075 * gst/rtsp-server/rtsp-sdp.c:
5076 * tests/check/gst/media.c:
5077 * tests/check/gst/rtspserver.c:
5078 rtsp-media/client: Reply to PLAY request with same type of Range
5079 Remember the type of Range from the PLAY request and use the same type for
5082 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
5084 * gst/rtsp-server/rtsp-client.c:
5085 * gst/rtsp-server/rtsp-client.h:
5086 * tests/check/gst/client.c:
5087 rtsp-client: expose uri
5089 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
5091 * tests/check/gst/mediafactory.c:
5092 tests: Hold ref while creating second media
5093 To test if the media aren't shared, make sure we keep the first one while creating a second
5094 otherwise the same memory address may be reused.
5096 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
5099 configure: remove out-of-date comment
5101 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
5104 .gitignore: ignore more build files
5106 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
5108 * tests/check/Makefile.am:
5109 tests: use right _LIBS variable for gst-plugins-base libs
5111 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5113 * tests/check/Makefile.am:
5114 check: add librtp to libs
5116 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
5118 * tests/check/gst/rtspserver.c:
5119 tests: Add test to check selecting a port the server will send from
5121 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
5123 * tests/check/gst/rtspserver.c:
5124 tests: Make sure packets are actually received
5126 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
5128 * gst/rtsp-server/rtsp-stream.c:
5129 stream: Select unicast address from pool if appropriate
5131 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
5133 * gst/rtsp-server/rtsp-stream.c:
5134 stream: Properties are always there in Gst 1.0
5136 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
5138 * tests/check/gst/addresspool.c:
5139 tests: Add tests for unicast addresses in pool
5141 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
5143 * gst/rtsp-server/rtsp-address-pool.c:
5144 * tests/check/gst/addresspool.c:
5145 address-pool: Verify that multicast addresses are used for multicast and vice-versa
5147 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
5149 * docs/libs/gst-rtsp-server-sections.txt:
5150 * gst/rtsp-server/rtsp-address-pool.c:
5151 * gst/rtsp-server/rtsp-address-pool.h:
5152 * gst/rtsp-server/rtsp-stream.c:
5153 * tests/check/gst/addresspool.c:
5154 address-pool: Add unicast addresses
5156 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
5159 * gst/rtsp-server/rtsp-server.c:
5160 * tests/check/gst/rtspserver.c:
5161 rtsp-server: Limit the number of threads per server instance
5162 If we exceed the maximum, just round robin the clients over the existing
5165 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
5167 * gst/rtsp-server/rtsp-server.c:
5168 rtsp-server: No need to store the GMainContext in the client context
5170 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
5172 * tests/check/gst/rtspserver.c:
5173 tests: Add test for client disconnection
5175 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
5177 * tests/check/gst/rtspserver.c:
5178 tests: Test client and session timeouts with multiple threads
5180 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
5182 * gst/rtsp-server/rtsp-address-pool.c:
5183 * gst/rtsp-server/rtsp-auth.c:
5184 * gst/rtsp-server/rtsp-client.c:
5185 * gst/rtsp-server/rtsp-media-factory-uri.c:
5186 * gst/rtsp-server/rtsp-media-factory.c:
5187 * gst/rtsp-server/rtsp-media.c:
5188 * gst/rtsp-server/rtsp-mount-points.c:
5189 * gst/rtsp-server/rtsp-server.c:
5190 * gst/rtsp-server/rtsp-session-media.c:
5191 * gst/rtsp-server/rtsp-session-pool.c:
5192 * gst/rtsp-server/rtsp-session.c:
5193 Document locking and its order
5195 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
5197 * tests/check/gst/rtspserver.c:
5198 tests: Test that slow DESCRIBE don't block other clients
5200 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
5202 * tests/check/gst/client.c:
5203 tests: Add tests for client-requested multicast address
5205 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
5207 * docs/libs/gst-rtsp-server-sections.txt:
5208 docs: Put the various functions in the right sections
5210 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
5212 * docs/libs/gst-rtsp-server-docs.sgml:
5213 * docs/libs/gst-rtsp-server-sections.txt:
5214 * gst/rtsp-server/rtsp-address-pool.c:
5215 * gst/rtsp-server/rtsp-address-pool.h:
5216 docs: Generate docs for GstRTSPAddressPool
5218 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
5220 * gst/rtsp-server/rtsp-client.c:
5221 * gst/rtsp-server/rtsp-stream.c:
5222 * gst/rtsp-server/rtsp-stream.h:
5223 client: Check client provided addresses against the address pool
5225 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
5227 * gst/rtsp-server/rtsp-address-pool.c:
5228 * gst/rtsp-server/rtsp-address-pool.h:
5229 * tests/check/gst/addresspool.c:
5230 address-pool: Add API to request a specific address from the pool
5231 Also add relevant unit tests.
5233 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
5235 * tests/check/gst/mediafactory.c:
5236 tests: Check the passing around of a RTSPAddressPool
5237 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
5238 way down to the stream.
5240 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
5242 * tests/check/gst/addresspool.c:
5243 tests: Add more tests for the address pool
5245 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
5247 * gst/rtsp-server/rtsp-address-pool.c:
5248 address-pool: Fix off by one error
5249 When splitting a port range, the port after a skip is not part of range.
5251 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
5254 Automatic update of common submodule
5255 From 2de221c to 04c7a1e
5257 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
5260 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
5261 AM_CONFIG_HEADER was removed in automake 1.13
5262 https://bugzilla.gnome.org/show_bug.cgi?id=693368
5264 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
5267 Automatic update of common submodule
5268 From a942293 to 2de221c
5270 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5272 * gst/rtsp-server/rtsp-client.c:
5273 client: make sure the watch exists while sending data
5274 Protect the send_func with a lock. This allows us to wait for sending
5275 to complete before changing the send_func and user_data. We add an
5276 extra ref to the watch to make sure that it remains valid during
5278 When closing the connection, set the send_func to NULL
5279 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
5281 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5283 * tests/check/Makefile.am:
5284 tests: use GST_*_1_0 environment variables everywhere
5285 The _1_0 suffixed environment variables override the
5286 non-suffixed ones, so if we're in an environment that
5287 sets the _1_0 suffixed ones, such as jhbuild, we need
5288 to set those to make sure ours actually always get
5291 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5294 Automatic update of common submodule
5295 From acb04d9 to a942293
5297 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5299 * gst/rtsp-server/rtsp-client.c:
5300 rtsp-client: set the client backlog
5301 Set the client backlog to a reasonable default
5303 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
5305 * gst/rtsp-server/rtsp-media.c:
5306 rtsp-media: Make the element a constructor parameter
5307 https://bugzilla.gnome.org/show_bug.cgi?id=689594
5309 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5311 * docs/libs/Makefile.am:
5312 docs: Link with gcov library when gcov is enabled
5313 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
5315 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5317 * gst/rtsp-server/rtsp-media.c:
5318 media: match prepare with unprepare
5319 Really unprepare when there were an equal amount of prepare calls.
5321 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5323 * gst/rtsp-server/rtsp-media.c:
5324 media: media has to be unprepared in finalize
5325 Because unprepare takes away the last ref on the media.
5327 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5329 * gst/rtsp-server/rtsp-client.c:
5330 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
5331 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
5332 We can't use the refcount to trigger unprepare because it is the unprepare call
5333 that removes the last refcount after all messages are consumed. What we should
5334 probably do is make a prepared refcount and only unprepare when the refcount
5337 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5339 * gst/rtsp-server/rtsp-media.c:
5340 media: let the source unref the last media ref
5341 the last ref to the media is held by the source so we don't need to add more ref
5342 and unrefs, we simply destroy the media when the source is gone.
5344 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5346 * gst/rtsp-server/rtsp-media.c:
5347 media: improve debug
5349 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5351 * gst/rtsp-server/rtsp-media.c:
5353 Make sure we are in the right state when collecting the position and duration.
5354 Only make ourselves PREPARED when we were previously PREPARING.
5356 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5358 * gst/rtsp-server/rtsp-media.c:
5359 media: use g_object_ref/unref for GObjects
5361 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
5363 * gst/rtsp-server/rtsp-client.c:
5364 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
5365 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
5366 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
5367 isn't being used anymore.
5369 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
5371 * gst/rtsp-server/rtsp-media.c:
5372 Fix compiler warning
5374 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
5376 * gst/rtsp-server/rtsp-media-factory-uri.c:
5377 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
5379 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5381 * gst/rtsp-server/rtsp-session-media.h:
5384 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5386 * gst/rtsp-server/rtsp-media.c:
5387 * tests/check/gst/media.c:
5388 media: avoid element leak
5390 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5392 * gst/rtsp-server/rtsp-media.c:
5393 media: require an element in media constructor
5395 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5397 * gst/rtsp-server/rtsp-client.c:
5398 Revert "client: TEARDOWN brings that state to Init again"
5399 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
5400 The object is already disposed, there is no point in setting the state.
5402 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5404 * gst/rtsp-server/rtsp-client.c:
5405 client: TEARDOWN brings that state to Init again
5407 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5409 * docs/libs/gst-rtsp-server-sections.txt:
5410 * examples/test-auth.c:
5411 * gst/rtsp-server/rtsp-auth.c:
5412 * gst/rtsp-server/rtsp-auth.h:
5413 * gst/rtsp-server/rtsp-client.c:
5414 * gst/rtsp-server/rtsp-client.h:
5415 * gst/rtsp-server/rtsp-media-factory-uri.c:
5416 * gst/rtsp-server/rtsp-media-factory-uri.h:
5417 * gst/rtsp-server/rtsp-media-factory.c:
5418 * gst/rtsp-server/rtsp-media-factory.h:
5419 * gst/rtsp-server/rtsp-media.c:
5420 * gst/rtsp-server/rtsp-media.h:
5421 * gst/rtsp-server/rtsp-mount-points.c:
5422 * gst/rtsp-server/rtsp-mount-points.h:
5423 * gst/rtsp-server/rtsp-sdp.c:
5424 * gst/rtsp-server/rtsp-server.c:
5425 * gst/rtsp-server/rtsp-server.h:
5426 * gst/rtsp-server/rtsp-session-media.c:
5427 * gst/rtsp-server/rtsp-session-media.h:
5428 * gst/rtsp-server/rtsp-session-pool.c:
5429 * gst/rtsp-server/rtsp-session-pool.h:
5430 * gst/rtsp-server/rtsp-session.c:
5431 * gst/rtsp-server/rtsp-session.h:
5432 * gst/rtsp-server/rtsp-stream-transport.c:
5433 * gst/rtsp-server/rtsp-stream-transport.h:
5434 * gst/rtsp-server/rtsp-stream.c:
5435 * gst/rtsp-server/rtsp-stream.h:
5436 * tests/check/gst/media.c:
5437 rtsp: make object details private
5438 Make all object details private
5439 Add methods to access private bits
5441 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5443 * tests/check/Makefile.am:
5444 * tests/check/gst/media.c:
5445 tests: add media tests
5447 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5449 * gst/rtsp-server/rtsp-media.c:
5450 media: check if prepared for some methods
5451 Check that the media object is prepared before doing seek and getting the
5452 current position etc.
5453 Add some g_return checks.
5455 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5457 * tests/check/Makefile.am:
5458 * tests/check/gst/mediafactory.c:
5459 tests: add mediafactory test
5461 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5463 * gst/rtsp-server/rtsp-stream.c:
5464 stream: improve debug
5466 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5468 * gst/rtsp-server/rtsp-media.c:
5469 * gst/rtsp-server/rtsp-media.h:
5470 media: unref pipeline in finalize to avoid leaking it
5472 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5474 * gst/rtsp-server/rtsp-media-factory-uri.c:
5475 * gst/rtsp-server/rtsp-media.c:
5476 rtsp: use gst_object_unref on GstObjects
5478 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5480 * gst/rtsp-server/rtsp-media-factory.c:
5481 media-factory: require an url
5483 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5485 * examples/test-uri.c:
5486 examples: fix include
5488 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5490 * gst/rtsp-server/rtsp-server.h:
5491 server: remove unused include
5493 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5495 * tests/check/Makefile.am:
5496 * tests/check/gst/mountpoints.c:
5497 tests: add test for mountpoints
5499 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5501 * gst/rtsp-server/rtsp-client.c:
5502 client: fix factory leak
5503 Keep the factory in the state object only for authorization checks and make
5504 sure we unref it on failure. Also don't keep invalid objects in the state
5507 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5509 * gst/rtsp-server/rtsp-mount-points.c:
5510 mounts: add g_return_if guards
5512 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5514 * tests/check/gst/client.c:
5515 tests: add more tests
5517 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5519 * gst/rtsp-server/rtsp-client.c:
5520 client: improve debug
5522 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5524 * gst/rtsp-server/rtsp-client.c:
5525 client: improve debug and fix leaks
5526 Cleanup the uri and session when there is a bad request.
5528 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5533 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5535 * tests/check/gst/client.c:
5536 test: add test for session in options request
5538 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5540 * gst/rtsp-server/rtsp-client.c:
5541 client: use 454 when session can't be found
5542 We should use 454 when a session can't be found because there was no session
5543 pool configured in the server. This is not a server configuration problem
5544 because the server on which the request is done might not be the same one that
5545 will keep the sessions for us and so it does not need to support sessions.
5547 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5549 * gst/rtsp-server/rtsp-client.c:
5550 client: only free connection when there is one
5551 It's possible that the client doesn't have a connection when we try to free it.
5553 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5555 * tests/check/Makefile.am:
5556 * tests/check/gst/client.c:
5557 tests: add unit test for the client object
5559 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5561 * gst/rtsp-server/rtsp-client.c:
5562 client: small cleanup
5564 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5566 * gst/rtsp-server/rtsp-client.h:
5567 client: remove unused include
5569 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5571 * gst/rtsp-server/rtsp-client.c:
5572 client: fix compilation
5574 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5576 * gst/rtsp-server/rtsp-client.c:
5577 client: call destroy without the lock
5579 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5581 * gst/rtsp-server/rtsp-client.c:
5582 * gst/rtsp-server/rtsp-client.h:
5583 client: make the client usable without a socket
5584 Make a method to let the client handle a message and a callback when the client
5585 wants us to send a response message back. This makes it possible to also use the
5586 client object without the sockets, which should make it easier to test.
5588 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5590 * gst/rtsp-server/rtsp-client.c:
5591 * gst/rtsp-server/rtsp-client.h:
5592 client: small cleanup
5594 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5596 * docs/libs/gst-rtsp-server-sections.txt:
5597 * gst/rtsp-server/rtsp-client.c:
5598 * gst/rtsp-server/rtsp-client.h:
5599 * gst/rtsp-server/rtsp-server.c:
5600 client: remove reference to server
5601 We don't need to keep a ref to the server
5603 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5605 * gst/rtsp-server/rtsp-client.c:
5606 * gst/rtsp-server/rtsp-client.h:
5608 Also add some g_return_if()
5610 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5612 * gst/rtsp-server/rtsp-client.c:
5613 client: log more errors
5615 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5617 * gst/rtsp-server/rtsp-client.c:
5618 client: fix compilation
5620 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5622 * gst/rtsp-server/rtsp-client.c:
5623 * gst/rtsp-server/rtsp-client.h:
5624 client: add generic close-after-send support
5625 Add a property to send_response() to close the connection after the response has
5626 been sent to the client.
5628 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5631 * docs/libs/gst-rtsp-server-docs.sgml:
5632 * docs/libs/gst-rtsp-server-sections.txt:
5633 * docs/libs/gst-rtsp-server.types:
5634 * examples/test-auth.c:
5635 * examples/test-launch.c:
5636 * examples/test-mp4.c:
5637 * examples/test-multicast.c:
5638 * examples/test-multicast2.c:
5639 * examples/test-ogg.c:
5640 * examples/test-readme.c:
5641 * examples/test-sdp.c:
5642 * examples/test-uri.c:
5643 * examples/test-video.c:
5644 * gst/rtsp-server/Makefile.am:
5645 * gst/rtsp-server/rtsp-auth.h:
5646 * gst/rtsp-server/rtsp-client.c:
5647 * gst/rtsp-server/rtsp-client.h:
5648 * gst/rtsp-server/rtsp-media-mapping.c:
5649 * gst/rtsp-server/rtsp-media-mapping.h:
5650 * gst/rtsp-server/rtsp-mount-points.c:
5651 * gst/rtsp-server/rtsp-mount-points.h:
5652 * gst/rtsp-server/rtsp-server.c:
5653 * gst/rtsp-server/rtsp-server.h:
5654 * gst/rtsp-server/rtsp-session-media.c:
5655 * gst/rtsp-server/rtsp-session-pool.c:
5656 * gst/rtsp-server/rtsp-session-pool.h:
5657 * tests/check/gst/rtspserver.c:
5658 MediaMapping -> MountPoints
5659 Describes better what the object manages.
5661 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5664 configure: bump required version of -base
5666 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5668 * gst/rtsp-server/rtsp-media.c:
5671 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5673 * gst/rtsp-server/rtsp-media.c:
5674 * gst/rtsp-server/rtsp-media.h:
5675 media: support more Range formats
5676 Use the new -base methods to convert the Range string into a seek start and stop
5679 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5681 * examples/test-launch.c:
5682 examples: fix whitespace
5684 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5686 * examples/test-auth.c:
5687 test-auth: add example of how to remove sessions
5688 Add an example of the session filter api.
5690 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5692 * examples/test-uri.c:
5693 test-uri: remove mapping example
5695 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5697 * examples/test-uri.c:
5698 test-uri: fix callback signature
5700 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5702 * gst/rtsp-server/rtsp-media-factory.c:
5703 factory: keep ref to factory while media active
5704 While the media from a factory is alive, keep a ref to the factory.
5705 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
5707 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5709 * gst/rtsp-server/rtsp-media-factory-uri.c:
5710 factory-uri: add some debug
5712 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5714 * gst/rtsp-server/rtsp-stream.c:
5715 stream: set udp sources to PLAYING
5716 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
5717 so that it doesn't cause our pipeline to produce ASYNC-DONE.
5719 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5721 * gst/rtsp-server/rtsp-media-factory-uri.c:
5722 factory-uri: take ref to factory
5723 Take a ref to the factory that we place in our list.
5725 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5727 * tests/Makefile.am:
5728 * tests/test-reuse.c:
5729 test: add test for server reuse
5730 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
5732 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
5734 * gst/rtsp-server/rtsp-server.c:
5735 server: start and stop multiple times
5736 Stop listening on the RTSP port when the GSource is removed, so clients
5737 can't connect and the server can be started again.
5738 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
5740 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5742 * gst/rtsp-server/rtsp-server.c:
5743 server: fix small leak
5745 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5747 * gst/rtsp-server/rtsp-media.c:
5748 media: unref source in finish_unprepare
5749 The source is created in prepare, unref it in finish_unprepare.
5750 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
5752 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
5754 * gst/rtsp-server/rtsp-client.c:
5755 * gst/rtsp-server/rtsp-media.c:
5756 rtsp-media: remove bus watch before finalizing
5757 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
5758 * An extra media ref is added for the bus watch. This extra ref is unreffed by
5759 the GDestroyNotify function.
5760 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
5761 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
5762 gst_rtsp_media_unprepare before unreffing the media.
5763 This way, the bus watch will be removed before the media is finalized.
5764 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
5766 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
5768 * gst/rtsp-server/rtsp-client.c:
5769 * gst/rtsp-server/rtsp-client.h:
5770 client: wait until the TEARDOWN response is sent to close the connection
5771 Responses can be sent async so we need to wait until the TEARDOWN response has
5772 been written before we close the connection to the client. This avoids the risk
5773 of writing/polling closed sockets.
5774 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
5776 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
5778 * gst/rtsp-server/rtsp-stream.c:
5779 rtsp-stream: plug socket leak
5780 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
5782 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
5785 Automatic update of common submodule
5786 From 6bb6951 to a72faea
5788 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
5790 * gst/rtsp-server/rtsp-media-factory-uri.c:
5791 rtsp-server: don't use deprecated API
5793 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
5795 * gst/rtsp-server/rtsp-client.c:
5796 rtsp-client: fix unused-but-set-variable compiler warning
5797 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
5799 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5802 * docs/libs/gst-rtsp-server-sections.txt:
5803 * gst/rtsp-server/rtsp-client.c:
5806 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5808 * examples/Makefile.am:
5809 * examples/test-multicast2.c:
5810 examples: add another multicast example
5811 Add an example for how to configure separate multicast ranges for each media
5814 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5816 * examples/test-multicast.c:
5819 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5821 * gst/rtsp-server/rtsp-client.c:
5822 * gst/rtsp-server/rtsp-media.c:
5823 * gst/rtsp-server/rtsp-session-media.c:
5824 * gst/rtsp-server/rtsp-session-media.h:
5825 * gst/rtsp-server/rtsp-stream-transport.c:
5826 * gst/rtsp-server/rtsp-stream-transport.h:
5827 stream: use the address managed by the stream
5828 Use the address managed by the stream for multicast. This allows us to have 1
5829 multicast address for each stream.
5830 Because the address is now managed by the stream we don't have to pass it around
5832 Set the address pool on the streams.
5834 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5836 * gst/rtsp-server/rtsp-client.c:
5837 * gst/rtsp-server/rtsp-media.c:
5838 * gst/rtsp-server/rtsp-stream.c:
5841 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5843 * gst/rtsp-server/rtsp-media.c:
5844 * gst/rtsp-server/rtsp-media.h:
5845 media: add signal for new streams
5846 This allows applications to listen for new streams and configure properties on
5847 them, like the address pool.
5849 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5851 * gst/rtsp-server/rtsp-media.c:
5852 media: configure address pool in new streams
5854 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5856 * gst/rtsp-server/rtsp-stream.c:
5857 * gst/rtsp-server/rtsp-stream.h:
5858 stream: add methods to deal with address pool
5859 Add methods to get and set the address pool for the stream
5860 Add method to allocate and get the multicast addresses for this stream.
5862 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5864 * docs/libs/gst-rtsp-server-sections.txt:
5865 * gst/rtsp-server/rtsp-media.c:
5866 * gst/rtsp-server/rtsp-media.h:
5867 media: remove MTU property
5868 It is a stream property
5870 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5872 * gst/rtsp-server/rtsp-client.c:
5873 client: set blocksize only on stream
5874 Set the blocksize only on the current stream.
5876 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5878 * gst/rtsp-server/rtsp-stream.c:
5879 stream: share src and sink sockets
5880 the allocated socket is in the used-socket property, not socket.
5882 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5884 * gst/rtsp-server/rtsp-address-pool.c:
5885 * gst/rtsp-server/rtsp-address-pool.h:
5886 * gst/rtsp-server/rtsp-client.c:
5887 * gst/rtsp-server/rtsp-session-media.c:
5888 * gst/rtsp-server/rtsp-session-media.h:
5889 * gst/rtsp-server/rtsp-stream-transport.c:
5890 * gst/rtsp-server/rtsp-stream-transport.h:
5891 * tests/check/gst/addresspool.c:
5892 rtsp: make address-pool return an address object
5893 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
5894 store more info in the structure and allows us to more easily return the address
5895 to the right pool when no longer needed.
5896 Pass the address to the StreamTransport so that we can return it to the pool
5897 when the stream transport is freed or changed.
5899 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5901 * examples/Makefile.am:
5902 * examples/test-multicast.c:
5903 examples: add multicast example
5904 Show how to set up the multicast address pool so that media can be
5905 server with multicast.
5907 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5909 * gst/rtsp-server/rtsp-client.c:
5910 * gst/rtsp-server/rtsp-media-factory.c:
5911 * gst/rtsp-server/rtsp-media-factory.h:
5912 * gst/rtsp-server/rtsp-media.c:
5913 * gst/rtsp-server/rtsp-media.h:
5914 rtsp: use AddressPool
5915 Remove the multicast_group property.
5916 Use the configured addresspool to allocate multicast addresses.
5918 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5920 * gst/rtsp-server/rtsp-address-pool.c:
5921 * gst/rtsp-server/rtsp-address-pool.h:
5922 address-pool: add clear method
5924 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5926 * gst/rtsp-server/rtsp-address-pool.c:
5927 address-pool: small cleanups
5929 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5931 * tests/check/Makefile.am:
5932 * tests/check/gst/addresspool.c:
5933 tests: add addresspool unit test
5935 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5937 * gst/rtsp-server/Makefile.am:
5938 * gst/rtsp-server/rtsp-address-pool.c:
5939 * gst/rtsp-server/rtsp-address-pool.h:
5940 address-pool: add object to manage multicast addresses
5941 Make an object that can manage a rage of multicast addresses and ports.
5943 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5945 * gst/rtsp-server/rtsp-server.c:
5946 server: set default max-threads property
5948 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5950 * gst/rtsp-server/rtsp-media.c:
5951 media: wait for concurrent _prepare
5952 If a prepare is busy, wait for the result.
5954 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5956 * gst/rtsp-server/rtsp-media.c:
5957 media: add lock around message handler
5958 We don't want to dispatch messages while we are still processing the result of
5961 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5963 * gst/rtsp-server/rtsp-media.c:
5964 * gst/rtsp-server/rtsp-media.h:
5965 media: add lock to protect state changes
5967 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5969 * gst/rtsp-server/rtsp-stream.c:
5970 * gst/rtsp-server/rtsp-stream.h:
5973 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5975 * gst/rtsp-server/rtsp-stream-transport.c:
5976 * gst/rtsp-server/rtsp-stream-transport.h:
5977 * gst/rtsp-server/rtsp-stream.c:
5978 stream-transport: add keep-alive method
5980 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5982 * gst/rtsp-server/rtsp-stream-transport.c:
5983 * gst/rtsp-server/rtsp-stream-transport.h:
5984 * gst/rtsp-server/rtsp-stream.c:
5985 stream-transport: add method to handle RTP/RTCP
5986 Call new methods instead of poking into the structures directly.
5988 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5990 * gst/rtsp-server/rtsp-session-media.c:
5991 * gst/rtsp-server/rtsp-session-media.h:
5992 session-media: add locking
5994 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5996 * gst/rtsp-server/rtsp-session.c:
5997 * gst/rtsp-server/rtsp-session.h:
5998 session: add locking
6000 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6002 * gst/rtsp-server/rtsp-server.c:
6003 server: free old socket
6005 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6007 * gst/rtsp-server/rtsp-media-mapping.c:
6008 * gst/rtsp-server/rtsp-media-mapping.h:
6009 mapping: add locking
6011 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6013 * gst/rtsp-server/rtsp-media-factory.c:
6014 media-factory: add locking
6016 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6018 * gst/rtsp-server/rtsp-auth.c:
6019 * gst/rtsp-server/rtsp-auth.h:
6022 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6024 * gst/rtsp-server/rtsp-server.c:
6025 * gst/rtsp-server/rtsp-server.h:
6026 server: add max-thread property
6028 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6030 * gst/rtsp-server/rtsp-server.c:
6031 * gst/rtsp-server/rtsp-server.h:
6032 server: use a threadpool for the mainloops
6034 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6036 * gst/rtsp-server/rtsp-client.c:
6037 * gst/rtsp-server/rtsp-client.h:
6038 client: rename method
6039 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
6040 don't really create the client from the socket, we use the socket for the
6043 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6045 * gst/rtsp-server/rtsp-client.c:
6046 * gst/rtsp-server/rtsp-client.h:
6047 * gst/rtsp-server/rtsp-server.c:
6048 server: rework maincontext handling in clients
6049 Make a separate method to attach a client to a MainContext.
6050 Let the server decide in what GMainContext the client will operate and give this
6051 context to the client in attach. Then the server can later decide to use a
6052 separate thread for each client or just use the mainthread.
6054 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6056 * gst/rtsp-server/rtsp-client.c:
6057 * gst/rtsp-server/rtsp-session.c:
6058 * gst/rtsp-server/rtsp-session.h:
6059 session: move session header code in session object
6061 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
6065 * examples/test-auth.c:
6066 * examples/test-launch.c:
6067 * examples/test-mp4.c:
6068 * examples/test-ogg.c:
6069 * examples/test-readme.c:
6070 * examples/test-sdp.c:
6071 * examples/test-uri.c:
6072 * examples/test-video.c:
6073 * gst/rtsp-server/rtsp-auth.c:
6074 * gst/rtsp-server/rtsp-auth.h:
6075 * gst/rtsp-server/rtsp-client.c:
6076 * gst/rtsp-server/rtsp-client.h:
6077 * gst/rtsp-server/rtsp-media-factory-uri.c:
6078 * gst/rtsp-server/rtsp-media-factory-uri.h:
6079 * gst/rtsp-server/rtsp-media-factory.c:
6080 * gst/rtsp-server/rtsp-media-factory.h:
6081 * gst/rtsp-server/rtsp-media-mapping.c:
6082 * gst/rtsp-server/rtsp-media-mapping.h:
6083 * gst/rtsp-server/rtsp-media.c:
6084 * gst/rtsp-server/rtsp-media.h:
6085 * gst/rtsp-server/rtsp-params.c:
6086 * gst/rtsp-server/rtsp-params.h:
6087 * gst/rtsp-server/rtsp-sdp.c:
6088 * gst/rtsp-server/rtsp-sdp.h:
6089 * gst/rtsp-server/rtsp-server.c:
6090 * gst/rtsp-server/rtsp-server.h:
6091 * gst/rtsp-server/rtsp-session-media.c:
6092 * gst/rtsp-server/rtsp-session-media.h:
6093 * gst/rtsp-server/rtsp-session-pool.c:
6094 * gst/rtsp-server/rtsp-session-pool.h:
6095 * gst/rtsp-server/rtsp-session.c:
6096 * gst/rtsp-server/rtsp-session.h:
6097 * gst/rtsp-server/rtsp-stream-transport.c:
6098 * gst/rtsp-server/rtsp-stream-transport.h:
6099 * gst/rtsp-server/rtsp-stream.c:
6100 * gst/rtsp-server/rtsp-stream.h:
6101 * tests/check/gst/rtspserver.c:
6102 * tests/test-cleanup.c:
6105 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6107 * gst/rtsp-server/rtsp-media.c:
6108 * gst/rtsp-server/rtsp-session-media.c:
6109 * gst/rtsp-server/rtsp-session.c:
6110 rtsp-server: added annotations to indicate type of ownership transfer of return values
6111 https://bugzilla.gnome.org/show_bug.cgi?id=680777
6113 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
6116 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
6118 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
6121 * bindings/Makefile.am:
6122 * bindings/vala/Makefile.am:
6123 * bindings/vala/gst-rtsp-server-0.10.deps:
6124 * bindings/vala/gst-rtsp-server-0.10.vapi:
6125 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
6126 * bindings/vala/packages/gst-rtsp-server-0.10.files:
6127 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6128 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6129 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
6131 bindings: remove vala bindings
6132 They'll be reunited with the other GStreamer bindings
6133 https://bugzilla.gnome.org/show_bug.cgi?id=680777
6135 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6137 * gst/rtsp-server/rtsp-client.c:
6138 * gst/rtsp-server/rtsp-session-media.c:
6139 * gst/rtsp-server/rtsp-session-media.h:
6140 * gst/rtsp-server/rtsp-stream-transport.c:
6141 * gst/rtsp-server/rtsp-stream-transport.h:
6142 rtsp: only create transport when needed
6143 Only create the StreamTransport when configured.
6145 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6147 * gst/rtsp-server/rtsp-client.c:
6148 client: small cleanup
6150 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6152 * gst/rtsp-server/rtsp-client.c:
6153 * gst/rtsp-server/rtsp-client.h:
6154 * gst/rtsp-server/rtsp-stream-transport.c:
6155 * gst/rtsp-server/rtsp-stream-transport.h:
6156 rtsp: refactor configuration of transport
6157 Move the configuration of the transport to a place where it makes
6160 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6162 * gst/rtsp-server/rtsp-client.c:
6163 client: refactor transport parsing
6165 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6167 * gst/rtsp-server/rtsp-client.c:
6168 client: refuse to change the MTU on shared media
6169 If we change the MTU of chared media, it changes for all clients.
6170 We don't want to set the MTU to something large for clients that
6173 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6175 * examples/test-mp4.c:
6176 * gst/rtsp-server/rtsp-media.c:
6177 small fixes to docs and debug
6179 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6181 * gst/rtsp-server/rtsp-stream.c:
6182 stream: transports must already have been removed
6184 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6186 * gst/rtsp-server/rtsp-media.c:
6187 * gst/rtsp-server/rtsp-stream.c:
6188 * gst/rtsp-server/rtsp-stream.h:
6189 stream: improve join and leave of the pipeline
6191 Do the cleanup properly
6194 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6196 * gst/rtsp-server/rtsp-media.c:
6197 media: move unprepare below default implementation
6198 Makes it easier to find the default implementation
6200 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6202 * gst/rtsp-server/rtsp-media.c:
6203 media: signal unprepared when we actually finish
6205 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6207 * gst/rtsp-server/rtsp-media.c:
6208 media: no need to unlock, unprepare does that when needed
6210 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6212 * docs/libs/gst-rtsp-server-sections.txt:
6213 * gst/rtsp-server/rtsp-media-factory.h:
6214 * gst/rtsp-server/rtsp-media-mapping.c:
6215 * gst/rtsp-server/rtsp-media.h:
6216 * gst/rtsp-server/rtsp-params.c:
6217 * gst/rtsp-server/rtsp-server.c:
6218 * gst/rtsp-server/rtsp-session-pool.h:
6219 * gst/rtsp-server/rtsp-session.c:
6220 * gst/rtsp-server/rtsp-session.h:
6221 * gst/rtsp-server/rtsp-stream-transport.h:
6222 * gst/rtsp-server/rtsp-stream.h:
6225 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6227 * gst/rtsp-server/rtsp-client.c:
6228 * gst/rtsp-server/rtsp-media-mapping.h:
6229 * gst/rtsp-server/rtsp-media.c:
6230 * gst/rtsp-server/rtsp-media.h:
6231 * gst/rtsp-server/rtsp-server.h:
6232 * gst/rtsp-server/rtsp-stream.c:
6233 * gst/rtsp-server/rtsp-stream.h:
6234 rtsp: fix MTU setting
6235 Fix setting of the MTU. There is no need for a vmethod.
6237 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6242 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6245 configure: bump version number after refactoring
6247 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6249 * gst/rtsp-server/Makefile.am:
6250 * gst/rtsp-server/rtsp-client.c:
6251 * gst/rtsp-server/rtsp-client.h:
6252 * gst/rtsp-server/rtsp-media-factory-uri.c:
6253 * gst/rtsp-server/rtsp-media-factory.c:
6254 * gst/rtsp-server/rtsp-media-factory.h:
6255 * gst/rtsp-server/rtsp-media.c:
6256 * gst/rtsp-server/rtsp-media.h:
6257 * gst/rtsp-server/rtsp-sdp.c:
6258 * gst/rtsp-server/rtsp-session-media.c:
6259 * gst/rtsp-server/rtsp-session-media.h:
6260 * gst/rtsp-server/rtsp-session.c:
6261 * gst/rtsp-server/rtsp-session.h:
6262 * gst/rtsp-server/rtsp-stream-transport.c:
6263 * gst/rtsp-server/rtsp-stream-transport.h:
6264 * gst/rtsp-server/rtsp-stream.c:
6265 * gst/rtsp-server/rtsp-stream.h:
6266 rtsp: massive refactoring
6267 Make GObjects from the remaining simple structures.
6268 Remove GstRTSPSessionStream, it's not needed.
6269 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
6270 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
6271 a GstRTSPStream should be transported to a client.
6272 Rename GstRTSPMediaFactory::get_element -> create_element because that
6273 more accurately describes what it does.
6274 Make nice methods instead of poking in the structures.
6275 Move some methods inside the relevant object source code.
6276 Use GPtrArray to store objects instead of plain arrays, it is more
6277 natural and allows us to more easily clean up.
6278 Move the allocation of udp ports to the Stream object. The Stream object
6279 contains the elements needed to stream the media to a client.
6280 Improve the prepare and unprepare methods. Unprepare should now undo
6281 everything prepare did. Improve also async unprepare when doing EOS on
6282 shutdown. Make sure we always unprepare correctly.
6284 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
6286 * gst/rtsp-server/rtsp-client.c:
6287 rtsp-client: Unref server address clients connected to
6288 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
6290 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
6292 * gst/rtsp-server/rtsp-server.c:
6293 rtsp-server: don't ref server socket if it is NULL
6294 Fixes test_bind_already_in_use unit test again after commit 6a497440.
6295 https://bugzilla.gnome.org/show_bug.cgi?id=686644
6297 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
6299 * tests/check/Makefile.am:
6300 tests: Add libgio link dependency
6301 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
6303 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6305 * gst/rtsp-server/rtsp-media-mapping.c:
6306 * gst/rtsp-server/rtsp-media-mapping.h:
6307 rtsp-media-mapping: rename find_media vfunc to find_factory
6308 The virtual method and class method should have the same name
6309 so it is correctly represented in GIR file
6310 https://bugzilla.gnome.org/show_bug.cgi?id=680777
6312 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6314 * gst/rtsp-server/rtsp-auth.c:
6315 * gst/rtsp-server/rtsp-client.c:
6316 * gst/rtsp-server/rtsp-media-factory-uri.c:
6317 * gst/rtsp-server/rtsp-media-factory.c:
6318 * gst/rtsp-server/rtsp-media-mapping.c:
6319 * gst/rtsp-server/rtsp-media.c:
6320 * gst/rtsp-server/rtsp-server.c:
6321 * gst/rtsp-server/rtsp-session-pool.c:
6322 * gst/rtsp-server/rtsp-session.c:
6323 rtsp-server: fixed comments and GIR annotations
6324 https://bugzilla.gnome.org/show_bug.cgi?id=680777
6326 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
6328 * gst/rtsp-server/rtsp-media-mapping.c:
6329 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
6331 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
6333 * gst/rtsp-server/rtsp-server.c:
6334 rtsp-server: allow binding on port 0 (binds on a random port)
6336 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
6338 * gst/rtsp-server/rtsp-server.c:
6339 * gst/rtsp-server/rtsp-server.h:
6340 rtsp-server: add bound-port property
6341 bound-port can be used to retrieve the port number when the server is bound on
6342 port 0, which binds on a random port.
6344 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
6346 * gst/rtsp-server/rtsp-media-factory.c:
6347 * gst/rtsp-server/rtsp-media-factory.h:
6348 rtsp-media-factory: make ::get_element overridable by GI bindings
6349 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
6350 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
6351 as the invoker for ::get_element(), making it overridable by GI generated
6354 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
6356 * gst/rtsp-server/rtsp-media-factory-uri.c:
6357 rtsp-media-factory-uri: don't autoplug parsers in a loop
6358 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
6361 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
6363 * gst/rtsp-server/Makefile.am:
6364 Explicitly link against gio. Fix link error on mac.
6366 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
6368 * gst/rtsp-server/rtsp-session.c:
6369 session: add ttl to the transport header in SETUP
6370 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
6372 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
6374 * gst/rtsp-server/rtsp-client.c:
6375 * gst/rtsp-server/rtsp-client.h:
6376 * gst/rtsp-server/rtsp-media.c:
6377 client: Use client transport settings for multicast if allowed.
6378 This patch makes it possible for the client to send transport settings for
6379 multicast (destination && ttl). Client settings must be explicitly allowed or
6380 the server will use its own settings.
6381 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
6383 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
6386 Automatic update of common submodule
6387 From 6c0b52c to 6bb6951
6389 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
6391 * gst/rtsp-server/rtsp-client.c:
6392 rtsp-client: do not destroy the rtsp watch
6393 Don't destroy the client watch while dispatching. The rtsp watch is
6394 automatically destroyed after the rtsp watch function closed() has
6396 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
6398 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
6401 Automatic update of common submodule
6402 From 4f962f7 to 6c0b52c
6404 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
6406 * gst/rtsp-server/rtsp-media.c:
6407 media: fix check for seekability
6409 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6411 * gst/rtsp-server/rtsp-client.c:
6412 client: use more GIO
6413 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
6415 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6417 * gst/rtsp-server/rtsp-server.c:
6418 server: remove obsolete includes
6420 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
6422 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
6423 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
6424 be available in "on_new_ssrc". The transports are added in
6425 gst_rtsp_media_set_state when going to PLAYING state. However,
6426 "on_new_ssrc" might be called before this happens.
6427 https://bugzilla.gnome.org/show_bug.cgi?id=683304
6429 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
6431 * gst/rtsp-server/rtsp-client.c:
6432 * gst/rtsp-server/rtsp-client.h:
6433 rtsp-client: add signals for rtsp requests (fixes #683287)
6435 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
6437 * gst/rtsp-server/rtsp-client.c:
6438 * gst/rtsp-server/rtsp-client.h:
6439 add new-session signal to rtsp-client (fixes #683058)
6441 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
6444 Automatic update of common submodule
6445 From 668acee to 4f962f7
6447 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
6449 * gst/rtsp-server/rtsp-server.c:
6450 * tests/check/gst/rtspserver.c:
6451 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
6452 Do not assume that *error is set in g_socket_address_enumerator_next.
6453 Added test_bind_already_in_use unit-test.
6454 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
6456 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
6459 Automatic update of common submodule
6460 From 94ccf4c to 668acee
6462 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
6464 * gst/rtsp-server/rtsp-client.c:
6465 * gst/rtsp-server/rtsp-client.h:
6466 rtsp-client: make create_sdp virtual method
6467 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
6469 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6472 Automatic update of common submodule
6473 From 98e386f to 94ccf4c
6475 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6477 * gst/rtsp-server/rtsp-client.c:
6480 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
6482 * gst/rtsp-server/rtsp-client.c:
6483 * gst/rtsp-server/rtsp-client.h:
6484 * gst/rtsp-server/rtsp-server.c:
6485 * gst/rtsp-server/rtsp-server.h:
6486 rtsp-server: use an existing socket to establish HTTP tunnel
6487 Make it possible to transfer a socket from an HTTP server to be used as
6488 an RTSP over HTTP tunnel.
6490 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
6492 * gst/rtsp-server/rtsp-client.c:
6493 * gst/rtsp-server/rtsp-media.c:
6494 * gst/rtsp-server/rtsp-media.h:
6495 rtsp: Handle the blocksize parameter
6496 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
6498 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
6500 * tests/check/Makefile.am:
6501 * tests/check/gst/rtspserver.c:
6502 Have unit test get header from source dir, not installed dir
6503 This makes compilation of unit tests work in a build directory other
6504 than the source directory.
6505 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
6507 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
6509 * gst/rtsp-server/rtsp-media.c:
6510 rtsp-media: update for gst_element_make_from_uri() changes
6512 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
6515 * tests/Makefile.am:
6516 * tests/check/Makefile.am:
6517 * tests/check/gst/rtspserver.c:
6519 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
6521 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
6523 * gst/rtsp-server/rtsp-media.c:
6524 rtsp-media: don't collect media stats when going to NULL
6525 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
6527 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6529 * gst/rtsp-server/rtsp-client.c:
6530 client: don't leak transports
6532 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
6534 * gst/rtsp-server/rtsp-client.c:
6535 rtsp-client: free transport on no_stream in SETUP handler
6537 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
6539 * gst/rtsp-server/rtsp-client.c:
6540 rtsp-client: changed session media iteration
6541 In client_unlink_session: now don't iterate in session->medias
6542 list where items are removed by gst_rtsp_session_release_media.
6543 Instead, repeatedly remove the first item.
6545 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
6547 * gst/rtsp-server/rtsp-client.c:
6548 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
6549 GstRTSPSessionMedia is not a GObject type. When the
6550 GstRTSPSession is freed, it will free the media.
6552 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
6554 * gst/rtsp-server/rtsp-media-factory.c:
6555 factory: plug pad leak in collect_streams
6556 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
6557 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
6558 will take one reference, and the other reference will otherwise
6561 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
6564 configure: suppress some warnings when debug is disabled
6565 Warnings about unused variables should be suppressed if core has the
6566 debug system disabled.
6567 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
6569 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6571 * docs/libs/Makefile.am:
6572 docs: fix build in uninstalled setup
6573 Include gst-plugins-base libs properly.
6575 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
6577 * docs/libs/gst-rtsp-server.types:
6578 docs: include headers defining rtsp-server object types
6579 Fixes compiler warnings during docs build.
6580 https://bugzilla.gnome.org/show_bug.cgi?id=676824
6582 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
6585 configure: Add warning flags for compiler when configuring
6586 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
6588 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6591 Automatic update of common submodule
6592 From 03a0e57 to 98e386f
6594 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6597 Automatic update of common submodule
6598 From 1fab359 to 03a0e57
6600 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
6602 * gst/rtsp-server/rtsp-client.c:
6603 client: fix GSocketAddress leak in gst_rtsp_client_accept
6604 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
6606 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6609 Automatic update of common submodule
6610 From f1b5a96 to 1fab359
6612 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6615 Automatic update of common submodule
6616 From 92b7266 to f1b5a96
6618 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6621 Automatic update of common submodule
6622 From ec1c4a8 to 92b7266
6624 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6627 Automatic update of common submodule
6628 From 3429ba6 to ec1c4a8
6630 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
6632 * gst/rtsp-server/rtsp-auth.c:
6633 * gst/rtsp-server/rtsp-client.c:
6634 * gst/rtsp-server/rtsp-media-factory-uri.c:
6635 * gst/rtsp-server/rtsp-server.c:
6636 rtsp: fix compiler warnings
6637 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
6639 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6642 Automatic update of common submodule
6643 From dc70203 to 3429ba6
6645 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6647 * gst/rtsp-server/rtsp-client.c:
6648 * gst/rtsp-server/rtsp-media-factory.c:
6649 * gst/rtsp-server/rtsp-media-factory.h:
6650 * gst/rtsp-server/rtsp-media.c:
6651 * gst/rtsp-server/rtsp-media.h:
6652 * gst/rtsp-server/rtsp-server.c:
6653 * gst/rtsp-server/rtsp-server.h:
6654 * gst/rtsp-server/rtsp-session-pool.c:
6655 * gst/rtsp-server/rtsp-session-pool.h:
6656 rtsp-server: port to new thread API
6658 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6661 Automatic update of common submodule
6662 From 6db25be to dc70203
6664 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6666 * gst/rtsp-server/rtsp-auth.c:
6667 * gst/rtsp-server/rtsp-auth.h:
6668 * gst/rtsp-server/rtsp-client.c:
6669 rtsp-server: Fix compilation and compiler warnings
6671 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6675 * gst/rtsp-server/Makefile.am:
6676 configure: Modernize autotools setup a bit
6677 Also we now only create tar.bz2 and tar.xz tarballs.
6679 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6682 Automatic update of common submodule
6683 From 464fe15 to 6db25be
6685 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6688 Automatic update of common submodule
6689 From 7fda524 to 464fe15
6691 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6694 * docs/libs/Makefile.am:
6695 * docs/version.entities.in:
6697 * gst/rtsp-server/Makefile.am:
6698 * pkgconfig/Makefile.am:
6699 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
6700 * pkgconfig/gstreamer-rtsp-server.pc.in:
6701 * tests/Makefile.am:
6702 rtsp-server: Update versioning
6704 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6706 Merge remote-tracking branch 'origin/0.10'
6708 gst/rtsp-server/rtsp-session-pool.c
6710 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6712 * gst/rtsp-server/rtsp-session-pool.c:
6713 rtsp-server: Don't use deprecated GLib API
6715 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6717 Replace master with 0.11
6719 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6721 Merge branch 'master' into 0.11
6723 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6725 Merge branch 'master' into 0.11
6727 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
6730 A couple minor typo fixes
6732 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6734 * gst/rtsp-server/rtsp-media.c:
6735 media: fix state of the appqueue
6737 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6739 * gst/rtsp-server/rtsp-media-factory-uri.c:
6740 factory: use videoconvert
6742 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6744 * gst/rtsp-server/rtsp-media-factory-uri.c:
6745 factory: change to new style caps
6747 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6749 * gst/rtsp-server/rtsp-client.c:
6750 * gst/rtsp-server/rtsp-client.h:
6751 * gst/rtsp-server/rtsp-media-factory-uri.c:
6752 * gst/rtsp-server/rtsp-media.c:
6753 * gst/rtsp-server/rtsp-server.c:
6754 * gst/rtsp-server/rtsp-server.h:
6755 * gst/rtsp-server/rtsp-session-pool.c:
6756 rtsp-server: port to GIO
6759 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6762 configure: fix build
6764 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6767 docs: fix for gst_rtsp_server_set_port() -> _set_service()
6768 https://bugzilla.gnome.org/show_bug.cgi?id=666548
6770 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6773 * examples/Makefile.am:
6774 First rule of gst-rtsp-server club: don't talk about gst-phonon
6776 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6779 * pkgconfig/Makefile.am:
6780 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
6781 * pkgconfig/gst-rtsp-server.pc.in:
6782 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
6783 * pkgconfig/gstreamer-rtsp-server.pc.in:
6784 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
6785 For consistency with all other modules.
6787 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6789 * gst/rtsp-server/rtsp-client.c:
6790 rtsp-client: update for new map API
6792 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6795 * bindings/Makefile.am:
6796 * bindings/python/Makefile.am:
6797 * bindings/python/arg-types.py:
6798 * bindings/python/codegen/Makefile.am:
6799 * bindings/python/codegen/__init__.py:
6800 * bindings/python/codegen/argtypes.py:
6801 * bindings/python/codegen/code-coverage.py:
6802 * bindings/python/codegen/codegen.py:
6803 * bindings/python/codegen/definitions.py:
6804 * bindings/python/codegen/defsparser.py:
6805 * bindings/python/codegen/docextract.py:
6806 * bindings/python/codegen/docgen.py:
6807 * bindings/python/codegen/fileprefix.override:
6808 * bindings/python/codegen/fileprefixmodule.c:
6809 * bindings/python/codegen/h2def.py:
6810 * bindings/python/codegen/mergedefs.py:
6811 * bindings/python/codegen/mkskel.py:
6812 * bindings/python/codegen/override.py:
6813 * bindings/python/codegen/reversewrapper.py:
6814 * bindings/python/codegen/scmexpr.py:
6815 * bindings/python/rtspserver-types.defs:
6816 * bindings/python/rtspserver.defs:
6817 * bindings/python/rtspserver.override:
6818 * bindings/python/rtspservermodule.c:
6819 * bindings/python/test.py:
6821 python: remove pygst-based python bindings
6822 pygi is the future, apparently.
6824 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
6827 Automatic update of common submodule
6828 From c463bc0 to 7fda524
6830 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6833 Automatic update of common submodule
6834 From 2a59016 to c463bc0
6836 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6839 Automatic update of common submodule
6840 From 0807187 to 2a59016
6842 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6845 Automatic update of common submodule
6846 From 11f0cd5 to 0807187
6848 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6850 * examples/test-auth.c:
6851 example: update for new caps
6853 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6855 * examples/test-video.c:
6856 * gst/rtsp-server/rtsp-client.c:
6857 * gst/rtsp-server/rtsp-media-factory-uri.c:
6858 * gst/rtsp-server/rtsp-media.c:
6859 * gst/rtsp-server/rtsp-media.h:
6860 * gst/rtsp-server/rtsp-session.c:
6861 * gst/rtsp-server/rtsp-session.h:
6862 rtsp-server: port some more to 0.11
6864 Remove bufferlist stuff
6866 Add queue before appsink now that preroll-queue-len is gone.
6867 Update for request pad changes.
6869 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6871 Merge branch 'master' into 0.11
6873 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
6875 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6876 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
6877 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6879 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
6881 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6882 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
6883 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6885 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6887 Merge branch 'master' into 0.11
6889 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6891 * gst/rtsp-server/rtsp-media.c:
6892 * gst/rtsp-server/rtsp-media.h:
6893 media: add a seekable boolean
6894 Maintain the seekable state with a new variable instead of reusing the
6897 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
6899 * gst/rtsp-server/rtsp-media.c:
6900 Disallow seek in live media
6902 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6904 Merge branch 'master' into 0.11
6906 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
6908 * gst/rtsp-server/rtsp-server.c:
6909 #ifdef statements for windows socket creation were missing
6911 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
6914 Automatic update of common submodule
6915 From a39eb83 to 11f0cd5
6917 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
6920 Automatic update of common submodule
6921 From 605cd9a to a39eb83
6923 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6925 Merge branch 'master' into 0.11
6927 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6929 * gst/rtsp-server/rtsp-client.c:
6930 client: use method to access property
6932 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6934 * gst/rtsp-server/rtsp-media-factory.c:
6935 * gst/rtsp-server/rtsp-media-factory.h:
6936 media-factory: add protocols property
6937 Add a property to configure the allowed protocols in the media created from the
6940 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6942 * gst/rtsp-server/rtsp-media-factory.c:
6943 * gst/rtsp-server/rtsp-media-factory.h:
6944 media-factory: add media-configure signal
6945 Add signal to allow the application to configure the media after it was created
6948 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6950 * gst/rtsp-server/rtsp-client.c:
6951 client: use method to access property
6953 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6955 * gst/rtsp-server/rtsp-media-factory.c:
6956 * gst/rtsp-server/rtsp-media-factory.h:
6957 media-factory: add protocols property
6958 Add a property to configure the allowed protocols in the media created from the
6961 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6963 * gst/rtsp-server/rtsp-media-factory.c:
6964 * gst/rtsp-server/rtsp-media-factory.h:
6965 media-factory: add media-configure signal
6966 Add signal to allow the application to configure the media after it was created
6969 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6971 Merge branch 'master' into 0.11
6973 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6975 * gst/rtsp-server/rtsp-client.c:
6976 client: use media multicast group
6978 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6980 * gst/rtsp-server/rtsp-media-factory.h:
6981 * gst/rtsp-server/rtsp-server.h:
6982 * gst/rtsp-server/rtsp-session-pool.h:
6983 * gst/rtsp-server/rtsp-session.h:
6986 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6988 * gst/rtsp-server/rtsp-client.c:
6989 * gst/rtsp-server/rtsp-sdp.h:
6990 sdp: copy and free the server ip address
6991 Copy and free the server ip address to make memory management easier later.
6993 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6995 * gst/rtsp-server/rtsp-media-factory.c:
6996 media-factory: configure multicast in media
6998 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7000 * gst/rtsp-server/rtsp-media.c:
7001 * gst/rtsp-server/rtsp-media.h:
7002 media: add property for multicast group
7003 Add a property to configure the multicast group in the media.
7004 Based on patches from Marc Leeman and Robert Krakora.
7006 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7008 * gst/rtsp-server/rtsp-media-factory.c:
7009 * gst/rtsp-server/rtsp-media-factory.h:
7010 media-factory: add property for multicast group
7011 Add a property to configure the multicast group in the media factory.
7012 Based on patches from Marc Leeman and Robert Krakora.
7014 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7016 * gst/rtsp-server/rtsp-client.c:
7017 client: do configuration of transport in one place
7018 Move the configuration of the transport destination address to where we also
7019 configure the other bits.
7021 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7023 * gst/rtsp-server/rtsp-client.c:
7024 client: use media multicast group
7026 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7028 * gst/rtsp-server/rtsp-media-factory.h:
7029 * gst/rtsp-server/rtsp-server.h:
7030 * gst/rtsp-server/rtsp-session-pool.h:
7031 * gst/rtsp-server/rtsp-session.h:
7034 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
7036 * gst/rtsp-server/rtsp-client.c:
7037 * gst/rtsp-server/rtsp-sdp.h:
7038 sdp: copy and free the server ip address
7039 Copy and free the server ip address to make memory management easier later.
7041 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7043 * gst/rtsp-server/rtsp-media-factory.c:
7044 media-factory: configure multicast in media
7046 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7048 * gst/rtsp-server/rtsp-media.c:
7049 * gst/rtsp-server/rtsp-media.h:
7050 media: add property for multicast group
7051 Add a property to configure the multicast group in the media.
7052 Based on patches from Marc Leeman and Robert Krakora.
7054 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7056 * gst/rtsp-server/rtsp-media-factory.c:
7057 * gst/rtsp-server/rtsp-media-factory.h:
7058 media-factory: add property for multicast group
7059 Add a property to configure the multicast group in the media factory.
7060 Based on patches from Marc Leeman and Robert Krakora.
7062 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7064 * gst/rtsp-server/rtsp-client.c:
7065 client: do configuration of transport in one place
7066 Move the configuration of the transport destination address to where we also
7067 configure the other bits.
7069 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7071 Merge branch 'master' into 0.11
7073 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
7075 * gst/rtsp-server/rtsp-client.c:
7076 client: destroy pipeline on client disconnect with no prior TEARDOWN.
7077 The problem occurs when the client abruptly closes the connection without
7078 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
7079 server is where the pipeline gets torn down. Since this handler is not called,
7080 the pipeline remains and is up and running. Subsequent clients get their own
7081 pipelines and if the do not issue TEARDOWNs then those pipelines will also
7082 remain up and running. This is a resource leak.
7084 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7086 Merge branch 'master' into 0.11
7088 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
7090 * gst/rtsp-server/rtsp-media-factory.c:
7091 * gst/rtsp-server/rtsp-media-factory.h:
7092 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
7093 For example, it can be used to retrieve source elements like appsrc, in a more
7094 convenient way than subclassing get_element.
7096 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7098 Merge branch 'master' into 0.11
7100 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
7102 * gst/rtsp-server/rtsp-server.c:
7103 rtsp-server: hold on to reference while using object
7105 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7107 * gst/rtsp-server/rtsp-media.c:
7110 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7113 configure: use unstable api
7115 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
7117 * gst/rtsp-server/rtsp-client.c:
7118 client: fix reference counting
7120 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
7122 * gst/rtsp-server/rtsp-client.c:
7123 * gst/rtsp-server/rtsp-media.c:
7124 fix compiler warnings about unused variables
7126 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
7128 * examples/test-launch.c:
7129 * examples/test-readme.c:
7130 * examples/test-uri.c:
7131 * examples/test-video.c:
7132 examples: tell rtsp uri when ready
7134 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
7137 Automatic update of common submodule
7138 From 69b981f to 605cd9a
7140 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7142 * gst/rtsp-server/rtsp-client.c:
7143 client: update for buffer API change
7145 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7147 * gst/rtsp-server/Makefile.am:
7148 Makefile.am: 0.10 => @GST_MAJORMINOR@
7150 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7152 * gst/rtsp-server/rtsp-media-factory-uri.c:
7153 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
7155 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7157 * gst/rtsp-server/.gitignore:
7158 .gitignore: 0.10 => 0.11
7160 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7162 * gst/rtsp-server/Makefile.am:
7163 Makefile.am: 0.10 => @GST_MAJORMINOR@
7165 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7167 Merge branch 'master' into 0.11
7169 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
7172 Automatic update of common submodule
7173 From 9e5bbd5 to 69b981f
7175 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
7178 Automatic update of common submodule
7179 From fd35073 to 9e5bbd5
7181 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
7184 Automatic update of common submodule
7185 From 46dfcea to fd35073
7187 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7189 * gst/rtsp-server/rtsp-media-factory-uri.c:
7190 * gst/rtsp-server/rtsp-media.c:
7191 media: port to new caps API
7193 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7195 Merge branch 'master' into 0.11
7197 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
7199 * bindings/vala/gst-rtsp-server-0.10.vapi:
7200 Updated Vala bindings.
7201 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
7203 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
7205 * gst/rtsp-server/rtsp-server.c:
7206 * gst/rtsp-server/rtsp-server.h:
7207 Add a signal for newly connected clients.
7208 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
7210 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
7212 * bindings/python/rtspserver.override:
7213 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
7215 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7217 * gst/rtsp-server/Makefile.am:
7218 * gst/rtsp-server/rtsp-client.c:
7219 * gst/rtsp-server/rtsp-funnel.c:
7220 * gst/rtsp-server/rtsp-funnel.h:
7221 * gst/rtsp-server/rtsp-media.c:
7222 rtsp-server: port to 0.11
7224 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7229 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7231 Merge branch 'master' into 0.11
7236 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7239 Automatic update of common submodule
7240 From c3cafe1 to 46dfcea
7242 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
7244 * bindings/python/Makefile.am:
7245 * bindings/python/rtspserver.defs:
7246 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
7248 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
7250 * bindings/python/arg-types.py:
7251 python bindings: add GstRTSPUrlParam
7252 Needed to implement MediaFactory virtual proxies
7254 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
7256 * bindings/python/arg-types.py:
7257 python bindings: fix returning GstRTSPUrl types
7259 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
7261 * bindings/python/arg-types.py:
7262 python bindings: add arg type for GstRTSPUrl
7264 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
7266 * bindings/python/rtspserver.defs:
7267 python bindings: fix the definition of MediaFactory.collect_stream
7269 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
7272 Automatic update of common submodule
7273 From 1ccbe09 to c3cafe1
7275 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7278 Automatic update of common submodule
7279 From 193b717 to 1ccbe09
7281 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
7284 Automatic update of common submodule
7285 From b77e2bf to 193b717
7287 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7290 build: Include lcov.mak to allow test coverage report generation
7292 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7295 Automatic update of common submodule
7296 From d8814b6 to b77e2bf
7298 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7301 Automatic update of common submodule
7302 From 6aaa286 to d8814b6
7304 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
7307 Automatic update of common submodule
7308 From 6aec6b9 to 6aaa286
7310 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
7313 autogen: wingo signed comment
7315 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
7317 * gst/rtsp-server/rtsp-session-pool.c:
7318 session: use full charset for RTSP session ID
7319 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
7320 session ID more difficult.
7321 https://bugzilla.gnome.org/show_bug.cgi?id=643812
7323 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7325 * gst/rtsp-server/Makefile.am:
7326 rtsp-server: Don't install the funnel header
7328 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
7331 Automatic update of common submodule
7332 From 1de7f6a to 6aec6b9
7334 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7337 configure: require core/base 0.10.31
7338 Needed at least for gst_plugin_feature_rank_compare_func().
7340 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
7343 Automatic update of common submodule
7344 From f94d739 to 1de7f6a
7346 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7348 * gst/rtsp-server/rtsp-media.c:
7349 media: remove more unused code
7351 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7353 * gst/rtsp-server/rtsp-media.c:
7354 * gst/rtsp-server/rtsp-media.h:
7355 media: remove duplicate filtering
7356 Remove the duplicate filtering code now that we have a released -good version.
7357 Give a warning instead.
7359 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7361 * gst/rtsp-server/rtsp-media-factory.c:
7362 * gst/rtsp-server/rtsp-media.c:
7363 media: fix default buffer size
7365 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7367 * gst/rtsp-server/rtsp-media-factory.c:
7368 * gst/rtsp-server/rtsp-media-factory.h:
7369 media-factory: add property to configure the buffer-size
7370 Add a property to configure the kernel UDP buffer size.
7372 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7374 * gst/rtsp-server/rtsp-media.c:
7375 * gst/rtsp-server/rtsp-media.h:
7376 media: add property to configure kernel buffer sizes
7377 Add a property to configure the kernel UDP buffer size.
7379 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7382 configure: set PYGOBJECT_REQ before using it
7383 https://bugzilla.gnome.org/show_bug.cgi?id=640641
7385 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7388 docs: recursive into sub-directories on 'make upload'
7390 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7392 * docs/libs/gst-rtsp-server-docs.sgml:
7393 * docs/version.entities.in:
7394 docs: mention full version these docs are for, not just major-minor
7396 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7401 === release 0.10.8 ===
7403 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7408 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7410 * gst/rtsp-server/rtsp-server.c:
7411 rtsp-server: clarify docs a little
7413 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7415 * gst/rtsp-server/rtsp-media.c:
7416 media: init debug category before starting thread
7418 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7420 * gst/rtsp-server/rtsp-auth.c:
7421 auth: add realm to make it more spec compliant
7423 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7425 * gst/rtsp-server/rtsp-server.c:
7426 * gst/rtsp-server/rtsp-server.h:
7429 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7431 * examples/test-video.c:
7432 example: improve example docs a little
7434 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7436 * gst/rtsp-server/rtsp-server.c:
7437 server: ensure the watch has a ref to the server
7439 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7441 * gst/rtsp-server/rtsp-server.c:
7442 server: simpify channel function
7444 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7446 * gst/rtsp-server/rtsp-server.c:
7447 * gst/rtsp-server/rtsp-server.h:
7448 server: simplify management of channel and source
7449 We don't need to keep around the channel and source objects. Let the mainloop
7450 and the source manage the source and channel respectively.
7452 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7458 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7461 * tests/Makefile.am:
7462 * tests/test-cleanup.c:
7463 tests: add tests directory and cleanup test
7465 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7467 * gst/rtsp-server/rtsp-media-factory-uri.c:
7468 * gst/rtsp-server/rtsp-media-factory.c:
7469 * gst/rtsp-server/rtsp-media-mapping.c:
7470 * gst/rtsp-server/rtsp-media.c:
7471 * gst/rtsp-server/rtsp-session-pool.c:
7472 * gst/rtsp-server/rtsp-session.c:
7473 server: improve debugging in various objects
7475 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7477 * gst/rtsp-server/rtsp-server.c:
7478 server: chain up to the parent finalize
7480 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
7482 * bindings/python/rtspserver-types.defs:
7483 * bindings/python/rtspserver.defs:
7484 * bindings/python/rtspserver.override:
7485 * bindings/python/test.py:
7486 gst-rtsp-server: update python bindings
7488 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7490 * gst/rtsp-server/rtsp-client.c:
7491 client: use the response from the clientstate
7492 Create the response object only once and store in the client state.
7493 Make all methods use the state response,
7495 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7497 * gst/rtsp-server/rtsp-server.c:
7498 server: use signal to keep track of clients
7499 Keep track of all the clients that the server creates and remove them when they
7500 fire the 'closed' signal.
7502 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7504 * gst/rtsp-server/rtsp-client.c:
7505 * gst/rtsp-server/rtsp-client.h:
7506 client: emit signal when closing
7508 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7510 * examples/.gitignore:
7511 * examples/Makefile.am:
7512 * examples/test-auth.c:
7513 * examples/test-video.c:
7514 * gst/rtsp-server/rtsp-auth.c:
7515 * gst/rtsp-server/rtsp-auth.h:
7516 * gst/rtsp-server/rtsp-client.c:
7517 * gst/rtsp-server/rtsp-media-factory.c:
7518 * gst/rtsp-server/rtsp-media.c:
7519 * gst/rtsp-server/rtsp-media.h:
7520 * gst/rtsp-server/rtsp-session-pool.h:
7521 * gst/rtsp-server/rtsp-session.h:
7522 media: enable per factory authorisations
7523 Allow for adding a GstRTSPAuth on the factory and media level and check
7524 permissions when accessing the factory.
7525 Add hints to the auth methods for future more fine grained authorisation.
7526 Add example application for per factory authentication.
7528 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7530 * gst/rtsp-server/rtsp-auth.c:
7531 * gst/rtsp-server/rtsp-auth.h:
7532 * gst/rtsp-server/rtsp-client.c:
7533 * gst/rtsp-server/rtsp-client.h:
7534 * gst/rtsp-server/rtsp-params.c:
7535 * gst/rtsp-server/rtsp-params.h:
7536 rtsp-server: Pass ClientState structure arround
7537 Pass the collected information for the ongoing request in a GstRTSPClientState
7538 structure that we can then pass around to simplify the method arguments. This
7539 will also be handy when we implement logging functionality.
7541 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7543 * gst/rtsp-server/rtsp-media-factory.c:
7544 * gst/rtsp-server/rtsp-media-factory.h:
7545 media-factory: add methods to configure authorisation
7547 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7549 * gst/rtsp-server/rtsp-client.c:
7550 client: unref auth in finalize
7552 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7554 * gst/rtsp-server/rtsp-server.c:
7555 server: unref auth in finalize
7557 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7559 * docs/libs/gst-rtsp-server-docs.sgml:
7560 * docs/libs/gst-rtsp-server-sections.txt:
7561 * docs/libs/gst-rtsp-server.types:
7564 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7566 * gst/rtsp-server/rtsp-server.c:
7567 * gst/rtsp-server/rtsp-server.h:
7568 server: separate create and accept
7569 Create separate create and accept methods so that subclasses can create custom
7571 Configure the server in the client object and prepare for keeping track of
7574 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7576 * gst/rtsp-server/rtsp-client.c:
7577 * gst/rtsp-server/rtsp-client.h:
7578 client: add support for setting the server.
7579 Add support for keeping a ref to the server that started this client
7582 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7584 * gst/rtsp-server/rtsp-auth.c:
7585 auth: fix memleak and add some docs
7586 Fix a memleak of the basic auth token.
7587 Add docs for the helper function
7589 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7591 * gst/rtsp-server/rtsp-auth.c:
7592 * gst/rtsp-server/rtsp-auth.h:
7593 * gst/rtsp-server/rtsp-client.c:
7594 client: delegate setup of auth to the manager
7595 Delegate the configuration of the authentication tokens to the manager object
7598 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7600 * examples/test-video.c:
7601 * gst/rtsp-server/Makefile.am:
7602 * gst/rtsp-server/rtsp-auth.c:
7603 * gst/rtsp-server/rtsp-auth.h:
7604 * gst/rtsp-server/rtsp-client.c:
7605 * gst/rtsp-server/rtsp-client.h:
7606 * gst/rtsp-server/rtsp-server.c:
7607 * gst/rtsp-server/rtsp-server.h:
7608 auth: add authentication object
7609 Add an object that can check the authorization of requests.
7610 Implement basic authentication.
7611 Add example authentication to test-video
7613 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7615 * gst/rtsp-server/rtsp-server.c:
7616 * gst/rtsp-server/rtsp-server.h:
7617 server: move includes back
7618 the includes are needed for sockaddr_in.
7620 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7622 * gst/rtsp-server/rtsp-client.c:
7623 * gst/rtsp-server/rtsp-client.h:
7624 * gst/rtsp-server/rtsp-server.c:
7625 * gst/rtsp-server/rtsp-server.h:
7626 rtsp: move network includes where they are needed
7628 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
7630 * gst/rtsp-server/rtsp-media.h:
7631 rtsp-media.h: Minor corrections in comments.
7634 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
7637 Automatic update of common submodule
7638 From e572c87 to f94d739
7640 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7644 * docs/libs/.gitignore:
7645 * examples/.gitignore:
7646 * gst/rtsp-server/.gitignore:
7649 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7651 * docs/libs/Makefile.am:
7652 docs: We don't build ps/pdf for API reference docs
7654 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7657 Automatic update of common submodule
7658 From ccbaa85 to e572c87
7660 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7663 Automatic update of common submodule
7664 From 46445ad to ccbaa85
7666 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7668 * gst/rtsp-server/Makefile.am:
7669 * gst/rtsp-server/fs-funnel.c:
7670 * gst/rtsp-server/fs-funnel.h:
7671 * gst/rtsp-server/rtsp-funnel.c:
7672 * gst/rtsp-server/rtsp-funnel.h:
7673 * gst/rtsp-server/rtsp-media.c:
7674 funnel: rename fsfunnel to rtspfunnel
7675 Rename the funnel to avoid conflicts with the farsight one.
7677 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7679 * gst/rtsp-server/Makefile.am:
7680 * gst/rtsp-server/fs-funnel.c:
7681 * gst/rtsp-server/fs-funnel.h:
7682 * gst/rtsp-server/rtsp-media.c:
7683 rtsp-media: add and use fsfunnel
7684 Add a copy of fsfunnel to the build because input-selector removed the (broken)
7685 select-all property that we need.
7687 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7689 * gst/rtsp-server/Makefile.am:
7690 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
7691 Use PKG_CONFIG_PATH specified at configure time (if any) as well
7692 for the g-ir-compiler, rather than just assuming the env var has
7695 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7702 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
7704 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7707 * gst/rtsp-server/Makefile.am:
7708 gobject-introspection: fix g-i build for uninstalled setup
7709 Requires gst-plugins-base git (> 0.10.31.2).
7711 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7713 * examples/test-uri.c:
7714 examples: add some more options and comments
7716 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7718 * gst/rtsp-server/rtsp-media-factory-uri.c:
7719 factory-uri: use right property type
7721 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7723 * gst/rtsp-server/rtsp-media-factory-uri.c:
7724 factory-uri: attempt to configure buffer-lists
7725 Attempt to configure buffer lists in the payloader for improved performance.
7727 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7729 * gst/rtsp-server/rtsp-media.c:
7730 media: attempt to configure bigger UDP buffers
7731 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
7732 send buffers with high bitrate streams.
7734 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
7736 * gst/rtsp-server/rtsp-client.c:
7737 client: use the socket length from getsockname
7738 Use the length returned by getsockname to perform the getnameinfo call because
7739 the size can depend on the socket type and platform.
7742 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7744 * docs/libs/gst-rtsp-server-docs.sgml:
7745 * docs/libs/gst-rtsp-server-sections.txt:
7746 docs: add uri factory to the docs
7748 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7750 * gst/rtsp-server/rtsp-client.c:
7751 * gst/rtsp-server/rtsp-media.h:
7754 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7756 * gst/rtsp-server/rtsp-client.c:
7757 * gst/rtsp-server/rtsp-media.c:
7758 * gst/rtsp-server/rtsp-media.h:
7759 * gst/rtsp-server/rtsp-session.c:
7760 * gst/rtsp-server/rtsp-session.h:
7761 rtsp-server: add support for buffer lists
7762 Add support for sending bufferlists received from appsink.
7765 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7767 * gst/rtsp-server/rtsp-client.c:
7768 * gst/rtsp-server/rtsp-media.c:
7769 * gst/rtsp-server/rtsp-media.h:
7770 * gst/rtsp-server/rtsp-sdp.c:
7771 media: make method to retrieve the play range
7772 Make a method to retrieve the playback range so that we can conditionally create
7773 a different range for the SDP and the PLAY requests.
7775 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7777 * gst/rtsp-server/rtsp-media.c:
7778 * gst/rtsp-server/rtsp-media.h:
7779 media: add signal to notify of state changes
7781 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7783 * gst/rtsp-server/rtsp-client.h:
7784 client: cleanup headers
7786 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7788 * gst/rtsp-server/rtsp-client.c:
7791 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7793 * gst/rtsp-server/rtsp-media-factory-uri.c:
7794 * gst/rtsp-server/rtsp-media-factory-uri.h:
7795 factory-uri: add support for gstpay
7796 Add an option to prefer gstpay over decoder + raw payloader.
7798 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7800 * gst/rtsp-server/rtsp-media-factory-uri.c:
7801 * gst/rtsp-server/rtsp-media-factory-uri.h:
7802 factory-uri: rework the autoplugger.
7803 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
7806 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7808 * gst/rtsp-server/rtsp-media-factory-uri.c:
7809 factory-uri: use better factory filter
7810 Make better payloader filter based on autoplug rank and RTP use case.
7812 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7815 Automatic update of common submodule
7816 From 169462a to 46445ad
7818 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7820 * gst/rtsp-server/rtsp-server.c:
7821 server: set SO_REUSEADDR before bind
7822 Set the SO_REUSEADDR _before_ bind() to make it actually work.
7824 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7826 * gst/rtsp-server/rtsp-media.c:
7827 * gst/rtsp-server/rtsp-media.h:
7828 media: emit prepared signal when prepared
7829 Make a 'prepared' signal and emit it when we successfully prepared the element.
7830 This signal can be used to configure the media object after it has been prepared
7833 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
7836 Automatic update of common submodule
7837 From 011bcc8 to 169462a
7839 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
7841 python an optional dependency
7842 * configure.ac: Move up valgrind and g-i checks. Make the python
7843 dependency optional, as it was before.
7845 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7847 Merge branch 'master' into 0.11
7852 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7854 * gst/rtsp-server/rtsp-media.c:
7855 media: update range when active clients changed
7856 When we changed the number of active clients, update the current range
7857 information because we want the second client connecting to a shared resource
7858 continue from where the stream currently.
7860 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7862 * gst/rtsp-server/rtsp-media-factory-uri.c:
7863 * gst/rtsp-server/rtsp-media-factory-uri.h:
7864 factory-uri: add colorspace and fix pt
7865 Rework the way we pass data to the autoplugger.
7866 When we have raw caps, plug a converter element to make pluggin to raw
7867 payloaders more successful.
7868 Make sure all dynamically plugged payloaders have a unique payload types.
7870 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7872 * examples/Makefile.am:
7873 * examples/test-uri.c:
7874 example: add example of the uri factory
7876 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7878 * gst/rtsp-server/Makefile.am:
7879 * gst/rtsp-server/rtsp-media-factory-uri.c:
7880 * gst/rtsp-server/rtsp-media-factory-uri.h:
7881 * gst/rtsp-server/rtsp-server.h:
7882 factory-uri: add a factory to stream any URI
7883 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
7886 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7888 * gst/rtsp-server/rtsp-media.c:
7889 * gst/rtsp-server/rtsp-media.h:
7890 media: ignore spurious ASYNC_DONE messages
7891 When we are dynamically adding pads, the addition of the udpsrc elements will
7892 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
7893 the real ASYNC_DONE when everything is prerolled.
7895 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7897 * gst/rtsp-server/rtsp-media-factory.c:
7898 * gst/rtsp-server/rtsp-media-factory.h:
7899 media-factory: make lock macro
7901 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
7903 * gst/rtsp-server/rtsp-client.c:
7904 rtsp-server: Remove unused variable and dead assignment
7906 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
7908 * examples/test-launch.c:
7909 * examples/test-mp4.c:
7910 * examples/test-ogg.c:
7911 * examples/test-readme.c:
7912 * examples/test-sdp.c:
7913 * examples/test-video.c:
7914 examples: Run gst-indent
7916 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
7918 * gst/rtsp-server/rtsp-client.c:
7919 * gst/rtsp-server/rtsp-media-factory.c:
7920 * gst/rtsp-server/rtsp-media-mapping.c:
7921 * gst/rtsp-server/rtsp-media.c:
7922 * gst/rtsp-server/rtsp-params.c:
7923 * gst/rtsp-server/rtsp-sdp.c:
7924 * gst/rtsp-server/rtsp-server.c:
7925 * gst/rtsp-server/rtsp-session-pool.c:
7926 * gst/rtsp-server/rtsp-session.c:
7927 rtsp-server: Run gst-indent
7928 Since it wasn't using the upstream common previously, there was no
7929 indentation check before commiting.
7931 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
7933 * gst/rtsp-server/rtsp-media-mapping.h:
7934 * gst/rtsp-server/rtsp-media.c:
7935 * gst/rtsp-server/rtsp-media.h:
7936 * gst/rtsp-server/rtsp-sdp.c:
7937 * gst/rtsp-server/rtsp-session-pool.h:
7938 * gst/rtsp-server/rtsp-session.c:
7939 * gst/rtsp-server/rtsp-session.h:
7940 rtsp-server: Some more doc fixups
7942 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7945 Makefile: Add cruft-cleaning support
7947 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7952 * docs/libs/Makefile.am:
7953 * docs/libs/gst-rtsp-server-docs.sgml:
7954 * docs/libs/gst-rtsp-server-sections.txt:
7955 * docs/libs/gst-rtsp-server.types:
7956 * docs/version.entities.in:
7957 docs: Add gtk-doc build system
7959 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7961 * gst/rtsp-server/Makefile.am:
7962 Makefile.am: Use standard GIR make behaviour
7964 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7968 autogen/configure: Bring more in sync to standard gst module behaviour
7970 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7972 * gst/rtsp-server/rtsp-media.c:
7973 media: warn and fail when gstrtpbin is not found
7975 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7978 configure: open 0.11 branch
7980 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
7984 Add common submodule
7986 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
7989 * common/Makefile.am:
7990 * common/c-to-xml.py:
7992 * common/coverage/coverage-report-entry.pl:
7993 * common/coverage/coverage-report.pl:
7994 * common/coverage/coverage-report.xsl:
7995 * common/coverage/lcov.mak:
7996 * common/gettext.patch:
7997 * common/glib-gen.mak:
7998 * common/gst-autogen.sh:
7999 * common/gst-xmlinspect.py:
8001 * common/gstdoc-scangobj:
8002 * common/gtk-doc-plugins.mak:
8003 * common/gtk-doc.mak:
8004 * common/m4/.gitignore:
8005 * common/m4/Makefile.am:
8007 * common/m4/as-ac-expand.m4:
8008 * common/m4/as-auto-alt.m4:
8009 * common/m4/as-compiler-flag.m4:
8010 * common/m4/as-compiler.m4:
8011 * common/m4/as-docbook.m4:
8012 * common/m4/as-libtool-tags.m4:
8013 * common/m4/as-libtool.m4:
8014 * common/m4/as-python.m4:
8015 * common/m4/as-scrub-include.m4:
8016 * common/m4/as-version.m4:
8017 * common/m4/ax_create_stdint_h.m4:
8018 * common/m4/check.m4:
8019 * common/m4/glib-gettext.m4:
8020 * common/m4/gst-arch.m4:
8021 * common/m4/gst-args.m4:
8022 * common/m4/gst-check.m4:
8023 * common/m4/gst-debuginfo.m4:
8024 * common/m4/gst-default.m4:
8025 * common/m4/gst-doc.m4:
8026 * common/m4/gst-error.m4:
8027 * common/m4/gst-feature.m4:
8028 * common/m4/gst-function.m4:
8029 * common/m4/gst-gettext.m4:
8030 * common/m4/gst-glib2.m4:
8031 * common/m4/gst-libxml2.m4:
8032 * common/m4/gst-plugindir.m4:
8033 * common/m4/gst-valgrind.m4:
8034 * common/m4/gtk-doc.m4:
8035 * common/m4/introspection.m4:
8037 * common/mangle-tmpl.py:
8038 * common/plugins.xsl:
8040 * common/release.mak:
8041 * common/scangobj-merge.py:
8042 * common/upload.mak:
8043 common: Remove static version
8045 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
8047 * common/m4/introspection.m4:
8048 Update introspection.m4 to match usage
8050 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8054 Remove old stuff from the README
8056 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8061 === release 0.10.7 ===
8063 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8068 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8070 * examples/test-ogg.c:
8071 test-ogg: remove parsers
8072 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
8073 buffers with timestamps. Using the parsers also seems to break things.
8075 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8077 * bindings/vala/gst-rtsp-server-0.10.vapi:
8078 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8079 Updated Vala bindings
8081 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8083 * common/m4/introspection.m4:
8085 * gst/rtsp-server/Makefile.am:
8086 Added initial gobject-introspection support
8088 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8090 * gst/rtsp-server/rtsp-media-factory.c:
8091 media-factory: don't use host for shared hash key
8092 When we generate the key to share made between connections, don't include the
8093 host used to connect so that we can share media even if between clients that
8094 connected with localhost and ones with the ip address.
8096 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8098 * bindings/vala/Makefile.am:
8099 build: fix distcheck
8101 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8103 * bindings/vala/gst-rtsp-server-0.10.vapi:
8104 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8105 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8106 Update Vala bindings
8108 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8110 * bindings/vala/Makefile.am:
8112 Fix configure checks and installation location for Vala bindings
8115 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8120 === release 0.10.6 ===
8122 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8125 configure: release 0.10.6
8127 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8129 * gst/rtsp-server/rtsp-media.c:
8130 media: help the compiler a little
8132 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8134 * gst/rtsp-server/rtsp-media.c:
8135 * gst/rtsp-server/rtsp-media.h:
8136 * gst/rtsp-server/rtsp-session.c:
8137 media: cleanup media transport before freeing
8138 Cleanup the media transport data before freeing. In particular, remove the qdata
8139 from the rtpsource object.
8141 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8143 * gst/rtsp-server/rtsp-media-factory.c:
8144 * gst/rtsp-server/rtsp-media-factory.h:
8145 * gst/rtsp-server/rtsp-media.c:
8146 * gst/rtsp-server/rtsp-media.h:
8147 media-factory: add eos-shutdown property
8148 Add an eos-shutdown property that will send an EOS to the pipeline before
8149 shutting it down. This allows for nice cleanup in case of a muxer.
8152 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8154 * gst/rtsp-server/rtsp-media.c:
8155 * gst/rtsp-server/rtsp-media.h:
8156 media: use multiudpsink send-duplicates when we can
8157 If we have a new enough multiudpsink with the send-duplicates property, use this
8158 instead of doing our own filtering. Our custom filtering code should eventually
8159 be removed when we can depend on a released -good.
8161 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8163 * gst/rtsp-server/rtsp-media.c:
8164 media: don't leak destinations
8165 Refactor and cleanup the destinations array when the stream is destroyed.
8167 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8169 * gst/rtsp-server/rtsp-media.c:
8170 * gst/rtsp-server/rtsp-media.h:
8171 media: don't add udp addresses multiple times
8172 Keep track of the udp addresses we added to udpsink and never add the same udp
8173 destination twice. This avoids duplicate packets when using multicast.
8175 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8177 * gst/rtsp-server/rtsp-server.c:
8178 server: disable use of SO_LINGER
8179 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
8180 server close()s the connection.
8182 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8184 * gst/rtsp-server/rtsp-server.c:
8185 server: use 5 second linger period in SO_LINGER
8186 Wait 5 seconds before clearing the send buffers and reseting the connection with
8187 the client when we do a close. This should be enough time to get the message to
8191 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
8193 * gst/rtsp-server/rtsp-server.c:
8194 server: use SO_LINGER
8195 SO_LINGER on the socket will make sure that any pending data on the socket is
8196 flushed ASAP and that the socket connection is reset. This makes sure that the
8197 socket can be reused immediately.
8200 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8203 README: add blurb about shared media factories
8205 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
8207 * gst/rtsp-server/rtsp-media.c:
8208 Add stdlib.h for atoi()
8210 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8212 * bindings/python/Makefile.am:
8213 * bindings/vala/Makefile.am:
8214 build: distcheck fixes
8215 Fix 'make distcheck', somewhat (it still fails because it tries to
8216 install files into /usr/share/vala/vapi/ irrespective of the
8219 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8222 configure: bump core/base requirements to released version
8223 Makes things less confusing for people.
8225 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8228 configure: fail if GStreamer core/base requirements are not met
8230 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8232 * gst/rtsp-server/rtsp-client.c:
8233 client: improve client cleanups
8234 Make sure the session does not timeout when using TCP. We need to do this
8235 because quicktime player does not send RTCP for some reason in tunneled
8237 Refactor some cleanup code.
8240 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8242 * gst/rtsp-server/rtsp-session.c:
8243 * gst/rtsp-server/rtsp-session.h:
8244 session: add support for prevent session timeouts
8245 Add an atomix counter to prevent session timeouts when we are, for example,
8248 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8250 * gst/rtsp-server/rtsp-client.c:
8251 client: fix unlink on session timeouts
8252 When our session times out, make sure we unlink all streams in this
8254 Remove the tunnelid when closing the connection.
8256 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8258 * gst/rtsp-server/rtsp-session.c:
8259 session: small cleanups
8261 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8263 * gst/rtsp-server/rtsp-client.c:
8264 client: handle lost_tunnel callbacks
8265 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
8266 hashtable so that we can reuse it for when the client reopens the POST
8268 Close the connection after a TEARDOWN.
8269 Make sure or watchid is cleared when the watch is removed.
8272 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8274 * gst/rtsp-server/rtsp-client.c:
8275 * gst/rtsp-server/rtsp-media.c:
8276 * gst/rtsp-server/rtsp-sdp.c:
8277 rtsp-server: add more support for multicast
8279 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8282 * gst/rtsp-server/rtsp-media.c:
8283 * gst/rtsp-server/rtsp-media.h:
8284 media: allow configuration of allowed lower transport
8286 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8288 * gst/rtsp-server/rtsp-client.h:
8289 * gst/rtsp-server/rtsp-media.c:
8290 * gst/rtsp-server/rtsp-media.h:
8291 * gst/rtsp-server/rtsp-sdp.c:
8292 * gst/rtsp-server/rtsp-sdp.h:
8293 * gst/rtsp-server/rtsp-server.c:
8294 rtsp: keep track of server ip and ipv6
8295 Keep track of how the client connected to the server and setup the udp ports
8296 with the same protocol.
8297 Copy the server ip address in the SDP so that clients can send RTCP back to
8300 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8302 * gst/rtsp-server/rtsp-session.c:
8305 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8307 * gst/rtsp-server/rtsp-client.c:
8308 client: use right size for malloc
8310 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8312 * gst/rtsp-server/rtsp-server.c:
8313 server: comment ipv6 server listening address
8315 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8317 * gst/rtsp-server/rtsp-media.c:
8318 media: allow for ipv6 sockets
8320 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8322 * gst/rtsp-server/rtsp-server.c:
8323 * gst/rtsp-server/rtsp-server.h:
8324 server: rework server part
8325 Allow setting a bind address, make sure we can deal with ipv6.
8326 Remove the port property and change with the service property.
8328 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8330 * gst/rtsp-server/rtsp-media.h:
8331 media: update comments a little
8333 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8335 * gst/rtsp-server/rtsp-client.c:
8336 client: make content-base better
8337 Use the URI formatting functions to make a content-base. Also make sure that
8338 there is a trailing / at the end.
8340 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8342 * gst/rtsp-server/rtsp-client.c:
8343 client: guard against invalid paths
8345 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8347 * examples/test-video.c:
8348 test: catch server bind errors
8350 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
8352 * gst/rtsp-server/rtsp-media.c:
8353 rtspmedia: emit "unprepared" if _prepare fails.
8354 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
8355 media object is removed from its factory's cache.
8357 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8359 * gst/rtsp-server/rtsp-media.c:
8360 media: collect media position when seek completes
8362 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
8364 * gst/rtsp-server/rtsp-client.c:
8365 client: call unlink_streams in client finalize
8368 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8370 * gst/rtsp-server/rtsp-media.c:
8371 media: limit the time to wait to something huge
8372 Avoid waiting forever but limit the timeout to 20 seconds.
8374 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8376 * gst/rtsp-server/rtsp-sdp.c:
8377 sdp: reindent and check for prepared status
8379 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8381 * gst/rtsp-server/rtsp-media.c:
8382 * gst/rtsp-server/rtsp-media.h:
8383 * gst/rtsp-server/rtsp-session.c:
8384 media: avoid doing _get_state() for state changes
8385 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
8386 until the media is prerolled or in error. This avoids doing a blocking call of
8387 gst_element_get_state() that can cause lockups when there is an error.
8390 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8392 * gst/rtsp-server/rtsp-media.c:
8395 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8397 * gst/rtsp-server/rtsp-media-factory.c:
8398 media-factory: better error handling
8399 Improve the error handling a bit.
8401 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8403 * gst/rtsp-server/rtsp-client.c:
8404 client: rework transport parsing
8405 Rework the transport parsing code so that we can ignore transports we don't
8406 support instead of just picking the first one we can parse.
8407 Configure a (for now hardcoded) destination for multicast transports.
8409 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8411 * gst/rtsp-server/rtsp-media.c:
8412 media: set multicast sink parameters
8413 Disable loop and automatic multicast join on the udpsink elements.
8414 Add some more debug info.
8415 Reset some state variables in the right place.
8416 Use the right port numbers for multicast.
8418 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8420 * gst/rtsp-server/rtsp-session.c:
8421 session: handle transport setup correctly
8422 Handle UDP, MCAST and TCP transport negotiation more correctly.
8423 Store the server session SSRC in the transport.
8425 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8427 * gst/rtsp-server/rtsp-client.c:
8428 rtsp-client: implement error_full
8429 Implement error_full to avoid some segfaults when the rtspconnection calls it.
8432 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8435 * gst/rtsp-server/rtsp-client.c:
8436 * gst/rtsp-server/rtsp-server.c:
8437 docs: update docs and comments
8439 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
8441 * gst/rtsp-server/rtsp-sdp.c:
8442 sdp: make server work better when behind a proxy
8444 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8446 * gst/rtsp-server/rtsp-client.c:
8447 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
8449 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8451 * gst/rtsp-server/rtsp-client.c:
8452 * gst/rtsp-server/rtsp-media-factory.c:
8453 * gst/rtsp-server/rtsp-media-mapping.c:
8454 * gst/rtsp-server/rtsp-media.c:
8455 * gst/rtsp-server/rtsp-server.c:
8456 * gst/rtsp-server/rtsp-session-pool.c:
8457 * gst/rtsp-server/rtsp-session.c:
8458 Use GStreamer's debugging subsystem
8460 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8462 * gst/rtsp-server/rtsp-media-factory.c:
8463 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
8465 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8470 === release 0.10.5 ===
8472 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8477 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8480 configure: bump required versions
8482 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
8484 * gst/rtsp-server/rtsp-client.c:
8485 client: call weak-unref on client->sessions from finalize
8488 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8490 * gst/rtsp-server/rtsp-media.c:
8491 media: Fixed crasher where caps got unref'ed too often
8493 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8496 * pkgconfig/.gitignore:
8497 * pkgconfig/Makefile.am:
8498 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
8499 Added pkg-config file to use gst-rtsp-server uninstalled
8501 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8503 * gst/rtsp-server/rtsp-media.c:
8504 media: add some docs
8506 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
8508 * gst/rtsp-server/rtsp-client.c:
8509 rtsp: Use gst_rtsp_watch_send_message().
8510 Use gst_rtsp_watch_send_message() since the old API which used
8511 gst_rtsp_watch_queue_message() has been deprecated.
8513 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8518 === release 0.10.4 ===
8520 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8525 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8527 * gst/rtsp-server/rtsp-client.c:
8528 * gst/rtsp-server/rtsp-session.c:
8529 * gst/rtsp-server/rtsp-session.h:
8530 rtsp: allocate channels in TCP mode
8531 When the client does not provide us with channels in TCP mode, allocate channels
8534 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8536 * gst/rtsp-server/rtsp-client.c:
8537 client: don't crash when tunnelid is missing
8538 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
8539 don't crash but return an error response to the client.
8542 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8544 * bindings/vala/gst-rtsp-server-0.10.vapi:
8545 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8546 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8547 bindings: update vala bindings with new method
8549 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8551 * gst/rtsp-server/rtsp-session-pool.c:
8552 * gst/rtsp-server/rtsp-session-pool.h:
8553 sessionpool: add function to filter sessions
8554 Add generic function to retrieve/remove sessions.
8556 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8559 configure: bump core/base requirements to release
8561 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8563 * gst/rtsp-server/rtsp-media.c:
8564 media: fix indentation
8566 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8568 * gst/rtsp-server/rtsp-media.c:
8569 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
8571 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8573 * gst/rtsp-server/rtsp-media.c:
8574 set state and remove elements of media in for loop
8576 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
8578 * bindings/vala/gst-rtsp-server-0.10.vapi:
8579 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8580 Added gst_rtsp_media_remove_elements function to Vala bindings
8582 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
8584 * gst/rtsp-server/rtsp-media.c:
8585 * gst/rtsp-server/rtsp-media.h:
8586 Added gst_rtsp_media_remove_elements function
8588 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
8590 * gst/rtsp-server/rtsp-media.c:
8591 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
8593 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8595 * bindings/vala/gst-rtsp-server-0.10.vapi:
8596 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8597 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8598 Updated Vala bindings
8600 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8602 * gst/rtsp-server/rtsp-media.c:
8603 * gst/rtsp-server/rtsp-media.h:
8604 Added vmethod unprepare to GstRTSPMedia
8605 The default implementation sets the state of the pipeline to GST_STATE_NULL
8607 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8609 * gst/rtsp-server/rtsp-media-factory.c:
8610 * gst/rtsp-server/rtsp-media-factory.h:
8611 Made collect_streams function public
8613 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8615 * gst/rtsp-server/rtsp-media-factory.c:
8616 * gst/rtsp-server/rtsp-media-factory.h:
8617 * gst/rtsp-server/rtsp-media.c:
8618 Added vmethod create_pipeline to GstRTSPMediaFactory
8619 The pipeline is created in this method and the GstRTSPMedia's element is added to it
8621 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8623 * gst/rtsp-server/rtsp-client.c:
8624 client: use g_source_destroy()
8625 We need to use g_source_destroy() because we might have added the source to a
8626 different main context than the default one.
8628 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8630 * gst/rtsp-server/Makefile.am:
8631 * gst/rtsp-server/rtsp-client.c:
8632 * gst/rtsp-server/rtsp-params.c:
8633 * gst/rtsp-server/rtsp-params.h:
8634 rtsp: prepare for handling GET/SET_PARAMETER
8635 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
8637 Fix return codes of handlers.
8639 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8641 * gst/rtsp-server/rtsp-media.c:
8642 media: don't leak session pads
8644 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8646 * gst/rtsp-server/rtsp-media.c:
8647 media: clean up the messages a bit
8649 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8651 * gst/rtsp-server/rtsp-sdp.c:
8652 sdp: warn and skip streams without media
8654 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8656 * bindings/vala/gst-rtsp-server-0.10.vapi:
8657 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8658 vala: Fixed typo in header file of RTSPMediaStream
8660 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8662 * gst/rtsp-server/rtsp-media.c:
8665 Make dumping RTCP stats configurable
8667 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8669 * gst/rtsp-server/rtsp-media.c:
8670 media: be less verbose and leak less
8672 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8674 * gst/rtsp-server/rtsp-media.c:
8675 media: don't leak the destination address
8677 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8679 * gst/rtsp-server/rtsp-client.c:
8680 * gst/rtsp-server/rtsp-media.c:
8681 * gst/rtsp-server/rtsp-media.h:
8682 * gst/rtsp-server/rtsp-session.c:
8683 * gst/rtsp-server/rtsp-session.h:
8684 rtsp: use RTCP to keep the session alive
8685 Use the RTCP rtcp-from stats field to find the associated session and use this
8686 to keep the session alive.
8688 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8690 * gst/rtsp-server/rtsp-session.c:
8691 session: add 5sec to the real session timeout
8692 Allow the session to live 5sec longer before really timing out. This should give
8693 clients some extra time to keep the session active.
8695 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8697 * gst/rtsp-server/rtsp-client.c:
8698 client: replay OK to GET/SET_PARAMETER
8699 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
8700 so that we return OK for those requests.
8702 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8704 * gst/rtsp-server/rtsp-media.c:
8705 * gst/rtsp-server/rtsp-media.h:
8706 media: keep track of active transports
8707 Keep track of which transport is active to avoid closing the connection too
8709 Remove the destination transport also when going to NULL.
8710 Print some stats about the SDES and other RTCP messages we receive from the
8713 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8715 * examples/.gitignore:
8716 * examples/Makefile.am:
8717 * examples/test-sdp.c:
8718 example: add SDP relay example
8720 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8722 * gst/rtsp-server/rtsp-media.c:
8723 media: also count active TCP connections
8725 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8727 * gst/rtsp-server/rtsp-media-factory.c:
8728 * gst/rtsp-server/rtsp-media.c:
8729 * gst/rtsp-server/rtsp-media.h:
8730 rtsp: add support for dynamic elements
8731 Add support for dynamic elements.
8732 Don't set live pipelines back to paused.
8734 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8736 * gst/rtsp-server/rtsp-sdp.c:
8737 sdp: don't add encoding name when absent in caps
8739 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8741 * gst/rtsp-server/rtsp-client.c:
8742 client: warn when we can't do RTP-Info
8744 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8746 * gst/rtsp-server/rtsp-media-factory.c:
8747 factory: factor out the stream construction
8749 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8751 * gst/rtsp-server/rtsp-client.c:
8752 client: only add RTP-Info when we have the info
8753 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
8756 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8761 === release 0.10.3 ===
8763 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8767 - Fixes a bug where it put the wrong verion in pkgconfig
8768 - Link RTP and RTCP sources
8770 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8772 * gst/rtsp-server/rtsp-media.c:
8773 * gst/rtsp-server/rtsp-media.h:
8774 media: link the RTP udpsrc to the session manager
8775 Link the RTP udpsrc and the appsrc to the session manager so that they don't
8776 shut down when the client sends a packet to open firewalls.
8778 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8780 * pkgconfig/gst-rtsp-server.pc.in:
8781 Don't use hard-coded version number in pkg-config file
8783 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8788 === release 0.10.2 ===
8790 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8795 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8798 * common/m4/.gitignore:
8799 * examples/.gitignore:
8800 * pkgconfig/.gitignore:
8801 add some .gitignore files
8803 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8805 * gst/rtsp-server/rtsp-media.c:
8806 media: seek to key frames
8808 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8810 * gst/rtsp-server/rtsp-media.c:
8811 media: emit the unprepared signal by id
8812 Emit the unprepared signal by id instead of name and set the media as
8815 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8817 * gst/rtsp-server/rtsp-media.c:
8818 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
8820 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8822 * gst/rtsp-server/rtsp-server.c:
8823 Added finalize function to GstRTPSPServer to unref session pool and media mapping
8825 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8827 * bindings/vala/gst-rtsp-server-0.10.vapi:
8828 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8829 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8830 Updated vala bindings
8832 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8834 * gst/rtsp-server/Makefile.am:
8835 * gst/rtsp-server/rtsp-client.c:
8836 * gst/rtsp-server/rtsp-media.c:
8837 server: use appsink and appsrc with the API
8838 Use the appsink/appsrc API instead of the signals for higher
8841 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8843 * examples/test-ogg.c:
8844 tests: set the payload type correctly
8846 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8848 * gst/rtsp-server/rtsp-media-factory.c:
8849 factory: connect to the unprepare signal
8850 Connect to the unprepare signal for non-reusable media so that we can remove
8851 them from the cache.
8853 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8855 * gst/rtsp-server/rtsp-media.c:
8856 * gst/rtsp-server/rtsp-media.h:
8857 media: add signal to notify of unprepare
8859 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8861 * gst/rtsp-server/rtsp-media.c:
8862 * gst/rtsp-server/rtsp-media.h:
8863 media: more work on making the media shared
8864 Add a reusable flag to medias, indicating that they can be reused after a state
8868 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8870 * examples/test-readme.c:
8871 examples: mark the example as shared for testing
8873 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8875 * gst/rtsp-server/rtsp-media.c:
8876 * gst/rtsp-server/rtsp-media.h:
8877 client: support shared media
8878 Always perform the state actions even if the target state of the pipeline is
8879 already correct, we still want to add/remove the transports when we are dealing
8881 Keep a counter of the number of active transports for a media so that we can use
8882 this to perform a state change when needed.
8883 Perform a state change of the pipeline only when the first transport was added
8884 or when there are no active transports.
8886 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8888 * gst/rtsp-server/rtsp-client.c:
8889 client: fix refcounting crasher
8890 Don't need to remove the weak refs in the finalize methods, they are already
8891 removed in the dispose.
8892 Don't register the callback with a DestroyNofity.
8894 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8896 * gst/rtsp-server/rtsp-client.c:
8897 Fix rtsp client refcount management in TCP mode.
8898 Don't unref a client ref we never had. Fixes an unref
8899 of an already-free client object after a client
8900 teardown request for me.
8902 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8904 * gst/rtsp-server/rtsp-session.c:
8905 docs: fix typo in API docs
8907 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8909 * gst/rtsp-server/rtsp-media.c:
8911 Keep the udp sources in playing even if we go to paused. unlock the sources when
8913 Add some more debug info.
8914 Only seek when we need to.
8915 Keep track of the position when we go to paused.
8917 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8919 * gst/rtsp-server/rtsp-client.c:
8920 * gst/rtsp-server/rtsp-media.c:
8921 * gst/rtsp-server/rtsp-media.h:
8922 Add beginnings of seeking.
8923 Parse the Range header and perform a seek on the pipeline for the requested
8924 position. It's disabled currently until I figure out what's going wrong.
8926 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8928 * gst/rtsp-server/rtsp-client.c:
8929 allow pause requests for now.
8932 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8934 * gst/rtsp-server/rtsp-client.c:
8935 Remove weak ref on the session in teardown
8936 We need to remove our weakref from the session when we do a teardown because
8937 else we close the TCP connection prematurely.
8939 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8941 * gst/rtsp-server/rtsp-client.c:
8942 * gst/rtsp-server/rtsp-client.h:
8943 * gst/rtsp-server/rtsp-session-pool.c:
8944 Do some more session cleanup
8945 Make session timeout kill the TCP connection that currently watches the
8947 Remove the client timeout property.
8949 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8951 * gst/rtsp-server/rtsp-client.c:
8952 * gst/rtsp-server/rtsp-client.h:
8953 * gst/rtsp-server/rtsp-media.c:
8954 * gst/rtsp-server/rtsp-media.h:
8955 * gst/rtsp-server/rtsp-server.c:
8956 * gst/rtsp-server/rtsp-session.c:
8957 * gst/rtsp-server/rtsp-session.h:
8959 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
8962 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8964 * examples/Makefile.am:
8965 * examples/test-launch.c:
8966 Add example server that takes launch lines
8967 Add an example server that streams any -launch line.
8969 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8971 * examples/test-readme.c:
8972 * gst/rtsp-server/rtsp-client.c:
8973 * gst/rtsp-server/rtsp-media.c:
8974 * gst/rtsp-server/rtsp-media.h:
8975 Add support for live streams
8976 Add support for live streams and ranges
8977 Start on handling TCP data transfer.
8979 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8981 * gst/rtsp-server/rtsp-media.c:
8982 Free the pipeline before other things
8985 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8987 * gst/rtsp-server/rtsp-client.c:
8988 Only free the pending tunnel if there is one
8991 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8993 * gst/rtsp-server/rtsp-client.c:
8994 * gst/rtsp-server/rtsp-client.h:
8995 * gst/rtsp-server/rtsp-media.c:
8996 rtsp-server: Add support for tunneling
8997 Add support for tunneling over HTTP.
8998 Use new connection methods to retrieve the url.
8999 Dispatch messages based on the message type instead of blindly
9000 assuming it's always a request.
9001 Keep track of the watch id so that we can remove it later.
9002 Set the media pipeline to NULL before unreffing the pipeline.
9004 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9006 * gst/rtsp-server/rtsp-client.c:
9007 * gst/rtsp-server/rtsp-client.h:
9008 Fix for channel -> watch rename in gstreamer
9009 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
9011 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9013 * gst/rtsp-server/rtsp-client.c:
9014 * gst/rtsp-server/rtsp-client.h:
9016 Use the async RTSP channels instead of spawning a new thread for each client.
9017 If a sessionid is specified in a request, fail if we don't have the session.
9019 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9021 * gst/rtsp-server/rtsp-media.c:
9022 Add better debug info
9023 Add some better debug info.
9025 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9027 * examples/test-video.c:
9029 Add support for session timeouts in the example.
9031 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9033 * gst/rtsp-server/rtsp-session-pool.c:
9034 * gst/rtsp-server/rtsp-session-pool.h:
9035 Pass GTimeVal around for performance reasons
9036 Get the current time only once and pass it around so that sessions don't have to
9037 get the current time anymore.
9038 Add experimental support for a GSource that dispatches when the session needs to
9041 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9043 * gst/rtsp-server/rtsp-session.c:
9044 * gst/rtsp-server/rtsp-session.h:
9045 Add better support for session timeouts
9046 Add a method to request the number of milliseconds when a session will timeout.
9048 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9050 * gst/rtsp-server/rtsp-media.c:
9051 * gst/rtsp-server/rtsp-media.h:
9052 Add suport for RTP manager monitoring
9053 Add the first stage in monitoring the rtp manager.
9054 Make sure we don't update the state to something we don't want.
9056 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9058 * gst/rtsp-server/rtsp-client.c:
9059 Add support for session keepalive
9060 Get and update the session timeout for all requests. get the session as early as
9063 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9065 * gst/rtsp-server/rtsp-media-factory.h:
9066 * gst/rtsp-server/rtsp-media.c:
9067 * gst/rtsp-server/rtsp-media.h:
9068 Handle media bus messages
9069 Handle media bus messages in a custom mainloop and dispatch them to the
9070 RTSPMedia objects. Let the default implementation handle some common messages.
9072 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9074 * gst/rtsp-server/rtsp-client.c:
9075 * gst/rtsp-server/rtsp-session-pool.c:
9076 * gst/rtsp-server/rtsp-session.c:
9077 Some more session timeout handling
9078 Move the session header setting code to a central place so that we always add
9079 the timeout parameter too.
9080 Handle timeouts by running the session cleanup code.
9081 Stop media before cleaning up.
9083 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9085 * gst/rtsp-server/rtsp-client.c:
9086 * gst/rtsp-server/rtsp-client.h:
9087 Add timeout property
9088 Add a timeout property ot the client and make the other properties into GObject
9091 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9093 * gst/rtsp-server/rtsp-session-pool.c:
9094 Use getters and setters in property code
9095 Use the getters and setters for the timeout property instead of locking
9098 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9100 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
9102 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9104 * gst/rtsp-server/rtsp-session-pool.c:
9105 * gst/rtsp-server/rtsp-session-pool.h:
9106 * gst/rtsp-server/rtsp-session.c:
9107 * gst/rtsp-server/rtsp-session.h:
9108 Add more timeout stuff
9109 Add method to check if a session is expired.
9110 Add method to perform cleanup on a session pool.
9112 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9114 * gst/rtsp-server/rtsp-client.c:
9115 * gst/rtsp-server/rtsp-session-pool.c:
9116 * gst/rtsp-server/rtsp-session-pool.h:
9117 * gst/rtsp-server/rtsp-session.c:
9118 * gst/rtsp-server/rtsp-session.h:
9119 Add beginnings of session timeouts and limits
9120 Add the timeout value to the Session header for unusual timeout values.
9121 Allow us to configure a limit to the amount of active sessions in a pool. Set a
9122 limit on the amount of retry we do after a sessionid collision.
9123 Add properties to the sessionid and the timeout of a session. Keep track of
9124 creation time and last access time for sessions.
9126 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9128 * gst/rtsp-server/rtsp-client.c:
9129 * gst/rtsp-server/rtsp-media.c:
9130 * gst/rtsp-server/rtsp-media.h:
9131 * gst/rtsp-server/rtsp-sdp.c:
9132 * gst/rtsp-server/rtsp-session-pool.c:
9133 * gst/rtsp-server/rtsp-session.c:
9134 * gst/rtsp-server/rtsp-session.h:
9135 Cleanup of sessions and more
9136 Fix the refcounting of media and sessions in the client. Properly clean up the
9137 session data when the client performs a teardown.
9138 Add Server header to responses.
9139 Allow for multiple uri setups in one session.
9140 Add Range header to the PLAY response and add the range attribute to the SDP
9142 Fix the session pool remove method, it used the wrong key in the hashtable. Also
9143 give the ownership of the sessionid to the session object.
9145 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9147 * gst/rtsp-server/rtsp-server.c:
9148 * gst/rtsp-server/rtsp-server.h:
9150 Rename the 'server_port' variable to simply 'port'.
9152 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9155 * gst/rtsp-server/rtsp-client.c:
9156 * gst/rtsp-server/rtsp-media.c:
9157 * gst/rtsp-server/rtsp-media.h:
9158 * gst/rtsp-server/rtsp-session.c:
9159 * gst/rtsp-server/rtsp-session.h:
9160 Rework the way we handle transports for streams
9161 Make the media accept an array of transports for the streams that we have
9162 configured for the play/pause requests.
9163 Implement server states for a client and its media.
9164 Require 0.10.22.1 (git HEAD) of gstreamer.
9166 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9168 * gst/rtsp-server/rtsp-client.c:
9169 * gst/rtsp-server/rtsp-media-factory.c:
9170 Drop const from functions dealing with urls
9171 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
9172 have the right const in them.
9174 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9176 * gst/rtsp-server/rtsp-client.c:
9177 * gst/rtsp-server/rtsp-media.c:
9178 * gst/rtsp-server/rtsp-sdp.c:
9182 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9184 * gst/rtsp-server/rtsp-client.c:
9185 * gst/rtsp-server/rtsp-media-factory.c:
9186 * gst/rtsp-server/rtsp-media.c:
9187 * gst/rtsp-server/rtsp-media.h:
9189 Don't keep a reference to the GstRTSPMedia in the stream.
9190 Free more things when freeing the GstRTSPMedia.
9192 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9195 * gst/rtsp-server/rtsp-media-factory.c:
9196 * gst/rtsp-server/rtsp-media-factory.h:
9197 * gst/rtsp-server/rtsp-media.c:
9198 * gst/rtsp-server/rtsp-media.h:
9199 * gst/rtsp-server/rtsp-server.c:
9200 * gst/rtsp-server/rtsp-server.h:
9201 More docs and small cleanups
9202 Add some more docs and update the README
9203 Cleanup some method names.
9204 Remove an unneeded idx field in the GstRTSPMediaStream
9206 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9209 * examples/Makefile.am:
9210 * examples/test-readme.c:
9211 Add a README and more example code
9212 Add a README file that contains a small introduction on how to use the server
9213 along with the example code explained in the readme.
9215 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9217 * gst/rtsp-server/rtsp-media.c:
9218 * gst/rtsp-server/rtsp-server.c:
9219 Fix some leaks and change default port
9220 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
9221 we finished the initial preroll. If we keep them locked, setting the pipeline to
9222 NULL will not stop and clean up the sources correctly.
9223 Change the default RTSP port to 8554 aka the official alternative RTSP port.
9225 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9227 * gst/rtsp-server/rtsp-session.c:
9228 * gst/rtsp-server/rtsp-session.h:
9229 Cleanups to the session object
9230 Remove some unneeded variables in the session state of a stream such as the
9231 owner media and the server transport.
9232 Get the configuration of a media stream in a session based on the media_stream
9233 in the original object instead of our cached index.
9234 Free more data in the finalize method.
9236 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9238 * gst/rtsp-server/rtsp-client.c:
9239 * gst/rtsp-server/rtsp-client.h:
9240 Cleanups and reuse media from DESCRIBE
9241 Handle thread create errors.
9242 Rename some internal methods to better match what they actually do.
9243 Handle misconfiguration of session_pool and media_mapping gracefully.
9244 Cache the DESCRIBE media and uri in the client connection and reuse them when
9245 we receive a SETUP request in the same connection for the same uri.
9246 Cleanup the client connection object.
9248 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9250 * gst/rtsp-server/rtsp-media-factory.c:
9251 * gst/rtsp-server/rtsp-media-factory.h:
9252 * gst/rtsp-server/rtsp-media.c:
9253 * gst/rtsp-server/rtsp-media.h:
9254 Add shared properties to media and factory
9255 Add the shared property to media.
9256 Implement some simple caching in the factory depending on if the media is shared
9259 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9261 * gst/rtsp-server/rtsp-client.c:
9262 Add a little comment
9263 Add some comment about the content-base header.
9265 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9267 * examples/Makefile.am:
9269 * examples/test-mp4.c:
9270 * examples/test-ogg.c:
9271 * examples/test-video.c:
9272 * gst/rtsp-server/Makefile.am:
9273 * gst/rtsp-server/rtsp-client.c:
9274 * gst/rtsp-server/rtsp-client.h:
9275 * gst/rtsp-server/rtsp-media-factory.c:
9276 * gst/rtsp-server/rtsp-media-factory.h:
9277 * gst/rtsp-server/rtsp-media.c:
9278 * gst/rtsp-server/rtsp-media.h:
9279 * gst/rtsp-server/rtsp-sdp.c:
9280 * gst/rtsp-server/rtsp-sdp.h:
9281 * gst/rtsp-server/rtsp-server.c:
9282 * gst/rtsp-server/rtsp-server.h:
9283 * gst/rtsp-server/rtsp-session.c:
9284 * gst/rtsp-server/rtsp-session.h:
9285 Reorganize things, prepare for media sharing
9286 Added various other test server examples
9287 Move the SDP message generation to a separate helper.
9288 Refactor common code for finding the session.
9289 Add content-base for realplayer compatibility
9290 Clean up request uris before processing for better vlc compatibility.
9291 Move prerolling and pipeline construction to the RTSPMedia object.
9292 Use multiudpsink for future pipeline reuse.
9294 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9300 === release 0.10.1 ===
9302 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9308 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9310 * bindings/vala/Makefile.am:
9312 Add more directories and files to the dist.
9314 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9316 * bindings/python/Makefile.am:
9317 * bindings/python/rtspserver.override:
9318 Fixed compile error of python bindings
9320 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9322 * bindings/vala/gst-rtsp-server-0.10.vapi:
9323 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9324 Marked values as nullable accordingly
9326 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9328 * bindings/vala/gst-rtsp-server-0.10.vapi:
9329 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
9330 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9331 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9332 Updated Vala bindings
9334 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9336 * gst/rtsp-server/rtsp-client.c:
9337 * gst/rtsp-server/rtsp-media-mapping.c:
9338 * gst/rtsp-server/rtsp-media-mapping.h:
9339 * gst/rtsp-server/rtsp-media.h:
9340 * gst/rtsp-server/rtsp-session-pool.h:
9341 Cleanups and doc updates
9342 Add some more documentation and do some minor cleanups here and there.
9344 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9346 * gst/rtsp-server/rtsp-client.c:
9347 * gst/rtsp-server/rtsp-media-factory.c:
9348 * gst/rtsp-server/rtsp-media-factory.h:
9349 * gst/rtsp-server/rtsp-media.c:
9350 * gst/rtsp-server/rtsp-media.h:
9351 * gst/rtsp-server/rtsp-session.c:
9352 * gst/rtsp-server/rtsp-session.h:
9354 Rename GstRTSPMediaBin to GstRTSPMedia
9355 Parse the request url into a GstRTSPUri object and pass this object to the
9356 various handlers and methods that require the uri.
9358 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9362 Add some more docs and remove some old code from the example.
9364 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9366 * gst/rtsp-server/rtsp-client.c:
9367 Handle state change failures better
9368 Handle state change failures better when changing the state of the pipeline to
9371 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9373 * gst/rtsp-server/rtsp-media-factory.c:
9374 * gst/rtsp-server/rtsp-media-factory.h:
9375 Make element creation more extendible
9376 Add get_element vmethod to the default MediaFactory so that subclasses can just
9377 override that method and still use the default logic for making a MediaBin from
9380 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9383 * gst/rtsp-server/Makefile.am:
9384 * gst/rtsp-server/rtsp-client.c:
9385 * gst/rtsp-server/rtsp-client.h:
9386 * gst/rtsp-server/rtsp-media-factory.c:
9387 * gst/rtsp-server/rtsp-media-factory.h:
9388 * gst/rtsp-server/rtsp-media-mapping.c:
9389 * gst/rtsp-server/rtsp-media-mapping.h:
9390 * gst/rtsp-server/rtsp-media.c:
9391 * gst/rtsp-server/rtsp-media.h:
9392 * gst/rtsp-server/rtsp-server.c:
9393 * gst/rtsp-server/rtsp-server.h:
9394 * gst/rtsp-server/rtsp-session.c:
9395 * gst/rtsp-server/rtsp-session.h:
9396 Make the server handle arbitrary pipelines
9397 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
9398 The GstMediaBin object has a handle to a bin with elements and to a list of
9399 GstMediaStream objects that this bin produces.
9400 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
9401 with methods to register and remove those mappings.
9402 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
9403 used by the server instance.
9404 Modify the example application so that it shows how to create custom pipelines
9405 attached to a specific mount point.
9406 Various misc cleanps.
9408 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9410 * gst/rtsp-server/rtsp-server.c:
9411 * gst/rtsp-server/rtsp-server.h:
9412 Allow setting a custom media factory for a server
9414 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9416 * gst/rtsp-server/rtsp-client.c:
9417 * gst/rtsp-server/rtsp-client.h:
9418 Allow setting a custom media factory for a client.
9420 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9422 * gst/rtsp-server/Makefile.am:
9423 Add Makefile entry for the media factory
9425 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9427 * gst/rtsp-server/rtsp-media-factory.c:
9428 * gst/rtsp-server/rtsp-media-factory.h:
9429 Add media factory to map urls to media pipeline objects.
9431 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9433 * gst/rtsp-server/rtsp-media.c:
9434 * gst/rtsp-server/rtsp-media.h:
9435 Add comments. Remove unused field
9437 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9439 * gst/rtsp-server/rtsp-session-pool.c:
9440 * gst/rtsp-server/rtsp-session-pool.h:
9441 Allow custom session pools to override the session id allocation algorithms Add some comments.
9443 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9445 * gst/rtsp-server/rtsp-session.h:
9448 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9450 * gst/rtsp-server/rtsp-client.c:
9451 * gst/rtsp-server/rtsp-client.h:
9452 Move the connection code in one place Add some comments
9454 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9456 * gst/rtsp-server/rtsp-server.c:
9457 * gst/rtsp-server/rtsp-server.h:
9458 Make vmethod to create and accept new clients. Add some docs.
9460 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9462 * gst/rtsp-server/rtsp-server.c:
9463 * gst/rtsp-server/rtsp-server.h:
9464 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
9466 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9468 * gst/rtsp-server/rtsp-client.c:
9469 * gst/rtsp-server/rtsp-client.h:
9470 Name the parameters more appropriately.
9472 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9474 * gst/rtsp-server/rtsp-session-pool.c:
9475 Do some more cleanup of the session pool.
9477 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9479 * gst/rtsp-server/Makefile.am:
9480 * gst/rtsp-server/rtsp-client.c:
9481 Check if return value of gst_rtsp_session_get_media is not NULL
9483 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9485 * gst/rtsp-server/Makefile.am:
9486 Install rtsp-session and rtsp-session-pool headers
9488 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9493 * bindings/python/Makefile.am:
9494 * bindings/python/arg-types.py:
9495 * bindings/python/codegen/Makefile.am:
9496 * bindings/python/codegen/__init__.py:
9497 * bindings/python/codegen/argtypes.py:
9498 * bindings/python/codegen/code-coverage.py:
9499 * bindings/python/codegen/codegen.py:
9500 * bindings/python/codegen/definitions.py:
9501 * bindings/python/codegen/defsparser.py:
9502 * bindings/python/codegen/docextract.py:
9503 * bindings/python/codegen/docgen.py:
9504 * bindings/python/codegen/fileprefix.override:
9505 * bindings/python/codegen/fileprefixmodule.c:
9506 * bindings/python/codegen/h2def.py:
9507 * bindings/python/codegen/mergedefs.py:
9508 * bindings/python/codegen/mkskel.py:
9509 * bindings/python/codegen/override.py:
9510 * bindings/python/codegen/reversewrapper.py:
9511 * bindings/python/codegen/scmexpr.py:
9512 * bindings/python/rtspserver-types.defs:
9513 * bindings/python/rtspserver.defs:
9514 * bindings/python/rtspserver.override:
9515 * bindings/python/rtspservermodule.c:
9517 Add python bindings.
9519 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9521 * bindings/Makefile.am:
9523 Don't go into python dir when requirements for python bindings are missing
9525 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9527 * bindings/Makefile.am:
9528 * bindings/vala/Makefile.am:
9530 Install Vala bindings if vala is available
9532 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9534 * bindings/vala/gst-rtsp-server-0.10.deps:
9535 * bindings/vala/gst-rtsp-server-0.10.vapi:
9536 * bindings/vala/gst-rtsp-server.vapi:
9537 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
9538 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
9539 * bindings/vala/packages/gst-rtsp-server-0.10.files:
9540 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9541 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9542 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
9543 * bindings/vala/packages/gst-rtsp-server.deps:
9544 * bindings/vala/packages/gst-rtsp-server.excludes:
9545 * bindings/vala/packages/gst-rtsp-server.files:
9546 * bindings/vala/packages/gst-rtsp-server.gi:
9547 * bindings/vala/packages/gst-rtsp-server.metadata:
9548 * bindings/vala/packages/gst-rtsp-server.namespace:
9549 Regenerated Vala bindings
9551 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9553 * bindings/vala/gst-rtsp-server.vapi:
9554 * bindings/vala/packages/gst-rtsp-server.metadata:
9555 Fixed typo in included headers for vala bindings
9557 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9561 * pkgconfig/Makefile.am:
9562 * pkgconfig/gst-rtsp-server.pc.in:
9563 Added pkgconfig file
9565 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
9567 * bindings/vala/gst-rtsp-server.vapi:
9568 * bindings/vala/packages/gst-rtsp-server.excludes:
9569 * bindings/vala/packages/gst-rtsp-server.gi:
9570 * bindings/vala/packages/gst-rtsp-server.metadata:
9571 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
9573 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
9575 * bindings/vala/gst-rtsp-server.vapi:
9576 * bindings/vala/packages/gst-rtsp-server.deps:
9577 * bindings/vala/packages/gst-rtsp-server.files:
9578 * bindings/vala/packages/gst-rtsp-server.gi:
9579 * bindings/vala/packages/gst-rtsp-server.metadata:
9580 * bindings/vala/packages/gst-rtsp-server.namespace:
9583 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
9585 * gst/rtsp-server/rtsp-session.c:
9586 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
9588 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
9590 * examples/Makefile.am:
9591 * gst/rtsp-server/Makefile.am:
9592 Put GStreamer version in library name
9594 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9596 * examples/Makefile.am:
9597 * gst/rtsp-server/Makefile.am:
9598 Fix some issues to pass distcheck
9600 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9602 * gst/rtsp-server/rtsp-server.c:
9603 Added port property to GstRTSPServer class.
9605 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9610 * examples/Makefile.am:
9613 * gst/rtsp-server/Makefile.am:
9614 * gst/rtsp-server/rtsp-client.c:
9615 * gst/rtsp-server/rtsp-client.h:
9616 * gst/rtsp-server/rtsp-media.c:
9617 * gst/rtsp-server/rtsp-media.h:
9618 * gst/rtsp-server/rtsp-server.c:
9619 * gst/rtsp-server/rtsp-server.h:
9620 * gst/rtsp-server/rtsp-session-pool.c:
9621 * gst/rtsp-server/rtsp-session-pool.h:
9622 * gst/rtsp-server/rtsp-session.c:
9623 * gst/rtsp-server/rtsp-session.h:
9626 * src/rtsp-client.c:
9627 * src/rtsp-client.h:
9630 * src/rtsp-server.c:
9631 * src/rtsp-server.h:
9632 * src/rtsp-session-pool.c:
9633 * src/rtsp-session-pool.h:
9634 * src/rtsp-session.c:
9635 * src/rtsp-session.h:
9636 Split in library and example program
9638 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
9640 * src/rtsp-client.h:
9641 Removed obsolete variable
9643 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
9645 * src/rtsp-client.c:
9646 * src/rtsp-client.h:
9647 Removed pipeline variable GstRTSPClient, because it's only used in one function
9649 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9652 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
9654 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
9656 * src/rtsp-session.c:
9657 Initialize some more vars.
9659 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
9661 * src/rtsp-session.c:
9662 Initialize variable to avoid compiler warning.
9664 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
9667 Add a reasonable generic .gitignore