3 2016-03-24 Sebastian Dröge <slomo@coaxion.net>
8 2016-03-16 23:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
10 * gst/rtsp-server/rtsp-stream.c:
11 rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
12 This would get us NO_PREROLL in the bin again and break seeking.
13 Thanks to Carlos Rafael Giani for helping to debug this!
14 https://bugzilla.gnome.org/show_bug.cgi?id=740509
16 === release 1.7.91 ===
18 2016-03-15 12:26:13 +0200 Sebastian Dröge <sebastian@centricular.com>
24 * gst-rtsp-server.doap:
27 2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
29 * gst/rtsp-server/rtsp-stream.c:
30 rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
31 Without this, RECORD pipelines are broken because
32 a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
33 added later. Previously it was there earlier and due to NO_PREROLL caused the
34 pipeline to preroll immediately
35 b) the udpsrc for the pipeline is added later and never set to PLAYING state,
36 as the corresponding code previously was only for PLAY pipelines.
37 https://bugzilla.gnome.org/show_bug.cgi?id=763281
39 2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
41 * gst/rtsp-server/rtsp-stream.c:
42 rtsp-stream: Fix typo in the docstring
43 gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
45 2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
47 * gst/rtsp-server/rtsp-stream.c:
48 rtsp-stream: Disable multicast loopback for all our sockets
49 On Windows this is a receiver-side setting, on Linux a sender-side setting. As
50 we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
51 loopback setting on the socket... while udpsink does which unfortunately has
52 no effect here on Windows but on Linux.
53 https://bugzilla.gnome.org/show_bug.cgi?id=757488
55 2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
57 * tests/check/gst/stream.c:
58 stream tests: added new tests
59 Test a case when the address pool only contains multicast addresses
60 and the client is requesting unicast udp.
61 Added tests for multicast ports allocation.
62 https://bugzilla.gnome.org/show_bug.cgi?id=757488
64 2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
66 * gst/rtsp-server/rtsp-stream.c:
67 rtsp-stream: Only bind multicast sockets to ANY on Windows
68 On Linux it is still needed to bind to the multicast address
69 to filter out random other packets, while on Windows binding
70 to multicast addresses just fails.
72 2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
74 * gst/rtsp-server/rtsp-stream.c:
75 rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
76 Otherwise we fail to allocate UDP ports if the pool only contains multicast
77 addresses, which is something that used to work before. For unicast addresses
78 if the pool contains none, we just allocate them as if there is no pool at
80 https://bugzilla.gnome.org/show_bug.cgi?id=757488
82 2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
84 * gst/rtsp-server/rtsp-client.c:
85 * gst/rtsp-server/rtsp-stream.c:
86 rtsp-server: Fix indentation
88 2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
90 * gst/rtsp-server/rtsp-stream.c:
91 rtsp-stream: Don't bind the sockets to multicast addresses
92 This works on Linux but fails completely on Windows. You're supposed
93 to bind to ANY and then join the multicast group.
94 https://bugzilla.gnome.org/show_bug.cgi?id=757488
96 === release 1.7.90 ===
98 2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
104 * gst-rtsp-server.doap:
107 2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
110 Automatic update of common submodule
111 From b64f03f to 6f2d209
113 2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
115 * gst/rtsp-sink/gstrtspclientsink.c:
116 * tests/check/gst/rtspclientsink.c:
117 rtspsink: Fix some leaks in rtspclientsink and the unit test.
118 https://bugzilla.gnome.org/show_bug.cgi?id=762525
120 2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
122 * tests/check/gst/media.c:
123 * tests/check/gst/rtspclientsink.c:
124 * tests/check/gst/rtspserver.c:
125 * tests/check/gst/stream.c:
126 tests: unit test fixes
127 Removed port allocation test from the media suite.
128 The port allocation failure is now in the stream suite.
130 Make sure that the media is suspended after the DESCRIBE request
131 before reconfiguring the UDP sinks.
133 In the RECORD case we have to set async property to false
134 for the appsink element in the test in order to make sure
135 that the media pipeline doesn't hang in start_preroll().
136 https://bugzilla.gnome.org/show_bug.cgi?id=757488
138 2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
140 * gst/rtsp-server/rtsp-client.c:
141 * gst/rtsp-server/rtsp-stream.c:
142 * gst/rtsp-server/rtsp-stream.h:
143 rtsp-stream: postpone UDP socket allocation until SETUP
144 Postpone the allocation of the UDP sockets until we know
145 what transport has been chosen by the client.
146 Both unicast and multicast UDP sources are created in one
148 https://bugzilla.gnome.org/show_bug.cgi?id=757488
150 2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
152 * gst/rtsp-server/rtsp-stream.c:
153 rtsp-stream: postpone the creation of the UDP sources
154 Code refactoring: allocate the UDP ports after the sender and
155 the reciver parts have been created.
156 We postpone the creation of the UDP sources until the UDP
157 ports have been allocated.
158 https://bugzilla.gnome.org/show_bug.cgi?id=757488
160 2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
162 * gst/rtsp-server/rtsp-stream.c:
163 rtsp-stream: added function for setting UDP sources to PLAYING state
164 Code refactoring: Introduced a function for setting UDP sources
166 https://bugzilla.gnome.org/show_bug.cgi?id=757488
168 2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
170 * gst/rtsp-server/rtsp-stream.c:
171 rtsp-stream: added function for creating and configuring UDP sources
172 Code refactoring: create and configure UDP sources in a separate function.
173 https://bugzilla.gnome.org/show_bug.cgi?id=757488
175 2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
177 * gst/rtsp-server/rtsp-stream.c:
178 rtsp-stream: added function for RTP/RTCP socket configuration
179 Code refactoring: configure RTP and RTCP sockets for UDP sinks
180 in a separate function.
181 https://bugzilla.gnome.org/show_bug.cgi?id=757488
183 2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
185 * gst/rtsp-server/rtsp-stream.c:
186 rtsp-stream: added function for creating and configuring UDP sinks
187 Code refactoring: create and configure UDP sinks in a separate function.
188 https://bugzilla.gnome.org/show_bug.cgi?id=757488
190 2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
192 * gst/rtsp-server/rtsp-stream.c:
193 rtsp-stream: added helper function for creating the sender/receiver parts
194 Code refactoring: introduced helper function for creating
195 the receiver and the sender parts of the streaming pipeline.
196 https://bugzilla.gnome.org/show_bug.cgi?id=757488
198 2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
203 === release 1.7.2 ===
205 2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
211 * gst-rtsp-server.doap:
214 2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
216 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
217 uninstalled.pc: add support for non libtool build systems
218 Currently the .la path is provided which requires to use libtool as
219 mentioned in the GStreamer manual section-helloworld-compilerun.html.
220 It is fine as long as the application is built using libtool.
221 So currently it is not possible to compile a GStreamer application
222 within gst-uninstalled with CMake or other build system different
224 This patch allows to do the following in gst-uninstalled env:
225 gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
226 gstreamer-rtsp-server-1.0)
227 Previously it required to prepend libtool --mode=link
228 https://bugzilla.gnome.org/show_bug.cgi?id=720778
230 2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
232 * gst/rtsp-sink/gstrtspclientsink.c:
233 rtspclientsink: remove check for impossible condition
234 Goto error label checks stream to see if it needs to be unreferenced before
235 returning, but this goto jumps happens before the stream is ever set, so it
236 will always be NULL in this error label.
239 2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
241 * gst/rtsp-sink/gstrtspclientsink.c:
242 rtspclientsink: clean switch statements
243 Coverity demands for fallthrough statements to be clearly commented,
244 to distinguish from accidental fall throughs. And it also needs all
245 cases to finish with a break, even if the break is never going to be
246 executed like in the case of a continue jump.
250 2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
252 * tests/check/Makefile.am:
253 tests: extend the AM_TESTS_ENVIRONMENT from check.mak
254 To get the CK_DEFAULT_TIMEOUT defined for all tests
255 Also removes a 120 seconds timeout that was set as default
256 explicitly in this module
257 https://bugzilla.gnome.org/show_bug.cgi?id=761472
259 2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
263 Automatic update of common submodule
264 From 86e4663 to b64f03f
266 2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
268 * gst/rtsp-server/rtsp-media.c:
269 rtsp-media: fix state_lock not locked again when preroll fails
270 https://bugzilla.gnome.org/show_bug.cgi?id=761399
272 2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
275 configure: Move plugin specific flags below all the others
276 They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
277 -no-undefined. And -no-undefined is required on Windows to build DLLs.
279 2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
281 * gst/rtsp-sink/gstrtspclientsink.c:
282 rtspclientsink: Simplify slightly using new -base API
283 Use the new Mikey and SDP API in the base plugins libs
284 to simplify some code.
285 https://bugzilla.gnome.org/show_bug.cgi?id=758180
287 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
292 * gst/rtsp-sink/Makefile.am:
293 * gst/rtsp-sink/gstrtspclientsink.c:
294 * gst/rtsp-sink/gstrtspclientsink.h:
295 * gst/rtsp-sink/plugin.c:
296 * tests/check/Makefile.am:
297 * tests/check/gst/rtspclientsink.c:
298 rtspsink: Add rtspclientsink element
299 Add an rtspclientsink element that accepts streams for which
300 there is a registered payloader and sends them to
301 an RTSP server using RECORD.
302 Sending is synchronised to the pipeline clock. Payload-types
303 are automatically selected. The 'new-payloader' signal is fired
304 for custom configuration of payloaders when they are created.
305 Can now stream a movie like this:
307 ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
308 decodebin name=depay1 ! audioconvert ! autoaudiosink )"
310 gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
311 queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
312 https://bugzilla.gnome.org/show_bug.cgi?id=758180
314 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
316 * gst/rtsp-server/rtsp-stream.c:
317 * gst/rtsp-server/rtsp-stream.h:
318 rtsp-stream: Add functions for using rtsp-stream from the client
319 Add a boolean to indicate that the rtsp-stream is running on the
320 'client' side of an RTSP connection, for sending streams via
321 RECORD. In that case, the roles of the client/server ports
322 in transport setup are swapped.
323 https://bugzilla.gnome.org/show_bug.cgi?id=758180
325 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
327 * gst/rtsp-server/rtsp-sdp.c:
328 * gst/rtsp-server/rtsp-sdp.h:
329 rtsp-sdp: Add gst_rtsp_sdp_from_stream()
330 A new function that adds info from a GstRTSPStream into an SDP message.
331 https://bugzilla.gnome.org/show_bug.cgi?id=758180
333 2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
335 * gst/rtsp-server/rtsp-media.c:
336 rtsp-media: Fix mutex beeing unlocked while they should be locked
337 https://bugzilla.gnome.org/show_bug.cgi?id=761226
339 2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
341 * gst/rtsp-server/rtsp-media-factory.c:
342 rtsp-media-factory: add missing break in "clock" property setter
345 2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
347 * gst/rtsp-server/rtsp-stream.c:
348 rtsp-stream: fixed assert during update transport
349 When RTSP server trying update transport during multicast, it throws an
350 assert. The assert is thrown because it is trying to get the parent of
351 an non-existing funnel element.
352 https://bugzilla.gnome.org/show_bug.cgi?id=760150
354 2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
356 * gst/rtsp-server/rtsp-permissions.h:
357 * gst/rtsp-server/rtsp-thread-pool.h:
358 * gst/rtsp-server/rtsp-token.h:
359 docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
360 gtk-doc can handle static inline functions just fine these days,
361 there's no need for this stuff any more.
363 2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
365 * gst/rtsp-server/rtsp-media.c:
366 * gst/rtsp-server/rtsp-sdp.c:
367 sdp: replace duplicated codes to call new base sdp apis
368 https://bugzilla.gnome.org/show_bug.cgi?id=745880
370 2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
372 * examples/test-netclock.c:
373 test-netclock: Use the new API to configure a clock directly
375 2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
377 * gst/rtsp-server/rtsp-media-factory.c:
378 * gst/rtsp-server/rtsp-media-factory.h:
379 * gst/rtsp-server/rtsp-media.c:
380 * gst/rtsp-server/rtsp-media.h:
381 rtsp-media: Add API to directly configure a clock on the media pipelines
383 2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
385 * gst/rtsp-server/rtsp-media.c:
386 rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
388 2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
390 * gst/rtsp-server/rtsp-media-factory.c:
391 rtsp-media-factory: Add FIXME for 2.0
393 2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
395 * gst/rtsp-server/rtsp-stream.c:
396 rtsp-stream: Fix indentation
398 2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
400 * gst/rtsp-server/rtsp-media.c:
401 rtsp-media: Do not prepare media after media times out
402 Deferred calls to start_prepare() can be deferred past the point until
403 which wait_preroll() and by proxy gst_rtsp_media_get_status() is
404 prepared to wait. Previously there was no lock and no check for this
405 situation. This meant that a media could be prepared and unprepared
406 simultaneously by two different threads. Now a lock is in place and a
407 suitable check is done.
408 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
410 2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
412 * gst/rtsp-server/rtsp-client.c:
413 * gst/rtsp-server/rtsp-media-factory.c:
414 * gst/rtsp-server/rtsp-media-factory.h:
415 * gst/rtsp-server/rtsp-media.c:
416 * gst/rtsp-server/rtsp-media.h:
417 rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
418 Without TEARDOWN it might be desireable to keep the media running and continue
419 sending data to the client, even if the RTSP connection itself is
421 Only do this for session medias that have only UDP transports. If there's at
422 least on TCP transport, it will stop working and cause problems when the
423 connection is disconnected.
424 https://bugzilla.gnome.org/show_bug.cgi?id=758999
426 2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
431 === release 1.7.1 ===
433 2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
439 * gst-rtsp-server.doap:
442 2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
445 configure: Make -Bsymbolic check work with clang.
446 Update the -Bsymbolic check with the version glib has. This version
448 https://bugzilla.gnome.org/show_bug.cgi?id=759713
450 2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
452 * gst/rtsp-server/rtsp-session-pool.c:
453 rtsp-session-pool: Avoid dollar sign ($) in session ids
454 Live555 in VLC strips off dollar signs and then gets very confused,
455 we don't loose too much entropy by just skipping it.
457 2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
459 * gst/rtsp-server/rtsp-address-pool.h:
460 * gst/rtsp-server/rtsp-auth.h:
461 * gst/rtsp-server/rtsp-client.h:
462 * gst/rtsp-server/rtsp-media-factory-uri.h:
463 * gst/rtsp-server/rtsp-media-factory.h:
464 * gst/rtsp-server/rtsp-media.h:
465 * gst/rtsp-server/rtsp-mount-points.h:
466 * gst/rtsp-server/rtsp-permissions.h:
467 * gst/rtsp-server/rtsp-server.h:
468 * gst/rtsp-server/rtsp-session-media.h:
469 * gst/rtsp-server/rtsp-session-pool.h:
470 * gst/rtsp-server/rtsp-session.h:
471 * gst/rtsp-server/rtsp-stream-transport.h:
472 * gst/rtsp-server/rtsp-stream.h:
473 * gst/rtsp-server/rtsp-thread-pool.h:
474 * gst/rtsp-server/rtsp-token.h:
475 rtsp-server: Add g_autoptr() support to all types
476 https://bugzilla.gnome.org/show_bug.cgi?id=754464
478 2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
480 * gst/rtsp-server/rtsp-stream.c:
481 rtsp-stream: fixed valgrind error
482 Fixed the valgrind error in unit test. The UDP source created during
483 gst_rtsp_stream_join_bin() was not released while destroying the rtp
485 https://bugzilla.gnome.org/show_bug.cgi?id=759010
487 2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
491 Automatic update of common submodule
492 From b319909 to 86e4663
494 2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
496 * gst/rtsp-server/rtsp-client.c:
497 rtsp-client: suspend media during setup request
498 SETUP request from clients needs to suspend the media to clear the
499 prerolled buffers. Otherwise it will not affect the prerolled buffer
500 and the prerolled buffers will be incorrect (for example block-size
501 from setup request will not affect the prerolled buffer unless the
503 https://bugzilla.gnome.org/show_bug.cgi?id=758268
505 2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
507 * gst/rtsp-server/rtsp-stream.c:
508 rtsp-stream: create stream pipeline based on transport
509 Based on the protocol, create the rtsp stream pipeline. If only TCP or
510 only UDP is set as the transport protocol, it will not add the extra tee
511 or queue element to the pipeline. Both these elements will be added, if
512 it supports both TCP and UDP protocols. This improves the pipeline
513 performance when one protocol is present.
514 https://bugzilla.gnome.org/show_bug.cgi?id=758179
516 2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
518 * gst/rtsp-server/rtsp-stream.c:
519 rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
520 Adding them when not needed will start some logic inside rtpbin that might be
521 problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
522 would start up a rtpjitterbuffer and behave in weird ways.
523 We still set up the UDP sources for RTP receiving for a sender media to be
524 able to receive any packets sent by the client for NAT traversal. They will
525 all go to a fakesink though.
526 Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
527 NO_PREROLL, which will cause deadlocks when seeking the media as it will never
528 receive ASYNC_DONE after a seek.
529 https://bugzilla.gnome.org/show_bug.cgi?id=758319
531 2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
533 * gst/rtsp-server/rtsp-stream.c:
534 rtsp-stream: Disable multicast loopback for the multicast udp sources too
535 On POSIX this setting is for sender sockets, on Windows for receiver sockets.
536 Previously we were only setting this for sender sockets, which caused looped
537 back packets to be received on Windows if a multicast transport was used.
539 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
541 * examples/test-record-auth.c:
542 * examples/test-record.c:
543 examples: Actually use the provided port in the record examples
545 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
547 * examples/test-record-auth.c:
548 test-record-auth: Add the option to build in TLS support
550 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
552 * examples/test-auth.c:
553 test-auth: Use an 'anonymous' user for unauthenticated default
554 There's a comment on one of the resources that 'user' and 'admin'
555 shouldn't even be able to see it, but they can if the default
556 token is 'admin2', since that gives them access anyway.
558 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
560 * examples/.gitignore:
561 * examples/Makefile.am:
562 * examples/test-record-auth.c:
563 Add test-record-auth example
565 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
567 * gst/rtsp-server/rtsp-client.c:
568 * tests/check/gst/client.c:
569 rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
571 2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
573 * gst/rtsp-server/rtsp-server.c:
574 rtsp-server: Change the logic so we don't pop a NULL context
575 When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
576 will sometimes fail. This call is made before any context is pushed
577 resulting in an attempt to pop a NULL context.
578 https://bugzilla.gnome.org/show_bug.cgi?id=757949
580 2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
582 * tests/check/gst/rtspserver.c:
583 rtspserver: Add udp-mcast transport SETUP test
584 Refactor utility functions in the test file so they can handle
585 more than UDP and TCP as lower transport.
586 https://bugzilla.gnome.org/show_bug.cgi?id=756969
588 2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
590 * gst/rtsp-server/rtsp-stream.c:
591 rtsp-stream: Always unref return value of gst_object_get_parent()
592 Fixes a leak of a GstBin in the udp-mcast case.
593 https://bugzilla.gnome.org/show_bug.cgi?id=756968
595 2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
598 Automatic update of common submodule
599 From b99800a to b319909
601 2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
604 Use new GST_ENABLE_EXTRA_CHECKS #define
605 https://bugzilla.gnome.org/show_bug.cgi?id=756870
607 2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
610 Automatic update of common submodule
611 From 6babecd to b99800a
613 2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
616 Update GLib dependency to 2.40.0
618 2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
620 * examples/test-mp4.c:
621 * gst/rtsp-server/rtsp-stream.c:
622 stream: listen to sender ssrc signals
623 https://bugzilla.gnome.org/show_bug.cgi?id=746747
625 2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
628 common: update for new suppression
629 Makes check-valgrind pass with glib 2.46
631 2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
633 * gst/rtsp-server/rtsp-media.c:
634 rtsp-media: Take reference to media that will be prepared
635 default_prepare() takes a transfer-none reference GstRTSPMedia object.
636 Later on a g_idle_source_new() is created and a pointer to the media
637 object is passed as user data. If the media is freed before the idle
638 source is dispatched the media object pointer is invalid, but the idle
639 source callback expects it to still be valid. To fix this a reference to
640 the media object is taken when registering the source callback function
641 and a corresponding release of the reference is done when the souce is
643 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
645 2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
647 * examples/test-launch.c:
648 * examples/test-mp4.c:
649 * examples/test-ogg.c:
650 * examples/test-record.c:
651 * examples/test-uri.c:
652 rtsp-server: Fix memory leaks when context parse fails
653 When g_option_context_parse fails, context and error variables are not getting free'd
654 which results in memory leaks. Free'ing the same.
655 And replacing g_error_free with g_clear_error, which checks if the error being passed
656 is not NULL and sets the variable to NULL on free'ing.
657 https://bugzilla.gnome.org/show_bug.cgi?id=753863
659 2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
664 === release 1.6.0 ===
666 2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
672 * gst-rtsp-server.doap:
675 === release 1.5.91 ===
677 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
683 * gst-rtsp-server.doap:
686 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
688 * docs/libs/gst-rtsp-server-sections.txt:
689 * gst/rtsp-server/rtsp-stream.c:
690 stream: fix docs for recently-added get/set_buffer_size API
691 https://bugzilla.gnome.org/show_bug.cgi?id=749095
693 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
695 * gst/rtsp-server/rtsp-media.c:
696 rtsp-media: Don't crash on encrypted RTX SDP
697 In parse_keymgmt(), don't mutate the input string that's been passed
698 as const, especially since we might need the original value again if
699 the same key info applies to multiple streams (RTX, for example).
700 https://bugzilla.gnome.org/show_bug.cgi?id=754753
702 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
704 * examples/test-mp4.c:
705 test-mp4: Support filenames with spaces in them. Error out on too few arguments
707 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
709 * examples/test-record.c:
710 test-record: Check parameter count and print out help
711 If no launch pipeline was supplied, print out some help
713 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
715 * gst/rtsp-server/rtsp-media.c:
716 * gst/rtsp-server/rtsp-stream.c:
717 * gst/rtsp-server/rtsp-stream.h:
718 rtsp-stream: Implement UDP buffer size setting.
719 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
721 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
722 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
724 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
726 * gst/rtsp-server/rtsp-media.h:
727 rtsp-media: Fix small typo causing gtk-doc to complain
729 === release 1.5.90 ===
731 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
737 * gst-rtsp-server.doap:
740 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
742 * gst/rtsp-server/rtsp-media-factory.c:
743 media-factory: get port number through gst_rtsp_url_get_port
744 https://bugzilla.gnome.org/show_bug.cgi?id=753473
746 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
748 * tests/check/gst/media.c:
749 media-test: Removing unnecessary assertion
750 https://bugzilla.gnome.org/show_bug.cgi?id=753385
752 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
754 * gst/rtsp-server/rtsp-server.c:
755 Document that source keeps a ref on server until it's destroyed
756 https://bugzilla.gnome.org/show_bug.cgi?id=749227
758 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
760 * tests/check/gst/media.c:
761 media-test: Test for multiple dynamic payload
762 https://bugzilla.gnome.org/show_bug.cgi?id=753385
764 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
766 * gst/rtsp-server/rtsp-media.c:
767 media: Only add fakesink once per pipeline
768 The intention is to prevent going PLAYING state before pads are created.
769 If there was mutilple dynamic payload, it would leak few fakesink and
770 actually prevent from ever reaching playing state.
771 https://bugzilla.gnome.org/show_bug.cgi?id=753385
773 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
775 * gst/rtsp-server/rtsp-media.c:
776 Revert "rtsp-media: Only add 1 fakesink per pipeline"
777 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
779 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
781 * gst/rtsp-server/rtsp-media.c:
782 rtsp-media: Only add 1 fakesink per pipeline
783 There should be only one fakesink per pipeline, not per dynpay. This
784 would lead to element naming clash.
786 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
788 * gst/rtsp-server/rtsp-media.c:
789 rtsp-media: assertion error due to wrong condition check
790 In media to caps function, reserved_keys array is being used for variable i,
791 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
792 changed it to variable j
793 https://bugzilla.gnome.org/show_bug.cgi?id=753009
795 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
797 * gst/rtsp-server/rtsp-media.c:
798 rtsp-media: Strip keys from the fmtp that we use internally in our caps
799 Skip keys from the fmtp, which we already use ourselves for the
800 caps. Some software is adding random things like clock-rate into
801 the fmtp, and we would otherwise here set a string-typed clock-rate
802 in the caps... and thus fail to create valid RTP caps
803 https://bugzilla.gnome.org/show_bug.cgi?id=753009
805 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
807 * gst/rtsp-server/rtsp-thread-pool.c:
808 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
809 https://bugzilla.gnome.org/show_bug.cgi?id=752640
811 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
814 Automatic update of common submodule
815 From f74b2df to 9aed1d7
817 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
822 === release 1.5.2 ===
824 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
830 * gst-rtsp-server.doap:
833 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
835 * gst/rtsp-server/rtsp-client.c:
836 * gst/rtsp-server/rtsp-client.h:
837 * tests/check/gst/client.c:
838 rtsp-client: allow application to decide what requirements are supported
839 Add "check-requirements" signal and vfunc to allow application
840 (and subclasses) to check the requirements.
841 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
842 https://bugzilla.gnome.org/show_bug.cgi?id=749417
844 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
847 Automatic update of common submodule
848 From 6015d26 to f74b2df
850 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
852 * gst/rtsp-server/rtsp-media.c:
853 rtsp-media: Always use real payloader when creating streams
854 A bin that contains the real payloader might be used as payloader. In this
855 case we have to get the real payloader for the various properties it provides.
856 Example use cases for this are bins that payload some media and then have
857 additional elements that add metadata or RTP extension headers to the stream.
858 https://bugzilla.gnome.org/show_bug.cgi?id=750800
860 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
862 * examples/test-netclock-client.c:
863 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
865 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
867 * examples/test-netclock-client.c:
868 * examples/test-netclock.c:
869 test-netclock: Use new ntp-time-source property on rtpbin
870 Select the clock time to be used as NTP time source. This allows proper
871 synchronization between receivers, independent of sharing base times, and just
872 requires them to use the same clock.
874 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
876 * examples/test-netclock-client.c:
877 * examples/test-netclock.c:
878 test-netclock: Setting the same base time on sender and receiver is not necessary
879 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
881 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
883 * gst/rtsp-server/rtsp-stream.c:
884 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
885 https://bugzilla.gnome.org/show_bug.cgi?id=750764
887 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
889 * docs/libs/gst-rtsp-server.types:
890 docs: add missing types
891 https://bugzilla.gnome.org/show_bug.cgi?id=750764
893 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
895 * docs/libs/gst-rtsp-server-sections.txt:
896 docs: add missing apis
897 https://bugzilla.gnome.org/show_bug.cgi?id=750764
899 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
901 * examples/test-netclock-client.c:
902 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
904 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
906 * docs/libs/gst-rtsp-server-sections.txt:
907 * gst/rtsp-server/rtsp-auth.c:
908 * gst/rtsp-server/rtsp-auth.h:
909 GstRTSPAuth: Add client certificate authentication support
910 https://bugzilla.gnome.org/show_bug.cgi?id=750471
912 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
914 * examples/test-netclock-client.c:
915 test-netclock-client: Use new GstClock API to wait for clock synchronization
917 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
919 * examples/test-netclock-client.c:
920 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
921 A mainloop is needed to get glimagesink to display something on OSX, and
922 the source-setup signal just makes things a little bit easier.
924 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
927 Automatic update of common submodule
928 From d9a3353 to 6015d26
930 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
933 Automatic update of common submodule
934 From d37af32 to d9a3353
936 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
939 Automatic update of common submodule
940 From 21ba2e5 to d37af32
942 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
945 Automatic update of common submodule
946 From c408583 to 21ba2e5
948 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
950 * docs/libs/Makefile.am:
951 docs: remove variables that we define in the snippet from common
952 This is syncing our Makefile.am with upstream gtkdoc.
954 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
957 Automatic update of common submodule
958 From 44a3517 to c408583
960 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
965 === release 1.5.1 ===
967 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
973 * gst-rtsp-server.doap:
976 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
978 * gst/rtsp-server/rtsp-client.c:
979 rtsp-client: No flush during Teardown.
980 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
981 backlog is empty it can happen that just a part of a message will be
982 sent and rest is in backlog queue. If then flush during teardown
983 just a part of message will be sent.This can lead to client miss
984 teardown response since it expect to get the last part of message.
985 The flushing during teardown was introduced to fix a deadlock that now
986 is fixed more generally in handle_request by temporary setting backlog
988 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
990 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
992 * tests/check/Makefile.am:
993 tests: Use AM_TESTS_ENVIRONMENT
994 Needed by the new automake test runner and the
995 current version of the common submodule.
997 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
999 * gst/rtsp-server/rtsp-media.h:
1000 * gst/rtsp-server/rtsp-stream.h:
1001 rtsp-server: Use single-include rtsp header to make sure we get all definitions
1003 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
1005 * gst/rtsp-server/rtsp-media.c:
1006 rtsp-media: Mark some more functions static
1008 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
1010 * gst/rtsp-server/rtsp-media.c:
1011 rtsp-media: Only unblock the media in suspend() when actually changing the state
1012 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
1014 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
1016 * examples/test-video-rtx.c:
1017 examples: Use AVPF profile for the RTX example
1019 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
1021 * gst/rtsp-server/rtsp-sdp.c:
1022 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
1024 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1026 * gst/rtsp-server/rtsp-stream.c:
1027 rtsp-stream: get valid clock-rate from last-sample
1028 clock-rate in last-sample's caps is integer, not unsigned.
1029 To get this value properly, variable needs to be type-casted to int.
1030 https://bugzilla.gnome.org/show_bug.cgi?id=747614
1032 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
1036 autogen.sh: only run autopoint if gettext requested in configure.ac
1037 Not just because there happens to be a po directory.
1038 https://bugzilla.gnome.org/show_bug.cgi?id=748058
1040 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
1043 Revert "configure.ac: uncomment gettext version setup"
1044 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
1045 We don't need a gettext setup here and there's no po
1046 directory either, so no reason why autopoint would be
1047 run in the first place.
1048 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
1050 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
1052 * examples/test-multicast.c:
1053 * examples/test-multicast2.c:
1054 * examples/test-sdp.c:
1055 * examples/test-video-rtx.c:
1056 * examples/test-video.c:
1057 * tests/test-cleanup.c:
1058 * tests/test-reuse.c:
1059 Fix timeout function signatures across tests and examples
1061 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
1063 * tests/check/Makefile.am:
1064 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
1065 Make sure the test environment is set up.
1066 https://bugzilla.gnome.org//show_bug.cgi?id=747624
1068 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
1071 configure: bump automake requirement to 1.14 and autoconf to 2.69
1072 This is only required for builds from git, people can still
1073 build tarballs if they only have older autotools.
1074 https://bugzilla.gnome.org//show_bug.cgi?id=747624
1076 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
1079 configure.ac: uncomment gettext version setup
1080 Fixes autogen.sh. It would run autopoint, which would complain
1081 that it could not find the gettext version in configure.ac.
1082 https://bugzilla.gnome.org/show_bug.cgi?id=748058
1084 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1086 * examples/test-video-rtx.c:
1087 test-video-rtx: set exact payload type to PCMA payloader
1088 Setting wrong payload type causes failure to do retransmission through audio stream
1089 https://bugzilla.gnome.org/show_bug.cgi?id=747839
1091 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1093 * gst/rtsp-server/rtsp-media.c:
1094 * gst/rtsp-server/rtsp-stream.c:
1095 * gst/rtsp-server/rtsp-stream.h:
1096 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
1097 Because of duplicated g_signal_connect for request-aux-sender signal,
1098 wrong stream pointer is passed to the signal handler.
1099 Instead of passing each stream, pass stream array and get the relevant stream.
1100 https://bugzilla.gnome.org/show_bug.cgi?id=747839
1102 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
1106 Update autogen.sh to latest version from common
1107 Fixes build after aclocal_check etc. helpers have been removed.
1109 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
1112 Automatic update of common submodule
1113 From bc76a8b to c8fb372
1115 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
1117 * gst/rtsp-server/rtsp-stream.c:
1118 rtsp-stream: Limit the queues to 1 buffer
1119 We only need them to be able to pre-roll, queueing up more data here
1120 is only going to harm latency and memory usage.
1122 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
1124 * gst/rtsp-server/rtsp-stream.c:
1125 rtsp-stream: Update comment and ASCII art to the latest code
1126 We have a queue in front of the udpsink too to prevent the pipeline from
1129 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
1131 * gst/rtsp-server/rtsp-stream.c:
1132 rtsp-media: Properly return first rtptime
1133 Instead we where returning first GstBuffer timestamp. This would result
1134 in clock skew and unwanted behaviour in RTSP playback.
1135 https://bugzilla.gnome.org/show_bug.cgi?id=746479
1137 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
1139 * gst/rtsp-server/rtsp-stream.c:
1140 rtsp-stream: Don't leave buffer mapped
1141 If the seq is NULL, the RTP buffer was left mapped. We should always
1144 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
1149 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
1151 * gst/rtsp-server/rtsp-media-factory.c:
1152 * tests/check/gst/client.c:
1153 Fix double semicolons
1155 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
1157 * gst/rtsp-server/rtsp-stream.c:
1158 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
1159 This gives more accurate values than asking the payloader. There might be
1160 queueing happening between the payloader and the sink.
1161 https://bugzilla.gnome.org/show_bug.cgi?id=745704
1163 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
1165 * gst/rtsp-server/rtsp-media.c:
1166 rtsp-media: Don't seek for PLAY if the position will not change
1167 https://bugzilla.gnome.org/show_bug.cgi?id=745704
1169 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
1171 * gst/rtsp-server/rtsp-media.c:
1172 rtsp-media: Don't include payload type in the caps for framesize
1173 When the sdp media attribute framesize are converted to caps
1174 the <payload> should not be included.
1175 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
1176 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
1178 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
1180 * gst/rtsp-server/rtsp-sdp.c:
1181 rtsp-sdp: add payload type to the sdp framesize attribute
1182 The sdp framesize attribute is desribed in RFC6064. It is specified
1183 for payloading of H263 and has the following form
1184 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
1185 should be added to the caps in a payloader and the <payload type> should
1186 be added by the rtsp-server.
1187 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
1189 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
1191 * examples/test-uri.c:
1192 examples: test-uri: fix tainted variable
1193 Insignificant but this keeps Coverity happy.
1196 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
1198 * examples/.gitignore:
1199 * examples/Makefile.am:
1200 * examples/test-netclock-client.c:
1201 * examples/test-netclock.c:
1202 examples: Add a simple example of network synch for live streams.
1203 An example server and client that works for synchronising live streams
1204 only - as it can't support pause/play.
1206 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
1208 * gst/rtsp-server/rtsp-media-factory.c:
1209 * gst/rtsp-server/rtsp-media-factory.h:
1210 rtsp-media-factory: Add functions to set/get the media gtype
1211 Allow specifying the GType of a GstRtspMedia subclass to create
1212 as a simpler way to get the factory to create a custom
1213 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
1215 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
1217 * gst/rtsp-server/rtsp-media.c:
1218 rtsp-media: fix double unlock in _get_buffer_size()
1219 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
1220 because of double g_mutex_unlock () usage.
1221 https://bugzilla.gnome.org/show_bug.cgi?id=745434
1223 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
1225 * gst/rtsp-server/rtsp-session-pool.c:
1226 * gst/rtsp-server/rtsp-session.c:
1227 * gst/rtsp-server/rtsp-session.h:
1228 rtsp-session: Use monotonic time for RTSP session timeout
1229 Changed RTSP session timeout handling to monotonic time
1230 and deprecating the API for current system time.
1231 This fixes timeouts when the system time changes.
1232 https://bugzilla.gnome.org/show_bug.cgi?id=743346
1234 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
1236 * gst/rtsp-server/rtsp-client.c:
1237 * gst/rtsp-server/rtsp-media.c:
1238 rtsp-client: Only error out in PLAY if seeking actually failed
1239 If the media was just not seekable, we continue from whatever position we are
1240 and let the client decide if that is what is wanted or not.
1241 Only if the actual seek failed, we can't really recover and should error out.
1243 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
1245 * gst/rtsp-server/rtsp-stream.c:
1246 rtsp-stream: Add necessary queues between tee and multiudpsink
1247 https://bugzilla.gnome.org/show_bug.cgi?id=744379
1249 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
1251 * gst/rtsp-server/rtsp-client.c:
1252 * gst/rtsp-server/rtsp-media.c:
1253 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
1254 Instead error out properly the same way as if the SEEKING query already
1257 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
1259 * gst/rtsp-server/rtsp-stream.h:
1260 rtsp-stream: minor code formatting fix
1262 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
1264 * gst/rtsp-server/rtsp-media.c:
1265 rtsp-media: fix logic for collect_streams
1266 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
1267 all streams it knows if it got any, and can check if the transport mode is OK.
1270 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
1272 * gst/rtsp-server/rtsp-media.c:
1273 rtsp-media: Don't set the transport mode based on what elements we find
1274 Just print a warning if the one that was set before disagrees with what
1275 elements we found. It must already be set to something before as this
1276 function is called after we received the SDP from ANNOUNCE in RECORD mode,
1277 and we would reject ANNOUNCE if the RECORD flag was not set.
1279 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
1281 * tests/check/gst/rtspserver.c:
1282 tests: rtspserver: rename shadowed variable
1283 We have two different 'sink' variables here,
1284 rename one of them for clarity.
1286 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
1288 * gst/rtsp-server/rtsp-client.c:
1289 rtsp-client: fix awkward if clause
1291 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
1293 * examples/test-uri.c:
1294 examples: test-uri: improve uri argument handling and accept file names
1295 Print an error if the argument passed is not a URI and can't
1296 be converted into one, or no arguments have been provided.
1298 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
1300 * examples/test-uri.c:
1301 examples: test-uri: don't remove mount point after 10 seconds
1302 It's very irritating when trying to test stuff repeatedly
1303 and serves no real purpose other than showing that it can
1306 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
1308 * examples/.gitignore:
1309 examples: add new test-record to .gitignore
1311 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
1313 * examples/test-record.c:
1314 * gst/rtsp-server/rtsp-client.c:
1315 * gst/rtsp-server/rtsp-media-factory.c:
1316 * gst/rtsp-server/rtsp-media-factory.h:
1317 * gst/rtsp-server/rtsp-media.c:
1318 * gst/rtsp-server/rtsp-media.h:
1319 * tests/check/gst/rtspserver.c:
1320 rtsp-media: Use flags to distinguish between PLAY and RECORD media
1322 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
1324 * examples/test-record.c:
1325 test-record: Set latency for playback-style example to 2s instead of 200ms
1327 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
1329 * tests/check/gst/rtspserver.c:
1330 tests: add some unit tests for ANNOUNCE and RECORD
1331 https://bugzilla.gnome.org/show_bug.cgi?id=743175
1333 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
1335 * gst/rtsp-server/rtsp-client.c:
1336 rtsp-client: fix a couple of leaks in handle_announce
1338 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
1340 * gst/rtsp-server/rtsp-media-factory.c:
1341 * gst/rtsp-server/rtsp-media-factory.h:
1342 * gst/rtsp-server/rtsp-media.c:
1343 * gst/rtsp-server/rtsp-media.h:
1344 rtsp-media: Expose latency setting for setting the rtpbin latency
1346 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
1348 * examples/test-record.c:
1349 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
1351 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
1353 * gst/rtsp-server/rtsp-stream.c:
1354 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
1356 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
1358 * examples/Makefile.am:
1359 * examples/test-record.c:
1360 * gst/rtsp-server/rtsp-client.c:
1361 * gst/rtsp-server/rtsp-client.h:
1362 * gst/rtsp-server/rtsp-media-factory.c:
1363 * gst/rtsp-server/rtsp-media-factory.h:
1364 * gst/rtsp-server/rtsp-media.c:
1365 * gst/rtsp-server/rtsp-media.h:
1366 * gst/rtsp-server/rtsp-session-media.c:
1367 * gst/rtsp-server/rtsp-stream.c:
1368 * gst/rtsp-server/rtsp-stream.h:
1369 Add initial support for RECORD
1370 We currently only support media that is RECORD or PLAY only, not both at once.
1371 https://bugzilla.gnome.org/show_bug.cgi?id=743175
1373 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
1375 * gst/rtsp-server/rtsp-stream.c:
1376 rtsp-stream: RTCP and RTP transport cache cookies seperated
1377 RTCP packets were not sent because the same tr_cache_cookie was used for
1378 both RTP and RTCP. So only one of the tr_cache lists were populated
1379 depending on which one was sent first. If the tr_cache list is not
1380 populated then no packets can be sent. Most often this happened to be
1381 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
1382 resulted in both the tr_cache_lists to be populated regardless of which
1384 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
1386 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
1388 * gst/rtsp-server/rtsp-stream.c:
1389 rtsp-stream: fix false compiler warning
1390 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
1392 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
1394 * gst/rtsp-server/rtsp-client.c:
1395 rtsp-client: log interleaved data received
1397 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
1399 * gst/rtsp-server/rtsp-client.c:
1400 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
1402 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
1404 * gst/rtsp-server/rtsp-client.c:
1405 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
1407 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
1409 * gst/rtsp-server/rtsp-client.c:
1410 rtsp-client: Use a random session ID in the SDP
1411 RFC4566 Section 5.2 says that it should make the username, session id,
1412 nettype, addrtype and unicast address tuple globally unique. Always using
1413 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
1414 Instead let's create a 64 bit random number, which at least brings us
1415 closer to the goal of global uniqueness.
1416 https://tools.ietf.org/html/rfc4566#section-5.2
1418 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
1420 * examples/test-launch.c:
1421 * examples/test-mp4.c:
1422 * examples/test-ogg.c:
1423 * examples/test-uri.c:
1424 examples: Don't call gst_init() and gst_get_option_group()
1425 The latter calls the former at the appropriate time.
1427 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
1429 * gst/rtsp-server/rtsp-client.c:
1430 rtsp-client: Drop trailing \0 of RTSP DATA messages
1431 We add a trailing \0 in GstRTSPConnection to make parsing of
1432 string message bodies easier (e.g. the SDP from DESCRIBE) but
1433 for actual data this means we have to drop it or otherwise
1434 create invalid data.
1436 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
1438 * gst/rtsp-server/rtsp-stream.c:
1439 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
1440 Fixes crash when two threads access handle_new_sample() at the same
1441 time, one for RTP, one for RTCP.
1442 Otherwise, when iterating over the transports cache, it might be modified by
1443 another thread at the same time if the transports cookie has changed.
1444 https://bugzilla.gnome.org/show_bug.cgi?id=742954
1446 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
1448 * gst/rtsp-server/rtsp-stream.c:
1449 rtsp-stream: Set format=TIME on our app sources for TCP
1451 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
1453 * gst/rtsp-server/rtsp-session-pool.c:
1454 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
1455 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
1456 RFC 2326 states that session IDs may consist of alphanumeric as well as
1457 the safe characters $-_.+ -- N.B. the percent character is not allowed.
1458 Previously the session ID was URI-escaped, this meant that any character
1459 which was not alphanumeric or any of the characters +-._~ would be
1460 percent encoded. While the RFC (surprisingly) mentions that linear white
1461 space in session IDs should be URI-escaped, it does not say anything
1462 about other characters. Moreover no white space is allowed in the
1463 session ID. Finally the percent character which is the result of
1464 URI-escaping is not allowed in a session ID.
1465 So there is no reason to do any URI-escaping, and now it is removed.
1466 https://bugzilla.gnome.org/show_bug.cgi?id=742869
1468 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
1471 Automatic update of common submodule
1472 From f2c6b95 to bc76a8b
1474 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
1477 Fix 'make check' from top-level directory
1479 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
1481 * examples/test-launch.c:
1482 * examples/test-mp4.c:
1483 * examples/test-ogg.c:
1484 * examples/test-uri.c:
1485 examples: Add command-line parsing and take a 'port' argument
1486 This allows users to run multiple servers on different ports for testing.
1487 Only done for examples that actually take arguments and hence are capable of
1488 outputting different streams for each instance on each port.
1489 https://bugzilla.gnome.org/show_bug.cgi?id=742115
1491 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
1493 * gst/rtsp-server/rtsp-client.c:
1494 * gst/rtsp-server/rtsp-client.h:
1495 rtsp-client: Add a send_message default signal handler
1496 This allows subclasses to easily hook into the response sending
1497 mechanism without doing everything from a signal, which seems
1498 awkward from subclasses.
1500 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
1503 Automatic update of common submodule
1504 From ef1ffdc to f2c6b95
1506 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1510 configure: add --disable-examples switch
1511 https://bugzilla.gnome.org/show_bug.cgi?id=741678
1513 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
1515 * examples/.gitignore:
1516 * examples/Makefile.am:
1517 * examples/test-video-rtx.c:
1518 examples: add a retransmisison example implementing RFC4588
1519 Currently only SSRC-multiplexed rtx streams are supported
1521 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
1523 * gst/rtsp-server/rtsp-stream.c:
1524 rtsp-stream: Fix some minor memory leaks
1526 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
1528 * gst/rtsp-server/rtsp-media.c:
1529 rtsp-media: Some minor cleanup
1531 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
1533 * gst/rtsp-server/rtsp-stream.c:
1534 rtsp-stream: Fix compiler warnings
1535 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
1536 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1538 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
1539 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1542 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
1544 * docs/libs/gst-rtsp-server-sections.txt:
1545 * gst/rtsp-server/rtsp-media-factory.c:
1546 * gst/rtsp-server/rtsp-media-factory.h:
1547 * gst/rtsp-server/rtsp-media.c:
1548 * gst/rtsp-server/rtsp-media.h:
1549 * gst/rtsp-server/rtsp-sdp.c:
1550 * gst/rtsp-server/rtsp-stream.c:
1551 * gst/rtsp-server/rtsp-stream.h:
1552 media: implement ssrc-multiplexed retransmission support
1553 based off RFC 4588 and the server-rtpaux example in -good
1555 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
1557 * gst/rtsp-server/rtsp-client.c:
1558 * gst/rtsp-server/rtsp-stream-transport.c:
1559 * gst/rtsp-server/rtsp-stream.c:
1560 rtsp: Ref transports in hash table.
1561 Also ref streams for transports.
1562 This solves a crash when reciving a rtcp after teardown but before
1564 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
1566 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
1569 Automatic update of common submodule
1570 From 7bb2bce to ef1ffdc
1572 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
1574 * gst/rtsp-server/rtsp-client.c:
1575 client: refactor cleanup of cached media
1577 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
1579 * tests/check/gst/client.c:
1581 The session leak is now fixed, lets remove those FIXME comments.
1583 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
1585 * tests/check/gst/rtspserver.c:
1586 tests: Test to setup two sessions on one connection
1587 https://bugzilla.gnome.org/show_bug.cgi?id=739112
1589 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
1591 * tests/check/gst/rtspserver.c:
1592 tests: Test setup with tcp transport
1593 https://bugzilla.gnome.org/show_bug.cgi?id=739112
1595 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
1597 * gst/rtsp-server/rtsp-client.c:
1598 client: Configure transport after creating session media
1599 The default implementation of configure_client_transport() in
1600 rtsp-client uses the session media when it chooses channels for
1601 interleaved traffic.
1602 https://bugzilla.gnome.org/show_bug.cgi?id=739112
1604 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
1606 * gst/rtsp-server/rtsp-client.c:
1607 * gst/rtsp-server/rtsp-session-media.c:
1608 client: Stop caching media in client when doing setup
1609 If the media has been managed by a session media, it should not be
1610 cached in the client any longer. The GstRTSPSessionMedia object is now
1611 responsible for unpreparing the GstRTSPMedia object using
1612 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
1614 https://bugzilla.gnome.org/show_bug.cgi?id=739112
1616 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1618 * gst/rtsp-server/rtsp-stream.c:
1619 rtsp-stream: unref srtp decoder when leaving bin
1620 https://bugzilla.gnome.org/show_bug.cgi?id=739481
1622 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1624 * gst/rtsp-server/rtsp-client.c:
1625 rtsp-client: mikey memory leaks
1626 https://bugzilla.gnome.org/show_bug.cgi?id=739383
1628 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
1631 Automatic update of common submodule
1632 From 84d06cd to 7bb2bce
1634 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
1637 Parallelise 'make check-valgrind'
1639 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
1642 Automatic update of common submodule
1643 From a8c8939 to 84d06cd
1645 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
1648 Automatic update of common submodule
1649 From 36388a1 to a8c8939
1651 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
1653 * gst/rtsp-server/rtsp-media.c:
1654 rtsp-media: deactivate media when shutting down from paused
1655 This was only done when going directly from playing.
1656 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
1658 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1660 * gst/rtsp-server/rtsp-client.c:
1661 * gst/rtsp-server/rtsp-context.h:
1662 rtsp-client: add stream transport to context
1663 We add the stream transport to the context so we can get the configured
1664 client stream transport in the setup request signal.
1665 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
1667 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1669 * gst/rtsp-server/rtsp-stream.c:
1670 stream: release lock even not all transports have been removed
1671 We don't want to keep the lock even we return FALSE because not all the
1672 transports have been removed. This could lead into a deadlock.
1673 https://bugzilla.gnome.org/show_bug.cgi?id=737797
1675 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
1677 * gst/rtsp-server/rtsp-sdp.c:
1678 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
1679 These were renamed in GstRTPBasePayload in 1.0
1681 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1683 * gst/rtsp-server/rtsp-client.c:
1684 client: set session media to NULL without the lock
1685 We need to set session medias to NULL without the client lock otherwise
1686 we can end up in a deadlock if another thread is waiting for the lock
1687 and media unprepare is also waiting for that thread to end.
1688 https://bugzilla.gnome.org/show_bug.cgi?id=737690
1690 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
1692 * gst/rtsp-server/rtsp-media.c:
1693 rtsp-media: Set state to UNPREPARING in all cases
1695 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
1697 * gst/rtsp-server/rtsp-media.c:
1698 media: set state to unpreparing when unprepare is initiated
1699 https://bugzilla.gnome.org/show_bug.cgi?id=737675
1701 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
1703 * gst/rtsp-server/rtsp-client.c:
1704 rtsp-client: Remove backlog limit while processings requests
1705 If the backlog limit is kept two cases of deadlocks may be
1706 encountered when streaming over TCP. Without the backlog
1707 limit this deadlocks can not happen, at the expence of
1709 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
1711 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
1713 * gst/rtsp-server/rtsp-client.c:
1714 rtsp-client: do not free main context before rtsp watch
1715 https://bugzilla.gnome.org/show_bug.cgi?id=737110
1717 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
1719 * tests/check/gst/rtspserver.c:
1720 tests: Extend unit test timeout to accomodate for valgrind
1721 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1723 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
1725 * gst/rtsp-server/rtsp-client.c:
1726 * gst/rtsp-server/rtsp-session.c:
1727 * gst/rtsp-server/rtsp-stream-transport.c:
1728 rtsp-*: Treat sending packets to clients as keepalive
1729 As long as gst-rtsp-server can successfully send RTP/RTCP data to
1730 clients then the client must be reading. This change makes the server
1731 timeout the connection if the client stops reading.
1732 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1734 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
1736 * gst/rtsp-server/rtsp-client.c:
1737 rtsp-client: Allow backlog to grow while expiring session
1738 Allow the send backlog in the RTSP watch to grow to unlimited size while
1739 attempting to bring the media pipeline to NULL due to a session
1740 expiring. Without this change the appsink element cannot change state
1741 because it is blocked while rendering data in the new_sample callback.
1742 This callback will block until it has successfully put the data into the
1743 send backlog. There is a chance that the send backlog is full at this
1744 point which means that the callback may block for a long time, possibly
1745 forever. Therefore the media pipeline may also be prevented from
1746 changing state for a long time.
1747 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1749 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
1751 * gst/rtsp-server/rtsp-client.c:
1752 rtsp-client: Make old compilers happy
1753 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
1754 Just in case that guint8 doesn't fit in a pointer. Just in case ...
1756 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
1758 * gst/rtsp-server/rtsp-client.c:
1759 client: raise the backlog limits before pausing
1760 We need to raise the backlog limits before pausing the pipeline or else
1761 the appsink might be blocking in the render method in wait_backlog() and
1762 we would deadlock waiting for paused.
1763 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
1765 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
1767 * gst/rtsp-server/rtsp-client.c:
1768 client: make define for the WATCH_BACKLOG
1769 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
1771 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
1773 * gst/rtsp-server/rtsp-client.c:
1774 client: simplify session transport handling
1775 link/unlink of the transport in a session was done to keep track of all
1776 TCP transports and to send RTP/RTCP data to the streams. We can simplify
1777 that by putting all the TCP transports in a hashtable indexed with the
1779 We also don't need to link/unlink the transports when we pause/resume
1780 the streams. The same effect is already achieved when we pause/play the
1781 media. Indeed, when we pause the media, the transport is removed from
1782 the media and the callbacks will not be called anymore.
1783 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
1785 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
1787 * gst/rtsp-server/rtsp-stream-transport.c:
1788 * gst/rtsp-server/rtsp-stream-transport.h:
1789 stream-transport: make method to handle received data
1790 Make a method to handle the data received on a channel. It sends the
1791 data to the stream of the transport on the RTP or RTCP pads based on
1794 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
1796 * examples/test-mp4.c:
1797 test: add example of dumping RTCP reports
1799 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
1801 * gst/rtsp-server/rtsp-media.c:
1802 * gst/rtsp-server/rtsp-stream.c:
1803 * gst/rtsp-server/rtsp-stream.h:
1804 rtsp-media: Make sure that sequence numbers are monotonic after pause
1805 The sequence number is not monotonic for RTP packets after pause. The
1806 reason is basepayloader generates a randon sequence number when the
1807 pipeline goes from ready to pause. With this fix generation of sequence
1808 number will be monotonic when going from pause to play request.
1809 https://bugzilla.gnome.org/show_bug.cgi?id=736017
1811 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
1813 * gst/rtsp-server/rtsp-client.c:
1814 rtsp-client: Protect saved clients watch with a mutex
1815 Fixes a crash when close() is called while merging clients
1816 in handle_tunnel(). In that case close() would destroy the
1817 watch while it is still being used in handle_tunnel().
1818 https://bugzilla.gnome.org/show_bug.cgi?id=735570
1820 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
1822 * gst/rtsp-server/rtsp-stream.c:
1823 rtsp-stream: Remove the multicast group udp sources when removing from the bin
1825 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
1827 * gst/rtsp-server/rtsp-media.c:
1828 * gst/rtsp-server/rtsp-stream.c:
1829 * gst/rtsp-server/rtsp-stream.h:
1830 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
1831 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
1832 seeking and will always continue counting the time. This leads to
1833 the NPT after a backwards seek to be something completely different
1834 to the actual seek position.
1835 https://bugzilla.gnome.org/show_bug.cgi?id=732644
1837 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
1839 * examples/test-appsrc.c:
1840 examples: fix another reference leak
1841 gst_rtsp_media_get_element() returns a new ref.
1843 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1845 * examples/test-appsrc.c:
1846 examples: unref element after usage
1847 gst_bin_get_by_name_recurse_up() returns an element
1848 reference that must be unreffed after usage.
1849 https://bugzilla.gnome.org/show_bug.cgi?id=734546
1851 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
1853 * gst/rtsp-server/rtsp-media.c:
1854 signals: Fix copy-pasto in target-state signal offset
1856 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
1860 Makefile: Add usage of build-checks step
1861 Allows building checks without running them
1863 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
1865 * gst/rtsp-server/rtsp-stream.c:
1866 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
1867 When a UDP multicast transport is used it is expected that the server listens
1868 for RTP and RTCP packets on the multicast group with the corresponding port.
1869 Without this we will never get RTCP packets from clients in multicast mode.
1870 https://bugzilla.gnome.org/show_bug.cgi?id=732238
1872 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
1877 === release 1.4.0 ===
1879 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
1885 * gst-rtsp-server.doap:
1888 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
1890 * gst/rtsp-server/rtsp-media.h:
1891 media: correct misspelled words in description
1892 https://bugzilla.gnome.org/show_bug.cgi?id=733244
1894 === release 1.3.91 ===
1896 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
1902 * gst-rtsp-server.doap:
1905 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
1907 * docs/libs/gst-rtsp-server-sections.txt:
1910 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
1912 * gst/rtsp-server/rtsp-server.c:
1913 server: implement client REMOVE filter
1915 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
1917 * gst/rtsp-server/rtsp-client.c:
1918 * gst/rtsp-server/rtsp-client.h:
1919 client: expose _close() method
1920 Expose a previously internal close method to close the client
1923 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
1925 * gst/rtsp-server/rtsp-session-pool.c:
1926 session-pool: signal session-removed outside of the lock
1927 Release the lock before emiting the session-removed signal.
1929 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
1931 * gst/rtsp-server/rtsp-client.c:
1932 * gst/rtsp-server/rtsp-server.c:
1933 * gst/rtsp-server/rtsp-session-pool.c:
1934 * gst/rtsp-server/rtsp-session.c:
1935 * gst/rtsp-server/rtsp-stream.c:
1936 filter: Release lock in filter functions
1937 Release the object lock before calling the filter functions. We need to
1938 keep a cookie to detect when the list changed during the filter
1939 callback. We also keep a hashtable to make sure we only call the filter
1940 function once for each object in case of concurrent modification.
1941 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
1943 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
1945 * gst/rtsp-server/rtsp-client.c:
1946 client: check if watch is set in handle_teardown()
1947 The unit tests run without a watch
1949 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
1951 * tests/check/gst/client.c:
1952 client tests: send teardown to cleanup session
1954 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
1956 * tests/check/gst/rtspserver.c:
1957 server tests: send teardown to cleanup session
1959 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
1961 * gst/rtsp-server/rtsp-client.c:
1962 client: keep ref to client for the session removed handler
1963 This extra ref will be dropped when all client sessions have been
1964 removed. A session is removed when a client sends teardown, closes its
1965 endpoint of the TCP connection or the sessions expires.
1966 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
1968 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
1970 * gst/rtsp-server/rtsp-client.c:
1971 * gst/rtsp-server/rtsp-session.c:
1972 * tests/check/gst/client.c:
1973 client: manage media in session as a last step
1974 Once we manage a media in a session, we can't unmanage it anymore
1975 without destroying it. Therefore, first check everything before we
1976 manage the media, otherwise if something is wrong we have no way to
1978 If we created a new session and something went wrong, remove the session
1979 again. Fixes a leak in the unit test.
1981 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
1983 * examples/test-mp4.c:
1984 * examples/test-ogg.c:
1985 examples: print 'stream ready at url' for mp4 and ogg example
1987 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
1989 * gst/rtsp-server/rtsp-client.c:
1990 * gst/rtsp-server/rtsp-sdp.c:
1991 rtsp: fix for MIKEY api change
1993 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
1995 * gst/rtsp-server/rtsp-client.c:
1996 client: free watch context only once
1997 The watch context is freed when the source is destroyed. Avoids
1998 a CRITICAL when we try to unref the context twice.
2000 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
2002 * gst/rtsp-server/rtsp-client.c:
2005 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
2007 * gst/rtsp-server/rtsp-client.c:
2008 client: protect sessions with lock
2009 Protect the list of sessions with the lock.
2010 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
2012 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
2014 * gst/rtsp-server/rtsp-client.c:
2015 Client: keep a ref to the session
2016 Don't just keep a weak ref to the session objects but use a hard ref. We
2017 will be notified when a session is removed from the pool (expired) with
2018 the new session-removed signal.
2019 Don't automatically close the RTSP connection when all the sessions of
2020 a client are removed, a client can continue to operate and it can create
2021 a new session if it wants. If you want to remove the client from the
2022 server, you have to use gst_rtsp_server_client_filter() now.
2023 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
2024 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
2026 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
2028 * gst/rtsp-server/rtsp-session-pool.c:
2029 * gst/rtsp-server/rtsp-session-pool.h:
2030 session-pool: add session-removed signal
2031 Add a signal to be notified when a session is removed from the pool.
2033 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
2035 * gst/rtsp-server/Makefile.am:
2036 * gst/rtsp-server/rtsp-server.h:
2037 Make rtsp-server.h a single-include header, use it for G-I
2038 https://bugzilla.gnome.org/show_bug.cgi?id=732411
2040 === release 1.3.90 ===
2042 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
2048 * gst-rtsp-server.doap:
2051 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
2053 * gst/rtsp-server/rtsp-stream.c:
2054 stream: crypto can be NULL
2056 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
2058 * gst/rtsp-server/rtsp-client.c:
2059 * gst/rtsp-server/rtsp-media.c:
2060 * gst/rtsp-server/rtsp-mount-points.c:
2061 introspection: add missing allow-none annotations
2062 https://bugzilla.gnome.org/show_bug.cgi?id=730952
2064 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
2066 * gst/rtsp-server/rtsp-address-pool.c:
2067 * gst/rtsp-server/rtsp-media.c:
2068 * gst/rtsp-server/rtsp-session-media.c:
2069 * gst/rtsp-server/rtsp-session-pool.c:
2070 * gst/rtsp-server/rtsp-stream-transport.c:
2071 * gst/rtsp-server/rtsp-stream.c:
2072 * gst/rtsp-server/rtsp-token.c:
2073 introspection: add (nullable) annotations to return values
2074 https://bugzilla.gnome.org/show_bug.cgi?id=730952
2076 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
2078 * gst/rtsp-server/rtsp-client.c:
2079 * gst/rtsp-server/rtsp-stream.c:
2080 gi: improve annotations
2081 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
2083 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
2085 * gst/rtsp-server/rtsp-client.c:
2086 * gst/rtsp-server/rtsp-media-factory.c:
2087 * gst/rtsp-server/rtsp-media.c:
2088 * gst/rtsp-server/rtsp-server.c:
2089 signals: use generic marshal function
2090 Use the generic C marshal function.
2091 Use more explicit type instead of G_TYPE_POINTER
2093 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
2095 * gst/rtsp-server/rtsp-context.h:
2096 context: add type macro
2098 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
2100 * gst/rtsp-server/rtsp-client.c:
2101 * gst/rtsp-server/rtsp-sdp.c:
2102 * gst/rtsp-server/rtsp-sdp.h:
2103 sdp: hide key length defines
2104 They don't have a namespace.
2106 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
2111 === release 1.3.3 ===
2113 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
2119 * gst-rtsp-server.doap:
2122 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2124 * gst/rtsp-server/rtsp-client.c:
2125 * gst/rtsp-server/rtsp-sdp.c:
2126 * gst/rtsp-server/rtsp-sdp.h:
2127 mikey: add different key length parameters
2128 Add encryption and authentication key length parameters to MIKEY. For
2129 the encoders, the key lengths are obtained from the cipher and auth
2130 algorithms set in the caps. For the decoders, they are obtained while
2131 parsing the key management from the client.
2132 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
2134 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
2136 * tests/check/gst/stream.c:
2137 stream tests: Make sure we get right multicast address from stream
2138 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
2140 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
2142 * gst/rtsp-server/rtsp-client.c:
2143 client: ref the context until rtsp watch is alive
2144 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
2146 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
2148 * gst/rtsp-server/rtsp-client.c:
2149 client: Destroy the rtsp watch after connection close
2151 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
2153 * gst/rtsp-server/rtsp-media.c:
2154 media: fix confusing comment
2156 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
2158 * gst/rtsp-server/rtsp-session.c:
2159 rtsp-session: Timeout in header.
2160 Adding the possbilty to always have timout in header.
2161 This is configurabe with setting "timeout-always-visible".
2162 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
2164 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
2169 === release 1.3.2 ===
2171 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
2178 * gst-rtsp-server.doap:
2181 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
2184 Automatic update of common submodule
2185 From 211fa5f to 1f5d3c3
2187 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
2189 * gst/rtsp-server/rtsp-client.c:
2190 client: store TCP ports in transport
2191 Store the TCP ports in the transport when we are doing RTSP over TCP.
2192 This way, we can easily get to the ports from the transport.
2193 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
2195 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2197 * gst/rtsp-server/rtsp-stream.c:
2198 stream: add signals for new RTP/RTCP encoders
2199 New signals to allow the user to configure the dynamically created
2201 https://bugzilla.gnome.org/show_bug.cgi?id=730228
2203 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
2205 * gst/rtsp-server/rtsp-media.c:
2206 * gst/rtsp-server/rtsp-media.h:
2207 media: Make suspend()/unsuspend() virtual
2208 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
2210 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2212 * gst/rtsp-server/rtsp-client.c:
2213 client: fix send-message signal marshaller
2214 Use generic marshalling for the send-message signal. It has
2215 two POINTER arguments, not just one.
2216 https://bugzilla.gnome.org/show_bug.cgi?id=729900
2218 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
2220 * tests/check/gst/media.c:
2221 tests: add and remove pads only once
2222 In this test we simulate a dynamic pad by watching the caps event.
2223 Because of renegotiation in the base payloader now, this caps is sent
2224 multiple times but we can only deal with 1 invocation, use a variable to
2225 only 'add and remove' the pad once.
2227 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
2229 * tests/check/gst/rtspserver.c:
2230 tests: add unit test for correct handling of Require headers
2231 https://bugzilla.gnome.org/show_bug.cgi?id=729426
2233 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
2235 * gst/rtsp-server/rtsp-client.c:
2236 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
2237 Servers must handle Require headers and must report a failure
2238 if they don't handle any of the Required options, see RFC 2326,
2239 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
2240 https://bugzilla.gnome.org/show_bug.cgi?id=729426
2242 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
2247 === release 1.3.1 ===
2249 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
2255 * gst-rtsp-server.doap:
2258 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
2261 Automatic update of common submodule
2262 From bcb1518 to 211fa5f
2264 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
2269 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
2271 * tests/check/gst/sessionmedia.c:
2272 tests: fix memory leak in sessionmedia unit test
2274 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
2276 * gst/rtsp-server/rtsp-client.c:
2277 client: emit a signal before sending a message
2278 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
2280 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
2282 * gst/rtsp-server/rtsp-client.c:
2283 client: pass context to send_message
2284 Pass the current context to send_message, we will need it later.
2286 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
2288 * gst/rtsp-server/rtsp-client.c:
2289 client: fix typo in comment
2291 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
2293 * gst/rtsp-server/rtsp-media.c:
2294 media: Do not stop thread twice if default_prepare() fails
2296 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
2298 * gst/rtsp-server/rtsp-client.c:
2299 client: set the watch to flushing before going to NULL
2300 First set the watch to flushing so that we unblock any current and
2301 future attempt to send data on the watch, Then set the pipeline to
2303 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
2305 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
2307 * gst/rtsp-server/rtsp-session-pool.c:
2308 * tests/check/gst/sessionpool.c:
2309 rtsp-session-pool: Fixes annotation
2310 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
2311 in the sessionpool test.
2312 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
2314 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
2316 * gst/rtsp-server/rtsp-media.c:
2317 * gst/rtsp-server/rtsp-media.h:
2318 media: make media_prepare virtual
2319 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
2321 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
2323 * gst/rtsp-server/rtsp-media.c:
2324 * tests/check/gst/media.c:
2325 media: stop the thread in more error cases
2327 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
2329 * gst/rtsp-server/rtsp-media.c:
2330 * tests/check/gst/media.c:
2331 media: allow NULL as the thread
2332 Use the default context whan passing a NULL thread.
2334 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
2336 * gst/rtsp-server/rtsp-client.c:
2337 rtsp-client: indent cleanup
2338 Coverity was moaning about unreachable code, and I think it was just
2339 confused by { being before the label. We'll see if it pops up again.
2342 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
2344 * gst/rtsp-server/rtsp-client.c:
2345 * gst/rtsp-server/rtsp-media.c:
2346 client: Add drop-backlog property
2347 When we have too many messages queued for a client (currently hardcoded
2348 to 100) we overflow and drop the messages. Add a drop-backlog property
2349 to control this behaviour. Setting this property to FALSE will retry
2350 to send the messages to the client by waiting for more room in the
2352 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
2354 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
2356 * gst/rtsp-server/rtsp-client.c:
2357 client: support for POST before GET when setting up a tunnel
2359 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
2361 * gst/rtsp-server/rtsp-client.c:
2362 client: remove watch of the second client after http tunnel setup
2363 The second client will be freed after the HTTP tunnel has been set up.
2364 Make sure it's RTSP watch is never dispatched again.
2365 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
2367 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
2369 * gst/rtsp-server/rtsp-media.c:
2370 * tests/check/gst/media.c:
2371 media: Make media_prepare() fail if port allocation fails
2372 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
2374 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
2376 * tests/check/gst/media.c:
2377 media test: cleanup the thread pool in tests
2379 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
2381 * gst/rtsp-server/rtsp-media.c:
2382 * tests/check/gst/media.c:
2383 rtsp-media: Unblock blocked streams in unprepare
2384 The streams will be blocked when a live media is prepared.
2385 The streams should be unblocked in gst_rtsp_media_unprepare.
2386 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
2388 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
2390 * gst/rtsp-server/rtsp-media.c:
2391 media: release the state lock when going to NULL
2392 Set our state to UNPREPARING and release the state-lock before
2393 setting the pipeline to the NULL state. This way, any pad-added
2394 callback will be able to take the state-lock and check that we are now
2395 unpreparing instead of deadlocking.
2396 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
2398 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
2400 * gst/rtsp-server/rtsp-media.c:
2401 media: protect status with lock
2402 Make sure we only update the status with the lock.
2404 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
2406 * gst/rtsp-server/rtsp-client.c:
2407 * gst/rtsp-server/rtsp-sdp.c:
2408 rtsp: update for MIKEY API changes
2410 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
2412 * gst/rtsp-server/rtsp-client.c:
2413 client: parse the mikey response from the client
2414 Parse the mikey response from the client and update the policy for
2417 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
2419 * gst/rtsp-server/rtsp-stream.c:
2420 * gst/rtsp-server/rtsp-stream.h:
2421 stream: add method to set crypto info
2422 Make a method to configure the crypto information of a stream.
2423 Set udpsrc in READY instead of PAUSED so that we can configure caps
2426 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
2428 * gst/rtsp-server/rtsp-client.c:
2429 client: cleanup error paths
2431 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
2433 * gst/rtsp-server/rtsp-media.c:
2436 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
2438 * examples/test-video.c:
2439 test: enable SRTP only on RTSPS
2440 We only want to enable SRTP when doing rtsp over TLS so that we can
2441 exchange the keys in a secure way.
2443 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
2445 * examples/test-video.c:
2446 test: print an error on failure
2448 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
2451 * examples/test-video.c:
2452 * gst/rtsp-server/rtsp-sdp.c:
2453 * gst/rtsp-server/rtsp-stream.c:
2454 * tests/check/Makefile.am:
2455 stream: add SRTP support
2456 Install srtp encoder and decoder elements in rtpbin
2459 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2461 * tests/check/Makefile.am:
2462 * tests/check/gst/sessionpool.c:
2463 tests: Add unit tests for sessionpool
2464 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
2466 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2468 * tests/check/gst/threadpool.c:
2469 tests: Improve code coverage of rtsp-threadpool tests
2470 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
2472 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2474 * tests/check/gst/sessionmedia.c:
2475 tests: Improve code coverage for rtsp-session-media
2476 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
2478 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2480 gobject-introspection: Add annotations to support language bindings
2481 In addition a few cosmetic changes:
2482 * Adjust the order of arguments
2483 * Fix typo: occured -> occurred
2484 * Fix indentation after Return:-clauses
2485 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
2487 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2489 * gst/rtsp-server/rtsp-stream.c:
2490 rtsp-stream: Don't mix IPv4 and IPv6 addresses
2491 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
2493 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
2495 * gst/rtsp-server/rtsp-stream.c:
2496 stream: take caps after the session manager
2497 Take the caps for the SDP after they leave the rtpbin so that we can
2498 also get the properties added by rtpbin elements.
2500 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
2502 * gst/rtsp-server/rtsp-stream.c:
2503 stream: release lock while pushing out packets
2504 Keep a cache of the transports and use this to iterate the transport
2505 while pushing packets. This allows us to release the lock early.
2506 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
2508 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
2510 * gst/rtsp-server/rtsp-client.c:
2511 * gst/rtsp-server/rtsp-client.h:
2512 rtsp-client: vmethod for modifying tunnel GET response
2513 Add a vmethod tunnel_http_response where the response to the HTTP GET
2514 for tunneled connections can be modified.
2515 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
2517 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
2519 * gst/rtsp-server/rtsp-sdp.c:
2520 sdp: make 1 media line per profile
2521 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
2522 line in the SDP for each profile. The client is then supposed to pick
2523 one of the profiles in the SETUP request. Because the m= lines have the
2524 same pt, the client also knows that only 1 option is possible.
2526 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
2528 * gst/rtsp-server/rtsp-media-factory.c:
2529 * gst/rtsp-server/rtsp-media-factory.h:
2530 * gst/rtsp-server/rtsp-media.c:
2531 factory: add profile property and pass to media and streams
2533 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
2535 * examples/test-multicast.c:
2536 * gst/rtsp-server/rtsp-sdp.c:
2537 sdp: pass multicast connection for multicast-only stream
2538 Pass the multicast address of the stream in the connection info in the
2539 SDP so that clients try a multicast connection first.
2540 Only allow multicast connections in the test-multicast example. Also
2541 increase the TTL a little.
2543 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2546 .gitignore: Ignore gcov intermediate files
2547 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
2549 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
2551 * gst/rtsp-server/rtsp-stream.c:
2552 stream: release some locks in error cases
2554 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2556 docs: Enable and fix gtk-doc warnings
2557 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
2558 * addresspool/mediafactory: Add missing annotation colon
2559 * stream: Annotate return value
2560 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
2562 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
2565 Automatic update of common submodule
2566 From fe1672e to bcb1518
2568 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
2571 Automatic update of common submodule
2572 From 1a07da9 to fe1672e
2574 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
2576 * examples/Makefile.am:
2577 examples: use LDADD for libs instead of LDFLAGS
2579 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
2582 configure: make sure releases are in .doap file
2584 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
2586 * examples/test-cgroups.c:
2587 examples: test-cgroups: don't put code with side effects into g_assert()
2588 The g_assert() might get compiled out with the right
2589 compiler/preprocessor flags.
2591 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
2593 * examples/.gitignore:
2594 examples: add cgroup test binary to .gitignore
2596 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
2598 * examples/test-cgroups.c:
2599 examples: fix cgroup test build
2600 Fixes build failure caused by compiler warning:
2601 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
2603 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
2606 .gitignore: ignore temp files created in the course of 'make check'
2608 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
2610 * gst/rtsp-server/rtsp-media.c:
2611 rtsp-media: don't loose frames handling new PLAY request
2612 If client supplied a range check if the range specifies the start point.
2613 If not, then do an accurate seek to the current position. If a start
2614 point was specified do do a key unit seek to make sure the streaming
2615 starts with decodeable frames.
2616 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
2618 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
2620 * gst/rtsp-server/rtsp-media.c:
2621 Revert "media: only flush when setting a new start position"
2622 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
2623 We need to do the flush in all cases, demuxer block currently for
2626 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
2628 * gst/rtsp-server/rtsp-media.c:
2629 media: only flush when setting a new start position
2630 Only flush the pipeline when we change the start position with
2632 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
2634 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
2636 * gst/rtsp-server/rtsp-stream.c:
2637 stream: set ttl-mc before adding the socket
2638 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
2639 never be set on socket.
2640 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
2642 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
2644 * gst/rtsp-server/rtsp-media.c:
2645 media: stop thread if media is already prepared
2646 in gst_rtsp_media_prepare() the thread is not used if media is already
2647 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
2649 https://bugzilla.gnome.org/show_bug.cgi?id=724182
2651 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
2654 build: Ship gst-rtsp-server.doap file
2656 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
2658 * tests/check/gst/rtspserver.c:
2659 tests: Fix another compiler warning with gcc
2661 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
2663 * gst/rtsp-server/rtsp-client.c:
2664 * gst/rtsp-server/rtsp-mount-points.c:
2665 * gst/rtsp-server/rtsp-stream.c:
2666 * tests/check/gst/client.c:
2667 rtsp-server: Fix lots of compiler warnings with clang
2669 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
2672 * gst-rtsp-server.doap:
2673 * tests/Makefile.am:
2674 configure: Synchronise with the configure scripts of the other modules
2676 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
2679 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
2681 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
2683 * gst/rtsp-server/rtsp-media.c:
2684 * gst/rtsp-server/rtsp-stream.c:
2685 Revert "rtsp-server: support build against last stable release"
2686 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
2687 Let us require 1.2.3 now, which is going to be released in a few
2690 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
2692 * gst/rtsp-server/rtsp-session-media.c:
2693 * gst/rtsp-server/rtsp-stream-transport.c:
2694 session: improve RTP-Info
2695 Ignore streams that can't generate RTP-Info instead of failing.
2696 Don't return the empty string when all streams are unconfigured but
2697 return NULL so that we don't generate and empty RTP-Info header.
2698 Improve docs a little.
2700 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
2702 * gst/rtsp-server/rtsp-session-media.c:
2703 Don't free rtpinfo GString when it is NULL
2704 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
2706 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
2708 * gst/rtsp-server/rtsp-media.c:
2709 media: only set keyframe flag when modifying start
2710 Only set the keyframe flag when we modify the start position. The
2711 keyframe flag should probably be ignored when no change is requested but
2712 until we can claim this is all documented properly and all demuxer
2713 implement this, avoid setting the flag.
2714 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
2716 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
2718 * gst/rtsp-server/rtsp-thread-pool.c:
2719 thread-pool: Unref source after mainloop has quit to avoid races in GLib
2720 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
2722 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
2724 * gst/rtsp-server/rtsp-stream.c:
2725 stream: handle NULL seqnum and rtptime arguments
2727 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
2729 * gst/rtsp-server/rtsp-thread-pool.c:
2730 * tests/check/gst/threadpool.c:
2731 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
2732 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
2734 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
2736 * gst/rtsp-server/rtsp-stream.c:
2737 stream: add fallback for missing stats property
2738 Use a fallback when the payloader does not have a stats property
2739 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
2741 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
2744 Automatic update of common submodule
2745 From f7bc1c3 to 1a07da9
2747 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
2749 * gst/rtsp-server/rtsp-stream.c:
2750 stream: don't leak stats structure
2751 Don't leak the stats structure and deal with NULL stats.
2753 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
2755 * gst/rtsp-server/rtsp-stream.c:
2756 stream: Get rtpinfo properties atomically from payloader
2757 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
2759 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
2761 * gst/rtsp-server/rtsp-media.c:
2762 media: refactor state change functions and signals
2763 Make functions to set the target state and the pipeline state and emit
2764 the signals from those functions.
2766 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
2768 * gst/rtsp-server/rtsp-media.c:
2769 * gst/rtsp-server/rtsp-media.h:
2770 media: add signal to notify of pending state changes
2772 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
2774 * gst/rtsp-server/rtsp-media.c:
2775 * gst/rtsp-server/rtsp-stream.c:
2776 rtsp-server: support build against last stable release
2777 Until 1.2.3 is out with the new get_type function and we
2780 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
2782 * gst/rtsp-server/rtsp-stream.c:
2783 stream: fix compilation
2785 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
2787 * gst/rtsp-server/rtsp-media.c:
2788 * gst/rtsp-server/rtsp-media.h:
2789 * gst/rtsp-server/rtsp-stream.c:
2790 * gst/rtsp-server/rtsp-stream.h:
2791 stream: add property to configure profiles
2793 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
2795 * gst/rtsp-server/rtsp-client.c:
2796 client: let stream check supported transport
2797 Delegate the check if a transport is allowed to the stream.
2798 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
2800 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
2802 * gst/rtsp-server/rtsp-stream.c:
2803 * gst/rtsp-server/rtsp-stream.h:
2804 stream: add method to check supported transport
2805 Add a method to check if a transport is supported
2807 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
2810 configure.ac: Only check for gstreamer-check, not check
2811 We include check in gstreamer-check since quite some time now.
2813 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
2815 * gst/rtsp-server/rtsp-session-media.c:
2816 * gst/rtsp-server/rtsp-stream-transport.c:
2817 * gst/rtsp-server/rtsp-stream.c:
2818 * gst/rtsp-server/rtsp-stream.h:
2819 stream: return clock-rate from get_rtpinfo
2820 And use it to correct the rtptime to the requested start-time.
2821 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
2823 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
2825 * gst/rtsp-server/rtsp-session-media.c:
2826 * gst/rtsp-server/rtsp-stream-transport.c:
2827 * gst/rtsp-server/rtsp-stream-transport.h:
2828 session-media: calculate start-time
2830 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
2832 * gst/rtsp-server/rtsp-stream-transport.c:
2833 * gst/rtsp-server/rtsp-stream.c:
2834 * gst/rtsp-server/rtsp-stream.h:
2835 stream: also return the running-time
2836 Return the running-time in the rtpinfo as well.
2838 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
2840 * gst/rtsp-server/rtsp-client.c:
2841 * gst/rtsp-server/rtsp-session-media.c:
2842 * gst/rtsp-server/rtsp-session-media.h:
2843 * gst/rtsp-server/rtsp-stream-transport.c:
2844 * gst/rtsp-server/rtsp-stream-transport.h:
2845 session-media: let the session-media make the RTPInfo
2846 Add method to create the RTPInfo for a stream-transport.
2847 Add method to create the RTPInfo for all stream-transports in a
2849 Use the session-media RTPInfo code in client. This allows us to refactor
2850 another method to link the TCP callbacks.
2852 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
2854 mount-points: sort sequence before g_sequence_lookup
2855 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
2856 sort sequence if dirty, otherwise lookup will fail.
2857 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
2859 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
2862 configure: rename package from gst-rtsp to gst-rtsp-server
2863 To match git module name and avoid confusion with the
2864 rtsp lib in gst-plugins-base and rtsp plugin in -good.
2866 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
2869 configure: bump core/base/good requirement to 1.2.0
2870 Bump to released stable version and make implicit
2871 requirements explicit.
2873 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
2878 Fix broken gettext setup which is not used anyway
2880 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
2883 Automatic update of common submodule
2884 From dbedaa0 to d48bed3
2886 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
2888 * gst/rtsp-server/rtsp-client.c:
2889 * gst/rtsp-server/rtsp-media.c:
2890 * gst/rtsp-server/rtsp-media.h:
2891 media: add setup_sdp vmethod
2892 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
2893 gst_rtsp_media_setup_sdp.
2894 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
2896 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
2898 * gst/rtsp-server/rtsp-stream.c:
2899 rtsp-stream: Check return value of sscanf
2900 streamid is only valid if sscanf matched something.
2902 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
2904 * gst/rtsp-server/rtsp-client.c:
2905 rtsp-client: Fix iteration
2906 Wouldn't even enter the code block otherwise (i++ was used as the check
2907 and not the postfix).
2909 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
2911 * gst/rtsp-server/rtsp-client.c:
2912 * gst/rtsp-server/rtsp-client.h:
2913 client: add vmethod to configure media and streams
2914 Implement a vmethod that can be used to configure the media and the
2915 streams based on the current context. Handle the blocksize handling in
2916 the default handler.
2917 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
2919 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
2922 Make git ignore more unit test binaries
2924 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
2926 * gst/rtsp-server/rtsp-address-pool.h:
2927 * gst/rtsp-server/rtsp-auth.h:
2928 * gst/rtsp-server/rtsp-client.h:
2929 * gst/rtsp-server/rtsp-context.h:
2930 * gst/rtsp-server/rtsp-media-factory-uri.h:
2931 * gst/rtsp-server/rtsp-media-factory.h:
2932 * gst/rtsp-server/rtsp-media.h:
2933 * gst/rtsp-server/rtsp-mount-points.h:
2934 * gst/rtsp-server/rtsp-server.h:
2935 * gst/rtsp-server/rtsp-session-media.h:
2936 * gst/rtsp-server/rtsp-session-pool.h:
2937 * gst/rtsp-server/rtsp-session.h:
2938 * gst/rtsp-server/rtsp-stream-transport.h:
2939 * gst/rtsp-server/rtsp-stream.h:
2940 * gst/rtsp-server/rtsp-thread-pool.h:
2941 * gst/rtsp-server/rtsp-token.h:
2942 rtsp-server: add padding to many public structures
2943 Not mini objects though, since they are not subclassable
2944 anyway, nor kept on the stack or inlined in a structure.
2946 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
2948 media: add new create_rtpbin vmethod
2949 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
2950 https://bugzilla.gnome.org/show_bug.cgi?id=719734
2952 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
2954 * tests/check/gst/media.c:
2955 tests: fix memory leak, free test's thread pool
2956 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
2958 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
2960 * gst/rtsp-server/rtsp-stream-transport.c:
2961 stream-transport: free url in finalize
2963 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
2965 * gst/rtsp-server/rtsp-media.c:
2966 media: also do state change in suspended state
2968 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
2970 * gst/rtsp-server/rtsp-client.c:
2971 * gst/rtsp-server/rtsp-media.c:
2972 media: also handle prepare and range in suspended state
2973 When we are suspended, we are already prepared.
2974 We can get the range in the suspended state.
2976 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
2978 * tests/check/Makefile.am:
2979 * tests/check/gst/sessionmedia.c:
2980 check: add test for uri in setup
2981 Added unit tests for the new functionality in GstRTSPStreamTransport.
2982 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
2984 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
2986 * gst/rtsp-server/rtsp-client.c:
2987 client: store setup uri and use in PLAY response
2988 Store the uri used when doing the setup and use that in the PLAY
2990 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
2992 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
2994 * gst/rtsp-server/rtsp-stream-transport.c:
2995 * gst/rtsp-server/rtsp-stream-transport.h:
2996 stream-transport: add method to get/set url
2998 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
3000 * gst/rtsp-server/rtsp-client.c:
3001 client: suspend after SDP and unsuspend before PLAYING
3002 Based on patches by Ognyan Tonchev <ognyan@axis.com>
3003 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
3005 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
3007 * gst/rtsp-server/rtsp-media-factory.c:
3008 * gst/rtsp-server/rtsp-media-factory.h:
3009 * gst/rtsp-server/rtsp-media.c:
3010 * gst/rtsp-server/rtsp-media.h:
3011 * gst/rtsp-server/rtsp-session-media.c:
3012 * gst/rtsp-server/rtsp-session.c:
3013 * tests/check/gst/media.c:
3014 * tests/check/gst/mediafactory.c:
3015 media: add suspend modes
3016 Add support for different suspend modes. The stream is suspended right after
3017 producing the SDP and after PAUSE. Different suspend modes are available that
3018 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
3019 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
3020 state and RESET will bring the pipeline to the NULL state.
3021 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
3022 this means that the pipeline needs to be prerolled again.
3023 Base on patches by Ognyan Tonchev <ognyan@axis.com>
3024 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3026 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
3028 * gst/rtsp-server/rtsp-media.c:
3029 media: start live streams in blocked state
3030 Start live streams in the blocked state and make them preroll using the
3031 messages. This ensure that no data is played by the sink until we explicitly
3032 unblock the stream right before going to PLAYING.
3033 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3035 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
3037 * gst/rtsp-server/rtsp-media.c:
3038 media: refactor starting and waiting for preroll
3039 Based on patches from Ognyan Tonchev <ognyan@axis.com>
3040 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3042 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
3044 * gst/rtsp-server/rtsp-stream.c:
3045 * gst/rtsp-server/rtsp-stream.h:
3046 stream: add API to block streams
3047 Add an API to block on the streams and make it post a message.
3048 Based on patch by Ognyan Tonchev <ognyan@axis.com>
3049 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3051 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
3053 * docs/libs/Makefile.am:
3054 docs: Specify the override file
3055 Even if it's empty (for now) it avoids make distcheck complaining
3057 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
3059 * gst/rtsp-server/rtsp-media.c:
3060 media: move default implementations to where they are used
3062 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
3064 * gst/rtsp-server/rtsp-media.c:
3065 media: take the right lock in gst_rtsp_media_set_pipeline_state()
3066 We need to take the state_lock when calling this method.
3068 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
3070 * gst/rtsp-server/rtsp-media.c:
3071 media: handle add-added on non-bins too
3072 Handle dynamic payloaders that are not bins, as used in the unit-test.
3074 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3076 * gst/rtsp-server/rtsp-media-factory.c:
3077 * gst/rtsp-server/rtsp-media-factory.h:
3078 * gst/rtsp-server/rtsp-media.c:
3079 rtsp-media/-factory: Fix request pad name comments
3080 These must be escaped for gtk-doc to parse the comments without warnings.
3082 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
3084 rtsp-media: remove transports if media is in error status
3085 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
3086 trying to change to GST_STATE_NULL and media is in error status, we
3087 remove all transports.
3088 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
3090 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
3092 * gst/rtsp-server/rtsp-media.c:
3093 rtsp-media: use element metadata to find payloader
3094 Use the element metadata to find the payloader instead of checking
3096 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
3098 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
3100 rtsp-stream: add getter for payload type
3101 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
3102 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
3103 element and create the stream with this one instead of the dynpay%d
3105 https://bugzilla.gnome.org/show_bug.cgi?id=712396
3107 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3109 * gst/rtsp-server/rtsp-client.c:
3110 * gst/rtsp-server/rtsp-context.h:
3111 * gst/rtsp-server/rtsp-media.c:
3112 * gst/rtsp-server/rtsp-mount-points.c:
3113 * gst/rtsp-server/rtsp-server.c:
3114 * gst/rtsp-server/rtsp-token.c:
3115 rtsp-*: Refer to NULL as a constant in comments
3117 https://bugzilla.gnome.org/show_bug.cgi?id=714988
3119 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3121 rtsp-*: Fix type name typos in comments
3122 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
3123 * rtsp-auth: Refer to part of constant name as text
3124 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
3125 * rtsp-session-media: Fix GstRTSPSessionMedia typo
3126 * rtsp-stream: Fix typo when refering to GstBin
3127 https://bugzilla.gnome.org/show_bug.cgi?id=714988
3129 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3132 * docs/libs/gst-rtsp-server-docs.sgml:
3133 * docs/libs/gst-rtsp-server-sections.txt:
3134 docs: Improve documentation
3135 * Include annotation-glossary to quiet gtk-doc
3136 * Rename remaining ClientState -> Context
3137 * Rename object hierarchy file
3138 * Remove stale chapter references
3139 * Add missing function and object references
3140 * Include missing GstRTSPAddressPoolResult
3141 https://bugzilla.gnome.org/show_bug.cgi?id=714988
3143 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
3145 * gst/rtsp-server/rtsp-client.c:
3146 * gst/rtsp-server/rtsp-server.c:
3147 * gst/rtsp-server/rtsp-session-pool.c:
3148 * gst/rtsp-server/rtsp-session.c:
3149 * gst/rtsp-server/rtsp-stream.c:
3150 rtsp-server: sprinkle some allow-none annotations for g-i
3152 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
3154 * gst/rtsp-server/rtsp-stream.c:
3155 * gst/rtsp-server/rtsp-stream.h:
3156 stream: add method to filter transports
3157 Add a method to safely iterate and collect the stream transports
3158 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
3160 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
3162 * gst/rtsp-server/rtsp-client.c:
3163 * gst/rtsp-server/rtsp-server.c:
3164 * gst/rtsp-server/rtsp-session-pool.c:
3165 * gst/rtsp-server/rtsp-session.c:
3166 rtsp: allow NULL func in filters
3167 Passing a null function make the filters return a list of
3170 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
3172 * gst/rtsp-server/rtsp-address-pool.c:
3173 * tests/check/gst/addresspool.c:
3174 address-pool: fix address increment
3175 Use a guint instead of guint8 to increment the address. It's still not
3176 completely correct because a guint might not be able to hold the complete
3177 address range, but that's an enhacement for later.
3178 Add unit test to test improved behaviour.
3179 https://bugzilla.gnome.org/show_bug.cgi?id=708237
3181 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
3183 * gst/rtsp-server/rtsp-client.c:
3184 * tests/check/gst/client.c:
3185 client: allow absolute path in requests
3186 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
3188 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
3190 * gst/rtsp-server/rtsp-client.c:
3191 * gst/rtsp-server/rtsp-client.h:
3192 client: make make_path_from_uri a vmethod
3194 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
3196 * docs/libs/gst-rtsp-server-sections.txt:
3197 * gst/rtsp-server/rtsp-stream.c:
3198 * gst/rtsp-server/rtsp-stream.h:
3199 * tests/check/Makefile.am:
3200 * tests/check/gst/stream.c:
3201 stream: Add functions to get rtp and rtcp sockets
3202 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
3204 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
3206 * gst/rtsp-server/rtsp-context.c:
3207 * gst/rtsp-server/rtsp-context.h:
3208 context: defing a GType for the context
3209 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
3211 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
3213 * gst/rtsp-server/Makefile.am:
3214 * gst/rtsp-server/rtsp-auth.c:
3215 * gst/rtsp-server/rtsp-context.c:
3216 * gst/rtsp-server/rtsp-media.c:
3217 * gst/rtsp-server/rtsp-mount-points.c:
3218 * gst/rtsp-server/rtsp-server.h:
3219 * gst/rtsp-server/rtsp-session-media.c:
3220 * gst/rtsp-server/rtsp-session.c:
3221 * gst/rtsp-server/rtsp-stream.c:
3222 Fixed several GIR warnings
3224 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
3226 * gst/rtsp-server/rtsp-auth.c:
3229 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3231 * tests/check/Makefile.am:
3232 * tests/check/gst/token.c:
3233 tests: Add unit tests for token
3234 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
3236 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3238 * gst/rtsp-server/rtsp-token.c:
3239 token: Validate args for gst_rtsp_token_is_allowed
3240 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
3242 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3244 * gst/rtsp-server/rtsp-token.c:
3245 token: Fix bug when creating empty token
3246 We always want to have a valid GstStructure in the token.
3247 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
3249 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
3251 * gst/rtsp-server/rtsp-thread-pool.c:
3252 thread-pool: avoid race in shutdown
3253 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
3254 don't actually stop the mainloop ever. Solve this race by adding an idle source
3255 to the mainloop that calls the _quit. This way we immediately exit the mainloop
3256 if quit was called before we started it.
3258 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3260 * tests/check/Makefile.am:
3261 * tests/check/gst/permissions.c:
3262 tests: Add unit tests for permissions
3263 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
3265 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3267 * tests/check/gst/mediafactory.c:
3268 tests: Test mediafactory permissions
3269 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
3271 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3273 * gst/rtsp-server/rtsp-permissions.c:
3274 permissions: Fix refcounting when adding/removing roles
3275 Previously a role that was removed was unreffed twice, and when
3276 replacing an existing role the replaced role was freed while still being
3277 referenced. Both bugs are now fixed.
3278 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
3280 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3282 * tests/check/gst/media.c:
3283 * tests/check/gst/mediafactory.c:
3284 * tests/check/gst/rtspserver.c:
3285 tests: Check gst_rtsp_url_parse return value
3286 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
3288 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
3291 Automatic update of common submodule
3292 From 865aa20 to dbedaa0
3294 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
3296 * gst/rtsp-server/rtsp-server.c:
3297 rtsp-server: Fix socket leak
3298 https://bugzilla.gnome.org/show_bug.cgi?id=710088
3300 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
3302 * gst/rtsp-server/rtsp-session-pool.c:
3303 rtsp-session-pool: Make sure session IDs are properly URI-escaped
3304 https://bugzilla.gnome.org/show_bug.cgi?id=643812
3306 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
3308 * examples/.gitignore:
3309 * examples/test-video.c:
3310 examples: fix compilation when WITH_AUTH is defined
3311 https://bugzilla.gnome.org/show_bug.cgi?id=710228
3313 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
3316 gitignore: Add new test binary
3318 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
3320 * tests/check/Makefile.am:
3321 * tests/check/gst/threadpool.c:
3322 thread-pool: Add unit test for the thread pools
3323 https://bugzilla.gnome.org/show_bug.cgi?id=710228
3325 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
3327 * gst/rtsp-server/rtsp-thread-pool.c:
3328 thread-pool: Fix thread leak when reusing threads
3329 https://bugzilla.gnome.org/show_bug.cgi?id=709730
3331 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
3333 * gst/rtsp-server/rtsp-server.c:
3334 * tests/check/gst/rtspserver.c:
3335 tests: fixed racy behavior in rtspserver tests
3336 https://bugzilla.gnome.org/show_bug.cgi?id=710078
3338 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3340 * tests/check/gst/addresspool.c:
3341 tests: Improve address pool unit tests
3342 Add a range with mixed IPV4 and IPV6 addresses to pool.
3343 Get an IPV4 address from an IPV6-only pool.
3344 Get an IPV6 address from an IPV4-only pool.
3345 Reserve a IPV6 address from an IPV4-only pool.
3346 Check for unicast addresses in multicast-only pool.
3347 Check for unicast addresses in uni-/multicast-mixed pool.
3348 https://bugzilla.gnome.org/show_bug.cgi?id=710128
3350 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3352 * gst/rtsp-server/rtsp-client.c:
3353 client: append query string in PAUSE/PLAY/TEARDOWN as well
3355 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
3357 * gst/rtsp-server/rtsp-client.c:
3358 client: Add query to control path
3359 If the SETUP url contains a query it must be appended to the control
3360 path so that it matches any already created stream in the media. The
3361 query will also be appended to the session media path.
3363 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3365 * gst/rtsp-server/rtsp-media.c:
3366 rtsp-media: remove old line
3368 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
3370 * gst/rtsp-server/rtsp-stream.c:
3371 stream: Correct control comparison
3372 https://bugzilla.gnome.org/show_bug.cgi?id=709176
3374 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3376 * gst/rtsp-server/rtsp-media.c:
3377 media: Check dynamically if the pipeline supports seeking
3378 We should not depend on whether or not the pipeline state change
3379 returned NO_PREROLL or not. A media could dynamically change its
3380 element and switch from seekable to non seekable so it's best to test
3381 the seekable nature of the pipeline dynamically when we try to do a seek.
3383 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3385 * gst/rtsp-server/rtsp-media.c:
3386 media: Return FALSE if seeking is not supported
3388 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3390 * gst/rtsp-server/rtsp-media.c:
3391 rtsp-media: don't seek accurate by default
3392 Accurate seeking is perhaps a little overkill in the most common situation and
3393 causes some formats (mp3) over slow media to seek extremely slowly.
3395 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
3397 * tests/check/gst/rtspserver.c:
3398 tests: fix unit test
3399 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
3401 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
3403 * gst/rtsp-server/rtsp-client.c:
3404 client: Reply 400 if media cannot be constructed
3405 Reply 400 Bad Request instead of 503 Service Unavailable if media
3406 cannot be constructed in SETUP.
3407 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
3409 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
3411 * gst/rtsp-server/rtsp-client.c:
3412 client: Send setup reply once only
3413 If find_media() failed in handle_setup_request() two replies was sent.
3414 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
3416 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
3419 Automatic update of common submodule
3420 From 6b03ba7 to 865aa20
3422 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
3424 * gst/rtsp-server/rtsp-server.c:
3425 server: Emit client-connected signal earlier
3426 Emit client-connected before the client ref is given to a GSource,
3427 otherwise client-connected can be emitted after the client object has
3430 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
3432 * gst/rtsp-server/rtsp-address-pool.c:
3433 * gst/rtsp-server/rtsp-address-pool.h:
3434 * gst/rtsp-server/rtsp-stream.c:
3435 * tests/check/gst/addresspool.c:
3436 addresspool: return reason of failure
3437 Let gst_rtsp_address_pool_reserve_address() return the reason why
3438 the address could not be reserved.
3439 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
3441 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
3444 autogen.sh: Sync behaviour with other GStreamer modules
3445 Allows building from outside of tree amongst other things
3447 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
3450 Automatic update of common submodule
3451 From b613661 to 6b03ba7
3453 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
3456 Automatic update of common submodule
3457 From 74a6857 to b613661
3459 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
3462 Automatic update of common submodule
3463 From 01a7a46 to 74a6857
3465 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
3467 * gst/rtsp-server/rtsp-client.c:
3468 client: Do not read beyond end of path string
3469 If the setup was done without a control url, make sure we don't try to read the
3470 non-existing control string and crash.
3472 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3474 * gst/rtsp-server/rtsp-client.c:
3475 client: Fix RTPInfo header
3476 Refactor the method to make the content_base.
3477 Use the content-base and the control url to construct the RTPInfo
3480 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3482 * gst/rtsp-server/rtsp-client.c:
3483 client: map url to path only in describe
3484 Only map the request url to a path in the DESCRIBE method. The SDP then
3485 contains the base and control urls that should be used to SETUP/PAUSE/
3486 PLAY/TEARDOWN the media.
3488 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3490 * gst/rtsp-server/rtsp-client.c:
3491 Revert "client: map URL to path in requests"
3492 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
3493 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
3494 contains the base and control urls which are used in the SETUP, PLAY,
3495 PAUSE and TEARDOWN requests.
3497 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3499 * gst/rtsp-server/rtsp-client.c:
3500 client: map URL to path in requests
3502 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3504 * gst/rtsp-server/rtsp-client.c:
3505 * gst/rtsp-server/rtsp-mount-points.c:
3506 * gst/rtsp-server/rtsp-mount-points.h:
3507 mount-points: make vmethod to make path from uri
3508 Make a vmethod to transform an url into a path. The path is then used to lookup
3509 the factory. This makes it possible to also use other bits of the url, such as
3510 the query parameters, to locate the factory.
3512 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
3514 * gst/rtsp-server/rtsp-thread-pool.c:
3515 * gst/rtsp-server/rtsp-thread-pool.h:
3516 thread-pool: Add cleanup to wait for the threadpool to finish
3517 Also fix race condition if two threads are asking for the first
3518 thread from the thread pool at once. This would case two internal
3519 GThreadPools to be created.
3520 https://bugzilla.gnome.org/show_bug.cgi?id=707753
3522 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
3524 * gst/rtsp-server/rtsp-client.c:
3525 * tests/check/gst/client.c:
3526 client: free threadpool
3527 https://bugzilla.gnome.org/show_bug.cgi?id=707638
3529 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
3531 * tests/check/gst/mountpoints.c:
3532 mountpoints tests: unref matched factories
3533 https://bugzilla.gnome.org/show_bug.cgi?id=707638
3535 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
3537 * tests/check/gst/media.c:
3538 media tests: unref thread pool and caps
3539 https://bugzilla.gnome.org/show_bug.cgi?id=707638
3541 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
3543 * gst/rtsp-server/rtsp-auth.c:
3544 * gst/rtsp-server/rtsp-media-factory.c:
3545 * gst/rtsp-server/rtsp-media.c:
3546 auth, media, media-factory: unref permissions
3547 https://bugzilla.gnome.org/show_bug.cgi?id=707638
3549 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3551 * examples/Makefile.am:
3552 Makefile: add rule for appsrc example
3554 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3556 * examples/test-appsrc.c:
3557 tests: add appsrc example
3558 Add an example on how to use appsrc to feed the server pipeline with data.
3560 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
3562 * gst/rtsp-server/rtsp-client.c:
3563 rtsp-client: remove query part from content-base string
3564 Make sure that after the control url has been resolved, it's
3565 not a part of the query-string.
3566 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
3568 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3570 * gst/rtsp-server/rtsp-client.c:
3571 client: don't check url in response
3572 There is no url or method in the response to check
3574 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3576 * gst/rtsp-server/rtsp-client.c:
3577 * gst/rtsp-server/rtsp-client.h:
3578 Add handle-response signal for when we receive a GET_PARAMETER response
3580 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3582 * gst/rtsp-server/rtsp-server.c:
3583 Fix gst_rtsp_server_client_filter, using wrong variable type
3585 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
3587 * gst/rtsp-server/rtsp-media-factory-uri.c:
3588 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
3589 For AAC we need to check for framed=true instead of parsed=true.
3590 https://bugzilla.gnome.org/show_bug.cgi?id=701384
3592 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3594 * gst/rtsp-server/rtsp-stream.c:
3595 stream: optimize pipeline for protocols
3596 When TCP is not an allowed protocol for the stream, avoid creating the
3597 appsrc/appsink/queue and tee elements.
3599 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3601 * gst/rtsp-server/rtsp-media.c:
3602 media: set protocols on streams
3604 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3606 * gst/rtsp-server/rtsp-client.c:
3607 client: use protocols supported by stream
3609 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3611 * gst/rtsp-server/rtsp-media-factory.c:
3612 * gst/rtsp-server/rtsp-media.c:
3613 * gst/rtsp-server/rtsp-stream.c:
3614 media-factory: allow all protocols
3616 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3618 * gst/rtsp-server/rtsp-media.c:
3619 media: configure protocols in new streams
3621 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3623 * gst/rtsp-server/rtsp-stream.c:
3624 * gst/rtsp-server/rtsp-stream.h:
3625 stream: add protocols property
3627 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3629 * gst/rtsp-server/rtsp-media.c:
3630 rtsp-media: send state in "new-state" signal
3631 https://bugzilla.gnome.org/show_bug.cgi?id=705110
3633 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
3636 build: add subdir-objects to AM_INIT_AUTOMAKE
3637 Fixes warnings with automake 1.14
3638 https://bugzilla.gnome.org/show_bug.cgi?id=705350
3640 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3642 * docs/libs/gst-rtsp-server-sections.txt:
3643 * gst/rtsp-server/rtsp-client.c:
3644 * gst/rtsp-server/rtsp-server.c:
3645 * gst/rtsp-server/rtsp-server.h:
3646 server: add method to iterate clients of server
3648 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3650 * gst/rtsp-server/rtsp-media.c:
3651 * gst/rtsp-server/rtsp-media.h:
3652 Add vmethod for rtsp-media subclass to access rtpbin
3654 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3656 * gst/rtsp-server/rtsp-client.h:
3657 small documentation fix
3659 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3661 * gst/rtsp-server/rtsp-client.c:
3662 Do not take range header if range is invalid
3664 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3666 * docs/libs/gst-rtsp-server-sections.txt:
3667 * gst/rtsp-server/rtsp-media.c:
3668 media: add docs for new method
3670 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3672 * gst/rtsp-server/rtsp-media.c:
3673 * gst/rtsp-server/rtsp-media.h:
3674 Add API to rtsp-media set the pipeline's state
3676 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3678 * gst/rtsp-server/rtsp-media.c:
3679 Update current position/duration when gst_rtsp_media_get_range_string is called
3681 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3683 * examples/test-cgroups.c:
3684 tests: add some more docs
3686 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3688 * examples/test-cgroups.c:
3689 * gst/rtsp-server/Makefile.am:
3690 * gst/rtsp-server/rtsp-auth.c:
3691 * gst/rtsp-server/rtsp-auth.h:
3692 * gst/rtsp-server/rtsp-client.c:
3693 * gst/rtsp-server/rtsp-client.h:
3694 * gst/rtsp-server/rtsp-context.c:
3695 * gst/rtsp-server/rtsp-context.h:
3696 * gst/rtsp-server/rtsp-params.c:
3697 * gst/rtsp-server/rtsp-params.h:
3698 * gst/rtsp-server/rtsp-server.c:
3699 * gst/rtsp-server/rtsp-thread-pool.c:
3700 * gst/rtsp-server/rtsp-thread-pool.h:
3701 * tests/check/gst/client.c:
3702 ClientState -> Context
3703 Rename the clientstate to context and put the code in a separate file.
3705 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3707 * examples/test-auth.c:
3708 * gst/rtsp-server/rtsp-auth.c:
3709 * gst/rtsp-server/rtsp-auth.h:
3710 auth: add support for default token
3711 The default token is used when the user is not authenticated and can be used to
3712 give minimal permissions.
3714 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3716 * examples/test-auth.c:
3717 * gst/rtsp-server/rtsp-auth.c:
3718 auth: use defines when possible
3720 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3722 * gst/rtsp-server/rtsp-address-pool.c:
3723 address-pool: improve docs
3725 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3727 * gst/rtsp-server/rtsp-permissions.c:
3728 permissions: add the role to the copy
3730 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
3732 * gst/rtsp-server/rtsp-permissions.c:
3733 permissions: Also copy the roles
3735 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
3737 * gst/rtsp-server/rtsp-permissions.c:
3738 permissions: Make it build
3740 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3742 * gst/rtsp-server/rtsp-address-pool.h:
3745 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3747 * docs/libs/gst-rtsp-server-sections.txt:
3748 * gst/rtsp-server/rtsp-auth.c:
3749 * gst/rtsp-server/rtsp-auth.h:
3750 * gst/rtsp-server/rtsp-media.c:
3751 * gst/rtsp-server/rtsp-session-media.c:
3752 * gst/rtsp-server/rtsp-stream-transport.c:
3753 * gst/rtsp-server/rtsp-stream-transport.h:
3754 * gst/rtsp-server/rtsp-stream.c:
3755 * tests/check/gst/client.c:
3758 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3760 * docs/libs/gst-rtsp-server-sections.txt:
3761 * gst/rtsp-server/rtsp-address-pool.c:
3762 * gst/rtsp-server/rtsp-address-pool.h:
3763 * tests/check/gst/addresspool.c:
3764 * tests/check/gst/rtspserver.c:
3765 address-pool: cleanups
3766 Remove redundant method, improve docs.
3768 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3770 * docs/libs/gst-rtsp-server-sections.txt:
3771 * gst/rtsp-server/rtsp-auth.h:
3772 * gst/rtsp-server/rtsp-permissions.c:
3773 * gst/rtsp-server/rtsp-permissions.h:
3774 * gst/rtsp-server/rtsp-token.c:
3777 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3779 * gst/rtsp-server/rtsp-permissions.c:
3780 permissions: implement _remove_role
3782 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3784 * gst/rtsp-server/rtsp-permissions.c:
3785 permissions: update docs
3787 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3789 * tests/check/gst/client.c:
3790 tests: simplify tests
3791 Client settings are now disabled by default so we don't need an auth
3792 module to disable them.
3794 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3796 * gst/rtsp-server/rtsp-auth.c:
3797 auth: add default authorizations
3798 When no auth module is specified, use our table of defaults to look up the
3799 default value of the check instead of always allowing everything. This was
3800 we can disallow client settings by default.
3802 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3805 README: update readme
3807 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3809 * gst/rtsp-server/rtsp-thread-pool.c:
3810 * gst/rtsp-server/rtsp-thread-pool.h:
3811 thread-pool: add more docs
3813 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3815 * gst/rtsp-server/rtsp-thread-pool.c:
3816 * gst/rtsp-server/rtsp-thread-pool.h:
3817 thread-pool: fix race in thread reuse
3818 If we try to reuse a thread right after we made it stop, we end up using a
3819 stopped thread. Catch this case and only reuse threads that are not stopping.
3821 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3823 * gst/rtsp-server/rtsp-server.c:
3824 server: add small debug
3826 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3828 * tests/check/gst/client.c:
3830 Add some permissions to media so we can use the auth and enable
3833 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3835 * gst/rtsp-server/rtsp-client.c:
3836 client: support pushed context in handle_request
3837 If we already have a pushed state, reuse it and add our own things. This makes
3838 it easier to write tests.
3840 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3842 * gst/rtsp-server/rtsp-auth.c:
3843 auth: don't auth on methods
3844 Don't authorize on methods anymore but on the resources that we
3845 try to access, this is more flexible.
3846 Move the authorization checks to where they are needed and let the
3847 check return the response on error.
3849 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3851 * gst/rtsp-server/rtsp-mount-points.c:
3852 mount-points: add some debug
3854 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3856 * tests/check/gst/client.c:
3857 tests: almost fix test
3859 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3861 * gst/rtsp-server/rtsp-auth.c:
3862 * gst/rtsp-server/rtsp-auth.h:
3863 * gst/rtsp-server/rtsp-client.c:
3864 * gst/rtsp-server/rtsp-client.h:
3865 * gst/rtsp-server/rtsp-server.c:
3866 * gst/rtsp-server/rtsp-server.h:
3867 auth: let the auth module check client_settings
3868 Let the auth module decide if client settings are allowed for the
3871 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3873 * gst/rtsp-server/rtsp-token.c:
3874 * gst/rtsp-server/rtsp-token.h:
3875 token: add method to check boolean permission
3877 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3879 * examples/test-auth.c:
3880 * examples/test-cgroups.c:
3881 * gst/rtsp-server/rtsp-token.c:
3882 * gst/rtsp-server/rtsp-token.h:
3883 token: simplify token constructor
3884 Use variable arguments to make easier API.
3886 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3888 * examples/test-auth.c:
3889 * examples/test-cgroups.c:
3890 * gst/rtsp-server/rtsp-media-factory.c:
3891 * gst/rtsp-server/rtsp-media-factory.h:
3892 media-factory: add convenience API for factory
3894 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3896 * examples/test-auth.c:
3897 * examples/test-cgroups.c:
3898 * gst/rtsp-server/rtsp-permissions.c:
3899 * gst/rtsp-server/rtsp-permissions.h:
3900 permissions: simplify API a little
3901 Avoid passing GstStructure in the add_role method, use varargs instead
3902 to construct the structure behind the scenes. We can then also use the
3903 structure name as the role and simplify some more logic.
3905 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3907 * gst/rtsp-server/rtsp-auth.c:
3910 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3912 * gst/rtsp-server/rtsp-auth.c:
3913 * gst/rtsp-server/rtsp-auth.h:
3914 * gst/rtsp-server/rtsp-client.c:
3915 auth: handle unauthorized response
3916 Move handling of the unauthorized response to the auth module, it can add
3917 the appropriate headers to request authorization for the required method
3918 much better than the client.
3920 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3922 * gst/rtsp-server/rtsp-client.c:
3923 * gst/rtsp-server/rtsp-client.h:
3924 client: allow for sending any message, not only requests
3925 Change the _send_request() method to _send_message() so that we
3926 can both send requests and replies.
3928 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3930 * docs/libs/gst-rtsp-server-sections.txt:
3931 * gst/rtsp-server/rtsp-server.h:
3934 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3936 * examples/test-video.c:
3937 * gst/rtsp-server/rtsp-auth.c:
3938 * gst/rtsp-server/rtsp-auth.h:
3939 * gst/rtsp-server/rtsp-server.c:
3940 * gst/rtsp-server/rtsp-server.h:
3941 auth: move TLS handling to auth module
3942 Remove the TLS settings on the server and move it to the auth module because
3943 that is where security related bits go.
3945 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3947 * gst/rtsp-server/rtsp-client.c:
3948 * gst/rtsp-server/rtsp-client.h:
3949 client: add state push/pop
3951 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3953 * gst/rtsp-server/rtsp-client.c:
3954 * gst/rtsp-server/rtsp-client.h:
3955 client: add connection to state
3957 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3959 * gst/rtsp-server/rtsp-mount-points.c:
3960 mount-points: fix debug
3962 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3964 * tests/check/gst/media.c:
3965 tests: fix media test
3967 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3969 * gst/rtsp-server/rtsp-thread-pool.c:
3970 thread-pool: we don't require a state
3972 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3974 * gst/rtsp-server/rtsp-server.c:
3975 server: let context ref the server
3976 So that we don't risk losing the server object early anc crash.
3978 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3980 * tests/check/gst/client.c:
3981 tests: fix client test
3983 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3986 * docs/libs/gst-rtsp-server-docs.sgml:
3987 * docs/libs/gst-rtsp-server-sections.txt:
3988 * gst/rtsp-server/rtsp-address-pool.c:
3989 * gst/rtsp-server/rtsp-auth.c:
3990 * gst/rtsp-server/rtsp-client.c:
3991 * gst/rtsp-server/rtsp-client.h:
3992 * gst/rtsp-server/rtsp-media-factory-uri.c:
3993 * gst/rtsp-server/rtsp-media-factory.c:
3994 * gst/rtsp-server/rtsp-media-factory.h:
3995 * gst/rtsp-server/rtsp-media.c:
3996 * gst/rtsp-server/rtsp-mount-points.c:
3997 * gst/rtsp-server/rtsp-params.c:
3998 * gst/rtsp-server/rtsp-permissions.c:
3999 * gst/rtsp-server/rtsp-sdp.c:
4000 * gst/rtsp-server/rtsp-server.c:
4001 * gst/rtsp-server/rtsp-server.h:
4002 * gst/rtsp-server/rtsp-session-media.c:
4003 * gst/rtsp-server/rtsp-session-pool.c:
4004 * gst/rtsp-server/rtsp-session.c:
4005 * gst/rtsp-server/rtsp-stream-transport.c:
4006 * gst/rtsp-server/rtsp-stream.c:
4007 * gst/rtsp-server/rtsp-thread-pool.c:
4008 * gst/rtsp-server/rtsp-token.c:
4011 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4013 * gst/rtsp-server/rtsp-session-pool.c:
4014 * gst/rtsp-server/rtsp-session-pool.h:
4015 session-pool: make vmethod to create a session
4016 Make a vmethod to create a sessions so that subclasses can create
4017 custom session objects
4019 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4021 * gst/rtsp-server/rtsp-auth.c:
4022 * gst/rtsp-server/rtsp-media-factory.h:
4023 * gst/rtsp-server/rtsp-media.h:
4024 * gst/rtsp-server/rtsp-mount-points.h:
4025 * gst/rtsp-server/rtsp-session-pool.h:
4026 * gst/rtsp-server/rtsp-stream.h:
4029 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4031 * docs/libs/gst-rtsp-server-docs.sgml:
4032 * docs/libs/gst-rtsp-server-sections.txt:
4033 * gst/rtsp-server/rtsp-address-pool.c:
4034 * gst/rtsp-server/rtsp-address-pool.h:
4035 * gst/rtsp-server/rtsp-auth.c:
4036 * gst/rtsp-server/rtsp-client.h:
4037 * gst/rtsp-server/rtsp-media-factory.h:
4038 * gst/rtsp-server/rtsp-media.c:
4039 * gst/rtsp-server/rtsp-media.h:
4040 * gst/rtsp-server/rtsp-permissions.c:
4041 * gst/rtsp-server/rtsp-permissions.h:
4042 * gst/rtsp-server/rtsp-server.h:
4043 * gst/rtsp-server/rtsp-session-media.c:
4044 * gst/rtsp-server/rtsp-session-media.h:
4045 * gst/rtsp-server/rtsp-session-pool.h:
4046 * gst/rtsp-server/rtsp-session.h:
4047 * gst/rtsp-server/rtsp-stream-transport.h:
4048 * gst/rtsp-server/rtsp-stream.c:
4049 * gst/rtsp-server/rtsp-thread-pool.h:
4052 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4055 * examples/Makefile.am:
4056 configure: compile cgroup example conditionally
4057 Only compile the cgroup example when we have libcgroup
4059 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4062 * examples/Makefile.am:
4063 * examples/test-cgroups.c:
4064 examples: add cgroups example
4066 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4068 * tests/check/gst/rtspserver.c:
4069 tests: fix compilation
4071 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4073 * gst/rtsp-server/rtsp-thread-pool.c:
4074 thread-pool: fix vmethod invocation
4076 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4078 * gst/rtsp-server/rtsp-thread-pool.c:
4079 * gst/rtsp-server/rtsp-thread-pool.h:
4080 thread-pool: store thread type in thread
4082 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4084 * gst/rtsp-server/rtsp-client.c:
4085 client: pass thread from pool to media _prepare
4086 Get a thread from the configured threadpool and pass it to the prepare method of
4089 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4091 * gst/rtsp-server/rtsp-media.c:
4092 * gst/rtsp-server/rtsp-media.h:
4093 media: Accept a thread in _prepare
4094 Remove out own threadpool handling and use the provided thread and
4095 maincontext for the bus messages and the state changes.
4097 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4099 * gst/rtsp-server/rtsp-server.c:
4100 server: configure client thread pool
4102 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4104 * gst/rtsp-server/rtsp-client.c:
4105 * gst/rtsp-server/rtsp-client.h:
4106 client: add method to configure thread pool
4108 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4110 * gst/rtsp-server/rtsp-client.h:
4111 * gst/rtsp-server/rtsp-server.c:
4112 * gst/rtsp-server/rtsp-server.h:
4113 server: use thread pool
4114 Use the thread pool instead of doing our own thing.
4116 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4118 * gst/rtsp-server/Makefile.am:
4119 * gst/rtsp-server/rtsp-thread-pool.c:
4120 * gst/rtsp-server/rtsp-thread-pool.h:
4121 thread-pool: add object to manage threads
4122 Add an object to manage the client and media threads.
4124 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4126 * gst/rtsp-server/rtsp-auth.c:
4127 auth: debug authorization check
4129 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4131 * gst/rtsp-server/rtsp-media.c:
4132 media: start media pipeline in context
4133 Start the media pipeline in the provided context (or our default one
4134 when NULL). This makes sure that we run the bus thread in this context and that
4135 all media threads are children of this context.
4137 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4139 * gst/rtsp-server/rtsp-media-factory.c:
4140 factory: pass permissions to media by default
4142 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4144 * examples/test-auth.c:
4145 test: add permissions to auth test
4146 Ass some permissions to the media factory in the test.
4148 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4150 * gst/rtsp-server/rtsp-auth.c:
4151 * gst/rtsp-server/rtsp-auth.h:
4152 * gst/rtsp-server/rtsp-client.c:
4153 auth: simplify auth checks
4154 Remove client from methods, it's now in the state
4155 Perform the check specified by the string, use the information from the
4156 thread local context.
4158 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4160 * gst/rtsp-server/rtsp-client.c:
4161 * gst/rtsp-server/rtsp-client.h:
4162 client: add state to current thread
4163 Add the client to the ClientState object.
4164 Place the ClientState on the current thread.
4166 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4168 * gst/rtsp-server/rtsp-media-factory.c:
4169 * gst/rtsp-server/rtsp-media-factory.h:
4170 * gst/rtsp-server/rtsp-media.c:
4171 * gst/rtsp-server/rtsp-media.h:
4172 media: make it possible to set permissions
4173 Make it possible to set permissions on media and media factory objects
4175 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4177 * gst/rtsp-server/Makefile.am:
4178 * gst/rtsp-server/rtsp-permissions.c:
4179 * gst/rtsp-server/rtsp-permissions.h:
4180 permissions: add permissions object
4181 Add a mini object to store permissions based on a role.
4183 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4185 * examples/test-auth.c:
4186 * gst/rtsp-server/rtsp-auth.c:
4187 * gst/rtsp-server/rtsp-auth.h:
4188 * gst/rtsp-server/rtsp-client.c:
4189 auth: add auth checks
4190 Add an enum with auth checks and implement the checks in the auth object.
4191 Perform the checks from the client.
4193 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4195 * examples/test-auth.c:
4196 * gst/rtsp-server/rtsp-auth.c:
4197 * gst/rtsp-server/rtsp-auth.h:
4198 * gst/rtsp-server/rtsp-client.h:
4199 auth: use the token after authentication
4200 After we authenticated a user, keep the Token around in the state.
4202 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4204 * gst/rtsp-server/rtsp-client.c:
4205 * gst/rtsp-server/rtsp-media.c:
4206 * gst/rtsp-server/rtsp-media.h:
4207 * tests/check/gst/media.c:
4208 media: add optional context for bus messages
4209 Add an optional mainloop to _prepare that will handle the bus messages instead
4210 of always using the shared mainloop.
4212 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4214 * gst/rtsp-server/Makefile.am:
4215 * gst/rtsp-server/rtsp-token.c:
4216 * gst/rtsp-server/rtsp-token.h:
4217 token: add authorization token
4218 Add a simply miniobject that contains the authorizations. The object contains a
4219 GstStructure that hold all authorization fields. When a user is authenticated,
4220 the auth module will create a Token for the user. The token is then used to
4221 check what operations the user is allowed to do and various other configuration
4224 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4226 * examples/test-auth.c:
4227 * gst/rtsp-server/rtsp-auth.c:
4228 * gst/rtsp-server/rtsp-auth.h:
4229 * gst/rtsp-server/rtsp-client.c:
4230 * gst/rtsp-server/rtsp-client.h:
4231 * gst/rtsp-server/rtsp-media-factory.c:
4232 * gst/rtsp-server/rtsp-media-factory.h:
4233 * gst/rtsp-server/rtsp-media.c:
4234 * gst/rtsp-server/rtsp-media.h:
4235 auth: remove auth from media and factory
4236 Remove the auth object from media and factory. We want to have the RTSPClient
4237 authenticate and authorize resources, there is no need to place another auth
4238 manager on the media/factory.
4240 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4242 * examples/test-auth.c:
4243 * gst/rtsp-server/rtsp-auth.c:
4244 * gst/rtsp-server/rtsp-auth.h:
4245 * gst/rtsp-server/rtsp-client.h:
4246 auth: add support for multiple basic auth tokens
4247 Make it possible to add multiple basic authorisation tokens to one authorization
4248 object. Associate with each token an authorization group that will define what
4249 capabilities are allowed.
4251 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4253 * gst/rtsp-server/rtsp-client.c:
4254 client: error out on non-aggregate control
4255 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
4257 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4259 * gst/rtsp-server/rtsp-client.c:
4260 client: rework setup request a little
4261 Cache the media in DESCRIBE based on the longest matching path with the uri
4262 that we can find in the mount points.
4263 Rework the setup request a little to get the media from the session or from
4264 the longest matching path, this way we can derive the control string as
4265 everything after the path instead of hardcoding it.
4266 Find the stream based on the control string and only open a session when all
4269 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4271 * gst/rtsp-server/rtsp-media.c:
4272 * gst/rtsp-server/rtsp-media.h:
4273 media: add method to find a stream by control url
4275 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4277 * gst/rtsp-server/rtsp-stream.c:
4278 * gst/rtsp-server/rtsp-stream.h:
4279 stream: add method to check control url of stream
4281 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4283 * gst/rtsp-server/rtsp-client.c:
4284 * gst/rtsp-server/rtsp-session-media.c:
4285 * gst/rtsp-server/rtsp-session-media.h:
4286 * gst/rtsp-server/rtsp-session.c:
4287 * gst/rtsp-server/rtsp-session.h:
4288 session: use path matching for session media
4289 Use a path string instead of a uri to lookup session media in the sessions. Also
4290 use path matching to find the largest possible path that matches.
4292 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4294 * gst/rtsp-server/rtsp-client.c:
4295 * gst/rtsp-server/rtsp-mount-points.c:
4296 * gst/rtsp-server/rtsp-mount-points.h:
4297 * tests/check/gst/mountpoints.c:
4298 mount-points: remove useless vmethod
4299 Making lookups in the mount points should not be done with a URL, if there is a
4300 mapping to be done from URL to mount points, we'll need to do it somewhere
4303 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4305 * gst/rtsp-server/rtsp-mount-points.c:
4306 * gst/rtsp-server/rtsp-mount-points.h:
4307 * tests/check/gst/mountpoints.c:
4308 mount-points: improve mount point searching
4309 Use a GSequence to keep track of the mount points.
4310 Match a URL to the longest matching registered mount point. This should be the
4311 URL to perform aggreagate control and the remainder is the stream specific
4313 Add some unit tests for this.
4315 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
4317 * gst/rtsp-server/Makefile.am:
4318 rtsp-server: Allow building of static library
4320 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4322 * tests/check/gst/mediafactory.c:
4323 tests: fix compilation
4325 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4327 * gst/rtsp-server/rtsp-sdp.c:
4328 sdp: get control string from stream
4329 Use the control string as configured in the stream.
4331 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4333 * gst/rtsp-server/rtsp-stream.c:
4334 * gst/rtsp-server/rtsp-stream.h:
4335 stream: add methods and property to set control string
4337 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4339 * gst/rtsp-server/rtsp-client.c:
4341 Rename variables for clarity
4342 Keep media in state when we can
4344 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4346 * gst/rtsp-server/rtsp-client.c:
4347 * gst/rtsp-server/rtsp-stream.c:
4348 * gst/rtsp-server/rtsp-stream.h:
4349 stream: add more support for IPv6
4350 Rename _get_address to _get_multicast_address in GstRTSPStream to
4351 make it clear that this function only deals with multicast.
4352 Make it possible to have both an IPv4 and IPv6 multicast address on
4353 a stream. Give the client an IPv4 or IPv6 address depending on the
4354 address it used to connect to the server.
4355 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
4357 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4359 * gst/rtsp-server/rtsp-client.c:
4362 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4364 * gst/rtsp-server/rtsp-stream.c:
4365 stream: handle failed port allocation
4366 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
4367 can't allocate any family at all. Also keep track of what port families we
4369 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
4371 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4373 * gst/rtsp-server/rtsp-stream.c:
4374 stream: improve docs
4376 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4378 * gst/rtsp-server/rtsp-stream-transport.c:
4379 stream-transport: remove old if 0 block
4381 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
4383 * tests/check/gst/client.c:
4385 gst_rtsp_client_get_uri() has been removed
4386 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
4388 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4390 * gst/rtsp-server/rtsp-client.c:
4391 * gst/rtsp-server/rtsp-client.h:
4392 client: add method to filter managed sessions
4393 Add a method to filter the sessions managed by this client connection.
4394 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
4396 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4398 * gst/rtsp-server/rtsp-client.c:
4399 * gst/rtsp-server/rtsp-client.h:
4400 client: remove _get_uri() method
4401 Remove the get_uri() method on the client. A client has no uri, the uri
4402 property is an internal property to manage the last cached media for
4405 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4407 * gst/rtsp-server/rtsp-media-factory.h:
4408 media-factory: fix typo
4410 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
4412 * gst/rtsp-server/rtsp-media.c:
4413 rtsp-media: Do not leak the query in default_query_stop
4414 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
4416 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4418 * gst/rtsp-server/rtsp-media.c:
4419 media: don't unlock when conversion fails
4420 Don't unlock the state lock when conversion fails because it was not locked.
4422 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4424 * gst/rtsp-server/rtsp-media.c:
4425 * gst/rtsp-server/rtsp-media.h:
4426 Add query_position and query_stop vmethods to rtsp-media
4428 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4430 * gst/rtsp-server/rtsp-media.c:
4431 Fix typo in property install for rtsp-media's time-provider
4433 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4435 * gst/rtsp-server/rtsp-client.c:
4436 * gst/rtsp-server/rtsp-client.h:
4437 client: clean some variables
4438 Clean some variables and add some guards to _send_request()
4440 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4442 * gst/rtsp-server/rtsp-client.c:
4443 * gst/rtsp-server/rtsp-client.h:
4444 Add gst_rtsp_client_send_request API
4445 This makes it possible to send arbitrary messages to a client, such as
4446 SET_PARAMETER or GET_PARAMETER
4448 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4450 * gst/rtsp-server/rtsp-media.c:
4451 * gst/rtsp-server/rtsp-media.h:
4452 media: add _get_element() method
4453 Add method to get the element used when creating the media.
4454 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
4456 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4458 * gst/rtsp-server/rtsp-media.c:
4461 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4463 * gst/rtsp-server/rtsp-stream.c:
4464 * gst/rtsp-server/rtsp-stream.h:
4465 stream: allow access to the rtp session
4466 https://bugzilla.gnome.org/show_bug.cgi?id=703004
4468 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
4470 * gst/rtsp-server/rtsp-stream.c:
4471 * gst/rtsp-server/rtsp-stream.h:
4472 dscp qos support in gst-rtsp-stream
4473 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
4475 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4477 * tests/check/gst/rtspserver.c:
4479 Actually do what the comment says. Also keep the old code around, not sure what
4480 should happen when you get a 454 from a TEARDOWN, does it close the connection?
4481 it currently doesn't.
4483 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4485 * gst/rtsp-server/rtsp-client.c:
4486 client: also watch newly created session
4487 When we newly created a session, start watching it immediately instead of
4488 on the next request.
4490 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
4492 * tests/check/gst/client.c:
4493 tests: add unit test for new-session
4494 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
4496 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4498 * gst/rtsp-server/rtsp-client.c:
4499 client: emit new-session when new session is created
4500 Only emit new-session when we created a new session for a client, not when a
4501 client picked up a previous session.
4502 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
4504 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
4506 * gst/rtsp-server/rtsp-client.c:
4507 client: handle asterisk as path in requests
4508 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
4510 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4512 * gst/rtsp-server/rtsp-media.c:
4513 media: handle segment query format mismatch
4514 It's possible that the segment query returns with a different format than what
4515 we asked for, handle this case also.
4517 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
4519 * gst/rtsp-server/rtsp-media.c:
4520 media: use segment stop in collect_media_stats
4521 Use segment stop instead of duration as range end point.
4522 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
4524 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
4526 * gst/rtsp-server/rtsp-media.c:
4527 * tests/check/gst/media.c:
4528 rtsp-media: Do not leak the element in take_pipeline
4529 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
4531 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
4533 * gst/rtsp-server/rtsp-client.c:
4534 * gst/rtsp-server/rtsp-client.h:
4535 rtsp-client: Make configure_client_transport virtual
4536 This patch makes configure_client_transport virtual. The functionality is
4537 needed to handle some weird clients sending multicast transport settings as url
4539 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
4541 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
4543 * gst/rtsp-server/rtsp-client.c:
4544 * gst/rtsp-server/rtsp-client.h:
4545 rtsp-client: Make param_set and param_get virtual
4546 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
4548 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
4550 * gst/rtsp-server/rtsp-client.c:
4551 * gst/rtsp-server/rtsp-media.c:
4552 * gst/rtsp-server/rtsp-media.h:
4553 media: convert_range replaces get_range_times
4554 get_range_times worked for handling UTC ranges for seeks, but we also
4555 need to convert back from NPT to the requested unit in
4556 get_range_string. convert_range is now used for both.
4557 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
4559 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4561 * gst/rtsp-server/rtsp-client.c:
4562 * gst/rtsp-server/rtsp-sdp.c:
4563 * gst/rtsp-server/rtsp-sdp.h:
4564 sdp: cleanup sdp info
4565 We don't need to pass the proto, we can more easily check a boolean.
4566 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
4568 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
4570 * gst/rtsp-server/rtsp-sdp.c:
4571 use 0.0.0.0 or :: for c= line instead of server address
4573 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
4575 * gst/rtsp-server/rtsp-client.c:
4576 use local address, not remote, in SDP
4577 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
4579 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4582 Automatic update of common submodule
4583 From 098c0d7 to 01a7a46
4585 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
4587 * gst/rtsp-server/rtsp-media.c:
4588 * gst/rtsp-server/rtsp-media.h:
4589 media: possibility to override range time conversion
4590 Make it possible to override the conversion from GstRTSPTimeRange to
4591 GstClockTimes, that is done before seeking on the media
4592 pipeline. Overriding can be useful for UTC ranges, where the default
4593 conversion gives nanoseconds since 1900.
4594 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
4596 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
4598 * gst/rtsp-server/rtsp-server.c:
4599 * gst/rtsp-server/rtsp-server.h:
4600 rtsp-server: Expose the use_client_settings API
4601 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
4603 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
4605 * gst/rtsp-server/rtsp-client.c:
4606 * gst/rtsp-server/rtsp-stream.c:
4607 * gst/rtsp-server/rtsp-stream.h:
4608 rtspstream: handle both ipv4 and ipv6 clients
4609 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
4611 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4613 * gst/rtsp-server/rtsp-sdp.c:
4614 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
4615 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
4616 We already have a way to place extra attributes in the SDP by using a string
4617 property with prefix x- or a- in the caps.
4619 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4621 * gst/rtsp-server/rtsp-sdp.c:
4622 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
4623 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
4624 We already have a way to place extra attributes in the SDP, just make a string
4625 property in the payloader with a- or x- prefix.
4627 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4629 * gst/rtsp-server/rtsp-sdp.c:
4630 rtsp: place a- and x- properties as attributes
4631 application/x-rtp has properties with a- and x- prefixes that should be
4632 placed as attributes in the SDP for the media instead of being added to the
4635 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4637 * examples/Makefile.am:
4638 * examples/test-video.c:
4639 example: add TLS example
4641 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4643 * gst/rtsp-server/rtsp-server.c:
4644 * gst/rtsp-server/rtsp-server.h:
4645 server: add support for TLS
4646 Add methods to set and get a TLS certificate.
4647 Add vmethod to configure a new connection. By default, configure the TLS
4648 certificate in a new connection if needed.
4650 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4652 * gst/rtsp-server/rtsp-server.c:
4653 * gst/rtsp-server/rtsp-server.h:
4654 server: remove accept_client vmethod
4655 This vmethod is not very useful so remove it.
4657 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4659 * gst/rtsp-server/rtsp-server.c:
4660 server: don't crash on NULL GError
4662 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
4664 * gst/rtsp-server/rtsp-session-pool.c:
4665 rtsp-session-pool: corrected session timeout detection
4666 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
4668 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4670 * gst/rtsp-server/rtsp-client.c:
4671 client: improve debug
4673 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4675 * gst/rtsp-server/rtsp-client.c:
4676 * gst/rtsp-server/rtsp-client.h:
4677 * gst/rtsp-server/rtsp-server.c:
4678 server: refactor connection setup
4679 Let the server accept the socket connection and construct a GstRTSPConnection
4680 from it. Remove the code from the client and let the client only deal with
4681 a fully configure GstRTSPConnection object.
4682 We will need this later when the server will configure the connection for
4685 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4687 * gst/rtsp-server/rtsp-stream.c:
4688 stream: keep the transport object alive
4689 Keep the transport object alive while we have it as qdata on the
4692 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
4694 * gst/rtsp-server/rtsp-client.c:
4695 * gst/rtsp-server/rtsp-server.c:
4696 rtsp-server: Do not crash on nmapping of server
4697 * generate error when gst_rtsp_connection_accept fails
4698 * do not stop accepting incoming connections because
4699 accepting a client fails
4700 https://bugzilla.gnome.org/show_bug.cgi?id=701072
4702 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
4704 * gst/rtsp-server/rtsp-client.c:
4705 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
4706 https://bugzilla.gnome.org/show_bug.cgi?id=700953
4708 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
4710 * gst/rtsp-server/rtsp-sdp.c:
4711 rtsp-sdp: Parse framerate caps field and set SDP attribute
4712 The SDP attribute and its format is described in RFC4566.
4713 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
4715 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
4717 * gst/rtsp-server/rtsp-sdp.c:
4718 rtsp-sdp: Parse width/height from caps and set SDP attribute
4719 The SDP attribute and its format is described in RFC6064.
4720 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
4722 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
4724 * gst/rtsp-server/rtsp-sdp.c:
4725 * tests/check/gst/client.c:
4726 rtsp-sdp: add bandwidth line
4727 https://bugzilla.gnome.org/show_bug.cgi?id=699220
4729 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4732 Automatic update of common submodule
4733 From 5edcd85 to 098c0d7
4735 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
4737 * tests/check/gst/media.c:
4738 tests: add dynamic payloader prepare/unprepare check
4740 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4742 * gst/rtsp-server/rtsp-media.c:
4743 media: release lock when removing fakesink
4745 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4747 * gst/rtsp-server/rtsp-stream.c:
4748 stream: set elements to NULL before removing
4749 When removing a stream, set the elements to NULL first. This avoids
4750 element-is-not-in-NULL-state errors when we dispose the elements.
4752 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
4755 Automatic update of common submodule
4756 From 3cb3d3c to 5edcd85
4758 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4760 * gst/rtsp-server/rtsp-media.c:
4761 * gst/rtsp-server/rtsp-media.h:
4762 media: listen to pad-removed signals
4763 Listen to the pad-removed signal and remove the stream associated with the
4765 Add signal to be notified of the removed pad.
4766 Remove the fakesink in unprepare()
4767 Fix signatures of the signal methods
4769 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4771 * examples/test-sdp.c:
4772 tests: add example of reusable pipelines
4774 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
4776 * gst/rtsp-server/rtsp-stream.c:
4777 * gst/rtsp-server/rtsp-stream.h:
4778 stream: add method to get the srcpad
4780 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
4782 * tests/check/gst/media.c:
4783 check: add media prepare/unprepare test
4784 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4786 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
4788 * gst/rtsp-server/rtsp-media.c:
4789 media: disconnect from signal handlers in unprepare()
4790 We connected to the pad-added and no-more-pads signals in prepare() so
4791 we need to disconnect from them in unprepare().
4792 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4794 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
4796 * gst/rtsp-server/rtsp-media.c:
4797 media: don't free streams array
4798 Don't free the streams array in the unprepare() method, they were not
4800 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4802 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
4804 * gst/rtsp-server/rtsp-media.c:
4805 media: don't unref the pipeline in unprepare
4806 Unprepare() should undo what prepare() does. Because the pipeline is
4807 not created in prepare(), we should not unref it in unprepare()
4809 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
4811 * gst/rtsp-server/rtsp-stream.c:
4812 stream: clear session and caps for reuse
4813 Set the session and caps to NULL after unref otherwise we might unref
4815 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4817 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
4819 * gst/rtsp-server/rtsp-client.c:
4820 client: send out teardown signal before tearing down
4821 The advantage is that in the signal handler you get direct access to
4822 information about what streams are about to get torn down (in the
4823 GstRTSPClientState).
4824 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
4826 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
4828 * gst/rtsp-server/rtsp-client.c:
4829 * gst/rtsp-server/rtsp-client.h:
4830 client: expose connection
4831 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
4833 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
4836 Automatic update of common submodule
4837 From aed87ae to 3cb3d3c
4839 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4841 * gst/rtsp-server/rtsp-media.c:
4842 * gst/rtsp-server/rtsp-media.h:
4843 * gst/rtsp-server/rtsp-session-media.c:
4844 * gst/rtsp-server/rtsp-session-media.h:
4845 media: add method to get the base_time of the pipeline
4846 Together with a shared clock, this base-time could eventually be sent to
4847 the client so that it can reconstruct the exact running-time of the clock
4850 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4852 * gst/rtsp-server/Makefile.am:
4853 * gst/rtsp-server/rtsp-media.c:
4854 * gst/rtsp-server/rtsp-media.h:
4855 * gst/rtsp-server/rtsp-sdp.c:
4856 media: add GstNetTimeProvider support
4857 Add a property to let the media provide a GstNetTimeProvider for its clock.
4858 Make methods to get the clock and nettimeprovider
4859 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
4860 provider and also the current time of the clock. This should make it possible
4861 for (GStreamer) clients to slave their clock to the server clock.
4863 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
4866 Automatic update of common submodule
4867 From 04c7a1e to aed87ae
4869 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4871 * gst/rtsp-server/rtsp-media.c:
4872 media: wait for buffering to complete
4873 Wait for buffering to complete before changing the state to the target state.
4875 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4877 * gst/rtsp-server/rtsp-media.c:
4878 media: small cleanup
4880 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
4882 * tests/check/gst/rtspserver.c:
4883 tests: remove extra unref in test_setup_non_existing_stream
4884 The unref is not needed anymore, teardown runs without it.
4885 https://bugzilla.gnome.org/show_bug.cgi?id=696542
4887 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
4889 * tests/check/gst/rtspserver.c:
4890 tests: GSocketService cleanup in test_bind_already_in_use
4891 Use g_socket_service_stop so the rtspserver test stops listening for
4892 incoming connections in test_bind_already_in_use.
4893 https://bugzilla.gnome.org/show_bug.cgi?id=696541
4895 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
4897 * gst/rtsp-server/rtsp-media-factory.c:
4898 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
4899 Instead use a GWeakRef which is safe to use
4900 This is a known GLib bug, see:
4901 https://bugzilla.gnome.org/show_bug.cgi?id=667145
4903 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
4905 * gst/rtsp-server/rtsp-client.c:
4906 * gst/rtsp-server/rtsp-media.c:
4907 * gst/rtsp-server/rtsp-media.h:
4908 * gst/rtsp-server/rtsp-sdp.c:
4909 * tests/check/gst/media.c:
4910 * tests/check/gst/rtspserver.c:
4911 rtsp-media/client: Reply to PLAY request with same type of Range
4912 Remember the type of Range from the PLAY request and use the same type for
4915 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
4917 * gst/rtsp-server/rtsp-client.c:
4918 * gst/rtsp-server/rtsp-client.h:
4919 * tests/check/gst/client.c:
4920 rtsp-client: expose uri
4922 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
4924 * tests/check/gst/mediafactory.c:
4925 tests: Hold ref while creating second media
4926 To test if the media aren't shared, make sure we keep the first one while creating a second
4927 otherwise the same memory address may be reused.
4929 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
4932 configure: remove out-of-date comment
4934 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
4937 .gitignore: ignore more build files
4939 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
4941 * tests/check/Makefile.am:
4942 tests: use right _LIBS variable for gst-plugins-base libs
4944 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4946 * tests/check/Makefile.am:
4947 check: add librtp to libs
4949 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
4951 * tests/check/gst/rtspserver.c:
4952 tests: Add test to check selecting a port the server will send from
4954 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
4956 * tests/check/gst/rtspserver.c:
4957 tests: Make sure packets are actually received
4959 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4961 * gst/rtsp-server/rtsp-stream.c:
4962 stream: Select unicast address from pool if appropriate
4964 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
4966 * gst/rtsp-server/rtsp-stream.c:
4967 stream: Properties are always there in Gst 1.0
4969 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4971 * tests/check/gst/addresspool.c:
4972 tests: Add tests for unicast addresses in pool
4974 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
4976 * gst/rtsp-server/rtsp-address-pool.c:
4977 * tests/check/gst/addresspool.c:
4978 address-pool: Verify that multicast addresses are used for multicast and vice-versa
4980 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
4982 * docs/libs/gst-rtsp-server-sections.txt:
4983 * gst/rtsp-server/rtsp-address-pool.c:
4984 * gst/rtsp-server/rtsp-address-pool.h:
4985 * gst/rtsp-server/rtsp-stream.c:
4986 * tests/check/gst/addresspool.c:
4987 address-pool: Add unicast addresses
4989 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
4992 * gst/rtsp-server/rtsp-server.c:
4993 * tests/check/gst/rtspserver.c:
4994 rtsp-server: Limit the number of threads per server instance
4995 If we exceed the maximum, just round robin the clients over the existing
4998 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
5000 * gst/rtsp-server/rtsp-server.c:
5001 rtsp-server: No need to store the GMainContext in the client context
5003 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
5005 * tests/check/gst/rtspserver.c:
5006 tests: Add test for client disconnection
5008 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
5010 * tests/check/gst/rtspserver.c:
5011 tests: Test client and session timeouts with multiple threads
5013 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
5015 * gst/rtsp-server/rtsp-address-pool.c:
5016 * gst/rtsp-server/rtsp-auth.c:
5017 * gst/rtsp-server/rtsp-client.c:
5018 * gst/rtsp-server/rtsp-media-factory-uri.c:
5019 * gst/rtsp-server/rtsp-media-factory.c:
5020 * gst/rtsp-server/rtsp-media.c:
5021 * gst/rtsp-server/rtsp-mount-points.c:
5022 * gst/rtsp-server/rtsp-server.c:
5023 * gst/rtsp-server/rtsp-session-media.c:
5024 * gst/rtsp-server/rtsp-session-pool.c:
5025 * gst/rtsp-server/rtsp-session.c:
5026 Document locking and its order
5028 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
5030 * tests/check/gst/rtspserver.c:
5031 tests: Test that slow DESCRIBE don't block other clients
5033 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
5035 * tests/check/gst/client.c:
5036 tests: Add tests for client-requested multicast address
5038 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
5040 * docs/libs/gst-rtsp-server-sections.txt:
5041 docs: Put the various functions in the right sections
5043 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
5045 * docs/libs/gst-rtsp-server-docs.sgml:
5046 * docs/libs/gst-rtsp-server-sections.txt:
5047 * gst/rtsp-server/rtsp-address-pool.c:
5048 * gst/rtsp-server/rtsp-address-pool.h:
5049 docs: Generate docs for GstRTSPAddressPool
5051 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
5053 * gst/rtsp-server/rtsp-client.c:
5054 * gst/rtsp-server/rtsp-stream.c:
5055 * gst/rtsp-server/rtsp-stream.h:
5056 client: Check client provided addresses against the address pool
5058 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
5060 * gst/rtsp-server/rtsp-address-pool.c:
5061 * gst/rtsp-server/rtsp-address-pool.h:
5062 * tests/check/gst/addresspool.c:
5063 address-pool: Add API to request a specific address from the pool
5064 Also add relevant unit tests.
5066 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
5068 * tests/check/gst/mediafactory.c:
5069 tests: Check the passing around of a RTSPAddressPool
5070 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
5071 way down to the stream.
5073 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
5075 * tests/check/gst/addresspool.c:
5076 tests: Add more tests for the address pool
5078 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
5080 * gst/rtsp-server/rtsp-address-pool.c:
5081 address-pool: Fix off by one error
5082 When splitting a port range, the port after a skip is not part of range.
5084 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
5087 Automatic update of common submodule
5088 From 2de221c to 04c7a1e
5090 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
5093 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
5094 AM_CONFIG_HEADER was removed in automake 1.13
5095 https://bugzilla.gnome.org/show_bug.cgi?id=693368
5097 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
5100 Automatic update of common submodule
5101 From a942293 to 2de221c
5103 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5105 * gst/rtsp-server/rtsp-client.c:
5106 client: make sure the watch exists while sending data
5107 Protect the send_func with a lock. This allows us to wait for sending
5108 to complete before changing the send_func and user_data. We add an
5109 extra ref to the watch to make sure that it remains valid during
5111 When closing the connection, set the send_func to NULL
5112 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
5114 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5116 * tests/check/Makefile.am:
5117 tests: use GST_*_1_0 environment variables everywhere
5118 The _1_0 suffixed environment variables override the
5119 non-suffixed ones, so if we're in an environment that
5120 sets the _1_0 suffixed ones, such as jhbuild, we need
5121 to set those to make sure ours actually always get
5124 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5127 Automatic update of common submodule
5128 From acb04d9 to a942293
5130 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5132 * gst/rtsp-server/rtsp-client.c:
5133 rtsp-client: set the client backlog
5134 Set the client backlog to a reasonable default
5136 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
5138 * gst/rtsp-server/rtsp-media.c:
5139 rtsp-media: Make the element a constructor parameter
5140 https://bugzilla.gnome.org/show_bug.cgi?id=689594
5142 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5144 * docs/libs/Makefile.am:
5145 docs: Link with gcov library when gcov is enabled
5146 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
5148 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5150 * gst/rtsp-server/rtsp-media.c:
5151 media: match prepare with unprepare
5152 Really unprepare when there were an equal amount of prepare calls.
5154 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5156 * gst/rtsp-server/rtsp-media.c:
5157 media: media has to be unprepared in finalize
5158 Because unprepare takes away the last ref on the media.
5160 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5162 * gst/rtsp-server/rtsp-client.c:
5163 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
5164 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
5165 We can't use the refcount to trigger unprepare because it is the unprepare call
5166 that removes the last refcount after all messages are consumed. What we should
5167 probably do is make a prepared refcount and only unprepare when the refcount
5170 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5172 * gst/rtsp-server/rtsp-media.c:
5173 media: let the source unref the last media ref
5174 the last ref to the media is held by the source so we don't need to add more ref
5175 and unrefs, we simply destroy the media when the source is gone.
5177 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5179 * gst/rtsp-server/rtsp-media.c:
5180 media: improve debug
5182 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5184 * gst/rtsp-server/rtsp-media.c:
5186 Make sure we are in the right state when collecting the position and duration.
5187 Only make ourselves PREPARED when we were previously PREPARING.
5189 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5191 * gst/rtsp-server/rtsp-media.c:
5192 media: use g_object_ref/unref for GObjects
5194 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
5196 * gst/rtsp-server/rtsp-client.c:
5197 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
5198 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
5199 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
5200 isn't being used anymore.
5202 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
5204 * gst/rtsp-server/rtsp-media.c:
5205 Fix compiler warning
5207 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
5209 * gst/rtsp-server/rtsp-media-factory-uri.c:
5210 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
5212 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5214 * gst/rtsp-server/rtsp-session-media.h:
5217 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5219 * gst/rtsp-server/rtsp-media.c:
5220 * tests/check/gst/media.c:
5221 media: avoid element leak
5223 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5225 * gst/rtsp-server/rtsp-media.c:
5226 media: require an element in media constructor
5228 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5230 * gst/rtsp-server/rtsp-client.c:
5231 Revert "client: TEARDOWN brings that state to Init again"
5232 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
5233 The object is already disposed, there is no point in setting the state.
5235 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5237 * gst/rtsp-server/rtsp-client.c:
5238 client: TEARDOWN brings that state to Init again
5240 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5242 * docs/libs/gst-rtsp-server-sections.txt:
5243 * examples/test-auth.c:
5244 * gst/rtsp-server/rtsp-auth.c:
5245 * gst/rtsp-server/rtsp-auth.h:
5246 * gst/rtsp-server/rtsp-client.c:
5247 * gst/rtsp-server/rtsp-client.h:
5248 * gst/rtsp-server/rtsp-media-factory-uri.c:
5249 * gst/rtsp-server/rtsp-media-factory-uri.h:
5250 * gst/rtsp-server/rtsp-media-factory.c:
5251 * gst/rtsp-server/rtsp-media-factory.h:
5252 * gst/rtsp-server/rtsp-media.c:
5253 * gst/rtsp-server/rtsp-media.h:
5254 * gst/rtsp-server/rtsp-mount-points.c:
5255 * gst/rtsp-server/rtsp-mount-points.h:
5256 * gst/rtsp-server/rtsp-sdp.c:
5257 * gst/rtsp-server/rtsp-server.c:
5258 * gst/rtsp-server/rtsp-server.h:
5259 * gst/rtsp-server/rtsp-session-media.c:
5260 * gst/rtsp-server/rtsp-session-media.h:
5261 * gst/rtsp-server/rtsp-session-pool.c:
5262 * gst/rtsp-server/rtsp-session-pool.h:
5263 * gst/rtsp-server/rtsp-session.c:
5264 * gst/rtsp-server/rtsp-session.h:
5265 * gst/rtsp-server/rtsp-stream-transport.c:
5266 * gst/rtsp-server/rtsp-stream-transport.h:
5267 * gst/rtsp-server/rtsp-stream.c:
5268 * gst/rtsp-server/rtsp-stream.h:
5269 * tests/check/gst/media.c:
5270 rtsp: make object details private
5271 Make all object details private
5272 Add methods to access private bits
5274 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5276 * tests/check/Makefile.am:
5277 * tests/check/gst/media.c:
5278 tests: add media tests
5280 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5282 * gst/rtsp-server/rtsp-media.c:
5283 media: check if prepared for some methods
5284 Check that the media object is prepared before doing seek and getting the
5285 current position etc.
5286 Add some g_return checks.
5288 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5290 * tests/check/Makefile.am:
5291 * tests/check/gst/mediafactory.c:
5292 tests: add mediafactory test
5294 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5296 * gst/rtsp-server/rtsp-stream.c:
5297 stream: improve debug
5299 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5301 * gst/rtsp-server/rtsp-media.c:
5302 * gst/rtsp-server/rtsp-media.h:
5303 media: unref pipeline in finalize to avoid leaking it
5305 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5307 * gst/rtsp-server/rtsp-media-factory-uri.c:
5308 * gst/rtsp-server/rtsp-media.c:
5309 rtsp: use gst_object_unref on GstObjects
5311 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5313 * gst/rtsp-server/rtsp-media-factory.c:
5314 media-factory: require an url
5316 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5318 * examples/test-uri.c:
5319 examples: fix include
5321 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5323 * gst/rtsp-server/rtsp-server.h:
5324 server: remove unused include
5326 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5328 * tests/check/Makefile.am:
5329 * tests/check/gst/mountpoints.c:
5330 tests: add test for mountpoints
5332 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5334 * gst/rtsp-server/rtsp-client.c:
5335 client: fix factory leak
5336 Keep the factory in the state object only for authorization checks and make
5337 sure we unref it on failure. Also don't keep invalid objects in the state
5340 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5342 * gst/rtsp-server/rtsp-mount-points.c:
5343 mounts: add g_return_if guards
5345 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5347 * tests/check/gst/client.c:
5348 tests: add more tests
5350 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5352 * gst/rtsp-server/rtsp-client.c:
5353 client: improve debug
5355 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5357 * gst/rtsp-server/rtsp-client.c:
5358 client: improve debug and fix leaks
5359 Cleanup the uri and session when there is a bad request.
5361 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5366 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5368 * tests/check/gst/client.c:
5369 test: add test for session in options request
5371 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5373 * gst/rtsp-server/rtsp-client.c:
5374 client: use 454 when session can't be found
5375 We should use 454 when a session can't be found because there was no session
5376 pool configured in the server. This is not a server configuration problem
5377 because the server on which the request is done might not be the same one that
5378 will keep the sessions for us and so it does not need to support sessions.
5380 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5382 * gst/rtsp-server/rtsp-client.c:
5383 client: only free connection when there is one
5384 It's possible that the client doesn't have a connection when we try to free it.
5386 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5388 * tests/check/Makefile.am:
5389 * tests/check/gst/client.c:
5390 tests: add unit test for the client object
5392 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5394 * gst/rtsp-server/rtsp-client.c:
5395 client: small cleanup
5397 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5399 * gst/rtsp-server/rtsp-client.h:
5400 client: remove unused include
5402 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5404 * gst/rtsp-server/rtsp-client.c:
5405 client: fix compilation
5407 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5409 * gst/rtsp-server/rtsp-client.c:
5410 client: call destroy without the lock
5412 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5414 * gst/rtsp-server/rtsp-client.c:
5415 * gst/rtsp-server/rtsp-client.h:
5416 client: make the client usable without a socket
5417 Make a method to let the client handle a message and a callback when the client
5418 wants us to send a response message back. This makes it possible to also use the
5419 client object without the sockets, which should make it easier to test.
5421 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5423 * gst/rtsp-server/rtsp-client.c:
5424 * gst/rtsp-server/rtsp-client.h:
5425 client: small cleanup
5427 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5429 * docs/libs/gst-rtsp-server-sections.txt:
5430 * gst/rtsp-server/rtsp-client.c:
5431 * gst/rtsp-server/rtsp-client.h:
5432 * gst/rtsp-server/rtsp-server.c:
5433 client: remove reference to server
5434 We don't need to keep a ref to the server
5436 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5438 * gst/rtsp-server/rtsp-client.c:
5439 * gst/rtsp-server/rtsp-client.h:
5441 Also add some g_return_if()
5443 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5445 * gst/rtsp-server/rtsp-client.c:
5446 client: log more errors
5448 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5450 * gst/rtsp-server/rtsp-client.c:
5451 client: fix compilation
5453 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5455 * gst/rtsp-server/rtsp-client.c:
5456 * gst/rtsp-server/rtsp-client.h:
5457 client: add generic close-after-send support
5458 Add a property to send_response() to close the connection after the response has
5459 been sent to the client.
5461 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5464 * docs/libs/gst-rtsp-server-docs.sgml:
5465 * docs/libs/gst-rtsp-server-sections.txt:
5466 * docs/libs/gst-rtsp-server.types:
5467 * examples/test-auth.c:
5468 * examples/test-launch.c:
5469 * examples/test-mp4.c:
5470 * examples/test-multicast.c:
5471 * examples/test-multicast2.c:
5472 * examples/test-ogg.c:
5473 * examples/test-readme.c:
5474 * examples/test-sdp.c:
5475 * examples/test-uri.c:
5476 * examples/test-video.c:
5477 * gst/rtsp-server/Makefile.am:
5478 * gst/rtsp-server/rtsp-auth.h:
5479 * gst/rtsp-server/rtsp-client.c:
5480 * gst/rtsp-server/rtsp-client.h:
5481 * gst/rtsp-server/rtsp-media-mapping.c:
5482 * gst/rtsp-server/rtsp-media-mapping.h:
5483 * gst/rtsp-server/rtsp-mount-points.c:
5484 * gst/rtsp-server/rtsp-mount-points.h:
5485 * gst/rtsp-server/rtsp-server.c:
5486 * gst/rtsp-server/rtsp-server.h:
5487 * gst/rtsp-server/rtsp-session-media.c:
5488 * gst/rtsp-server/rtsp-session-pool.c:
5489 * gst/rtsp-server/rtsp-session-pool.h:
5490 * tests/check/gst/rtspserver.c:
5491 MediaMapping -> MountPoints
5492 Describes better what the object manages.
5494 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5497 configure: bump required version of -base
5499 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5501 * gst/rtsp-server/rtsp-media.c:
5504 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5506 * gst/rtsp-server/rtsp-media.c:
5507 * gst/rtsp-server/rtsp-media.h:
5508 media: support more Range formats
5509 Use the new -base methods to convert the Range string into a seek start and stop
5512 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5514 * examples/test-launch.c:
5515 examples: fix whitespace
5517 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5519 * examples/test-auth.c:
5520 test-auth: add example of how to remove sessions
5521 Add an example of the session filter api.
5523 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5525 * examples/test-uri.c:
5526 test-uri: remove mapping example
5528 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5530 * examples/test-uri.c:
5531 test-uri: fix callback signature
5533 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5535 * gst/rtsp-server/rtsp-media-factory.c:
5536 factory: keep ref to factory while media active
5537 While the media from a factory is alive, keep a ref to the factory.
5538 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
5540 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5542 * gst/rtsp-server/rtsp-media-factory-uri.c:
5543 factory-uri: add some debug
5545 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5547 * gst/rtsp-server/rtsp-stream.c:
5548 stream: set udp sources to PLAYING
5549 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
5550 so that it doesn't cause our pipeline to produce ASYNC-DONE.
5552 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5554 * gst/rtsp-server/rtsp-media-factory-uri.c:
5555 factory-uri: take ref to factory
5556 Take a ref to the factory that we place in our list.
5558 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5560 * tests/Makefile.am:
5561 * tests/test-reuse.c:
5562 test: add test for server reuse
5563 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
5565 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
5567 * gst/rtsp-server/rtsp-server.c:
5568 server: start and stop multiple times
5569 Stop listening on the RTSP port when the GSource is removed, so clients
5570 can't connect and the server can be started again.
5571 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
5573 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5575 * gst/rtsp-server/rtsp-server.c:
5576 server: fix small leak
5578 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5580 * gst/rtsp-server/rtsp-media.c:
5581 media: unref source in finish_unprepare
5582 The source is created in prepare, unref it in finish_unprepare.
5583 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
5585 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
5587 * gst/rtsp-server/rtsp-client.c:
5588 * gst/rtsp-server/rtsp-media.c:
5589 rtsp-media: remove bus watch before finalizing
5590 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
5591 * An extra media ref is added for the bus watch. This extra ref is unreffed by
5592 the GDestroyNotify function.
5593 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
5594 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
5595 gst_rtsp_media_unprepare before unreffing the media.
5596 This way, the bus watch will be removed before the media is finalized.
5597 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
5599 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
5601 * gst/rtsp-server/rtsp-client.c:
5602 * gst/rtsp-server/rtsp-client.h:
5603 client: wait until the TEARDOWN response is sent to close the connection
5604 Responses can be sent async so we need to wait until the TEARDOWN response has
5605 been written before we close the connection to the client. This avoids the risk
5606 of writing/polling closed sockets.
5607 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
5609 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
5611 * gst/rtsp-server/rtsp-stream.c:
5612 rtsp-stream: plug socket leak
5613 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
5615 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
5618 Automatic update of common submodule
5619 From 6bb6951 to a72faea
5621 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
5623 * gst/rtsp-server/rtsp-media-factory-uri.c:
5624 rtsp-server: don't use deprecated API
5626 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
5628 * gst/rtsp-server/rtsp-client.c:
5629 rtsp-client: fix unused-but-set-variable compiler warning
5630 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
5632 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5635 * docs/libs/gst-rtsp-server-sections.txt:
5636 * gst/rtsp-server/rtsp-client.c:
5639 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5641 * examples/Makefile.am:
5642 * examples/test-multicast2.c:
5643 examples: add another multicast example
5644 Add an example for how to configure separate multicast ranges for each media
5647 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5649 * examples/test-multicast.c:
5652 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5654 * gst/rtsp-server/rtsp-client.c:
5655 * gst/rtsp-server/rtsp-media.c:
5656 * gst/rtsp-server/rtsp-session-media.c:
5657 * gst/rtsp-server/rtsp-session-media.h:
5658 * gst/rtsp-server/rtsp-stream-transport.c:
5659 * gst/rtsp-server/rtsp-stream-transport.h:
5660 stream: use the address managed by the stream
5661 Use the address managed by the stream for multicast. This allows us to have 1
5662 multicast address for each stream.
5663 Because the address is now managed by the stream we don't have to pass it around
5665 Set the address pool on the streams.
5667 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5669 * gst/rtsp-server/rtsp-client.c:
5670 * gst/rtsp-server/rtsp-media.c:
5671 * gst/rtsp-server/rtsp-stream.c:
5674 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5676 * gst/rtsp-server/rtsp-media.c:
5677 * gst/rtsp-server/rtsp-media.h:
5678 media: add signal for new streams
5679 This allows applications to listen for new streams and configure properties on
5680 them, like the address pool.
5682 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5684 * gst/rtsp-server/rtsp-media.c:
5685 media: configure address pool in new streams
5687 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5689 * gst/rtsp-server/rtsp-stream.c:
5690 * gst/rtsp-server/rtsp-stream.h:
5691 stream: add methods to deal with address pool
5692 Add methods to get and set the address pool for the stream
5693 Add method to allocate and get the multicast addresses for this stream.
5695 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5697 * docs/libs/gst-rtsp-server-sections.txt:
5698 * gst/rtsp-server/rtsp-media.c:
5699 * gst/rtsp-server/rtsp-media.h:
5700 media: remove MTU property
5701 It is a stream property
5703 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5705 * gst/rtsp-server/rtsp-client.c:
5706 client: set blocksize only on stream
5707 Set the blocksize only on the current stream.
5709 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5711 * gst/rtsp-server/rtsp-stream.c:
5712 stream: share src and sink sockets
5713 the allocated socket is in the used-socket property, not socket.
5715 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5717 * gst/rtsp-server/rtsp-address-pool.c:
5718 * gst/rtsp-server/rtsp-address-pool.h:
5719 * gst/rtsp-server/rtsp-client.c:
5720 * gst/rtsp-server/rtsp-session-media.c:
5721 * gst/rtsp-server/rtsp-session-media.h:
5722 * gst/rtsp-server/rtsp-stream-transport.c:
5723 * gst/rtsp-server/rtsp-stream-transport.h:
5724 * tests/check/gst/addresspool.c:
5725 rtsp: make address-pool return an address object
5726 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
5727 store more info in the structure and allows us to more easily return the address
5728 to the right pool when no longer needed.
5729 Pass the address to the StreamTransport so that we can return it to the pool
5730 when the stream transport is freed or changed.
5732 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5734 * examples/Makefile.am:
5735 * examples/test-multicast.c:
5736 examples: add multicast example
5737 Show how to set up the multicast address pool so that media can be
5738 server with multicast.
5740 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5742 * gst/rtsp-server/rtsp-client.c:
5743 * gst/rtsp-server/rtsp-media-factory.c:
5744 * gst/rtsp-server/rtsp-media-factory.h:
5745 * gst/rtsp-server/rtsp-media.c:
5746 * gst/rtsp-server/rtsp-media.h:
5747 rtsp: use AddressPool
5748 Remove the multicast_group property.
5749 Use the configured addresspool to allocate multicast addresses.
5751 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5753 * gst/rtsp-server/rtsp-address-pool.c:
5754 * gst/rtsp-server/rtsp-address-pool.h:
5755 address-pool: add clear method
5757 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5759 * gst/rtsp-server/rtsp-address-pool.c:
5760 address-pool: small cleanups
5762 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5764 * tests/check/Makefile.am:
5765 * tests/check/gst/addresspool.c:
5766 tests: add addresspool unit test
5768 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5770 * gst/rtsp-server/Makefile.am:
5771 * gst/rtsp-server/rtsp-address-pool.c:
5772 * gst/rtsp-server/rtsp-address-pool.h:
5773 address-pool: add object to manage multicast addresses
5774 Make an object that can manage a rage of multicast addresses and ports.
5776 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5778 * gst/rtsp-server/rtsp-server.c:
5779 server: set default max-threads property
5781 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5783 * gst/rtsp-server/rtsp-media.c:
5784 media: wait for concurrent _prepare
5785 If a prepare is busy, wait for the result.
5787 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5789 * gst/rtsp-server/rtsp-media.c:
5790 media: add lock around message handler
5791 We don't want to dispatch messages while we are still processing the result of
5794 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5796 * gst/rtsp-server/rtsp-media.c:
5797 * gst/rtsp-server/rtsp-media.h:
5798 media: add lock to protect state changes
5800 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5802 * gst/rtsp-server/rtsp-stream.c:
5803 * gst/rtsp-server/rtsp-stream.h:
5806 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5808 * gst/rtsp-server/rtsp-stream-transport.c:
5809 * gst/rtsp-server/rtsp-stream-transport.h:
5810 * gst/rtsp-server/rtsp-stream.c:
5811 stream-transport: add keep-alive method
5813 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5815 * gst/rtsp-server/rtsp-stream-transport.c:
5816 * gst/rtsp-server/rtsp-stream-transport.h:
5817 * gst/rtsp-server/rtsp-stream.c:
5818 stream-transport: add method to handle RTP/RTCP
5819 Call new methods instead of poking into the structures directly.
5821 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5823 * gst/rtsp-server/rtsp-session-media.c:
5824 * gst/rtsp-server/rtsp-session-media.h:
5825 session-media: add locking
5827 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5829 * gst/rtsp-server/rtsp-session.c:
5830 * gst/rtsp-server/rtsp-session.h:
5831 session: add locking
5833 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5835 * gst/rtsp-server/rtsp-server.c:
5836 server: free old socket
5838 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5840 * gst/rtsp-server/rtsp-media-mapping.c:
5841 * gst/rtsp-server/rtsp-media-mapping.h:
5842 mapping: add locking
5844 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5846 * gst/rtsp-server/rtsp-media-factory.c:
5847 media-factory: add locking
5849 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5851 * gst/rtsp-server/rtsp-auth.c:
5852 * gst/rtsp-server/rtsp-auth.h:
5855 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5857 * gst/rtsp-server/rtsp-server.c:
5858 * gst/rtsp-server/rtsp-server.h:
5859 server: add max-thread property
5861 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5863 * gst/rtsp-server/rtsp-server.c:
5864 * gst/rtsp-server/rtsp-server.h:
5865 server: use a threadpool for the mainloops
5867 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5869 * gst/rtsp-server/rtsp-client.c:
5870 * gst/rtsp-server/rtsp-client.h:
5871 client: rename method
5872 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
5873 don't really create the client from the socket, we use the socket for the
5876 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5878 * gst/rtsp-server/rtsp-client.c:
5879 * gst/rtsp-server/rtsp-client.h:
5880 * gst/rtsp-server/rtsp-server.c:
5881 server: rework maincontext handling in clients
5882 Make a separate method to attach a client to a MainContext.
5883 Let the server decide in what GMainContext the client will operate and give this
5884 context to the client in attach. Then the server can later decide to use a
5885 separate thread for each client or just use the mainthread.
5887 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5889 * gst/rtsp-server/rtsp-client.c:
5890 * gst/rtsp-server/rtsp-session.c:
5891 * gst/rtsp-server/rtsp-session.h:
5892 session: move session header code in session object
5894 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
5898 * examples/test-auth.c:
5899 * examples/test-launch.c:
5900 * examples/test-mp4.c:
5901 * examples/test-ogg.c:
5902 * examples/test-readme.c:
5903 * examples/test-sdp.c:
5904 * examples/test-uri.c:
5905 * examples/test-video.c:
5906 * gst/rtsp-server/rtsp-auth.c:
5907 * gst/rtsp-server/rtsp-auth.h:
5908 * gst/rtsp-server/rtsp-client.c:
5909 * gst/rtsp-server/rtsp-client.h:
5910 * gst/rtsp-server/rtsp-media-factory-uri.c:
5911 * gst/rtsp-server/rtsp-media-factory-uri.h:
5912 * gst/rtsp-server/rtsp-media-factory.c:
5913 * gst/rtsp-server/rtsp-media-factory.h:
5914 * gst/rtsp-server/rtsp-media-mapping.c:
5915 * gst/rtsp-server/rtsp-media-mapping.h:
5916 * gst/rtsp-server/rtsp-media.c:
5917 * gst/rtsp-server/rtsp-media.h:
5918 * gst/rtsp-server/rtsp-params.c:
5919 * gst/rtsp-server/rtsp-params.h:
5920 * gst/rtsp-server/rtsp-sdp.c:
5921 * gst/rtsp-server/rtsp-sdp.h:
5922 * gst/rtsp-server/rtsp-server.c:
5923 * gst/rtsp-server/rtsp-server.h:
5924 * gst/rtsp-server/rtsp-session-media.c:
5925 * gst/rtsp-server/rtsp-session-media.h:
5926 * gst/rtsp-server/rtsp-session-pool.c:
5927 * gst/rtsp-server/rtsp-session-pool.h:
5928 * gst/rtsp-server/rtsp-session.c:
5929 * gst/rtsp-server/rtsp-session.h:
5930 * gst/rtsp-server/rtsp-stream-transport.c:
5931 * gst/rtsp-server/rtsp-stream-transport.h:
5932 * gst/rtsp-server/rtsp-stream.c:
5933 * gst/rtsp-server/rtsp-stream.h:
5934 * tests/check/gst/rtspserver.c:
5935 * tests/test-cleanup.c:
5938 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
5940 * gst/rtsp-server/rtsp-media.c:
5941 * gst/rtsp-server/rtsp-session-media.c:
5942 * gst/rtsp-server/rtsp-session.c:
5943 rtsp-server: added annotations to indicate type of ownership transfer of return values
5944 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5946 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
5949 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
5951 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
5954 * bindings/Makefile.am:
5955 * bindings/vala/Makefile.am:
5956 * bindings/vala/gst-rtsp-server-0.10.deps:
5957 * bindings/vala/gst-rtsp-server-0.10.vapi:
5958 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
5959 * bindings/vala/packages/gst-rtsp-server-0.10.files:
5960 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
5961 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
5962 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
5964 bindings: remove vala bindings
5965 They'll be reunited with the other GStreamer bindings
5966 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5968 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5970 * gst/rtsp-server/rtsp-client.c:
5971 * gst/rtsp-server/rtsp-session-media.c:
5972 * gst/rtsp-server/rtsp-session-media.h:
5973 * gst/rtsp-server/rtsp-stream-transport.c:
5974 * gst/rtsp-server/rtsp-stream-transport.h:
5975 rtsp: only create transport when needed
5976 Only create the StreamTransport when configured.
5978 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5980 * gst/rtsp-server/rtsp-client.c:
5981 client: small cleanup
5983 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5985 * gst/rtsp-server/rtsp-client.c:
5986 * gst/rtsp-server/rtsp-client.h:
5987 * gst/rtsp-server/rtsp-stream-transport.c:
5988 * gst/rtsp-server/rtsp-stream-transport.h:
5989 rtsp: refactor configuration of transport
5990 Move the configuration of the transport to a place where it makes
5993 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5995 * gst/rtsp-server/rtsp-client.c:
5996 client: refactor transport parsing
5998 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6000 * gst/rtsp-server/rtsp-client.c:
6001 client: refuse to change the MTU on shared media
6002 If we change the MTU of chared media, it changes for all clients.
6003 We don't want to set the MTU to something large for clients that
6006 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6008 * examples/test-mp4.c:
6009 * gst/rtsp-server/rtsp-media.c:
6010 small fixes to docs and debug
6012 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6014 * gst/rtsp-server/rtsp-stream.c:
6015 stream: transports must already have been removed
6017 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6019 * gst/rtsp-server/rtsp-media.c:
6020 * gst/rtsp-server/rtsp-stream.c:
6021 * gst/rtsp-server/rtsp-stream.h:
6022 stream: improve join and leave of the pipeline
6024 Do the cleanup properly
6027 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6029 * gst/rtsp-server/rtsp-media.c:
6030 media: move unprepare below default implementation
6031 Makes it easier to find the default implementation
6033 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6035 * gst/rtsp-server/rtsp-media.c:
6036 media: signal unprepared when we actually finish
6038 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6040 * gst/rtsp-server/rtsp-media.c:
6041 media: no need to unlock, unprepare does that when needed
6043 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6045 * docs/libs/gst-rtsp-server-sections.txt:
6046 * gst/rtsp-server/rtsp-media-factory.h:
6047 * gst/rtsp-server/rtsp-media-mapping.c:
6048 * gst/rtsp-server/rtsp-media.h:
6049 * gst/rtsp-server/rtsp-params.c:
6050 * gst/rtsp-server/rtsp-server.c:
6051 * gst/rtsp-server/rtsp-session-pool.h:
6052 * gst/rtsp-server/rtsp-session.c:
6053 * gst/rtsp-server/rtsp-session.h:
6054 * gst/rtsp-server/rtsp-stream-transport.h:
6055 * gst/rtsp-server/rtsp-stream.h:
6058 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6060 * gst/rtsp-server/rtsp-client.c:
6061 * gst/rtsp-server/rtsp-media-mapping.h:
6062 * gst/rtsp-server/rtsp-media.c:
6063 * gst/rtsp-server/rtsp-media.h:
6064 * gst/rtsp-server/rtsp-server.h:
6065 * gst/rtsp-server/rtsp-stream.c:
6066 * gst/rtsp-server/rtsp-stream.h:
6067 rtsp: fix MTU setting
6068 Fix setting of the MTU. There is no need for a vmethod.
6070 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6075 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6078 configure: bump version number after refactoring
6080 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6082 * gst/rtsp-server/Makefile.am:
6083 * gst/rtsp-server/rtsp-client.c:
6084 * gst/rtsp-server/rtsp-client.h:
6085 * gst/rtsp-server/rtsp-media-factory-uri.c:
6086 * gst/rtsp-server/rtsp-media-factory.c:
6087 * gst/rtsp-server/rtsp-media-factory.h:
6088 * gst/rtsp-server/rtsp-media.c:
6089 * gst/rtsp-server/rtsp-media.h:
6090 * gst/rtsp-server/rtsp-sdp.c:
6091 * gst/rtsp-server/rtsp-session-media.c:
6092 * gst/rtsp-server/rtsp-session-media.h:
6093 * gst/rtsp-server/rtsp-session.c:
6094 * gst/rtsp-server/rtsp-session.h:
6095 * gst/rtsp-server/rtsp-stream-transport.c:
6096 * gst/rtsp-server/rtsp-stream-transport.h:
6097 * gst/rtsp-server/rtsp-stream.c:
6098 * gst/rtsp-server/rtsp-stream.h:
6099 rtsp: massive refactoring
6100 Make GObjects from the remaining simple structures.
6101 Remove GstRTSPSessionStream, it's not needed.
6102 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
6103 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
6104 a GstRTSPStream should be transported to a client.
6105 Rename GstRTSPMediaFactory::get_element -> create_element because that
6106 more accurately describes what it does.
6107 Make nice methods instead of poking in the structures.
6108 Move some methods inside the relevant object source code.
6109 Use GPtrArray to store objects instead of plain arrays, it is more
6110 natural and allows us to more easily clean up.
6111 Move the allocation of udp ports to the Stream object. The Stream object
6112 contains the elements needed to stream the media to a client.
6113 Improve the prepare and unprepare methods. Unprepare should now undo
6114 everything prepare did. Improve also async unprepare when doing EOS on
6115 shutdown. Make sure we always unprepare correctly.
6117 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
6119 * gst/rtsp-server/rtsp-client.c:
6120 rtsp-client: Unref server address clients connected to
6121 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
6123 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
6125 * gst/rtsp-server/rtsp-server.c:
6126 rtsp-server: don't ref server socket if it is NULL
6127 Fixes test_bind_already_in_use unit test again after commit 6a497440.
6128 https://bugzilla.gnome.org/show_bug.cgi?id=686644
6130 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
6132 * tests/check/Makefile.am:
6133 tests: Add libgio link dependency
6134 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
6136 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6138 * gst/rtsp-server/rtsp-media-mapping.c:
6139 * gst/rtsp-server/rtsp-media-mapping.h:
6140 rtsp-media-mapping: rename find_media vfunc to find_factory
6141 The virtual method and class method should have the same name
6142 so it is correctly represented in GIR file
6143 https://bugzilla.gnome.org/show_bug.cgi?id=680777
6145 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6147 * gst/rtsp-server/rtsp-auth.c:
6148 * gst/rtsp-server/rtsp-client.c:
6149 * gst/rtsp-server/rtsp-media-factory-uri.c:
6150 * gst/rtsp-server/rtsp-media-factory.c:
6151 * gst/rtsp-server/rtsp-media-mapping.c:
6152 * gst/rtsp-server/rtsp-media.c:
6153 * gst/rtsp-server/rtsp-server.c:
6154 * gst/rtsp-server/rtsp-session-pool.c:
6155 * gst/rtsp-server/rtsp-session.c:
6156 rtsp-server: fixed comments and GIR annotations
6157 https://bugzilla.gnome.org/show_bug.cgi?id=680777
6159 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
6161 * gst/rtsp-server/rtsp-media-mapping.c:
6162 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
6164 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
6166 * gst/rtsp-server/rtsp-server.c:
6167 rtsp-server: allow binding on port 0 (binds on a random port)
6169 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
6171 * gst/rtsp-server/rtsp-server.c:
6172 * gst/rtsp-server/rtsp-server.h:
6173 rtsp-server: add bound-port property
6174 bound-port can be used to retrieve the port number when the server is bound on
6175 port 0, which binds on a random port.
6177 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
6179 * gst/rtsp-server/rtsp-media-factory.c:
6180 * gst/rtsp-server/rtsp-media-factory.h:
6181 rtsp-media-factory: make ::get_element overridable by GI bindings
6182 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
6183 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
6184 as the invoker for ::get_element(), making it overridable by GI generated
6187 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
6189 * gst/rtsp-server/rtsp-media-factory-uri.c:
6190 rtsp-media-factory-uri: don't autoplug parsers in a loop
6191 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
6194 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
6196 * gst/rtsp-server/Makefile.am:
6197 Explicitly link against gio. Fix link error on mac.
6199 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
6201 * gst/rtsp-server/rtsp-session.c:
6202 session: add ttl to the transport header in SETUP
6203 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
6205 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
6207 * gst/rtsp-server/rtsp-client.c:
6208 * gst/rtsp-server/rtsp-client.h:
6209 * gst/rtsp-server/rtsp-media.c:
6210 client: Use client transport settings for multicast if allowed.
6211 This patch makes it possible for the client to send transport settings for
6212 multicast (destination && ttl). Client settings must be explicitly allowed or
6213 the server will use its own settings.
6214 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
6216 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
6219 Automatic update of common submodule
6220 From 6c0b52c to 6bb6951
6222 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
6224 * gst/rtsp-server/rtsp-client.c:
6225 rtsp-client: do not destroy the rtsp watch
6226 Don't destroy the client watch while dispatching. The rtsp watch is
6227 automatically destroyed after the rtsp watch function closed() has
6229 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
6231 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
6234 Automatic update of common submodule
6235 From 4f962f7 to 6c0b52c
6237 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
6239 * gst/rtsp-server/rtsp-media.c:
6240 media: fix check for seekability
6242 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6244 * gst/rtsp-server/rtsp-client.c:
6245 client: use more GIO
6246 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
6248 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6250 * gst/rtsp-server/rtsp-server.c:
6251 server: remove obsolete includes
6253 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
6255 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
6256 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
6257 be available in "on_new_ssrc". The transports are added in
6258 gst_rtsp_media_set_state when going to PLAYING state. However,
6259 "on_new_ssrc" might be called before this happens.
6260 https://bugzilla.gnome.org/show_bug.cgi?id=683304
6262 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
6264 * gst/rtsp-server/rtsp-client.c:
6265 * gst/rtsp-server/rtsp-client.h:
6266 rtsp-client: add signals for rtsp requests (fixes #683287)
6268 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
6270 * gst/rtsp-server/rtsp-client.c:
6271 * gst/rtsp-server/rtsp-client.h:
6272 add new-session signal to rtsp-client (fixes #683058)
6274 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
6277 Automatic update of common submodule
6278 From 668acee to 4f962f7
6280 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
6282 * gst/rtsp-server/rtsp-server.c:
6283 * tests/check/gst/rtspserver.c:
6284 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
6285 Do not assume that *error is set in g_socket_address_enumerator_next.
6286 Added test_bind_already_in_use unit-test.
6287 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
6289 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
6292 Automatic update of common submodule
6293 From 94ccf4c to 668acee
6295 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
6297 * gst/rtsp-server/rtsp-client.c:
6298 * gst/rtsp-server/rtsp-client.h:
6299 rtsp-client: make create_sdp virtual method
6300 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
6302 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6305 Automatic update of common submodule
6306 From 98e386f to 94ccf4c
6308 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6310 * gst/rtsp-server/rtsp-client.c:
6313 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
6315 * gst/rtsp-server/rtsp-client.c:
6316 * gst/rtsp-server/rtsp-client.h:
6317 * gst/rtsp-server/rtsp-server.c:
6318 * gst/rtsp-server/rtsp-server.h:
6319 rtsp-server: use an existing socket to establish HTTP tunnel
6320 Make it possible to transfer a socket from an HTTP server to be used as
6321 an RTSP over HTTP tunnel.
6323 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
6325 * gst/rtsp-server/rtsp-client.c:
6326 * gst/rtsp-server/rtsp-media.c:
6327 * gst/rtsp-server/rtsp-media.h:
6328 rtsp: Handle the blocksize parameter
6329 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
6331 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
6333 * tests/check/Makefile.am:
6334 * tests/check/gst/rtspserver.c:
6335 Have unit test get header from source dir, not installed dir
6336 This makes compilation of unit tests work in a build directory other
6337 than the source directory.
6338 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
6340 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
6342 * gst/rtsp-server/rtsp-media.c:
6343 rtsp-media: update for gst_element_make_from_uri() changes
6345 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
6348 * tests/Makefile.am:
6349 * tests/check/Makefile.am:
6350 * tests/check/gst/rtspserver.c:
6352 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
6354 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
6356 * gst/rtsp-server/rtsp-media.c:
6357 rtsp-media: don't collect media stats when going to NULL
6358 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
6360 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6362 * gst/rtsp-server/rtsp-client.c:
6363 client: don't leak transports
6365 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
6367 * gst/rtsp-server/rtsp-client.c:
6368 rtsp-client: free transport on no_stream in SETUP handler
6370 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
6372 * gst/rtsp-server/rtsp-client.c:
6373 rtsp-client: changed session media iteration
6374 In client_unlink_session: now don't iterate in session->medias
6375 list where items are removed by gst_rtsp_session_release_media.
6376 Instead, repeatedly remove the first item.
6378 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
6380 * gst/rtsp-server/rtsp-client.c:
6381 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
6382 GstRTSPSessionMedia is not a GObject type. When the
6383 GstRTSPSession is freed, it will free the media.
6385 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
6387 * gst/rtsp-server/rtsp-media-factory.c:
6388 factory: plug pad leak in collect_streams
6389 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
6390 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
6391 will take one reference, and the other reference will otherwise
6394 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
6397 configure: suppress some warnings when debug is disabled
6398 Warnings about unused variables should be suppressed if core has the
6399 debug system disabled.
6400 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
6402 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6404 * docs/libs/Makefile.am:
6405 docs: fix build in uninstalled setup
6406 Include gst-plugins-base libs properly.
6408 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
6410 * docs/libs/gst-rtsp-server.types:
6411 docs: include headers defining rtsp-server object types
6412 Fixes compiler warnings during docs build.
6413 https://bugzilla.gnome.org/show_bug.cgi?id=676824
6415 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
6418 configure: Add warning flags for compiler when configuring
6419 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
6421 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6424 Automatic update of common submodule
6425 From 03a0e57 to 98e386f
6427 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6430 Automatic update of common submodule
6431 From 1fab359 to 03a0e57
6433 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
6435 * gst/rtsp-server/rtsp-client.c:
6436 client: fix GSocketAddress leak in gst_rtsp_client_accept
6437 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
6439 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6442 Automatic update of common submodule
6443 From f1b5a96 to 1fab359
6445 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6448 Automatic update of common submodule
6449 From 92b7266 to f1b5a96
6451 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6454 Automatic update of common submodule
6455 From ec1c4a8 to 92b7266
6457 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6460 Automatic update of common submodule
6461 From 3429ba6 to ec1c4a8
6463 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
6465 * gst/rtsp-server/rtsp-auth.c:
6466 * gst/rtsp-server/rtsp-client.c:
6467 * gst/rtsp-server/rtsp-media-factory-uri.c:
6468 * gst/rtsp-server/rtsp-server.c:
6469 rtsp: fix compiler warnings
6470 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
6472 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6475 Automatic update of common submodule
6476 From dc70203 to 3429ba6
6478 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6480 * gst/rtsp-server/rtsp-client.c:
6481 * gst/rtsp-server/rtsp-media-factory.c:
6482 * gst/rtsp-server/rtsp-media-factory.h:
6483 * gst/rtsp-server/rtsp-media.c:
6484 * gst/rtsp-server/rtsp-media.h:
6485 * gst/rtsp-server/rtsp-server.c:
6486 * gst/rtsp-server/rtsp-server.h:
6487 * gst/rtsp-server/rtsp-session-pool.c:
6488 * gst/rtsp-server/rtsp-session-pool.h:
6489 rtsp-server: port to new thread API
6491 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6494 Automatic update of common submodule
6495 From 6db25be to dc70203
6497 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6499 * gst/rtsp-server/rtsp-auth.c:
6500 * gst/rtsp-server/rtsp-auth.h:
6501 * gst/rtsp-server/rtsp-client.c:
6502 rtsp-server: Fix compilation and compiler warnings
6504 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6508 * gst/rtsp-server/Makefile.am:
6509 configure: Modernize autotools setup a bit
6510 Also we now only create tar.bz2 and tar.xz tarballs.
6512 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6515 Automatic update of common submodule
6516 From 464fe15 to 6db25be
6518 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6521 Automatic update of common submodule
6522 From 7fda524 to 464fe15
6524 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6527 * docs/libs/Makefile.am:
6528 * docs/version.entities.in:
6530 * gst/rtsp-server/Makefile.am:
6531 * pkgconfig/Makefile.am:
6532 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
6533 * pkgconfig/gstreamer-rtsp-server.pc.in:
6534 * tests/Makefile.am:
6535 rtsp-server: Update versioning
6537 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6539 Merge remote-tracking branch 'origin/0.10'
6541 gst/rtsp-server/rtsp-session-pool.c
6543 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6545 * gst/rtsp-server/rtsp-session-pool.c:
6546 rtsp-server: Don't use deprecated GLib API
6548 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6550 Replace master with 0.11
6552 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6554 Merge branch 'master' into 0.11
6556 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6558 Merge branch 'master' into 0.11
6560 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
6563 A couple minor typo fixes
6565 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6567 * gst/rtsp-server/rtsp-media.c:
6568 media: fix state of the appqueue
6570 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6572 * gst/rtsp-server/rtsp-media-factory-uri.c:
6573 factory: use videoconvert
6575 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6577 * gst/rtsp-server/rtsp-media-factory-uri.c:
6578 factory: change to new style caps
6580 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6582 * gst/rtsp-server/rtsp-client.c:
6583 * gst/rtsp-server/rtsp-client.h:
6584 * gst/rtsp-server/rtsp-media-factory-uri.c:
6585 * gst/rtsp-server/rtsp-media.c:
6586 * gst/rtsp-server/rtsp-server.c:
6587 * gst/rtsp-server/rtsp-server.h:
6588 * gst/rtsp-server/rtsp-session-pool.c:
6589 rtsp-server: port to GIO
6592 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6595 configure: fix build
6597 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6600 docs: fix for gst_rtsp_server_set_port() -> _set_service()
6601 https://bugzilla.gnome.org/show_bug.cgi?id=666548
6603 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6606 * examples/Makefile.am:
6607 First rule of gst-rtsp-server club: don't talk about gst-phonon
6609 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6612 * pkgconfig/Makefile.am:
6613 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
6614 * pkgconfig/gst-rtsp-server.pc.in:
6615 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
6616 * pkgconfig/gstreamer-rtsp-server.pc.in:
6617 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
6618 For consistency with all other modules.
6620 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6622 * gst/rtsp-server/rtsp-client.c:
6623 rtsp-client: update for new map API
6625 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6628 * bindings/Makefile.am:
6629 * bindings/python/Makefile.am:
6630 * bindings/python/arg-types.py:
6631 * bindings/python/codegen/Makefile.am:
6632 * bindings/python/codegen/__init__.py:
6633 * bindings/python/codegen/argtypes.py:
6634 * bindings/python/codegen/code-coverage.py:
6635 * bindings/python/codegen/codegen.py:
6636 * bindings/python/codegen/definitions.py:
6637 * bindings/python/codegen/defsparser.py:
6638 * bindings/python/codegen/docextract.py:
6639 * bindings/python/codegen/docgen.py:
6640 * bindings/python/codegen/fileprefix.override:
6641 * bindings/python/codegen/fileprefixmodule.c:
6642 * bindings/python/codegen/h2def.py:
6643 * bindings/python/codegen/mergedefs.py:
6644 * bindings/python/codegen/mkskel.py:
6645 * bindings/python/codegen/override.py:
6646 * bindings/python/codegen/reversewrapper.py:
6647 * bindings/python/codegen/scmexpr.py:
6648 * bindings/python/rtspserver-types.defs:
6649 * bindings/python/rtspserver.defs:
6650 * bindings/python/rtspserver.override:
6651 * bindings/python/rtspservermodule.c:
6652 * bindings/python/test.py:
6654 python: remove pygst-based python bindings
6655 pygi is the future, apparently.
6657 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
6660 Automatic update of common submodule
6661 From c463bc0 to 7fda524
6663 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6666 Automatic update of common submodule
6667 From 2a59016 to c463bc0
6669 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6672 Automatic update of common submodule
6673 From 0807187 to 2a59016
6675 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6678 Automatic update of common submodule
6679 From 11f0cd5 to 0807187
6681 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6683 * examples/test-auth.c:
6684 example: update for new caps
6686 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6688 * examples/test-video.c:
6689 * gst/rtsp-server/rtsp-client.c:
6690 * gst/rtsp-server/rtsp-media-factory-uri.c:
6691 * gst/rtsp-server/rtsp-media.c:
6692 * gst/rtsp-server/rtsp-media.h:
6693 * gst/rtsp-server/rtsp-session.c:
6694 * gst/rtsp-server/rtsp-session.h:
6695 rtsp-server: port some more to 0.11
6697 Remove bufferlist stuff
6699 Add queue before appsink now that preroll-queue-len is gone.
6700 Update for request pad changes.
6702 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6704 Merge branch 'master' into 0.11
6706 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
6708 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6709 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
6710 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6712 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
6714 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6715 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
6716 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6718 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6720 Merge branch 'master' into 0.11
6722 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6724 * gst/rtsp-server/rtsp-media.c:
6725 * gst/rtsp-server/rtsp-media.h:
6726 media: add a seekable boolean
6727 Maintain the seekable state with a new variable instead of reusing the
6730 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
6732 * gst/rtsp-server/rtsp-media.c:
6733 Disallow seek in live media
6735 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6737 Merge branch 'master' into 0.11
6739 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
6741 * gst/rtsp-server/rtsp-server.c:
6742 #ifdef statements for windows socket creation were missing
6744 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
6747 Automatic update of common submodule
6748 From a39eb83 to 11f0cd5
6750 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
6753 Automatic update of common submodule
6754 From 605cd9a to a39eb83
6756 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6758 Merge branch 'master' into 0.11
6760 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6762 * gst/rtsp-server/rtsp-client.c:
6763 client: use method to access property
6765 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6767 * gst/rtsp-server/rtsp-media-factory.c:
6768 * gst/rtsp-server/rtsp-media-factory.h:
6769 media-factory: add protocols property
6770 Add a property to configure the allowed protocols in the media created from the
6773 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6775 * gst/rtsp-server/rtsp-media-factory.c:
6776 * gst/rtsp-server/rtsp-media-factory.h:
6777 media-factory: add media-configure signal
6778 Add signal to allow the application to configure the media after it was created
6781 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6783 * gst/rtsp-server/rtsp-client.c:
6784 client: use method to access property
6786 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6788 * gst/rtsp-server/rtsp-media-factory.c:
6789 * gst/rtsp-server/rtsp-media-factory.h:
6790 media-factory: add protocols property
6791 Add a property to configure the allowed protocols in the media created from the
6794 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6796 * gst/rtsp-server/rtsp-media-factory.c:
6797 * gst/rtsp-server/rtsp-media-factory.h:
6798 media-factory: add media-configure signal
6799 Add signal to allow the application to configure the media after it was created
6802 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6804 Merge branch 'master' into 0.11
6806 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6808 * gst/rtsp-server/rtsp-client.c:
6809 client: use media multicast group
6811 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6813 * gst/rtsp-server/rtsp-media-factory.h:
6814 * gst/rtsp-server/rtsp-server.h:
6815 * gst/rtsp-server/rtsp-session-pool.h:
6816 * gst/rtsp-server/rtsp-session.h:
6819 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6821 * gst/rtsp-server/rtsp-client.c:
6822 * gst/rtsp-server/rtsp-sdp.h:
6823 sdp: copy and free the server ip address
6824 Copy and free the server ip address to make memory management easier later.
6826 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6828 * gst/rtsp-server/rtsp-media-factory.c:
6829 media-factory: configure multicast in media
6831 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6833 * gst/rtsp-server/rtsp-media.c:
6834 * gst/rtsp-server/rtsp-media.h:
6835 media: add property for multicast group
6836 Add a property to configure the multicast group in the media.
6837 Based on patches from Marc Leeman and Robert Krakora.
6839 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6841 * gst/rtsp-server/rtsp-media-factory.c:
6842 * gst/rtsp-server/rtsp-media-factory.h:
6843 media-factory: add property for multicast group
6844 Add a property to configure the multicast group in the media factory.
6845 Based on patches from Marc Leeman and Robert Krakora.
6847 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6849 * gst/rtsp-server/rtsp-client.c:
6850 client: do configuration of transport in one place
6851 Move the configuration of the transport destination address to where we also
6852 configure the other bits.
6854 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6856 * gst/rtsp-server/rtsp-client.c:
6857 client: use media multicast group
6859 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6861 * gst/rtsp-server/rtsp-media-factory.h:
6862 * gst/rtsp-server/rtsp-server.h:
6863 * gst/rtsp-server/rtsp-session-pool.h:
6864 * gst/rtsp-server/rtsp-session.h:
6867 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6869 * gst/rtsp-server/rtsp-client.c:
6870 * gst/rtsp-server/rtsp-sdp.h:
6871 sdp: copy and free the server ip address
6872 Copy and free the server ip address to make memory management easier later.
6874 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6876 * gst/rtsp-server/rtsp-media-factory.c:
6877 media-factory: configure multicast in media
6879 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6881 * gst/rtsp-server/rtsp-media.c:
6882 * gst/rtsp-server/rtsp-media.h:
6883 media: add property for multicast group
6884 Add a property to configure the multicast group in the media.
6885 Based on patches from Marc Leeman and Robert Krakora.
6887 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6889 * gst/rtsp-server/rtsp-media-factory.c:
6890 * gst/rtsp-server/rtsp-media-factory.h:
6891 media-factory: add property for multicast group
6892 Add a property to configure the multicast group in the media factory.
6893 Based on patches from Marc Leeman and Robert Krakora.
6895 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6897 * gst/rtsp-server/rtsp-client.c:
6898 client: do configuration of transport in one place
6899 Move the configuration of the transport destination address to where we also
6900 configure the other bits.
6902 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6904 Merge branch 'master' into 0.11
6906 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6908 * gst/rtsp-server/rtsp-client.c:
6909 client: destroy pipeline on client disconnect with no prior TEARDOWN.
6910 The problem occurs when the client abruptly closes the connection without
6911 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
6912 server is where the pipeline gets torn down. Since this handler is not called,
6913 the pipeline remains and is up and running. Subsequent clients get their own
6914 pipelines and if the do not issue TEARDOWNs then those pipelines will also
6915 remain up and running. This is a resource leak.
6917 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6919 Merge branch 'master' into 0.11
6921 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
6923 * gst/rtsp-server/rtsp-media-factory.c:
6924 * gst/rtsp-server/rtsp-media-factory.h:
6925 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
6926 For example, it can be used to retrieve source elements like appsrc, in a more
6927 convenient way than subclassing get_element.
6929 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6931 Merge branch 'master' into 0.11
6933 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
6935 * gst/rtsp-server/rtsp-server.c:
6936 rtsp-server: hold on to reference while using object
6938 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6940 * gst/rtsp-server/rtsp-media.c:
6943 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6946 configure: use unstable api
6948 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
6950 * gst/rtsp-server/rtsp-client.c:
6951 client: fix reference counting
6953 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
6955 * gst/rtsp-server/rtsp-client.c:
6956 * gst/rtsp-server/rtsp-media.c:
6957 fix compiler warnings about unused variables
6959 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
6961 * examples/test-launch.c:
6962 * examples/test-readme.c:
6963 * examples/test-uri.c:
6964 * examples/test-video.c:
6965 examples: tell rtsp uri when ready
6967 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
6970 Automatic update of common submodule
6971 From 69b981f to 605cd9a
6973 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6975 * gst/rtsp-server/rtsp-client.c:
6976 client: update for buffer API change
6978 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6980 * gst/rtsp-server/Makefile.am:
6981 Makefile.am: 0.10 => @GST_MAJORMINOR@
6983 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6985 * gst/rtsp-server/rtsp-media-factory-uri.c:
6986 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
6988 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6990 * gst/rtsp-server/.gitignore:
6991 .gitignore: 0.10 => 0.11
6993 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6995 * gst/rtsp-server/Makefile.am:
6996 Makefile.am: 0.10 => @GST_MAJORMINOR@
6998 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7000 Merge branch 'master' into 0.11
7002 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
7005 Automatic update of common submodule
7006 From 9e5bbd5 to 69b981f
7008 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
7011 Automatic update of common submodule
7012 From fd35073 to 9e5bbd5
7014 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
7017 Automatic update of common submodule
7018 From 46dfcea to fd35073
7020 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7022 * gst/rtsp-server/rtsp-media-factory-uri.c:
7023 * gst/rtsp-server/rtsp-media.c:
7024 media: port to new caps API
7026 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7028 Merge branch 'master' into 0.11
7030 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
7032 * bindings/vala/gst-rtsp-server-0.10.vapi:
7033 Updated Vala bindings.
7034 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
7036 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
7038 * gst/rtsp-server/rtsp-server.c:
7039 * gst/rtsp-server/rtsp-server.h:
7040 Add a signal for newly connected clients.
7041 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
7043 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
7045 * bindings/python/rtspserver.override:
7046 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
7048 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7050 * gst/rtsp-server/Makefile.am:
7051 * gst/rtsp-server/rtsp-client.c:
7052 * gst/rtsp-server/rtsp-funnel.c:
7053 * gst/rtsp-server/rtsp-funnel.h:
7054 * gst/rtsp-server/rtsp-media.c:
7055 rtsp-server: port to 0.11
7057 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7062 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7064 Merge branch 'master' into 0.11
7069 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7072 Automatic update of common submodule
7073 From c3cafe1 to 46dfcea
7075 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
7077 * bindings/python/Makefile.am:
7078 * bindings/python/rtspserver.defs:
7079 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
7081 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
7083 * bindings/python/arg-types.py:
7084 python bindings: add GstRTSPUrlParam
7085 Needed to implement MediaFactory virtual proxies
7087 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
7089 * bindings/python/arg-types.py:
7090 python bindings: fix returning GstRTSPUrl types
7092 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
7094 * bindings/python/arg-types.py:
7095 python bindings: add arg type for GstRTSPUrl
7097 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
7099 * bindings/python/rtspserver.defs:
7100 python bindings: fix the definition of MediaFactory.collect_stream
7102 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
7105 Automatic update of common submodule
7106 From 1ccbe09 to c3cafe1
7108 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7111 Automatic update of common submodule
7112 From 193b717 to 1ccbe09
7114 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
7117 Automatic update of common submodule
7118 From b77e2bf to 193b717
7120 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7123 build: Include lcov.mak to allow test coverage report generation
7125 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7128 Automatic update of common submodule
7129 From d8814b6 to b77e2bf
7131 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7134 Automatic update of common submodule
7135 From 6aaa286 to d8814b6
7137 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
7140 Automatic update of common submodule
7141 From 6aec6b9 to 6aaa286
7143 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
7146 autogen: wingo signed comment
7148 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
7150 * gst/rtsp-server/rtsp-session-pool.c:
7151 session: use full charset for RTSP session ID
7152 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
7153 session ID more difficult.
7154 https://bugzilla.gnome.org/show_bug.cgi?id=643812
7156 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7158 * gst/rtsp-server/Makefile.am:
7159 rtsp-server: Don't install the funnel header
7161 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
7164 Automatic update of common submodule
7165 From 1de7f6a to 6aec6b9
7167 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7170 configure: require core/base 0.10.31
7171 Needed at least for gst_plugin_feature_rank_compare_func().
7173 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
7176 Automatic update of common submodule
7177 From f94d739 to 1de7f6a
7179 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7181 * gst/rtsp-server/rtsp-media.c:
7182 media: remove more unused code
7184 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7186 * gst/rtsp-server/rtsp-media.c:
7187 * gst/rtsp-server/rtsp-media.h:
7188 media: remove duplicate filtering
7189 Remove the duplicate filtering code now that we have a released -good version.
7190 Give a warning instead.
7192 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7194 * gst/rtsp-server/rtsp-media-factory.c:
7195 * gst/rtsp-server/rtsp-media.c:
7196 media: fix default buffer size
7198 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7200 * gst/rtsp-server/rtsp-media-factory.c:
7201 * gst/rtsp-server/rtsp-media-factory.h:
7202 media-factory: add property to configure the buffer-size
7203 Add a property to configure the kernel UDP buffer size.
7205 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7207 * gst/rtsp-server/rtsp-media.c:
7208 * gst/rtsp-server/rtsp-media.h:
7209 media: add property to configure kernel buffer sizes
7210 Add a property to configure the kernel UDP buffer size.
7212 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7215 configure: set PYGOBJECT_REQ before using it
7216 https://bugzilla.gnome.org/show_bug.cgi?id=640641
7218 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7221 docs: recursive into sub-directories on 'make upload'
7223 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7225 * docs/libs/gst-rtsp-server-docs.sgml:
7226 * docs/version.entities.in:
7227 docs: mention full version these docs are for, not just major-minor
7229 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7234 === release 0.10.8 ===
7236 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7241 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7243 * gst/rtsp-server/rtsp-server.c:
7244 rtsp-server: clarify docs a little
7246 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7248 * gst/rtsp-server/rtsp-media.c:
7249 media: init debug category before starting thread
7251 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7253 * gst/rtsp-server/rtsp-auth.c:
7254 auth: add realm to make it more spec compliant
7256 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7258 * gst/rtsp-server/rtsp-server.c:
7259 * gst/rtsp-server/rtsp-server.h:
7262 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7264 * examples/test-video.c:
7265 example: improve example docs a little
7267 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7269 * gst/rtsp-server/rtsp-server.c:
7270 server: ensure the watch has a ref to the server
7272 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7274 * gst/rtsp-server/rtsp-server.c:
7275 server: simpify channel function
7277 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7279 * gst/rtsp-server/rtsp-server.c:
7280 * gst/rtsp-server/rtsp-server.h:
7281 server: simplify management of channel and source
7282 We don't need to keep around the channel and source objects. Let the mainloop
7283 and the source manage the source and channel respectively.
7285 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7291 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7294 * tests/Makefile.am:
7295 * tests/test-cleanup.c:
7296 tests: add tests directory and cleanup test
7298 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7300 * gst/rtsp-server/rtsp-media-factory-uri.c:
7301 * gst/rtsp-server/rtsp-media-factory.c:
7302 * gst/rtsp-server/rtsp-media-mapping.c:
7303 * gst/rtsp-server/rtsp-media.c:
7304 * gst/rtsp-server/rtsp-session-pool.c:
7305 * gst/rtsp-server/rtsp-session.c:
7306 server: improve debugging in various objects
7308 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7310 * gst/rtsp-server/rtsp-server.c:
7311 server: chain up to the parent finalize
7313 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
7315 * bindings/python/rtspserver-types.defs:
7316 * bindings/python/rtspserver.defs:
7317 * bindings/python/rtspserver.override:
7318 * bindings/python/test.py:
7319 gst-rtsp-server: update python bindings
7321 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7323 * gst/rtsp-server/rtsp-client.c:
7324 client: use the response from the clientstate
7325 Create the response object only once and store in the client state.
7326 Make all methods use the state response,
7328 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7330 * gst/rtsp-server/rtsp-server.c:
7331 server: use signal to keep track of clients
7332 Keep track of all the clients that the server creates and remove them when they
7333 fire the 'closed' signal.
7335 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7337 * gst/rtsp-server/rtsp-client.c:
7338 * gst/rtsp-server/rtsp-client.h:
7339 client: emit signal when closing
7341 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7343 * examples/.gitignore:
7344 * examples/Makefile.am:
7345 * examples/test-auth.c:
7346 * examples/test-video.c:
7347 * gst/rtsp-server/rtsp-auth.c:
7348 * gst/rtsp-server/rtsp-auth.h:
7349 * gst/rtsp-server/rtsp-client.c:
7350 * gst/rtsp-server/rtsp-media-factory.c:
7351 * gst/rtsp-server/rtsp-media.c:
7352 * gst/rtsp-server/rtsp-media.h:
7353 * gst/rtsp-server/rtsp-session-pool.h:
7354 * gst/rtsp-server/rtsp-session.h:
7355 media: enable per factory authorisations
7356 Allow for adding a GstRTSPAuth on the factory and media level and check
7357 permissions when accessing the factory.
7358 Add hints to the auth methods for future more fine grained authorisation.
7359 Add example application for per factory authentication.
7361 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7363 * gst/rtsp-server/rtsp-auth.c:
7364 * gst/rtsp-server/rtsp-auth.h:
7365 * gst/rtsp-server/rtsp-client.c:
7366 * gst/rtsp-server/rtsp-client.h:
7367 * gst/rtsp-server/rtsp-params.c:
7368 * gst/rtsp-server/rtsp-params.h:
7369 rtsp-server: Pass ClientState structure arround
7370 Pass the collected information for the ongoing request in a GstRTSPClientState
7371 structure that we can then pass around to simplify the method arguments. This
7372 will also be handy when we implement logging functionality.
7374 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7376 * gst/rtsp-server/rtsp-media-factory.c:
7377 * gst/rtsp-server/rtsp-media-factory.h:
7378 media-factory: add methods to configure authorisation
7380 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7382 * gst/rtsp-server/rtsp-client.c:
7383 client: unref auth in finalize
7385 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7387 * gst/rtsp-server/rtsp-server.c:
7388 server: unref auth in finalize
7390 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7392 * docs/libs/gst-rtsp-server-docs.sgml:
7393 * docs/libs/gst-rtsp-server-sections.txt:
7394 * docs/libs/gst-rtsp-server.types:
7397 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7399 * gst/rtsp-server/rtsp-server.c:
7400 * gst/rtsp-server/rtsp-server.h:
7401 server: separate create and accept
7402 Create separate create and accept methods so that subclasses can create custom
7404 Configure the server in the client object and prepare for keeping track of
7407 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7409 * gst/rtsp-server/rtsp-client.c:
7410 * gst/rtsp-server/rtsp-client.h:
7411 client: add support for setting the server.
7412 Add support for keeping a ref to the server that started this client
7415 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7417 * gst/rtsp-server/rtsp-auth.c:
7418 auth: fix memleak and add some docs
7419 Fix a memleak of the basic auth token.
7420 Add docs for the helper function
7422 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7424 * gst/rtsp-server/rtsp-auth.c:
7425 * gst/rtsp-server/rtsp-auth.h:
7426 * gst/rtsp-server/rtsp-client.c:
7427 client: delegate setup of auth to the manager
7428 Delegate the configuration of the authentication tokens to the manager object
7431 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7433 * examples/test-video.c:
7434 * gst/rtsp-server/Makefile.am:
7435 * gst/rtsp-server/rtsp-auth.c:
7436 * gst/rtsp-server/rtsp-auth.h:
7437 * gst/rtsp-server/rtsp-client.c:
7438 * gst/rtsp-server/rtsp-client.h:
7439 * gst/rtsp-server/rtsp-server.c:
7440 * gst/rtsp-server/rtsp-server.h:
7441 auth: add authentication object
7442 Add an object that can check the authorization of requests.
7443 Implement basic authentication.
7444 Add example authentication to test-video
7446 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7448 * gst/rtsp-server/rtsp-server.c:
7449 * gst/rtsp-server/rtsp-server.h:
7450 server: move includes back
7451 the includes are needed for sockaddr_in.
7453 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7455 * gst/rtsp-server/rtsp-client.c:
7456 * gst/rtsp-server/rtsp-client.h:
7457 * gst/rtsp-server/rtsp-server.c:
7458 * gst/rtsp-server/rtsp-server.h:
7459 rtsp: move network includes where they are needed
7461 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
7463 * gst/rtsp-server/rtsp-media.h:
7464 rtsp-media.h: Minor corrections in comments.
7467 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
7470 Automatic update of common submodule
7471 From e572c87 to f94d739
7473 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7477 * docs/libs/.gitignore:
7478 * examples/.gitignore:
7479 * gst/rtsp-server/.gitignore:
7482 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7484 * docs/libs/Makefile.am:
7485 docs: We don't build ps/pdf for API reference docs
7487 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7490 Automatic update of common submodule
7491 From ccbaa85 to e572c87
7493 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7496 Automatic update of common submodule
7497 From 46445ad to ccbaa85
7499 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7501 * gst/rtsp-server/Makefile.am:
7502 * gst/rtsp-server/fs-funnel.c:
7503 * gst/rtsp-server/fs-funnel.h:
7504 * gst/rtsp-server/rtsp-funnel.c:
7505 * gst/rtsp-server/rtsp-funnel.h:
7506 * gst/rtsp-server/rtsp-media.c:
7507 funnel: rename fsfunnel to rtspfunnel
7508 Rename the funnel to avoid conflicts with the farsight one.
7510 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7512 * gst/rtsp-server/Makefile.am:
7513 * gst/rtsp-server/fs-funnel.c:
7514 * gst/rtsp-server/fs-funnel.h:
7515 * gst/rtsp-server/rtsp-media.c:
7516 rtsp-media: add and use fsfunnel
7517 Add a copy of fsfunnel to the build because input-selector removed the (broken)
7518 select-all property that we need.
7520 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7522 * gst/rtsp-server/Makefile.am:
7523 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
7524 Use PKG_CONFIG_PATH specified at configure time (if any) as well
7525 for the g-ir-compiler, rather than just assuming the env var has
7528 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7535 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
7537 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7540 * gst/rtsp-server/Makefile.am:
7541 gobject-introspection: fix g-i build for uninstalled setup
7542 Requires gst-plugins-base git (> 0.10.31.2).
7544 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7546 * examples/test-uri.c:
7547 examples: add some more options and comments
7549 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7551 * gst/rtsp-server/rtsp-media-factory-uri.c:
7552 factory-uri: use right property type
7554 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7556 * gst/rtsp-server/rtsp-media-factory-uri.c:
7557 factory-uri: attempt to configure buffer-lists
7558 Attempt to configure buffer lists in the payloader for improved performance.
7560 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7562 * gst/rtsp-server/rtsp-media.c:
7563 media: attempt to configure bigger UDP buffers
7564 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
7565 send buffers with high bitrate streams.
7567 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
7569 * gst/rtsp-server/rtsp-client.c:
7570 client: use the socket length from getsockname
7571 Use the length returned by getsockname to perform the getnameinfo call because
7572 the size can depend on the socket type and platform.
7575 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7577 * docs/libs/gst-rtsp-server-docs.sgml:
7578 * docs/libs/gst-rtsp-server-sections.txt:
7579 docs: add uri factory to the docs
7581 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7583 * gst/rtsp-server/rtsp-client.c:
7584 * gst/rtsp-server/rtsp-media.h:
7587 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7589 * gst/rtsp-server/rtsp-client.c:
7590 * gst/rtsp-server/rtsp-media.c:
7591 * gst/rtsp-server/rtsp-media.h:
7592 * gst/rtsp-server/rtsp-session.c:
7593 * gst/rtsp-server/rtsp-session.h:
7594 rtsp-server: add support for buffer lists
7595 Add support for sending bufferlists received from appsink.
7598 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7600 * gst/rtsp-server/rtsp-client.c:
7601 * gst/rtsp-server/rtsp-media.c:
7602 * gst/rtsp-server/rtsp-media.h:
7603 * gst/rtsp-server/rtsp-sdp.c:
7604 media: make method to retrieve the play range
7605 Make a method to retrieve the playback range so that we can conditionally create
7606 a different range for the SDP and the PLAY requests.
7608 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7610 * gst/rtsp-server/rtsp-media.c:
7611 * gst/rtsp-server/rtsp-media.h:
7612 media: add signal to notify of state changes
7614 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7616 * gst/rtsp-server/rtsp-client.h:
7617 client: cleanup headers
7619 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7621 * gst/rtsp-server/rtsp-client.c:
7624 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7626 * gst/rtsp-server/rtsp-media-factory-uri.c:
7627 * gst/rtsp-server/rtsp-media-factory-uri.h:
7628 factory-uri: add support for gstpay
7629 Add an option to prefer gstpay over decoder + raw payloader.
7631 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7633 * gst/rtsp-server/rtsp-media-factory-uri.c:
7634 * gst/rtsp-server/rtsp-media-factory-uri.h:
7635 factory-uri: rework the autoplugger.
7636 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
7639 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7641 * gst/rtsp-server/rtsp-media-factory-uri.c:
7642 factory-uri: use better factory filter
7643 Make better payloader filter based on autoplug rank and RTP use case.
7645 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7648 Automatic update of common submodule
7649 From 169462a to 46445ad
7651 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7653 * gst/rtsp-server/rtsp-server.c:
7654 server: set SO_REUSEADDR before bind
7655 Set the SO_REUSEADDR _before_ bind() to make it actually work.
7657 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7659 * gst/rtsp-server/rtsp-media.c:
7660 * gst/rtsp-server/rtsp-media.h:
7661 media: emit prepared signal when prepared
7662 Make a 'prepared' signal and emit it when we successfully prepared the element.
7663 This signal can be used to configure the media object after it has been prepared
7666 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
7669 Automatic update of common submodule
7670 From 011bcc8 to 169462a
7672 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
7674 python an optional dependency
7675 * configure.ac: Move up valgrind and g-i checks. Make the python
7676 dependency optional, as it was before.
7678 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7680 Merge branch 'master' into 0.11
7685 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7687 * gst/rtsp-server/rtsp-media.c:
7688 media: update range when active clients changed
7689 When we changed the number of active clients, update the current range
7690 information because we want the second client connecting to a shared resource
7691 continue from where the stream currently.
7693 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7695 * gst/rtsp-server/rtsp-media-factory-uri.c:
7696 * gst/rtsp-server/rtsp-media-factory-uri.h:
7697 factory-uri: add colorspace and fix pt
7698 Rework the way we pass data to the autoplugger.
7699 When we have raw caps, plug a converter element to make pluggin to raw
7700 payloaders more successful.
7701 Make sure all dynamically plugged payloaders have a unique payload types.
7703 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7705 * examples/Makefile.am:
7706 * examples/test-uri.c:
7707 example: add example of the uri factory
7709 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7711 * gst/rtsp-server/Makefile.am:
7712 * gst/rtsp-server/rtsp-media-factory-uri.c:
7713 * gst/rtsp-server/rtsp-media-factory-uri.h:
7714 * gst/rtsp-server/rtsp-server.h:
7715 factory-uri: add a factory to stream any URI
7716 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
7719 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7721 * gst/rtsp-server/rtsp-media.c:
7722 * gst/rtsp-server/rtsp-media.h:
7723 media: ignore spurious ASYNC_DONE messages
7724 When we are dynamically adding pads, the addition of the udpsrc elements will
7725 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
7726 the real ASYNC_DONE when everything is prerolled.
7728 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7730 * gst/rtsp-server/rtsp-media-factory.c:
7731 * gst/rtsp-server/rtsp-media-factory.h:
7732 media-factory: make lock macro
7734 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
7736 * gst/rtsp-server/rtsp-client.c:
7737 rtsp-server: Remove unused variable and dead assignment
7739 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
7741 * examples/test-launch.c:
7742 * examples/test-mp4.c:
7743 * examples/test-ogg.c:
7744 * examples/test-readme.c:
7745 * examples/test-sdp.c:
7746 * examples/test-video.c:
7747 examples: Run gst-indent
7749 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
7751 * gst/rtsp-server/rtsp-client.c:
7752 * gst/rtsp-server/rtsp-media-factory.c:
7753 * gst/rtsp-server/rtsp-media-mapping.c:
7754 * gst/rtsp-server/rtsp-media.c:
7755 * gst/rtsp-server/rtsp-params.c:
7756 * gst/rtsp-server/rtsp-sdp.c:
7757 * gst/rtsp-server/rtsp-server.c:
7758 * gst/rtsp-server/rtsp-session-pool.c:
7759 * gst/rtsp-server/rtsp-session.c:
7760 rtsp-server: Run gst-indent
7761 Since it wasn't using the upstream common previously, there was no
7762 indentation check before commiting.
7764 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
7766 * gst/rtsp-server/rtsp-media-mapping.h:
7767 * gst/rtsp-server/rtsp-media.c:
7768 * gst/rtsp-server/rtsp-media.h:
7769 * gst/rtsp-server/rtsp-sdp.c:
7770 * gst/rtsp-server/rtsp-session-pool.h:
7771 * gst/rtsp-server/rtsp-session.c:
7772 * gst/rtsp-server/rtsp-session.h:
7773 rtsp-server: Some more doc fixups
7775 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7778 Makefile: Add cruft-cleaning support
7780 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7785 * docs/libs/Makefile.am:
7786 * docs/libs/gst-rtsp-server-docs.sgml:
7787 * docs/libs/gst-rtsp-server-sections.txt:
7788 * docs/libs/gst-rtsp-server.types:
7789 * docs/version.entities.in:
7790 docs: Add gtk-doc build system
7792 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7794 * gst/rtsp-server/Makefile.am:
7795 Makefile.am: Use standard GIR make behaviour
7797 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7801 autogen/configure: Bring more in sync to standard gst module behaviour
7803 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7805 * gst/rtsp-server/rtsp-media.c:
7806 media: warn and fail when gstrtpbin is not found
7808 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7811 configure: open 0.11 branch
7813 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
7817 Add common submodule
7819 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
7822 * common/Makefile.am:
7823 * common/c-to-xml.py:
7825 * common/coverage/coverage-report-entry.pl:
7826 * common/coverage/coverage-report.pl:
7827 * common/coverage/coverage-report.xsl:
7828 * common/coverage/lcov.mak:
7829 * common/gettext.patch:
7830 * common/glib-gen.mak:
7831 * common/gst-autogen.sh:
7832 * common/gst-xmlinspect.py:
7834 * common/gstdoc-scangobj:
7835 * common/gtk-doc-plugins.mak:
7836 * common/gtk-doc.mak:
7837 * common/m4/.gitignore:
7838 * common/m4/Makefile.am:
7840 * common/m4/as-ac-expand.m4:
7841 * common/m4/as-auto-alt.m4:
7842 * common/m4/as-compiler-flag.m4:
7843 * common/m4/as-compiler.m4:
7844 * common/m4/as-docbook.m4:
7845 * common/m4/as-libtool-tags.m4:
7846 * common/m4/as-libtool.m4:
7847 * common/m4/as-python.m4:
7848 * common/m4/as-scrub-include.m4:
7849 * common/m4/as-version.m4:
7850 * common/m4/ax_create_stdint_h.m4:
7851 * common/m4/check.m4:
7852 * common/m4/glib-gettext.m4:
7853 * common/m4/gst-arch.m4:
7854 * common/m4/gst-args.m4:
7855 * common/m4/gst-check.m4:
7856 * common/m4/gst-debuginfo.m4:
7857 * common/m4/gst-default.m4:
7858 * common/m4/gst-doc.m4:
7859 * common/m4/gst-error.m4:
7860 * common/m4/gst-feature.m4:
7861 * common/m4/gst-function.m4:
7862 * common/m4/gst-gettext.m4:
7863 * common/m4/gst-glib2.m4:
7864 * common/m4/gst-libxml2.m4:
7865 * common/m4/gst-plugindir.m4:
7866 * common/m4/gst-valgrind.m4:
7867 * common/m4/gtk-doc.m4:
7868 * common/m4/introspection.m4:
7870 * common/mangle-tmpl.py:
7871 * common/plugins.xsl:
7873 * common/release.mak:
7874 * common/scangobj-merge.py:
7875 * common/upload.mak:
7876 common: Remove static version
7878 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
7880 * common/m4/introspection.m4:
7881 Update introspection.m4 to match usage
7883 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7887 Remove old stuff from the README
7889 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7894 === release 0.10.7 ===
7896 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7901 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7903 * examples/test-ogg.c:
7904 test-ogg: remove parsers
7905 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
7906 buffers with timestamps. Using the parsers also seems to break things.
7908 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7910 * bindings/vala/gst-rtsp-server-0.10.vapi:
7911 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7912 Updated Vala bindings
7914 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7916 * common/m4/introspection.m4:
7918 * gst/rtsp-server/Makefile.am:
7919 Added initial gobject-introspection support
7921 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7923 * gst/rtsp-server/rtsp-media-factory.c:
7924 media-factory: don't use host for shared hash key
7925 When we generate the key to share made between connections, don't include the
7926 host used to connect so that we can share media even if between clients that
7927 connected with localhost and ones with the ip address.
7929 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7931 * bindings/vala/Makefile.am:
7932 build: fix distcheck
7934 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7936 * bindings/vala/gst-rtsp-server-0.10.vapi:
7937 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7938 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7939 Update Vala bindings
7941 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7943 * bindings/vala/Makefile.am:
7945 Fix configure checks and installation location for Vala bindings
7948 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7953 === release 0.10.6 ===
7955 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7958 configure: release 0.10.6
7960 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7962 * gst/rtsp-server/rtsp-media.c:
7963 media: help the compiler a little
7965 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7967 * gst/rtsp-server/rtsp-media.c:
7968 * gst/rtsp-server/rtsp-media.h:
7969 * gst/rtsp-server/rtsp-session.c:
7970 media: cleanup media transport before freeing
7971 Cleanup the media transport data before freeing. In particular, remove the qdata
7972 from the rtpsource object.
7974 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7976 * gst/rtsp-server/rtsp-media-factory.c:
7977 * gst/rtsp-server/rtsp-media-factory.h:
7978 * gst/rtsp-server/rtsp-media.c:
7979 * gst/rtsp-server/rtsp-media.h:
7980 media-factory: add eos-shutdown property
7981 Add an eos-shutdown property that will send an EOS to the pipeline before
7982 shutting it down. This allows for nice cleanup in case of a muxer.
7985 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7987 * gst/rtsp-server/rtsp-media.c:
7988 * gst/rtsp-server/rtsp-media.h:
7989 media: use multiudpsink send-duplicates when we can
7990 If we have a new enough multiudpsink with the send-duplicates property, use this
7991 instead of doing our own filtering. Our custom filtering code should eventually
7992 be removed when we can depend on a released -good.
7994 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7996 * gst/rtsp-server/rtsp-media.c:
7997 media: don't leak destinations
7998 Refactor and cleanup the destinations array when the stream is destroyed.
8000 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8002 * gst/rtsp-server/rtsp-media.c:
8003 * gst/rtsp-server/rtsp-media.h:
8004 media: don't add udp addresses multiple times
8005 Keep track of the udp addresses we added to udpsink and never add the same udp
8006 destination twice. This avoids duplicate packets when using multicast.
8008 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8010 * gst/rtsp-server/rtsp-server.c:
8011 server: disable use of SO_LINGER
8012 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
8013 server close()s the connection.
8015 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8017 * gst/rtsp-server/rtsp-server.c:
8018 server: use 5 second linger period in SO_LINGER
8019 Wait 5 seconds before clearing the send buffers and reseting the connection with
8020 the client when we do a close. This should be enough time to get the message to
8024 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
8026 * gst/rtsp-server/rtsp-server.c:
8027 server: use SO_LINGER
8028 SO_LINGER on the socket will make sure that any pending data on the socket is
8029 flushed ASAP and that the socket connection is reset. This makes sure that the
8030 socket can be reused immediately.
8033 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8036 README: add blurb about shared media factories
8038 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
8040 * gst/rtsp-server/rtsp-media.c:
8041 Add stdlib.h for atoi()
8043 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8045 * bindings/python/Makefile.am:
8046 * bindings/vala/Makefile.am:
8047 build: distcheck fixes
8048 Fix 'make distcheck', somewhat (it still fails because it tries to
8049 install files into /usr/share/vala/vapi/ irrespective of the
8052 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8055 configure: bump core/base requirements to released version
8056 Makes things less confusing for people.
8058 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8061 configure: fail if GStreamer core/base requirements are not met
8063 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8065 * gst/rtsp-server/rtsp-client.c:
8066 client: improve client cleanups
8067 Make sure the session does not timeout when using TCP. We need to do this
8068 because quicktime player does not send RTCP for some reason in tunneled
8070 Refactor some cleanup code.
8073 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8075 * gst/rtsp-server/rtsp-session.c:
8076 * gst/rtsp-server/rtsp-session.h:
8077 session: add support for prevent session timeouts
8078 Add an atomix counter to prevent session timeouts when we are, for example,
8081 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8083 * gst/rtsp-server/rtsp-client.c:
8084 client: fix unlink on session timeouts
8085 When our session times out, make sure we unlink all streams in this
8087 Remove the tunnelid when closing the connection.
8089 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8091 * gst/rtsp-server/rtsp-session.c:
8092 session: small cleanups
8094 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8096 * gst/rtsp-server/rtsp-client.c:
8097 client: handle lost_tunnel callbacks
8098 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
8099 hashtable so that we can reuse it for when the client reopens the POST
8101 Close the connection after a TEARDOWN.
8102 Make sure or watchid is cleared when the watch is removed.
8105 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8107 * gst/rtsp-server/rtsp-client.c:
8108 * gst/rtsp-server/rtsp-media.c:
8109 * gst/rtsp-server/rtsp-sdp.c:
8110 rtsp-server: add more support for multicast
8112 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8115 * gst/rtsp-server/rtsp-media.c:
8116 * gst/rtsp-server/rtsp-media.h:
8117 media: allow configuration of allowed lower transport
8119 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8121 * gst/rtsp-server/rtsp-client.h:
8122 * gst/rtsp-server/rtsp-media.c:
8123 * gst/rtsp-server/rtsp-media.h:
8124 * gst/rtsp-server/rtsp-sdp.c:
8125 * gst/rtsp-server/rtsp-sdp.h:
8126 * gst/rtsp-server/rtsp-server.c:
8127 rtsp: keep track of server ip and ipv6
8128 Keep track of how the client connected to the server and setup the udp ports
8129 with the same protocol.
8130 Copy the server ip address in the SDP so that clients can send RTCP back to
8133 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8135 * gst/rtsp-server/rtsp-session.c:
8138 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8140 * gst/rtsp-server/rtsp-client.c:
8141 client: use right size for malloc
8143 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8145 * gst/rtsp-server/rtsp-server.c:
8146 server: comment ipv6 server listening address
8148 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8150 * gst/rtsp-server/rtsp-media.c:
8151 media: allow for ipv6 sockets
8153 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8155 * gst/rtsp-server/rtsp-server.c:
8156 * gst/rtsp-server/rtsp-server.h:
8157 server: rework server part
8158 Allow setting a bind address, make sure we can deal with ipv6.
8159 Remove the port property and change with the service property.
8161 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8163 * gst/rtsp-server/rtsp-media.h:
8164 media: update comments a little
8166 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8168 * gst/rtsp-server/rtsp-client.c:
8169 client: make content-base better
8170 Use the URI formatting functions to make a content-base. Also make sure that
8171 there is a trailing / at the end.
8173 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8175 * gst/rtsp-server/rtsp-client.c:
8176 client: guard against invalid paths
8178 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8180 * examples/test-video.c:
8181 test: catch server bind errors
8183 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
8185 * gst/rtsp-server/rtsp-media.c:
8186 rtspmedia: emit "unprepared" if _prepare fails.
8187 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
8188 media object is removed from its factory's cache.
8190 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8192 * gst/rtsp-server/rtsp-media.c:
8193 media: collect media position when seek completes
8195 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
8197 * gst/rtsp-server/rtsp-client.c:
8198 client: call unlink_streams in client finalize
8201 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8203 * gst/rtsp-server/rtsp-media.c:
8204 media: limit the time to wait to something huge
8205 Avoid waiting forever but limit the timeout to 20 seconds.
8207 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8209 * gst/rtsp-server/rtsp-sdp.c:
8210 sdp: reindent and check for prepared status
8212 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8214 * gst/rtsp-server/rtsp-media.c:
8215 * gst/rtsp-server/rtsp-media.h:
8216 * gst/rtsp-server/rtsp-session.c:
8217 media: avoid doing _get_state() for state changes
8218 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
8219 until the media is prerolled or in error. This avoids doing a blocking call of
8220 gst_element_get_state() that can cause lockups when there is an error.
8223 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8225 * gst/rtsp-server/rtsp-media.c:
8228 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8230 * gst/rtsp-server/rtsp-media-factory.c:
8231 media-factory: better error handling
8232 Improve the error handling a bit.
8234 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8236 * gst/rtsp-server/rtsp-client.c:
8237 client: rework transport parsing
8238 Rework the transport parsing code so that we can ignore transports we don't
8239 support instead of just picking the first one we can parse.
8240 Configure a (for now hardcoded) destination for multicast transports.
8242 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8244 * gst/rtsp-server/rtsp-media.c:
8245 media: set multicast sink parameters
8246 Disable loop and automatic multicast join on the udpsink elements.
8247 Add some more debug info.
8248 Reset some state variables in the right place.
8249 Use the right port numbers for multicast.
8251 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8253 * gst/rtsp-server/rtsp-session.c:
8254 session: handle transport setup correctly
8255 Handle UDP, MCAST and TCP transport negotiation more correctly.
8256 Store the server session SSRC in the transport.
8258 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8260 * gst/rtsp-server/rtsp-client.c:
8261 rtsp-client: implement error_full
8262 Implement error_full to avoid some segfaults when the rtspconnection calls it.
8265 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8268 * gst/rtsp-server/rtsp-client.c:
8269 * gst/rtsp-server/rtsp-server.c:
8270 docs: update docs and comments
8272 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
8274 * gst/rtsp-server/rtsp-sdp.c:
8275 sdp: make server work better when behind a proxy
8277 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8279 * gst/rtsp-server/rtsp-client.c:
8280 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
8282 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8284 * gst/rtsp-server/rtsp-client.c:
8285 * gst/rtsp-server/rtsp-media-factory.c:
8286 * gst/rtsp-server/rtsp-media-mapping.c:
8287 * gst/rtsp-server/rtsp-media.c:
8288 * gst/rtsp-server/rtsp-server.c:
8289 * gst/rtsp-server/rtsp-session-pool.c:
8290 * gst/rtsp-server/rtsp-session.c:
8291 Use GStreamer's debugging subsystem
8293 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8295 * gst/rtsp-server/rtsp-media-factory.c:
8296 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
8298 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8303 === release 0.10.5 ===
8305 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8310 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8313 configure: bump required versions
8315 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
8317 * gst/rtsp-server/rtsp-client.c:
8318 client: call weak-unref on client->sessions from finalize
8321 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8323 * gst/rtsp-server/rtsp-media.c:
8324 media: Fixed crasher where caps got unref'ed too often
8326 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8329 * pkgconfig/.gitignore:
8330 * pkgconfig/Makefile.am:
8331 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
8332 Added pkg-config file to use gst-rtsp-server uninstalled
8334 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8336 * gst/rtsp-server/rtsp-media.c:
8337 media: add some docs
8339 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
8341 * gst/rtsp-server/rtsp-client.c:
8342 rtsp: Use gst_rtsp_watch_send_message().
8343 Use gst_rtsp_watch_send_message() since the old API which used
8344 gst_rtsp_watch_queue_message() has been deprecated.
8346 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8351 === release 0.10.4 ===
8353 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8358 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8360 * gst/rtsp-server/rtsp-client.c:
8361 * gst/rtsp-server/rtsp-session.c:
8362 * gst/rtsp-server/rtsp-session.h:
8363 rtsp: allocate channels in TCP mode
8364 When the client does not provide us with channels in TCP mode, allocate channels
8367 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8369 * gst/rtsp-server/rtsp-client.c:
8370 client: don't crash when tunnelid is missing
8371 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
8372 don't crash but return an error response to the client.
8375 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8377 * bindings/vala/gst-rtsp-server-0.10.vapi:
8378 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8379 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8380 bindings: update vala bindings with new method
8382 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8384 * gst/rtsp-server/rtsp-session-pool.c:
8385 * gst/rtsp-server/rtsp-session-pool.h:
8386 sessionpool: add function to filter sessions
8387 Add generic function to retrieve/remove sessions.
8389 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8392 configure: bump core/base requirements to release
8394 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8396 * gst/rtsp-server/rtsp-media.c:
8397 media: fix indentation
8399 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8401 * gst/rtsp-server/rtsp-media.c:
8402 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
8404 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8406 * gst/rtsp-server/rtsp-media.c:
8407 set state and remove elements of media in for loop
8409 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
8411 * bindings/vala/gst-rtsp-server-0.10.vapi:
8412 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8413 Added gst_rtsp_media_remove_elements function to Vala bindings
8415 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
8417 * gst/rtsp-server/rtsp-media.c:
8418 * gst/rtsp-server/rtsp-media.h:
8419 Added gst_rtsp_media_remove_elements function
8421 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
8423 * gst/rtsp-server/rtsp-media.c:
8424 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
8426 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8428 * bindings/vala/gst-rtsp-server-0.10.vapi:
8429 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8430 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8431 Updated Vala bindings
8433 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8435 * gst/rtsp-server/rtsp-media.c:
8436 * gst/rtsp-server/rtsp-media.h:
8437 Added vmethod unprepare to GstRTSPMedia
8438 The default implementation sets the state of the pipeline to GST_STATE_NULL
8440 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8442 * gst/rtsp-server/rtsp-media-factory.c:
8443 * gst/rtsp-server/rtsp-media-factory.h:
8444 Made collect_streams function public
8446 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8448 * gst/rtsp-server/rtsp-media-factory.c:
8449 * gst/rtsp-server/rtsp-media-factory.h:
8450 * gst/rtsp-server/rtsp-media.c:
8451 Added vmethod create_pipeline to GstRTSPMediaFactory
8452 The pipeline is created in this method and the GstRTSPMedia's element is added to it
8454 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8456 * gst/rtsp-server/rtsp-client.c:
8457 client: use g_source_destroy()
8458 We need to use g_source_destroy() because we might have added the source to a
8459 different main context than the default one.
8461 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8463 * gst/rtsp-server/Makefile.am:
8464 * gst/rtsp-server/rtsp-client.c:
8465 * gst/rtsp-server/rtsp-params.c:
8466 * gst/rtsp-server/rtsp-params.h:
8467 rtsp: prepare for handling GET/SET_PARAMETER
8468 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
8470 Fix return codes of handlers.
8472 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8474 * gst/rtsp-server/rtsp-media.c:
8475 media: don't leak session pads
8477 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8479 * gst/rtsp-server/rtsp-media.c:
8480 media: clean up the messages a bit
8482 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8484 * gst/rtsp-server/rtsp-sdp.c:
8485 sdp: warn and skip streams without media
8487 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8489 * bindings/vala/gst-rtsp-server-0.10.vapi:
8490 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8491 vala: Fixed typo in header file of RTSPMediaStream
8493 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8495 * gst/rtsp-server/rtsp-media.c:
8498 Make dumping RTCP stats configurable
8500 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8502 * gst/rtsp-server/rtsp-media.c:
8503 media: be less verbose and leak less
8505 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8507 * gst/rtsp-server/rtsp-media.c:
8508 media: don't leak the destination address
8510 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8512 * gst/rtsp-server/rtsp-client.c:
8513 * gst/rtsp-server/rtsp-media.c:
8514 * gst/rtsp-server/rtsp-media.h:
8515 * gst/rtsp-server/rtsp-session.c:
8516 * gst/rtsp-server/rtsp-session.h:
8517 rtsp: use RTCP to keep the session alive
8518 Use the RTCP rtcp-from stats field to find the associated session and use this
8519 to keep the session alive.
8521 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8523 * gst/rtsp-server/rtsp-session.c:
8524 session: add 5sec to the real session timeout
8525 Allow the session to live 5sec longer before really timing out. This should give
8526 clients some extra time to keep the session active.
8528 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8530 * gst/rtsp-server/rtsp-client.c:
8531 client: replay OK to GET/SET_PARAMETER
8532 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
8533 so that we return OK for those requests.
8535 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8537 * gst/rtsp-server/rtsp-media.c:
8538 * gst/rtsp-server/rtsp-media.h:
8539 media: keep track of active transports
8540 Keep track of which transport is active to avoid closing the connection too
8542 Remove the destination transport also when going to NULL.
8543 Print some stats about the SDES and other RTCP messages we receive from the
8546 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8548 * examples/.gitignore:
8549 * examples/Makefile.am:
8550 * examples/test-sdp.c:
8551 example: add SDP relay example
8553 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8555 * gst/rtsp-server/rtsp-media.c:
8556 media: also count active TCP connections
8558 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8560 * gst/rtsp-server/rtsp-media-factory.c:
8561 * gst/rtsp-server/rtsp-media.c:
8562 * gst/rtsp-server/rtsp-media.h:
8563 rtsp: add support for dynamic elements
8564 Add support for dynamic elements.
8565 Don't set live pipelines back to paused.
8567 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8569 * gst/rtsp-server/rtsp-sdp.c:
8570 sdp: don't add encoding name when absent in caps
8572 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8574 * gst/rtsp-server/rtsp-client.c:
8575 client: warn when we can't do RTP-Info
8577 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8579 * gst/rtsp-server/rtsp-media-factory.c:
8580 factory: factor out the stream construction
8582 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8584 * gst/rtsp-server/rtsp-client.c:
8585 client: only add RTP-Info when we have the info
8586 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
8589 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8594 === release 0.10.3 ===
8596 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8600 - Fixes a bug where it put the wrong verion in pkgconfig
8601 - Link RTP and RTCP sources
8603 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8605 * gst/rtsp-server/rtsp-media.c:
8606 * gst/rtsp-server/rtsp-media.h:
8607 media: link the RTP udpsrc to the session manager
8608 Link the RTP udpsrc and the appsrc to the session manager so that they don't
8609 shut down when the client sends a packet to open firewalls.
8611 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8613 * pkgconfig/gst-rtsp-server.pc.in:
8614 Don't use hard-coded version number in pkg-config file
8616 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8621 === release 0.10.2 ===
8623 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8628 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8631 * common/m4/.gitignore:
8632 * examples/.gitignore:
8633 * pkgconfig/.gitignore:
8634 add some .gitignore files
8636 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8638 * gst/rtsp-server/rtsp-media.c:
8639 media: seek to key frames
8641 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8643 * gst/rtsp-server/rtsp-media.c:
8644 media: emit the unprepared signal by id
8645 Emit the unprepared signal by id instead of name and set the media as
8648 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8650 * gst/rtsp-server/rtsp-media.c:
8651 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
8653 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8655 * gst/rtsp-server/rtsp-server.c:
8656 Added finalize function to GstRTPSPServer to unref session pool and media mapping
8658 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8660 * bindings/vala/gst-rtsp-server-0.10.vapi:
8661 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8662 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8663 Updated vala bindings
8665 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8667 * gst/rtsp-server/Makefile.am:
8668 * gst/rtsp-server/rtsp-client.c:
8669 * gst/rtsp-server/rtsp-media.c:
8670 server: use appsink and appsrc with the API
8671 Use the appsink/appsrc API instead of the signals for higher
8674 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8676 * examples/test-ogg.c:
8677 tests: set the payload type correctly
8679 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8681 * gst/rtsp-server/rtsp-media-factory.c:
8682 factory: connect to the unprepare signal
8683 Connect to the unprepare signal for non-reusable media so that we can remove
8684 them from the cache.
8686 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8688 * gst/rtsp-server/rtsp-media.c:
8689 * gst/rtsp-server/rtsp-media.h:
8690 media: add signal to notify of unprepare
8692 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8694 * gst/rtsp-server/rtsp-media.c:
8695 * gst/rtsp-server/rtsp-media.h:
8696 media: more work on making the media shared
8697 Add a reusable flag to medias, indicating that they can be reused after a state
8701 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8703 * examples/test-readme.c:
8704 examples: mark the example as shared for testing
8706 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8708 * gst/rtsp-server/rtsp-media.c:
8709 * gst/rtsp-server/rtsp-media.h:
8710 client: support shared media
8711 Always perform the state actions even if the target state of the pipeline is
8712 already correct, we still want to add/remove the transports when we are dealing
8714 Keep a counter of the number of active transports for a media so that we can use
8715 this to perform a state change when needed.
8716 Perform a state change of the pipeline only when the first transport was added
8717 or when there are no active transports.
8719 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8721 * gst/rtsp-server/rtsp-client.c:
8722 client: fix refcounting crasher
8723 Don't need to remove the weak refs in the finalize methods, they are already
8724 removed in the dispose.
8725 Don't register the callback with a DestroyNofity.
8727 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8729 * gst/rtsp-server/rtsp-client.c:
8730 Fix rtsp client refcount management in TCP mode.
8731 Don't unref a client ref we never had. Fixes an unref
8732 of an already-free client object after a client
8733 teardown request for me.
8735 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8737 * gst/rtsp-server/rtsp-session.c:
8738 docs: fix typo in API docs
8740 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8742 * gst/rtsp-server/rtsp-media.c:
8744 Keep the udp sources in playing even if we go to paused. unlock the sources when
8746 Add some more debug info.
8747 Only seek when we need to.
8748 Keep track of the position when we go to paused.
8750 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8752 * gst/rtsp-server/rtsp-client.c:
8753 * gst/rtsp-server/rtsp-media.c:
8754 * gst/rtsp-server/rtsp-media.h:
8755 Add beginnings of seeking.
8756 Parse the Range header and perform a seek on the pipeline for the requested
8757 position. It's disabled currently until I figure out what's going wrong.
8759 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8761 * gst/rtsp-server/rtsp-client.c:
8762 allow pause requests for now.
8765 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8767 * gst/rtsp-server/rtsp-client.c:
8768 Remove weak ref on the session in teardown
8769 We need to remove our weakref from the session when we do a teardown because
8770 else we close the TCP connection prematurely.
8772 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8774 * gst/rtsp-server/rtsp-client.c:
8775 * gst/rtsp-server/rtsp-client.h:
8776 * gst/rtsp-server/rtsp-session-pool.c:
8777 Do some more session cleanup
8778 Make session timeout kill the TCP connection that currently watches the
8780 Remove the client timeout property.
8782 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8784 * gst/rtsp-server/rtsp-client.c:
8785 * gst/rtsp-server/rtsp-client.h:
8786 * gst/rtsp-server/rtsp-media.c:
8787 * gst/rtsp-server/rtsp-media.h:
8788 * gst/rtsp-server/rtsp-server.c:
8789 * gst/rtsp-server/rtsp-session.c:
8790 * gst/rtsp-server/rtsp-session.h:
8792 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
8795 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8797 * examples/Makefile.am:
8798 * examples/test-launch.c:
8799 Add example server that takes launch lines
8800 Add an example server that streams any -launch line.
8802 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8804 * examples/test-readme.c:
8805 * gst/rtsp-server/rtsp-client.c:
8806 * gst/rtsp-server/rtsp-media.c:
8807 * gst/rtsp-server/rtsp-media.h:
8808 Add support for live streams
8809 Add support for live streams and ranges
8810 Start on handling TCP data transfer.
8812 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8814 * gst/rtsp-server/rtsp-media.c:
8815 Free the pipeline before other things
8818 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8820 * gst/rtsp-server/rtsp-client.c:
8821 Only free the pending tunnel if there is one
8824 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8826 * gst/rtsp-server/rtsp-client.c:
8827 * gst/rtsp-server/rtsp-client.h:
8828 * gst/rtsp-server/rtsp-media.c:
8829 rtsp-server: Add support for tunneling
8830 Add support for tunneling over HTTP.
8831 Use new connection methods to retrieve the url.
8832 Dispatch messages based on the message type instead of blindly
8833 assuming it's always a request.
8834 Keep track of the watch id so that we can remove it later.
8835 Set the media pipeline to NULL before unreffing the pipeline.
8837 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8839 * gst/rtsp-server/rtsp-client.c:
8840 * gst/rtsp-server/rtsp-client.h:
8841 Fix for channel -> watch rename in gstreamer
8842 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
8844 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8846 * gst/rtsp-server/rtsp-client.c:
8847 * gst/rtsp-server/rtsp-client.h:
8849 Use the async RTSP channels instead of spawning a new thread for each client.
8850 If a sessionid is specified in a request, fail if we don't have the session.
8852 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8854 * gst/rtsp-server/rtsp-media.c:
8855 Add better debug info
8856 Add some better debug info.
8858 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8860 * examples/test-video.c:
8862 Add support for session timeouts in the example.
8864 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8866 * gst/rtsp-server/rtsp-session-pool.c:
8867 * gst/rtsp-server/rtsp-session-pool.h:
8868 Pass GTimeVal around for performance reasons
8869 Get the current time only once and pass it around so that sessions don't have to
8870 get the current time anymore.
8871 Add experimental support for a GSource that dispatches when the session needs to
8874 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8876 * gst/rtsp-server/rtsp-session.c:
8877 * gst/rtsp-server/rtsp-session.h:
8878 Add better support for session timeouts
8879 Add a method to request the number of milliseconds when a session will timeout.
8881 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8883 * gst/rtsp-server/rtsp-media.c:
8884 * gst/rtsp-server/rtsp-media.h:
8885 Add suport for RTP manager monitoring
8886 Add the first stage in monitoring the rtp manager.
8887 Make sure we don't update the state to something we don't want.
8889 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8891 * gst/rtsp-server/rtsp-client.c:
8892 Add support for session keepalive
8893 Get and update the session timeout for all requests. get the session as early as
8896 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8898 * gst/rtsp-server/rtsp-media-factory.h:
8899 * gst/rtsp-server/rtsp-media.c:
8900 * gst/rtsp-server/rtsp-media.h:
8901 Handle media bus messages
8902 Handle media bus messages in a custom mainloop and dispatch them to the
8903 RTSPMedia objects. Let the default implementation handle some common messages.
8905 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8907 * gst/rtsp-server/rtsp-client.c:
8908 * gst/rtsp-server/rtsp-session-pool.c:
8909 * gst/rtsp-server/rtsp-session.c:
8910 Some more session timeout handling
8911 Move the session header setting code to a central place so that we always add
8912 the timeout parameter too.
8913 Handle timeouts by running the session cleanup code.
8914 Stop media before cleaning up.
8916 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8918 * gst/rtsp-server/rtsp-client.c:
8919 * gst/rtsp-server/rtsp-client.h:
8920 Add timeout property
8921 Add a timeout property ot the client and make the other properties into GObject
8924 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8926 * gst/rtsp-server/rtsp-session-pool.c:
8927 Use getters and setters in property code
8928 Use the getters and setters for the timeout property instead of locking
8931 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8933 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
8935 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8937 * gst/rtsp-server/rtsp-session-pool.c:
8938 * gst/rtsp-server/rtsp-session-pool.h:
8939 * gst/rtsp-server/rtsp-session.c:
8940 * gst/rtsp-server/rtsp-session.h:
8941 Add more timeout stuff
8942 Add method to check if a session is expired.
8943 Add method to perform cleanup on a session pool.
8945 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8947 * gst/rtsp-server/rtsp-client.c:
8948 * gst/rtsp-server/rtsp-session-pool.c:
8949 * gst/rtsp-server/rtsp-session-pool.h:
8950 * gst/rtsp-server/rtsp-session.c:
8951 * gst/rtsp-server/rtsp-session.h:
8952 Add beginnings of session timeouts and limits
8953 Add the timeout value to the Session header for unusual timeout values.
8954 Allow us to configure a limit to the amount of active sessions in a pool. Set a
8955 limit on the amount of retry we do after a sessionid collision.
8956 Add properties to the sessionid and the timeout of a session. Keep track of
8957 creation time and last access time for sessions.
8959 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8961 * gst/rtsp-server/rtsp-client.c:
8962 * gst/rtsp-server/rtsp-media.c:
8963 * gst/rtsp-server/rtsp-media.h:
8964 * gst/rtsp-server/rtsp-sdp.c:
8965 * gst/rtsp-server/rtsp-session-pool.c:
8966 * gst/rtsp-server/rtsp-session.c:
8967 * gst/rtsp-server/rtsp-session.h:
8968 Cleanup of sessions and more
8969 Fix the refcounting of media and sessions in the client. Properly clean up the
8970 session data when the client performs a teardown.
8971 Add Server header to responses.
8972 Allow for multiple uri setups in one session.
8973 Add Range header to the PLAY response and add the range attribute to the SDP
8975 Fix the session pool remove method, it used the wrong key in the hashtable. Also
8976 give the ownership of the sessionid to the session object.
8978 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8980 * gst/rtsp-server/rtsp-server.c:
8981 * gst/rtsp-server/rtsp-server.h:
8983 Rename the 'server_port' variable to simply 'port'.
8985 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8988 * gst/rtsp-server/rtsp-client.c:
8989 * gst/rtsp-server/rtsp-media.c:
8990 * gst/rtsp-server/rtsp-media.h:
8991 * gst/rtsp-server/rtsp-session.c:
8992 * gst/rtsp-server/rtsp-session.h:
8993 Rework the way we handle transports for streams
8994 Make the media accept an array of transports for the streams that we have
8995 configured for the play/pause requests.
8996 Implement server states for a client and its media.
8997 Require 0.10.22.1 (git HEAD) of gstreamer.
8999 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9001 * gst/rtsp-server/rtsp-client.c:
9002 * gst/rtsp-server/rtsp-media-factory.c:
9003 Drop const from functions dealing with urls
9004 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
9005 have the right const in them.
9007 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9009 * gst/rtsp-server/rtsp-client.c:
9010 * gst/rtsp-server/rtsp-media.c:
9011 * gst/rtsp-server/rtsp-sdp.c:
9015 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9017 * gst/rtsp-server/rtsp-client.c:
9018 * gst/rtsp-server/rtsp-media-factory.c:
9019 * gst/rtsp-server/rtsp-media.c:
9020 * gst/rtsp-server/rtsp-media.h:
9022 Don't keep a reference to the GstRTSPMedia in the stream.
9023 Free more things when freeing the GstRTSPMedia.
9025 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9028 * gst/rtsp-server/rtsp-media-factory.c:
9029 * gst/rtsp-server/rtsp-media-factory.h:
9030 * gst/rtsp-server/rtsp-media.c:
9031 * gst/rtsp-server/rtsp-media.h:
9032 * gst/rtsp-server/rtsp-server.c:
9033 * gst/rtsp-server/rtsp-server.h:
9034 More docs and small cleanups
9035 Add some more docs and update the README
9036 Cleanup some method names.
9037 Remove an unneeded idx field in the GstRTSPMediaStream
9039 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9042 * examples/Makefile.am:
9043 * examples/test-readme.c:
9044 Add a README and more example code
9045 Add a README file that contains a small introduction on how to use the server
9046 along with the example code explained in the readme.
9048 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9050 * gst/rtsp-server/rtsp-media.c:
9051 * gst/rtsp-server/rtsp-server.c:
9052 Fix some leaks and change default port
9053 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
9054 we finished the initial preroll. If we keep them locked, setting the pipeline to
9055 NULL will not stop and clean up the sources correctly.
9056 Change the default RTSP port to 8554 aka the official alternative RTSP port.
9058 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9060 * gst/rtsp-server/rtsp-session.c:
9061 * gst/rtsp-server/rtsp-session.h:
9062 Cleanups to the session object
9063 Remove some unneeded variables in the session state of a stream such as the
9064 owner media and the server transport.
9065 Get the configuration of a media stream in a session based on the media_stream
9066 in the original object instead of our cached index.
9067 Free more data in the finalize method.
9069 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9071 * gst/rtsp-server/rtsp-client.c:
9072 * gst/rtsp-server/rtsp-client.h:
9073 Cleanups and reuse media from DESCRIBE
9074 Handle thread create errors.
9075 Rename some internal methods to better match what they actually do.
9076 Handle misconfiguration of session_pool and media_mapping gracefully.
9077 Cache the DESCRIBE media and uri in the client connection and reuse them when
9078 we receive a SETUP request in the same connection for the same uri.
9079 Cleanup the client connection object.
9081 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9083 * gst/rtsp-server/rtsp-media-factory.c:
9084 * gst/rtsp-server/rtsp-media-factory.h:
9085 * gst/rtsp-server/rtsp-media.c:
9086 * gst/rtsp-server/rtsp-media.h:
9087 Add shared properties to media and factory
9088 Add the shared property to media.
9089 Implement some simple caching in the factory depending on if the media is shared
9092 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9094 * gst/rtsp-server/rtsp-client.c:
9095 Add a little comment
9096 Add some comment about the content-base header.
9098 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9100 * examples/Makefile.am:
9102 * examples/test-mp4.c:
9103 * examples/test-ogg.c:
9104 * examples/test-video.c:
9105 * gst/rtsp-server/Makefile.am:
9106 * gst/rtsp-server/rtsp-client.c:
9107 * gst/rtsp-server/rtsp-client.h:
9108 * gst/rtsp-server/rtsp-media-factory.c:
9109 * gst/rtsp-server/rtsp-media-factory.h:
9110 * gst/rtsp-server/rtsp-media.c:
9111 * gst/rtsp-server/rtsp-media.h:
9112 * gst/rtsp-server/rtsp-sdp.c:
9113 * gst/rtsp-server/rtsp-sdp.h:
9114 * gst/rtsp-server/rtsp-server.c:
9115 * gst/rtsp-server/rtsp-server.h:
9116 * gst/rtsp-server/rtsp-session.c:
9117 * gst/rtsp-server/rtsp-session.h:
9118 Reorganize things, prepare for media sharing
9119 Added various other test server examples
9120 Move the SDP message generation to a separate helper.
9121 Refactor common code for finding the session.
9122 Add content-base for realplayer compatibility
9123 Clean up request uris before processing for better vlc compatibility.
9124 Move prerolling and pipeline construction to the RTSPMedia object.
9125 Use multiudpsink for future pipeline reuse.
9127 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9133 === release 0.10.1 ===
9135 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9141 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9143 * bindings/vala/Makefile.am:
9145 Add more directories and files to the dist.
9147 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9149 * bindings/python/Makefile.am:
9150 * bindings/python/rtspserver.override:
9151 Fixed compile error of python bindings
9153 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9155 * bindings/vala/gst-rtsp-server-0.10.vapi:
9156 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9157 Marked values as nullable accordingly
9159 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9161 * bindings/vala/gst-rtsp-server-0.10.vapi:
9162 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
9163 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9164 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9165 Updated Vala bindings
9167 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9169 * gst/rtsp-server/rtsp-client.c:
9170 * gst/rtsp-server/rtsp-media-mapping.c:
9171 * gst/rtsp-server/rtsp-media-mapping.h:
9172 * gst/rtsp-server/rtsp-media.h:
9173 * gst/rtsp-server/rtsp-session-pool.h:
9174 Cleanups and doc updates
9175 Add some more documentation and do some minor cleanups here and there.
9177 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9179 * gst/rtsp-server/rtsp-client.c:
9180 * gst/rtsp-server/rtsp-media-factory.c:
9181 * gst/rtsp-server/rtsp-media-factory.h:
9182 * gst/rtsp-server/rtsp-media.c:
9183 * gst/rtsp-server/rtsp-media.h:
9184 * gst/rtsp-server/rtsp-session.c:
9185 * gst/rtsp-server/rtsp-session.h:
9187 Rename GstRTSPMediaBin to GstRTSPMedia
9188 Parse the request url into a GstRTSPUri object and pass this object to the
9189 various handlers and methods that require the uri.
9191 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9195 Add some more docs and remove some old code from the example.
9197 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9199 * gst/rtsp-server/rtsp-client.c:
9200 Handle state change failures better
9201 Handle state change failures better when changing the state of the pipeline to
9204 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9206 * gst/rtsp-server/rtsp-media-factory.c:
9207 * gst/rtsp-server/rtsp-media-factory.h:
9208 Make element creation more extendible
9209 Add get_element vmethod to the default MediaFactory so that subclasses can just
9210 override that method and still use the default logic for making a MediaBin from
9213 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9216 * gst/rtsp-server/Makefile.am:
9217 * gst/rtsp-server/rtsp-client.c:
9218 * gst/rtsp-server/rtsp-client.h:
9219 * gst/rtsp-server/rtsp-media-factory.c:
9220 * gst/rtsp-server/rtsp-media-factory.h:
9221 * gst/rtsp-server/rtsp-media-mapping.c:
9222 * gst/rtsp-server/rtsp-media-mapping.h:
9223 * gst/rtsp-server/rtsp-media.c:
9224 * gst/rtsp-server/rtsp-media.h:
9225 * gst/rtsp-server/rtsp-server.c:
9226 * gst/rtsp-server/rtsp-server.h:
9227 * gst/rtsp-server/rtsp-session.c:
9228 * gst/rtsp-server/rtsp-session.h:
9229 Make the server handle arbitrary pipelines
9230 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
9231 The GstMediaBin object has a handle to a bin with elements and to a list of
9232 GstMediaStream objects that this bin produces.
9233 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
9234 with methods to register and remove those mappings.
9235 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
9236 used by the server instance.
9237 Modify the example application so that it shows how to create custom pipelines
9238 attached to a specific mount point.
9239 Various misc cleanps.
9241 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9243 * gst/rtsp-server/rtsp-server.c:
9244 * gst/rtsp-server/rtsp-server.h:
9245 Allow setting a custom media factory for a server
9247 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9249 * gst/rtsp-server/rtsp-client.c:
9250 * gst/rtsp-server/rtsp-client.h:
9251 Allow setting a custom media factory for a client.
9253 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9255 * gst/rtsp-server/Makefile.am:
9256 Add Makefile entry for the media factory
9258 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9260 * gst/rtsp-server/rtsp-media-factory.c:
9261 * gst/rtsp-server/rtsp-media-factory.h:
9262 Add media factory to map urls to media pipeline objects.
9264 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9266 * gst/rtsp-server/rtsp-media.c:
9267 * gst/rtsp-server/rtsp-media.h:
9268 Add comments. Remove unused field
9270 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9272 * gst/rtsp-server/rtsp-session-pool.c:
9273 * gst/rtsp-server/rtsp-session-pool.h:
9274 Allow custom session pools to override the session id allocation algorithms Add some comments.
9276 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9278 * gst/rtsp-server/rtsp-session.h:
9281 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9283 * gst/rtsp-server/rtsp-client.c:
9284 * gst/rtsp-server/rtsp-client.h:
9285 Move the connection code in one place Add some comments
9287 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9289 * gst/rtsp-server/rtsp-server.c:
9290 * gst/rtsp-server/rtsp-server.h:
9291 Make vmethod to create and accept new clients. Add some docs.
9293 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9295 * gst/rtsp-server/rtsp-server.c:
9296 * gst/rtsp-server/rtsp-server.h:
9297 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
9299 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9301 * gst/rtsp-server/rtsp-client.c:
9302 * gst/rtsp-server/rtsp-client.h:
9303 Name the parameters more appropriately.
9305 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9307 * gst/rtsp-server/rtsp-session-pool.c:
9308 Do some more cleanup of the session pool.
9310 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9312 * gst/rtsp-server/Makefile.am:
9313 * gst/rtsp-server/rtsp-client.c:
9314 Check if return value of gst_rtsp_session_get_media is not NULL
9316 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9318 * gst/rtsp-server/Makefile.am:
9319 Install rtsp-session and rtsp-session-pool headers
9321 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9326 * bindings/python/Makefile.am:
9327 * bindings/python/arg-types.py:
9328 * bindings/python/codegen/Makefile.am:
9329 * bindings/python/codegen/__init__.py:
9330 * bindings/python/codegen/argtypes.py:
9331 * bindings/python/codegen/code-coverage.py:
9332 * bindings/python/codegen/codegen.py:
9333 * bindings/python/codegen/definitions.py:
9334 * bindings/python/codegen/defsparser.py:
9335 * bindings/python/codegen/docextract.py:
9336 * bindings/python/codegen/docgen.py:
9337 * bindings/python/codegen/fileprefix.override:
9338 * bindings/python/codegen/fileprefixmodule.c:
9339 * bindings/python/codegen/h2def.py:
9340 * bindings/python/codegen/mergedefs.py:
9341 * bindings/python/codegen/mkskel.py:
9342 * bindings/python/codegen/override.py:
9343 * bindings/python/codegen/reversewrapper.py:
9344 * bindings/python/codegen/scmexpr.py:
9345 * bindings/python/rtspserver-types.defs:
9346 * bindings/python/rtspserver.defs:
9347 * bindings/python/rtspserver.override:
9348 * bindings/python/rtspservermodule.c:
9350 Add python bindings.
9352 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9354 * bindings/Makefile.am:
9356 Don't go into python dir when requirements for python bindings are missing
9358 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9360 * bindings/Makefile.am:
9361 * bindings/vala/Makefile.am:
9363 Install Vala bindings if vala is available
9365 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9367 * bindings/vala/gst-rtsp-server-0.10.deps:
9368 * bindings/vala/gst-rtsp-server-0.10.vapi:
9369 * bindings/vala/gst-rtsp-server.vapi:
9370 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
9371 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
9372 * bindings/vala/packages/gst-rtsp-server-0.10.files:
9373 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9374 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9375 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
9376 * bindings/vala/packages/gst-rtsp-server.deps:
9377 * bindings/vala/packages/gst-rtsp-server.excludes:
9378 * bindings/vala/packages/gst-rtsp-server.files:
9379 * bindings/vala/packages/gst-rtsp-server.gi:
9380 * bindings/vala/packages/gst-rtsp-server.metadata:
9381 * bindings/vala/packages/gst-rtsp-server.namespace:
9382 Regenerated Vala bindings
9384 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9386 * bindings/vala/gst-rtsp-server.vapi:
9387 * bindings/vala/packages/gst-rtsp-server.metadata:
9388 Fixed typo in included headers for vala bindings
9390 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9394 * pkgconfig/Makefile.am:
9395 * pkgconfig/gst-rtsp-server.pc.in:
9396 Added pkgconfig file
9398 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
9400 * bindings/vala/gst-rtsp-server.vapi:
9401 * bindings/vala/packages/gst-rtsp-server.excludes:
9402 * bindings/vala/packages/gst-rtsp-server.gi:
9403 * bindings/vala/packages/gst-rtsp-server.metadata:
9404 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
9406 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
9408 * bindings/vala/gst-rtsp-server.vapi:
9409 * bindings/vala/packages/gst-rtsp-server.deps:
9410 * bindings/vala/packages/gst-rtsp-server.files:
9411 * bindings/vala/packages/gst-rtsp-server.gi:
9412 * bindings/vala/packages/gst-rtsp-server.metadata:
9413 * bindings/vala/packages/gst-rtsp-server.namespace:
9416 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
9418 * gst/rtsp-server/rtsp-session.c:
9419 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
9421 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
9423 * examples/Makefile.am:
9424 * gst/rtsp-server/Makefile.am:
9425 Put GStreamer version in library name
9427 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9429 * examples/Makefile.am:
9430 * gst/rtsp-server/Makefile.am:
9431 Fix some issues to pass distcheck
9433 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9435 * gst/rtsp-server/rtsp-server.c:
9436 Added port property to GstRTSPServer class.
9438 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9443 * examples/Makefile.am:
9446 * gst/rtsp-server/Makefile.am:
9447 * gst/rtsp-server/rtsp-client.c:
9448 * gst/rtsp-server/rtsp-client.h:
9449 * gst/rtsp-server/rtsp-media.c:
9450 * gst/rtsp-server/rtsp-media.h:
9451 * gst/rtsp-server/rtsp-server.c:
9452 * gst/rtsp-server/rtsp-server.h:
9453 * gst/rtsp-server/rtsp-session-pool.c:
9454 * gst/rtsp-server/rtsp-session-pool.h:
9455 * gst/rtsp-server/rtsp-session.c:
9456 * gst/rtsp-server/rtsp-session.h:
9459 * src/rtsp-client.c:
9460 * src/rtsp-client.h:
9463 * src/rtsp-server.c:
9464 * src/rtsp-server.h:
9465 * src/rtsp-session-pool.c:
9466 * src/rtsp-session-pool.h:
9467 * src/rtsp-session.c:
9468 * src/rtsp-session.h:
9469 Split in library and example program
9471 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
9473 * src/rtsp-client.h:
9474 Removed obsolete variable
9476 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
9478 * src/rtsp-client.c:
9479 * src/rtsp-client.h:
9480 Removed pipeline variable GstRTSPClient, because it's only used in one function
9482 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9485 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
9487 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
9489 * src/rtsp-session.c:
9490 Initialize some more vars.
9492 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
9494 * src/rtsp-session.c:
9495 Initialize variable to avoid compiler warning.
9497 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
9500 Add a reasonable generic .gitignore