3 2016-03-15 Sebastian Dröge <slomo@coaxion.net>
8 2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
10 * gst/rtsp-server/rtsp-stream.c:
11 rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
12 Without this, RECORD pipelines are broken because
13 a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
14 added later. Previously it was there earlier and due to NO_PREROLL caused the
15 pipeline to preroll immediately
16 b) the udpsrc for the pipeline is added later and never set to PLAYING state,
17 as the corresponding code previously was only for PLAY pipelines.
18 https://bugzilla.gnome.org/show_bug.cgi?id=763281
20 2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
22 * gst/rtsp-server/rtsp-stream.c:
23 rtsp-stream: Fix typo in the docstring
24 gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
26 2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
28 * gst/rtsp-server/rtsp-stream.c:
29 rtsp-stream: Disable multicast loopback for all our sockets
30 On Windows this is a receiver-side setting, on Linux a sender-side setting. As
31 we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
32 loopback setting on the socket... while udpsink does which unfortunately has
33 no effect here on Windows but on Linux.
34 https://bugzilla.gnome.org/show_bug.cgi?id=757488
36 2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
38 * tests/check/gst/stream.c:
39 stream tests: added new tests
40 Test a case when the address pool only contains multicast addresses
41 and the client is requesting unicast udp.
42 Added tests for multicast ports allocation.
43 https://bugzilla.gnome.org/show_bug.cgi?id=757488
45 2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
47 * gst/rtsp-server/rtsp-stream.c:
48 rtsp-stream: Only bind multicast sockets to ANY on Windows
49 On Linux it is still needed to bind to the multicast address
50 to filter out random other packets, while on Windows binding
51 to multicast addresses just fails.
53 2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
55 * gst/rtsp-server/rtsp-stream.c:
56 rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
57 Otherwise we fail to allocate UDP ports if the pool only contains multicast
58 addresses, which is something that used to work before. For unicast addresses
59 if the pool contains none, we just allocate them as if there is no pool at
61 https://bugzilla.gnome.org/show_bug.cgi?id=757488
63 2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
65 * gst/rtsp-server/rtsp-client.c:
66 * gst/rtsp-server/rtsp-stream.c:
67 rtsp-server: Fix indentation
69 2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
71 * gst/rtsp-server/rtsp-stream.c:
72 rtsp-stream: Don't bind the sockets to multicast addresses
73 This works on Linux but fails completely on Windows. You're supposed
74 to bind to ANY and then join the multicast group.
75 https://bugzilla.gnome.org/show_bug.cgi?id=757488
77 === release 1.7.90 ===
79 2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
85 * gst-rtsp-server.doap:
88 2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
91 Automatic update of common submodule
92 From b64f03f to 6f2d209
94 2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
96 * gst/rtsp-sink/gstrtspclientsink.c:
97 * tests/check/gst/rtspclientsink.c:
98 rtspsink: Fix some leaks in rtspclientsink and the unit test.
99 https://bugzilla.gnome.org/show_bug.cgi?id=762525
101 2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
103 * tests/check/gst/media.c:
104 * tests/check/gst/rtspclientsink.c:
105 * tests/check/gst/rtspserver.c:
106 * tests/check/gst/stream.c:
107 tests: unit test fixes
108 Removed port allocation test from the media suite.
109 The port allocation failure is now in the stream suite.
111 Make sure that the media is suspended after the DESCRIBE request
112 before reconfiguring the UDP sinks.
114 In the RECORD case we have to set async property to false
115 for the appsink element in the test in order to make sure
116 that the media pipeline doesn't hang in start_preroll().
117 https://bugzilla.gnome.org/show_bug.cgi?id=757488
119 2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
121 * gst/rtsp-server/rtsp-client.c:
122 * gst/rtsp-server/rtsp-stream.c:
123 * gst/rtsp-server/rtsp-stream.h:
124 rtsp-stream: postpone UDP socket allocation until SETUP
125 Postpone the allocation of the UDP sockets until we know
126 what transport has been chosen by the client.
127 Both unicast and multicast UDP sources are created in one
129 https://bugzilla.gnome.org/show_bug.cgi?id=757488
131 2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
133 * gst/rtsp-server/rtsp-stream.c:
134 rtsp-stream: postpone the creation of the UDP sources
135 Code refactoring: allocate the UDP ports after the sender and
136 the reciver parts have been created.
137 We postpone the creation of the UDP sources until the UDP
138 ports have been allocated.
139 https://bugzilla.gnome.org/show_bug.cgi?id=757488
141 2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
143 * gst/rtsp-server/rtsp-stream.c:
144 rtsp-stream: added function for setting UDP sources to PLAYING state
145 Code refactoring: Introduced a function for setting UDP sources
147 https://bugzilla.gnome.org/show_bug.cgi?id=757488
149 2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
151 * gst/rtsp-server/rtsp-stream.c:
152 rtsp-stream: added function for creating and configuring UDP sources
153 Code refactoring: create and configure UDP sources in a separate function.
154 https://bugzilla.gnome.org/show_bug.cgi?id=757488
156 2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
158 * gst/rtsp-server/rtsp-stream.c:
159 rtsp-stream: added function for RTP/RTCP socket configuration
160 Code refactoring: configure RTP and RTCP sockets for UDP sinks
161 in a separate function.
162 https://bugzilla.gnome.org/show_bug.cgi?id=757488
164 2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
166 * gst/rtsp-server/rtsp-stream.c:
167 rtsp-stream: added function for creating and configuring UDP sinks
168 Code refactoring: create and configure UDP sinks in a separate function.
169 https://bugzilla.gnome.org/show_bug.cgi?id=757488
171 2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
173 * gst/rtsp-server/rtsp-stream.c:
174 rtsp-stream: added helper function for creating the sender/receiver parts
175 Code refactoring: introduced helper function for creating
176 the receiver and the sender parts of the streaming pipeline.
177 https://bugzilla.gnome.org/show_bug.cgi?id=757488
179 2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
184 === release 1.7.2 ===
186 2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
192 * gst-rtsp-server.doap:
195 2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
197 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
198 uninstalled.pc: add support for non libtool build systems
199 Currently the .la path is provided which requires to use libtool as
200 mentioned in the GStreamer manual section-helloworld-compilerun.html.
201 It is fine as long as the application is built using libtool.
202 So currently it is not possible to compile a GStreamer application
203 within gst-uninstalled with CMake or other build system different
205 This patch allows to do the following in gst-uninstalled env:
206 gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
207 gstreamer-rtsp-server-1.0)
208 Previously it required to prepend libtool --mode=link
209 https://bugzilla.gnome.org/show_bug.cgi?id=720778
211 2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
213 * gst/rtsp-sink/gstrtspclientsink.c:
214 rtspclientsink: remove check for impossible condition
215 Goto error label checks stream to see if it needs to be unreferenced before
216 returning, but this goto jumps happens before the stream is ever set, so it
217 will always be NULL in this error label.
220 2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
222 * gst/rtsp-sink/gstrtspclientsink.c:
223 rtspclientsink: clean switch statements
224 Coverity demands for fallthrough statements to be clearly commented,
225 to distinguish from accidental fall throughs. And it also needs all
226 cases to finish with a break, even if the break is never going to be
227 executed like in the case of a continue jump.
231 2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
233 * tests/check/Makefile.am:
234 tests: extend the AM_TESTS_ENVIRONMENT from check.mak
235 To get the CK_DEFAULT_TIMEOUT defined for all tests
236 Also removes a 120 seconds timeout that was set as default
237 explicitly in this module
238 https://bugzilla.gnome.org/show_bug.cgi?id=761472
240 2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
244 Automatic update of common submodule
245 From 86e4663 to b64f03f
247 2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
249 * gst/rtsp-server/rtsp-media.c:
250 rtsp-media: fix state_lock not locked again when preroll fails
251 https://bugzilla.gnome.org/show_bug.cgi?id=761399
253 2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
256 configure: Move plugin specific flags below all the others
257 They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
258 -no-undefined. And -no-undefined is required on Windows to build DLLs.
260 2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
262 * gst/rtsp-sink/gstrtspclientsink.c:
263 rtspclientsink: Simplify slightly using new -base API
264 Use the new Mikey and SDP API in the base plugins libs
265 to simplify some code.
266 https://bugzilla.gnome.org/show_bug.cgi?id=758180
268 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
273 * gst/rtsp-sink/Makefile.am:
274 * gst/rtsp-sink/gstrtspclientsink.c:
275 * gst/rtsp-sink/gstrtspclientsink.h:
276 * gst/rtsp-sink/plugin.c:
277 * tests/check/Makefile.am:
278 * tests/check/gst/rtspclientsink.c:
279 rtspsink: Add rtspclientsink element
280 Add an rtspclientsink element that accepts streams for which
281 there is a registered payloader and sends them to
282 an RTSP server using RECORD.
283 Sending is synchronised to the pipeline clock. Payload-types
284 are automatically selected. The 'new-payloader' signal is fired
285 for custom configuration of payloaders when they are created.
286 Can now stream a movie like this:
288 ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
289 decodebin name=depay1 ! audioconvert ! autoaudiosink )"
291 gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
292 queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
293 https://bugzilla.gnome.org/show_bug.cgi?id=758180
295 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
297 * gst/rtsp-server/rtsp-stream.c:
298 * gst/rtsp-server/rtsp-stream.h:
299 rtsp-stream: Add functions for using rtsp-stream from the client
300 Add a boolean to indicate that the rtsp-stream is running on the
301 'client' side of an RTSP connection, for sending streams via
302 RECORD. In that case, the roles of the client/server ports
303 in transport setup are swapped.
304 https://bugzilla.gnome.org/show_bug.cgi?id=758180
306 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
308 * gst/rtsp-server/rtsp-sdp.c:
309 * gst/rtsp-server/rtsp-sdp.h:
310 rtsp-sdp: Add gst_rtsp_sdp_from_stream()
311 A new function that adds info from a GstRTSPStream into an SDP message.
312 https://bugzilla.gnome.org/show_bug.cgi?id=758180
314 2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
316 * gst/rtsp-server/rtsp-media.c:
317 rtsp-media: Fix mutex beeing unlocked while they should be locked
318 https://bugzilla.gnome.org/show_bug.cgi?id=761226
320 2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
322 * gst/rtsp-server/rtsp-media-factory.c:
323 rtsp-media-factory: add missing break in "clock" property setter
326 2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
328 * gst/rtsp-server/rtsp-stream.c:
329 rtsp-stream: fixed assert during update transport
330 When RTSP server trying update transport during multicast, it throws an
331 assert. The assert is thrown because it is trying to get the parent of
332 an non-existing funnel element.
333 https://bugzilla.gnome.org/show_bug.cgi?id=760150
335 2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
337 * gst/rtsp-server/rtsp-permissions.h:
338 * gst/rtsp-server/rtsp-thread-pool.h:
339 * gst/rtsp-server/rtsp-token.h:
340 docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
341 gtk-doc can handle static inline functions just fine these days,
342 there's no need for this stuff any more.
344 2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
346 * gst/rtsp-server/rtsp-media.c:
347 * gst/rtsp-server/rtsp-sdp.c:
348 sdp: replace duplicated codes to call new base sdp apis
349 https://bugzilla.gnome.org/show_bug.cgi?id=745880
351 2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
353 * examples/test-netclock.c:
354 test-netclock: Use the new API to configure a clock directly
356 2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
358 * gst/rtsp-server/rtsp-media-factory.c:
359 * gst/rtsp-server/rtsp-media-factory.h:
360 * gst/rtsp-server/rtsp-media.c:
361 * gst/rtsp-server/rtsp-media.h:
362 rtsp-media: Add API to directly configure a clock on the media pipelines
364 2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
366 * gst/rtsp-server/rtsp-media.c:
367 rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
369 2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
371 * gst/rtsp-server/rtsp-media-factory.c:
372 rtsp-media-factory: Add FIXME for 2.0
374 2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
376 * gst/rtsp-server/rtsp-stream.c:
377 rtsp-stream: Fix indentation
379 2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
381 * gst/rtsp-server/rtsp-media.c:
382 rtsp-media: Do not prepare media after media times out
383 Deferred calls to start_prepare() can be deferred past the point until
384 which wait_preroll() and by proxy gst_rtsp_media_get_status() is
385 prepared to wait. Previously there was no lock and no check for this
386 situation. This meant that a media could be prepared and unprepared
387 simultaneously by two different threads. Now a lock is in place and a
388 suitable check is done.
389 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
391 2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
393 * gst/rtsp-server/rtsp-client.c:
394 * gst/rtsp-server/rtsp-media-factory.c:
395 * gst/rtsp-server/rtsp-media-factory.h:
396 * gst/rtsp-server/rtsp-media.c:
397 * gst/rtsp-server/rtsp-media.h:
398 rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
399 Without TEARDOWN it might be desireable to keep the media running and continue
400 sending data to the client, even if the RTSP connection itself is
402 Only do this for session medias that have only UDP transports. If there's at
403 least on TCP transport, it will stop working and cause problems when the
404 connection is disconnected.
405 https://bugzilla.gnome.org/show_bug.cgi?id=758999
407 2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
412 === release 1.7.1 ===
414 2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
420 * gst-rtsp-server.doap:
423 2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
426 configure: Make -Bsymbolic check work with clang.
427 Update the -Bsymbolic check with the version glib has. This version
429 https://bugzilla.gnome.org/show_bug.cgi?id=759713
431 2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
433 * gst/rtsp-server/rtsp-session-pool.c:
434 rtsp-session-pool: Avoid dollar sign ($) in session ids
435 Live555 in VLC strips off dollar signs and then gets very confused,
436 we don't loose too much entropy by just skipping it.
438 2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
440 * gst/rtsp-server/rtsp-address-pool.h:
441 * gst/rtsp-server/rtsp-auth.h:
442 * gst/rtsp-server/rtsp-client.h:
443 * gst/rtsp-server/rtsp-media-factory-uri.h:
444 * gst/rtsp-server/rtsp-media-factory.h:
445 * gst/rtsp-server/rtsp-media.h:
446 * gst/rtsp-server/rtsp-mount-points.h:
447 * gst/rtsp-server/rtsp-permissions.h:
448 * gst/rtsp-server/rtsp-server.h:
449 * gst/rtsp-server/rtsp-session-media.h:
450 * gst/rtsp-server/rtsp-session-pool.h:
451 * gst/rtsp-server/rtsp-session.h:
452 * gst/rtsp-server/rtsp-stream-transport.h:
453 * gst/rtsp-server/rtsp-stream.h:
454 * gst/rtsp-server/rtsp-thread-pool.h:
455 * gst/rtsp-server/rtsp-token.h:
456 rtsp-server: Add g_autoptr() support to all types
457 https://bugzilla.gnome.org/show_bug.cgi?id=754464
459 2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
461 * gst/rtsp-server/rtsp-stream.c:
462 rtsp-stream: fixed valgrind error
463 Fixed the valgrind error in unit test. The UDP source created during
464 gst_rtsp_stream_join_bin() was not released while destroying the rtp
466 https://bugzilla.gnome.org/show_bug.cgi?id=759010
468 2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
472 Automatic update of common submodule
473 From b319909 to 86e4663
475 2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
477 * gst/rtsp-server/rtsp-client.c:
478 rtsp-client: suspend media during setup request
479 SETUP request from clients needs to suspend the media to clear the
480 prerolled buffers. Otherwise it will not affect the prerolled buffer
481 and the prerolled buffers will be incorrect (for example block-size
482 from setup request will not affect the prerolled buffer unless the
484 https://bugzilla.gnome.org/show_bug.cgi?id=758268
486 2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
488 * gst/rtsp-server/rtsp-stream.c:
489 rtsp-stream: create stream pipeline based on transport
490 Based on the protocol, create the rtsp stream pipeline. If only TCP or
491 only UDP is set as the transport protocol, it will not add the extra tee
492 or queue element to the pipeline. Both these elements will be added, if
493 it supports both TCP and UDP protocols. This improves the pipeline
494 performance when one protocol is present.
495 https://bugzilla.gnome.org/show_bug.cgi?id=758179
497 2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
499 * gst/rtsp-server/rtsp-stream.c:
500 rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
501 Adding them when not needed will start some logic inside rtpbin that might be
502 problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
503 would start up a rtpjitterbuffer and behave in weird ways.
504 We still set up the UDP sources for RTP receiving for a sender media to be
505 able to receive any packets sent by the client for NAT traversal. They will
506 all go to a fakesink though.
507 Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
508 NO_PREROLL, which will cause deadlocks when seeking the media as it will never
509 receive ASYNC_DONE after a seek.
510 https://bugzilla.gnome.org/show_bug.cgi?id=758319
512 2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
514 * gst/rtsp-server/rtsp-stream.c:
515 rtsp-stream: Disable multicast loopback for the multicast udp sources too
516 On POSIX this setting is for sender sockets, on Windows for receiver sockets.
517 Previously we were only setting this for sender sockets, which caused looped
518 back packets to be received on Windows if a multicast transport was used.
520 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
522 * examples/test-record-auth.c:
523 * examples/test-record.c:
524 examples: Actually use the provided port in the record examples
526 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
528 * examples/test-record-auth.c:
529 test-record-auth: Add the option to build in TLS support
531 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
533 * examples/test-auth.c:
534 test-auth: Use an 'anonymous' user for unauthenticated default
535 There's a comment on one of the resources that 'user' and 'admin'
536 shouldn't even be able to see it, but they can if the default
537 token is 'admin2', since that gives them access anyway.
539 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
541 * examples/.gitignore:
542 * examples/Makefile.am:
543 * examples/test-record-auth.c:
544 Add test-record-auth example
546 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
548 * gst/rtsp-server/rtsp-client.c:
549 * tests/check/gst/client.c:
550 rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
552 2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
554 * gst/rtsp-server/rtsp-server.c:
555 rtsp-server: Change the logic so we don't pop a NULL context
556 When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
557 will sometimes fail. This call is made before any context is pushed
558 resulting in an attempt to pop a NULL context.
559 https://bugzilla.gnome.org/show_bug.cgi?id=757949
561 2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
563 * tests/check/gst/rtspserver.c:
564 rtspserver: Add udp-mcast transport SETUP test
565 Refactor utility functions in the test file so they can handle
566 more than UDP and TCP as lower transport.
567 https://bugzilla.gnome.org/show_bug.cgi?id=756969
569 2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
571 * gst/rtsp-server/rtsp-stream.c:
572 rtsp-stream: Always unref return value of gst_object_get_parent()
573 Fixes a leak of a GstBin in the udp-mcast case.
574 https://bugzilla.gnome.org/show_bug.cgi?id=756968
576 2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
579 Automatic update of common submodule
580 From b99800a to b319909
582 2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
585 Use new GST_ENABLE_EXTRA_CHECKS #define
586 https://bugzilla.gnome.org/show_bug.cgi?id=756870
588 2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
591 Automatic update of common submodule
592 From 6babecd to b99800a
594 2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
597 Update GLib dependency to 2.40.0
599 2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
601 * examples/test-mp4.c:
602 * gst/rtsp-server/rtsp-stream.c:
603 stream: listen to sender ssrc signals
604 https://bugzilla.gnome.org/show_bug.cgi?id=746747
606 2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
609 common: update for new suppression
610 Makes check-valgrind pass with glib 2.46
612 2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
614 * gst/rtsp-server/rtsp-media.c:
615 rtsp-media: Take reference to media that will be prepared
616 default_prepare() takes a transfer-none reference GstRTSPMedia object.
617 Later on a g_idle_source_new() is created and a pointer to the media
618 object is passed as user data. If the media is freed before the idle
619 source is dispatched the media object pointer is invalid, but the idle
620 source callback expects it to still be valid. To fix this a reference to
621 the media object is taken when registering the source callback function
622 and a corresponding release of the reference is done when the souce is
624 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
626 2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
628 * examples/test-launch.c:
629 * examples/test-mp4.c:
630 * examples/test-ogg.c:
631 * examples/test-record.c:
632 * examples/test-uri.c:
633 rtsp-server: Fix memory leaks when context parse fails
634 When g_option_context_parse fails, context and error variables are not getting free'd
635 which results in memory leaks. Free'ing the same.
636 And replacing g_error_free with g_clear_error, which checks if the error being passed
637 is not NULL and sets the variable to NULL on free'ing.
638 https://bugzilla.gnome.org/show_bug.cgi?id=753863
640 2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
645 === release 1.6.0 ===
647 2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
653 * gst-rtsp-server.doap:
656 === release 1.5.91 ===
658 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
664 * gst-rtsp-server.doap:
667 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
669 * docs/libs/gst-rtsp-server-sections.txt:
670 * gst/rtsp-server/rtsp-stream.c:
671 stream: fix docs for recently-added get/set_buffer_size API
672 https://bugzilla.gnome.org/show_bug.cgi?id=749095
674 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
676 * gst/rtsp-server/rtsp-media.c:
677 rtsp-media: Don't crash on encrypted RTX SDP
678 In parse_keymgmt(), don't mutate the input string that's been passed
679 as const, especially since we might need the original value again if
680 the same key info applies to multiple streams (RTX, for example).
681 https://bugzilla.gnome.org/show_bug.cgi?id=754753
683 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
685 * examples/test-mp4.c:
686 test-mp4: Support filenames with spaces in them. Error out on too few arguments
688 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
690 * examples/test-record.c:
691 test-record: Check parameter count and print out help
692 If no launch pipeline was supplied, print out some help
694 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
696 * gst/rtsp-server/rtsp-media.c:
697 * gst/rtsp-server/rtsp-stream.c:
698 * gst/rtsp-server/rtsp-stream.h:
699 rtsp-stream: Implement UDP buffer size setting.
700 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
702 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
703 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
705 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
707 * gst/rtsp-server/rtsp-media.h:
708 rtsp-media: Fix small typo causing gtk-doc to complain
710 === release 1.5.90 ===
712 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
718 * gst-rtsp-server.doap:
721 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
723 * gst/rtsp-server/rtsp-media-factory.c:
724 media-factory: get port number through gst_rtsp_url_get_port
725 https://bugzilla.gnome.org/show_bug.cgi?id=753473
727 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
729 * tests/check/gst/media.c:
730 media-test: Removing unnecessary assertion
731 https://bugzilla.gnome.org/show_bug.cgi?id=753385
733 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
735 * gst/rtsp-server/rtsp-server.c:
736 Document that source keeps a ref on server until it's destroyed
737 https://bugzilla.gnome.org/show_bug.cgi?id=749227
739 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
741 * tests/check/gst/media.c:
742 media-test: Test for multiple dynamic payload
743 https://bugzilla.gnome.org/show_bug.cgi?id=753385
745 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
747 * gst/rtsp-server/rtsp-media.c:
748 media: Only add fakesink once per pipeline
749 The intention is to prevent going PLAYING state before pads are created.
750 If there was mutilple dynamic payload, it would leak few fakesink and
751 actually prevent from ever reaching playing state.
752 https://bugzilla.gnome.org/show_bug.cgi?id=753385
754 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
756 * gst/rtsp-server/rtsp-media.c:
757 Revert "rtsp-media: Only add 1 fakesink per pipeline"
758 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
760 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
762 * gst/rtsp-server/rtsp-media.c:
763 rtsp-media: Only add 1 fakesink per pipeline
764 There should be only one fakesink per pipeline, not per dynpay. This
765 would lead to element naming clash.
767 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
769 * gst/rtsp-server/rtsp-media.c:
770 rtsp-media: assertion error due to wrong condition check
771 In media to caps function, reserved_keys array is being used for variable i,
772 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
773 changed it to variable j
774 https://bugzilla.gnome.org/show_bug.cgi?id=753009
776 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
778 * gst/rtsp-server/rtsp-media.c:
779 rtsp-media: Strip keys from the fmtp that we use internally in our caps
780 Skip keys from the fmtp, which we already use ourselves for the
781 caps. Some software is adding random things like clock-rate into
782 the fmtp, and we would otherwise here set a string-typed clock-rate
783 in the caps... and thus fail to create valid RTP caps
784 https://bugzilla.gnome.org/show_bug.cgi?id=753009
786 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
788 * gst/rtsp-server/rtsp-thread-pool.c:
789 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
790 https://bugzilla.gnome.org/show_bug.cgi?id=752640
792 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
795 Automatic update of common submodule
796 From f74b2df to 9aed1d7
798 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
803 === release 1.5.2 ===
805 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
811 * gst-rtsp-server.doap:
814 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
816 * gst/rtsp-server/rtsp-client.c:
817 * gst/rtsp-server/rtsp-client.h:
818 * tests/check/gst/client.c:
819 rtsp-client: allow application to decide what requirements are supported
820 Add "check-requirements" signal and vfunc to allow application
821 (and subclasses) to check the requirements.
822 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
823 https://bugzilla.gnome.org/show_bug.cgi?id=749417
825 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
828 Automatic update of common submodule
829 From 6015d26 to f74b2df
831 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
833 * gst/rtsp-server/rtsp-media.c:
834 rtsp-media: Always use real payloader when creating streams
835 A bin that contains the real payloader might be used as payloader. In this
836 case we have to get the real payloader for the various properties it provides.
837 Example use cases for this are bins that payload some media and then have
838 additional elements that add metadata or RTP extension headers to the stream.
839 https://bugzilla.gnome.org/show_bug.cgi?id=750800
841 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
843 * examples/test-netclock-client.c:
844 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
846 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
848 * examples/test-netclock-client.c:
849 * examples/test-netclock.c:
850 test-netclock: Use new ntp-time-source property on rtpbin
851 Select the clock time to be used as NTP time source. This allows proper
852 synchronization between receivers, independent of sharing base times, and just
853 requires them to use the same clock.
855 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
857 * examples/test-netclock-client.c:
858 * examples/test-netclock.c:
859 test-netclock: Setting the same base time on sender and receiver is not necessary
860 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
862 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
864 * gst/rtsp-server/rtsp-stream.c:
865 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
866 https://bugzilla.gnome.org/show_bug.cgi?id=750764
868 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
870 * docs/libs/gst-rtsp-server.types:
871 docs: add missing types
872 https://bugzilla.gnome.org/show_bug.cgi?id=750764
874 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
876 * docs/libs/gst-rtsp-server-sections.txt:
877 docs: add missing apis
878 https://bugzilla.gnome.org/show_bug.cgi?id=750764
880 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
882 * examples/test-netclock-client.c:
883 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
885 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
887 * docs/libs/gst-rtsp-server-sections.txt:
888 * gst/rtsp-server/rtsp-auth.c:
889 * gst/rtsp-server/rtsp-auth.h:
890 GstRTSPAuth: Add client certificate authentication support
891 https://bugzilla.gnome.org/show_bug.cgi?id=750471
893 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
895 * examples/test-netclock-client.c:
896 test-netclock-client: Use new GstClock API to wait for clock synchronization
898 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
900 * examples/test-netclock-client.c:
901 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
902 A mainloop is needed to get glimagesink to display something on OSX, and
903 the source-setup signal just makes things a little bit easier.
905 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
908 Automatic update of common submodule
909 From d9a3353 to 6015d26
911 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
914 Automatic update of common submodule
915 From d37af32 to d9a3353
917 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
920 Automatic update of common submodule
921 From 21ba2e5 to d37af32
923 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
926 Automatic update of common submodule
927 From c408583 to 21ba2e5
929 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
931 * docs/libs/Makefile.am:
932 docs: remove variables that we define in the snippet from common
933 This is syncing our Makefile.am with upstream gtkdoc.
935 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
938 Automatic update of common submodule
939 From 44a3517 to c408583
941 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
946 === release 1.5.1 ===
948 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
954 * gst-rtsp-server.doap:
957 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
959 * gst/rtsp-server/rtsp-client.c:
960 rtsp-client: No flush during Teardown.
961 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
962 backlog is empty it can happen that just a part of a message will be
963 sent and rest is in backlog queue. If then flush during teardown
964 just a part of message will be sent.This can lead to client miss
965 teardown response since it expect to get the last part of message.
966 The flushing during teardown was introduced to fix a deadlock that now
967 is fixed more generally in handle_request by temporary setting backlog
969 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
971 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
973 * tests/check/Makefile.am:
974 tests: Use AM_TESTS_ENVIRONMENT
975 Needed by the new automake test runner and the
976 current version of the common submodule.
978 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
980 * gst/rtsp-server/rtsp-media.h:
981 * gst/rtsp-server/rtsp-stream.h:
982 rtsp-server: Use single-include rtsp header to make sure we get all definitions
984 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
986 * gst/rtsp-server/rtsp-media.c:
987 rtsp-media: Mark some more functions static
989 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
991 * gst/rtsp-server/rtsp-media.c:
992 rtsp-media: Only unblock the media in suspend() when actually changing the state
993 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
995 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
997 * examples/test-video-rtx.c:
998 examples: Use AVPF profile for the RTX example
1000 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
1002 * gst/rtsp-server/rtsp-sdp.c:
1003 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
1005 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1007 * gst/rtsp-server/rtsp-stream.c:
1008 rtsp-stream: get valid clock-rate from last-sample
1009 clock-rate in last-sample's caps is integer, not unsigned.
1010 To get this value properly, variable needs to be type-casted to int.
1011 https://bugzilla.gnome.org/show_bug.cgi?id=747614
1013 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
1017 autogen.sh: only run autopoint if gettext requested in configure.ac
1018 Not just because there happens to be a po directory.
1019 https://bugzilla.gnome.org/show_bug.cgi?id=748058
1021 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
1024 Revert "configure.ac: uncomment gettext version setup"
1025 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
1026 We don't need a gettext setup here and there's no po
1027 directory either, so no reason why autopoint would be
1028 run in the first place.
1029 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
1031 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
1033 * examples/test-multicast.c:
1034 * examples/test-multicast2.c:
1035 * examples/test-sdp.c:
1036 * examples/test-video-rtx.c:
1037 * examples/test-video.c:
1038 * tests/test-cleanup.c:
1039 * tests/test-reuse.c:
1040 Fix timeout function signatures across tests and examples
1042 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
1044 * tests/check/Makefile.am:
1045 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
1046 Make sure the test environment is set up.
1047 https://bugzilla.gnome.org//show_bug.cgi?id=747624
1049 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
1052 configure: bump automake requirement to 1.14 and autoconf to 2.69
1053 This is only required for builds from git, people can still
1054 build tarballs if they only have older autotools.
1055 https://bugzilla.gnome.org//show_bug.cgi?id=747624
1057 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
1060 configure.ac: uncomment gettext version setup
1061 Fixes autogen.sh. It would run autopoint, which would complain
1062 that it could not find the gettext version in configure.ac.
1063 https://bugzilla.gnome.org/show_bug.cgi?id=748058
1065 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1067 * examples/test-video-rtx.c:
1068 test-video-rtx: set exact payload type to PCMA payloader
1069 Setting wrong payload type causes failure to do retransmission through audio stream
1070 https://bugzilla.gnome.org/show_bug.cgi?id=747839
1072 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1074 * gst/rtsp-server/rtsp-media.c:
1075 * gst/rtsp-server/rtsp-stream.c:
1076 * gst/rtsp-server/rtsp-stream.h:
1077 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
1078 Because of duplicated g_signal_connect for request-aux-sender signal,
1079 wrong stream pointer is passed to the signal handler.
1080 Instead of passing each stream, pass stream array and get the relevant stream.
1081 https://bugzilla.gnome.org/show_bug.cgi?id=747839
1083 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
1087 Update autogen.sh to latest version from common
1088 Fixes build after aclocal_check etc. helpers have been removed.
1090 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
1093 Automatic update of common submodule
1094 From bc76a8b to c8fb372
1096 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
1098 * gst/rtsp-server/rtsp-stream.c:
1099 rtsp-stream: Limit the queues to 1 buffer
1100 We only need them to be able to pre-roll, queueing up more data here
1101 is only going to harm latency and memory usage.
1103 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
1105 * gst/rtsp-server/rtsp-stream.c:
1106 rtsp-stream: Update comment and ASCII art to the latest code
1107 We have a queue in front of the udpsink too to prevent the pipeline from
1110 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
1112 * gst/rtsp-server/rtsp-stream.c:
1113 rtsp-media: Properly return first rtptime
1114 Instead we where returning first GstBuffer timestamp. This would result
1115 in clock skew and unwanted behaviour in RTSP playback.
1116 https://bugzilla.gnome.org/show_bug.cgi?id=746479
1118 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
1120 * gst/rtsp-server/rtsp-stream.c:
1121 rtsp-stream: Don't leave buffer mapped
1122 If the seq is NULL, the RTP buffer was left mapped. We should always
1125 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
1130 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
1132 * gst/rtsp-server/rtsp-media-factory.c:
1133 * tests/check/gst/client.c:
1134 Fix double semicolons
1136 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
1138 * gst/rtsp-server/rtsp-stream.c:
1139 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
1140 This gives more accurate values than asking the payloader. There might be
1141 queueing happening between the payloader and the sink.
1142 https://bugzilla.gnome.org/show_bug.cgi?id=745704
1144 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
1146 * gst/rtsp-server/rtsp-media.c:
1147 rtsp-media: Don't seek for PLAY if the position will not change
1148 https://bugzilla.gnome.org/show_bug.cgi?id=745704
1150 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
1152 * gst/rtsp-server/rtsp-media.c:
1153 rtsp-media: Don't include payload type in the caps for framesize
1154 When the sdp media attribute framesize are converted to caps
1155 the <payload> should not be included.
1156 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
1157 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
1159 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
1161 * gst/rtsp-server/rtsp-sdp.c:
1162 rtsp-sdp: add payload type to the sdp framesize attribute
1163 The sdp framesize attribute is desribed in RFC6064. It is specified
1164 for payloading of H263 and has the following form
1165 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
1166 should be added to the caps in a payloader and the <payload type> should
1167 be added by the rtsp-server.
1168 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
1170 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
1172 * examples/test-uri.c:
1173 examples: test-uri: fix tainted variable
1174 Insignificant but this keeps Coverity happy.
1177 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
1179 * examples/.gitignore:
1180 * examples/Makefile.am:
1181 * examples/test-netclock-client.c:
1182 * examples/test-netclock.c:
1183 examples: Add a simple example of network synch for live streams.
1184 An example server and client that works for synchronising live streams
1185 only - as it can't support pause/play.
1187 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
1189 * gst/rtsp-server/rtsp-media-factory.c:
1190 * gst/rtsp-server/rtsp-media-factory.h:
1191 rtsp-media-factory: Add functions to set/get the media gtype
1192 Allow specifying the GType of a GstRtspMedia subclass to create
1193 as a simpler way to get the factory to create a custom
1194 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
1196 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
1198 * gst/rtsp-server/rtsp-media.c:
1199 rtsp-media: fix double unlock in _get_buffer_size()
1200 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
1201 because of double g_mutex_unlock () usage.
1202 https://bugzilla.gnome.org/show_bug.cgi?id=745434
1204 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
1206 * gst/rtsp-server/rtsp-session-pool.c:
1207 * gst/rtsp-server/rtsp-session.c:
1208 * gst/rtsp-server/rtsp-session.h:
1209 rtsp-session: Use monotonic time for RTSP session timeout
1210 Changed RTSP session timeout handling to monotonic time
1211 and deprecating the API for current system time.
1212 This fixes timeouts when the system time changes.
1213 https://bugzilla.gnome.org/show_bug.cgi?id=743346
1215 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
1217 * gst/rtsp-server/rtsp-client.c:
1218 * gst/rtsp-server/rtsp-media.c:
1219 rtsp-client: Only error out in PLAY if seeking actually failed
1220 If the media was just not seekable, we continue from whatever position we are
1221 and let the client decide if that is what is wanted or not.
1222 Only if the actual seek failed, we can't really recover and should error out.
1224 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
1226 * gst/rtsp-server/rtsp-stream.c:
1227 rtsp-stream: Add necessary queues between tee and multiudpsink
1228 https://bugzilla.gnome.org/show_bug.cgi?id=744379
1230 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
1232 * gst/rtsp-server/rtsp-client.c:
1233 * gst/rtsp-server/rtsp-media.c:
1234 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
1235 Instead error out properly the same way as if the SEEKING query already
1238 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
1240 * gst/rtsp-server/rtsp-stream.h:
1241 rtsp-stream: minor code formatting fix
1243 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
1245 * gst/rtsp-server/rtsp-media.c:
1246 rtsp-media: fix logic for collect_streams
1247 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
1248 all streams it knows if it got any, and can check if the transport mode is OK.
1251 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
1253 * gst/rtsp-server/rtsp-media.c:
1254 rtsp-media: Don't set the transport mode based on what elements we find
1255 Just print a warning if the one that was set before disagrees with what
1256 elements we found. It must already be set to something before as this
1257 function is called after we received the SDP from ANNOUNCE in RECORD mode,
1258 and we would reject ANNOUNCE if the RECORD flag was not set.
1260 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
1262 * tests/check/gst/rtspserver.c:
1263 tests: rtspserver: rename shadowed variable
1264 We have two different 'sink' variables here,
1265 rename one of them for clarity.
1267 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
1269 * gst/rtsp-server/rtsp-client.c:
1270 rtsp-client: fix awkward if clause
1272 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
1274 * examples/test-uri.c:
1275 examples: test-uri: improve uri argument handling and accept file names
1276 Print an error if the argument passed is not a URI and can't
1277 be converted into one, or no arguments have been provided.
1279 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
1281 * examples/test-uri.c:
1282 examples: test-uri: don't remove mount point after 10 seconds
1283 It's very irritating when trying to test stuff repeatedly
1284 and serves no real purpose other than showing that it can
1287 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
1289 * examples/.gitignore:
1290 examples: add new test-record to .gitignore
1292 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
1294 * examples/test-record.c:
1295 * gst/rtsp-server/rtsp-client.c:
1296 * gst/rtsp-server/rtsp-media-factory.c:
1297 * gst/rtsp-server/rtsp-media-factory.h:
1298 * gst/rtsp-server/rtsp-media.c:
1299 * gst/rtsp-server/rtsp-media.h:
1300 * tests/check/gst/rtspserver.c:
1301 rtsp-media: Use flags to distinguish between PLAY and RECORD media
1303 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
1305 * examples/test-record.c:
1306 test-record: Set latency for playback-style example to 2s instead of 200ms
1308 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
1310 * tests/check/gst/rtspserver.c:
1311 tests: add some unit tests for ANNOUNCE and RECORD
1312 https://bugzilla.gnome.org/show_bug.cgi?id=743175
1314 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
1316 * gst/rtsp-server/rtsp-client.c:
1317 rtsp-client: fix a couple of leaks in handle_announce
1319 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
1321 * gst/rtsp-server/rtsp-media-factory.c:
1322 * gst/rtsp-server/rtsp-media-factory.h:
1323 * gst/rtsp-server/rtsp-media.c:
1324 * gst/rtsp-server/rtsp-media.h:
1325 rtsp-media: Expose latency setting for setting the rtpbin latency
1327 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
1329 * examples/test-record.c:
1330 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
1332 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
1334 * gst/rtsp-server/rtsp-stream.c:
1335 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
1337 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
1339 * examples/Makefile.am:
1340 * examples/test-record.c:
1341 * gst/rtsp-server/rtsp-client.c:
1342 * gst/rtsp-server/rtsp-client.h:
1343 * gst/rtsp-server/rtsp-media-factory.c:
1344 * gst/rtsp-server/rtsp-media-factory.h:
1345 * gst/rtsp-server/rtsp-media.c:
1346 * gst/rtsp-server/rtsp-media.h:
1347 * gst/rtsp-server/rtsp-session-media.c:
1348 * gst/rtsp-server/rtsp-stream.c:
1349 * gst/rtsp-server/rtsp-stream.h:
1350 Add initial support for RECORD
1351 We currently only support media that is RECORD or PLAY only, not both at once.
1352 https://bugzilla.gnome.org/show_bug.cgi?id=743175
1354 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
1356 * gst/rtsp-server/rtsp-stream.c:
1357 rtsp-stream: RTCP and RTP transport cache cookies seperated
1358 RTCP packets were not sent because the same tr_cache_cookie was used for
1359 both RTP and RTCP. So only one of the tr_cache lists were populated
1360 depending on which one was sent first. If the tr_cache list is not
1361 populated then no packets can be sent. Most often this happened to be
1362 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
1363 resulted in both the tr_cache_lists to be populated regardless of which
1365 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
1367 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
1369 * gst/rtsp-server/rtsp-stream.c:
1370 rtsp-stream: fix false compiler warning
1371 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
1373 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
1375 * gst/rtsp-server/rtsp-client.c:
1376 rtsp-client: log interleaved data received
1378 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
1380 * gst/rtsp-server/rtsp-client.c:
1381 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
1383 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
1385 * gst/rtsp-server/rtsp-client.c:
1386 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
1388 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
1390 * gst/rtsp-server/rtsp-client.c:
1391 rtsp-client: Use a random session ID in the SDP
1392 RFC4566 Section 5.2 says that it should make the username, session id,
1393 nettype, addrtype and unicast address tuple globally unique. Always using
1394 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
1395 Instead let's create a 64 bit random number, which at least brings us
1396 closer to the goal of global uniqueness.
1397 https://tools.ietf.org/html/rfc4566#section-5.2
1399 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
1401 * examples/test-launch.c:
1402 * examples/test-mp4.c:
1403 * examples/test-ogg.c:
1404 * examples/test-uri.c:
1405 examples: Don't call gst_init() and gst_get_option_group()
1406 The latter calls the former at the appropriate time.
1408 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
1410 * gst/rtsp-server/rtsp-client.c:
1411 rtsp-client: Drop trailing \0 of RTSP DATA messages
1412 We add a trailing \0 in GstRTSPConnection to make parsing of
1413 string message bodies easier (e.g. the SDP from DESCRIBE) but
1414 for actual data this means we have to drop it or otherwise
1415 create invalid data.
1417 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
1419 * gst/rtsp-server/rtsp-stream.c:
1420 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
1421 Fixes crash when two threads access handle_new_sample() at the same
1422 time, one for RTP, one for RTCP.
1423 Otherwise, when iterating over the transports cache, it might be modified by
1424 another thread at the same time if the transports cookie has changed.
1425 https://bugzilla.gnome.org/show_bug.cgi?id=742954
1427 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
1429 * gst/rtsp-server/rtsp-stream.c:
1430 rtsp-stream: Set format=TIME on our app sources for TCP
1432 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
1434 * gst/rtsp-server/rtsp-session-pool.c:
1435 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
1436 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
1437 RFC 2326 states that session IDs may consist of alphanumeric as well as
1438 the safe characters $-_.+ -- N.B. the percent character is not allowed.
1439 Previously the session ID was URI-escaped, this meant that any character
1440 which was not alphanumeric or any of the characters +-._~ would be
1441 percent encoded. While the RFC (surprisingly) mentions that linear white
1442 space in session IDs should be URI-escaped, it does not say anything
1443 about other characters. Moreover no white space is allowed in the
1444 session ID. Finally the percent character which is the result of
1445 URI-escaping is not allowed in a session ID.
1446 So there is no reason to do any URI-escaping, and now it is removed.
1447 https://bugzilla.gnome.org/show_bug.cgi?id=742869
1449 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
1452 Automatic update of common submodule
1453 From f2c6b95 to bc76a8b
1455 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
1458 Fix 'make check' from top-level directory
1460 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
1462 * examples/test-launch.c:
1463 * examples/test-mp4.c:
1464 * examples/test-ogg.c:
1465 * examples/test-uri.c:
1466 examples: Add command-line parsing and take a 'port' argument
1467 This allows users to run multiple servers on different ports for testing.
1468 Only done for examples that actually take arguments and hence are capable of
1469 outputting different streams for each instance on each port.
1470 https://bugzilla.gnome.org/show_bug.cgi?id=742115
1472 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
1474 * gst/rtsp-server/rtsp-client.c:
1475 * gst/rtsp-server/rtsp-client.h:
1476 rtsp-client: Add a send_message default signal handler
1477 This allows subclasses to easily hook into the response sending
1478 mechanism without doing everything from a signal, which seems
1479 awkward from subclasses.
1481 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
1484 Automatic update of common submodule
1485 From ef1ffdc to f2c6b95
1487 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1491 configure: add --disable-examples switch
1492 https://bugzilla.gnome.org/show_bug.cgi?id=741678
1494 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
1496 * examples/.gitignore:
1497 * examples/Makefile.am:
1498 * examples/test-video-rtx.c:
1499 examples: add a retransmisison example implementing RFC4588
1500 Currently only SSRC-multiplexed rtx streams are supported
1502 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
1504 * gst/rtsp-server/rtsp-stream.c:
1505 rtsp-stream: Fix some minor memory leaks
1507 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
1509 * gst/rtsp-server/rtsp-media.c:
1510 rtsp-media: Some minor cleanup
1512 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
1514 * gst/rtsp-server/rtsp-stream.c:
1515 rtsp-stream: Fix compiler warnings
1516 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
1517 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1519 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
1520 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1523 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
1525 * docs/libs/gst-rtsp-server-sections.txt:
1526 * gst/rtsp-server/rtsp-media-factory.c:
1527 * gst/rtsp-server/rtsp-media-factory.h:
1528 * gst/rtsp-server/rtsp-media.c:
1529 * gst/rtsp-server/rtsp-media.h:
1530 * gst/rtsp-server/rtsp-sdp.c:
1531 * gst/rtsp-server/rtsp-stream.c:
1532 * gst/rtsp-server/rtsp-stream.h:
1533 media: implement ssrc-multiplexed retransmission support
1534 based off RFC 4588 and the server-rtpaux example in -good
1536 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
1538 * gst/rtsp-server/rtsp-client.c:
1539 * gst/rtsp-server/rtsp-stream-transport.c:
1540 * gst/rtsp-server/rtsp-stream.c:
1541 rtsp: Ref transports in hash table.
1542 Also ref streams for transports.
1543 This solves a crash when reciving a rtcp after teardown but before
1545 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
1547 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
1550 Automatic update of common submodule
1551 From 7bb2bce to ef1ffdc
1553 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
1555 * gst/rtsp-server/rtsp-client.c:
1556 client: refactor cleanup of cached media
1558 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
1560 * tests/check/gst/client.c:
1562 The session leak is now fixed, lets remove those FIXME comments.
1564 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
1566 * tests/check/gst/rtspserver.c:
1567 tests: Test to setup two sessions on one connection
1568 https://bugzilla.gnome.org/show_bug.cgi?id=739112
1570 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
1572 * tests/check/gst/rtspserver.c:
1573 tests: Test setup with tcp transport
1574 https://bugzilla.gnome.org/show_bug.cgi?id=739112
1576 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
1578 * gst/rtsp-server/rtsp-client.c:
1579 client: Configure transport after creating session media
1580 The default implementation of configure_client_transport() in
1581 rtsp-client uses the session media when it chooses channels for
1582 interleaved traffic.
1583 https://bugzilla.gnome.org/show_bug.cgi?id=739112
1585 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
1587 * gst/rtsp-server/rtsp-client.c:
1588 * gst/rtsp-server/rtsp-session-media.c:
1589 client: Stop caching media in client when doing setup
1590 If the media has been managed by a session media, it should not be
1591 cached in the client any longer. The GstRTSPSessionMedia object is now
1592 responsible for unpreparing the GstRTSPMedia object using
1593 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
1595 https://bugzilla.gnome.org/show_bug.cgi?id=739112
1597 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1599 * gst/rtsp-server/rtsp-stream.c:
1600 rtsp-stream: unref srtp decoder when leaving bin
1601 https://bugzilla.gnome.org/show_bug.cgi?id=739481
1603 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1605 * gst/rtsp-server/rtsp-client.c:
1606 rtsp-client: mikey memory leaks
1607 https://bugzilla.gnome.org/show_bug.cgi?id=739383
1609 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
1612 Automatic update of common submodule
1613 From 84d06cd to 7bb2bce
1615 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
1618 Parallelise 'make check-valgrind'
1620 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
1623 Automatic update of common submodule
1624 From a8c8939 to 84d06cd
1626 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
1629 Automatic update of common submodule
1630 From 36388a1 to a8c8939
1632 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
1634 * gst/rtsp-server/rtsp-media.c:
1635 rtsp-media: deactivate media when shutting down from paused
1636 This was only done when going directly from playing.
1637 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
1639 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1641 * gst/rtsp-server/rtsp-client.c:
1642 * gst/rtsp-server/rtsp-context.h:
1643 rtsp-client: add stream transport to context
1644 We add the stream transport to the context so we can get the configured
1645 client stream transport in the setup request signal.
1646 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
1648 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1650 * gst/rtsp-server/rtsp-stream.c:
1651 stream: release lock even not all transports have been removed
1652 We don't want to keep the lock even we return FALSE because not all the
1653 transports have been removed. This could lead into a deadlock.
1654 https://bugzilla.gnome.org/show_bug.cgi?id=737797
1656 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
1658 * gst/rtsp-server/rtsp-sdp.c:
1659 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
1660 These were renamed in GstRTPBasePayload in 1.0
1662 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1664 * gst/rtsp-server/rtsp-client.c:
1665 client: set session media to NULL without the lock
1666 We need to set session medias to NULL without the client lock otherwise
1667 we can end up in a deadlock if another thread is waiting for the lock
1668 and media unprepare is also waiting for that thread to end.
1669 https://bugzilla.gnome.org/show_bug.cgi?id=737690
1671 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
1673 * gst/rtsp-server/rtsp-media.c:
1674 rtsp-media: Set state to UNPREPARING in all cases
1676 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
1678 * gst/rtsp-server/rtsp-media.c:
1679 media: set state to unpreparing when unprepare is initiated
1680 https://bugzilla.gnome.org/show_bug.cgi?id=737675
1682 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
1684 * gst/rtsp-server/rtsp-client.c:
1685 rtsp-client: Remove backlog limit while processings requests
1686 If the backlog limit is kept two cases of deadlocks may be
1687 encountered when streaming over TCP. Without the backlog
1688 limit this deadlocks can not happen, at the expence of
1690 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
1692 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
1694 * gst/rtsp-server/rtsp-client.c:
1695 rtsp-client: do not free main context before rtsp watch
1696 https://bugzilla.gnome.org/show_bug.cgi?id=737110
1698 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
1700 * tests/check/gst/rtspserver.c:
1701 tests: Extend unit test timeout to accomodate for valgrind
1702 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1704 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
1706 * gst/rtsp-server/rtsp-client.c:
1707 * gst/rtsp-server/rtsp-session.c:
1708 * gst/rtsp-server/rtsp-stream-transport.c:
1709 rtsp-*: Treat sending packets to clients as keepalive
1710 As long as gst-rtsp-server can successfully send RTP/RTCP data to
1711 clients then the client must be reading. This change makes the server
1712 timeout the connection if the client stops reading.
1713 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1715 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
1717 * gst/rtsp-server/rtsp-client.c:
1718 rtsp-client: Allow backlog to grow while expiring session
1719 Allow the send backlog in the RTSP watch to grow to unlimited size while
1720 attempting to bring the media pipeline to NULL due to a session
1721 expiring. Without this change the appsink element cannot change state
1722 because it is blocked while rendering data in the new_sample callback.
1723 This callback will block until it has successfully put the data into the
1724 send backlog. There is a chance that the send backlog is full at this
1725 point which means that the callback may block for a long time, possibly
1726 forever. Therefore the media pipeline may also be prevented from
1727 changing state for a long time.
1728 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1730 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
1732 * gst/rtsp-server/rtsp-client.c:
1733 rtsp-client: Make old compilers happy
1734 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
1735 Just in case that guint8 doesn't fit in a pointer. Just in case ...
1737 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
1739 * gst/rtsp-server/rtsp-client.c:
1740 client: raise the backlog limits before pausing
1741 We need to raise the backlog limits before pausing the pipeline or else
1742 the appsink might be blocking in the render method in wait_backlog() and
1743 we would deadlock waiting for paused.
1744 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
1746 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
1748 * gst/rtsp-server/rtsp-client.c:
1749 client: make define for the WATCH_BACKLOG
1750 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
1752 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
1754 * gst/rtsp-server/rtsp-client.c:
1755 client: simplify session transport handling
1756 link/unlink of the transport in a session was done to keep track of all
1757 TCP transports and to send RTP/RTCP data to the streams. We can simplify
1758 that by putting all the TCP transports in a hashtable indexed with the
1760 We also don't need to link/unlink the transports when we pause/resume
1761 the streams. The same effect is already achieved when we pause/play the
1762 media. Indeed, when we pause the media, the transport is removed from
1763 the media and the callbacks will not be called anymore.
1764 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
1766 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
1768 * gst/rtsp-server/rtsp-stream-transport.c:
1769 * gst/rtsp-server/rtsp-stream-transport.h:
1770 stream-transport: make method to handle received data
1771 Make a method to handle the data received on a channel. It sends the
1772 data to the stream of the transport on the RTP or RTCP pads based on
1775 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
1777 * examples/test-mp4.c:
1778 test: add example of dumping RTCP reports
1780 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
1782 * gst/rtsp-server/rtsp-media.c:
1783 * gst/rtsp-server/rtsp-stream.c:
1784 * gst/rtsp-server/rtsp-stream.h:
1785 rtsp-media: Make sure that sequence numbers are monotonic after pause
1786 The sequence number is not monotonic for RTP packets after pause. The
1787 reason is basepayloader generates a randon sequence number when the
1788 pipeline goes from ready to pause. With this fix generation of sequence
1789 number will be monotonic when going from pause to play request.
1790 https://bugzilla.gnome.org/show_bug.cgi?id=736017
1792 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
1794 * gst/rtsp-server/rtsp-client.c:
1795 rtsp-client: Protect saved clients watch with a mutex
1796 Fixes a crash when close() is called while merging clients
1797 in handle_tunnel(). In that case close() would destroy the
1798 watch while it is still being used in handle_tunnel().
1799 https://bugzilla.gnome.org/show_bug.cgi?id=735570
1801 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
1803 * gst/rtsp-server/rtsp-stream.c:
1804 rtsp-stream: Remove the multicast group udp sources when removing from the bin
1806 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
1808 * gst/rtsp-server/rtsp-media.c:
1809 * gst/rtsp-server/rtsp-stream.c:
1810 * gst/rtsp-server/rtsp-stream.h:
1811 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
1812 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
1813 seeking and will always continue counting the time. This leads to
1814 the NPT after a backwards seek to be something completely different
1815 to the actual seek position.
1816 https://bugzilla.gnome.org/show_bug.cgi?id=732644
1818 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
1820 * examples/test-appsrc.c:
1821 examples: fix another reference leak
1822 gst_rtsp_media_get_element() returns a new ref.
1824 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1826 * examples/test-appsrc.c:
1827 examples: unref element after usage
1828 gst_bin_get_by_name_recurse_up() returns an element
1829 reference that must be unreffed after usage.
1830 https://bugzilla.gnome.org/show_bug.cgi?id=734546
1832 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
1834 * gst/rtsp-server/rtsp-media.c:
1835 signals: Fix copy-pasto in target-state signal offset
1837 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
1841 Makefile: Add usage of build-checks step
1842 Allows building checks without running them
1844 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
1846 * gst/rtsp-server/rtsp-stream.c:
1847 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
1848 When a UDP multicast transport is used it is expected that the server listens
1849 for RTP and RTCP packets on the multicast group with the corresponding port.
1850 Without this we will never get RTCP packets from clients in multicast mode.
1851 https://bugzilla.gnome.org/show_bug.cgi?id=732238
1853 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
1858 === release 1.4.0 ===
1860 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
1866 * gst-rtsp-server.doap:
1869 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
1871 * gst/rtsp-server/rtsp-media.h:
1872 media: correct misspelled words in description
1873 https://bugzilla.gnome.org/show_bug.cgi?id=733244
1875 === release 1.3.91 ===
1877 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
1883 * gst-rtsp-server.doap:
1886 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
1888 * docs/libs/gst-rtsp-server-sections.txt:
1891 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
1893 * gst/rtsp-server/rtsp-server.c:
1894 server: implement client REMOVE filter
1896 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
1898 * gst/rtsp-server/rtsp-client.c:
1899 * gst/rtsp-server/rtsp-client.h:
1900 client: expose _close() method
1901 Expose a previously internal close method to close the client
1904 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
1906 * gst/rtsp-server/rtsp-session-pool.c:
1907 session-pool: signal session-removed outside of the lock
1908 Release the lock before emiting the session-removed signal.
1910 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
1912 * gst/rtsp-server/rtsp-client.c:
1913 * gst/rtsp-server/rtsp-server.c:
1914 * gst/rtsp-server/rtsp-session-pool.c:
1915 * gst/rtsp-server/rtsp-session.c:
1916 * gst/rtsp-server/rtsp-stream.c:
1917 filter: Release lock in filter functions
1918 Release the object lock before calling the filter functions. We need to
1919 keep a cookie to detect when the list changed during the filter
1920 callback. We also keep a hashtable to make sure we only call the filter
1921 function once for each object in case of concurrent modification.
1922 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
1924 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
1926 * gst/rtsp-server/rtsp-client.c:
1927 client: check if watch is set in handle_teardown()
1928 The unit tests run without a watch
1930 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
1932 * tests/check/gst/client.c:
1933 client tests: send teardown to cleanup session
1935 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
1937 * tests/check/gst/rtspserver.c:
1938 server tests: send teardown to cleanup session
1940 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
1942 * gst/rtsp-server/rtsp-client.c:
1943 client: keep ref to client for the session removed handler
1944 This extra ref will be dropped when all client sessions have been
1945 removed. A session is removed when a client sends teardown, closes its
1946 endpoint of the TCP connection or the sessions expires.
1947 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
1949 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
1951 * gst/rtsp-server/rtsp-client.c:
1952 * gst/rtsp-server/rtsp-session.c:
1953 * tests/check/gst/client.c:
1954 client: manage media in session as a last step
1955 Once we manage a media in a session, we can't unmanage it anymore
1956 without destroying it. Therefore, first check everything before we
1957 manage the media, otherwise if something is wrong we have no way to
1959 If we created a new session and something went wrong, remove the session
1960 again. Fixes a leak in the unit test.
1962 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
1964 * examples/test-mp4.c:
1965 * examples/test-ogg.c:
1966 examples: print 'stream ready at url' for mp4 and ogg example
1968 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
1970 * gst/rtsp-server/rtsp-client.c:
1971 * gst/rtsp-server/rtsp-sdp.c:
1972 rtsp: fix for MIKEY api change
1974 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
1976 * gst/rtsp-server/rtsp-client.c:
1977 client: free watch context only once
1978 The watch context is freed when the source is destroyed. Avoids
1979 a CRITICAL when we try to unref the context twice.
1981 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
1983 * gst/rtsp-server/rtsp-client.c:
1986 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
1988 * gst/rtsp-server/rtsp-client.c:
1989 client: protect sessions with lock
1990 Protect the list of sessions with the lock.
1991 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
1993 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
1995 * gst/rtsp-server/rtsp-client.c:
1996 Client: keep a ref to the session
1997 Don't just keep a weak ref to the session objects but use a hard ref. We
1998 will be notified when a session is removed from the pool (expired) with
1999 the new session-removed signal.
2000 Don't automatically close the RTSP connection when all the sessions of
2001 a client are removed, a client can continue to operate and it can create
2002 a new session if it wants. If you want to remove the client from the
2003 server, you have to use gst_rtsp_server_client_filter() now.
2004 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
2005 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
2007 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
2009 * gst/rtsp-server/rtsp-session-pool.c:
2010 * gst/rtsp-server/rtsp-session-pool.h:
2011 session-pool: add session-removed signal
2012 Add a signal to be notified when a session is removed from the pool.
2014 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
2016 * gst/rtsp-server/Makefile.am:
2017 * gst/rtsp-server/rtsp-server.h:
2018 Make rtsp-server.h a single-include header, use it for G-I
2019 https://bugzilla.gnome.org/show_bug.cgi?id=732411
2021 === release 1.3.90 ===
2023 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
2029 * gst-rtsp-server.doap:
2032 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
2034 * gst/rtsp-server/rtsp-stream.c:
2035 stream: crypto can be NULL
2037 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
2039 * gst/rtsp-server/rtsp-client.c:
2040 * gst/rtsp-server/rtsp-media.c:
2041 * gst/rtsp-server/rtsp-mount-points.c:
2042 introspection: add missing allow-none annotations
2043 https://bugzilla.gnome.org/show_bug.cgi?id=730952
2045 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
2047 * gst/rtsp-server/rtsp-address-pool.c:
2048 * gst/rtsp-server/rtsp-media.c:
2049 * gst/rtsp-server/rtsp-session-media.c:
2050 * gst/rtsp-server/rtsp-session-pool.c:
2051 * gst/rtsp-server/rtsp-stream-transport.c:
2052 * gst/rtsp-server/rtsp-stream.c:
2053 * gst/rtsp-server/rtsp-token.c:
2054 introspection: add (nullable) annotations to return values
2055 https://bugzilla.gnome.org/show_bug.cgi?id=730952
2057 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
2059 * gst/rtsp-server/rtsp-client.c:
2060 * gst/rtsp-server/rtsp-stream.c:
2061 gi: improve annotations
2062 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
2064 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
2066 * gst/rtsp-server/rtsp-client.c:
2067 * gst/rtsp-server/rtsp-media-factory.c:
2068 * gst/rtsp-server/rtsp-media.c:
2069 * gst/rtsp-server/rtsp-server.c:
2070 signals: use generic marshal function
2071 Use the generic C marshal function.
2072 Use more explicit type instead of G_TYPE_POINTER
2074 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
2076 * gst/rtsp-server/rtsp-context.h:
2077 context: add type macro
2079 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
2081 * gst/rtsp-server/rtsp-client.c:
2082 * gst/rtsp-server/rtsp-sdp.c:
2083 * gst/rtsp-server/rtsp-sdp.h:
2084 sdp: hide key length defines
2085 They don't have a namespace.
2087 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
2092 === release 1.3.3 ===
2094 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
2100 * gst-rtsp-server.doap:
2103 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2105 * gst/rtsp-server/rtsp-client.c:
2106 * gst/rtsp-server/rtsp-sdp.c:
2107 * gst/rtsp-server/rtsp-sdp.h:
2108 mikey: add different key length parameters
2109 Add encryption and authentication key length parameters to MIKEY. For
2110 the encoders, the key lengths are obtained from the cipher and auth
2111 algorithms set in the caps. For the decoders, they are obtained while
2112 parsing the key management from the client.
2113 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
2115 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
2117 * tests/check/gst/stream.c:
2118 stream tests: Make sure we get right multicast address from stream
2119 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
2121 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
2123 * gst/rtsp-server/rtsp-client.c:
2124 client: ref the context until rtsp watch is alive
2125 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
2127 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
2129 * gst/rtsp-server/rtsp-client.c:
2130 client: Destroy the rtsp watch after connection close
2132 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
2134 * gst/rtsp-server/rtsp-media.c:
2135 media: fix confusing comment
2137 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
2139 * gst/rtsp-server/rtsp-session.c:
2140 rtsp-session: Timeout in header.
2141 Adding the possbilty to always have timout in header.
2142 This is configurabe with setting "timeout-always-visible".
2143 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
2145 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
2150 === release 1.3.2 ===
2152 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
2159 * gst-rtsp-server.doap:
2162 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
2165 Automatic update of common submodule
2166 From 211fa5f to 1f5d3c3
2168 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
2170 * gst/rtsp-server/rtsp-client.c:
2171 client: store TCP ports in transport
2172 Store the TCP ports in the transport when we are doing RTSP over TCP.
2173 This way, we can easily get to the ports from the transport.
2174 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
2176 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2178 * gst/rtsp-server/rtsp-stream.c:
2179 stream: add signals for new RTP/RTCP encoders
2180 New signals to allow the user to configure the dynamically created
2182 https://bugzilla.gnome.org/show_bug.cgi?id=730228
2184 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
2186 * gst/rtsp-server/rtsp-media.c:
2187 * gst/rtsp-server/rtsp-media.h:
2188 media: Make suspend()/unsuspend() virtual
2189 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
2191 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2193 * gst/rtsp-server/rtsp-client.c:
2194 client: fix send-message signal marshaller
2195 Use generic marshalling for the send-message signal. It has
2196 two POINTER arguments, not just one.
2197 https://bugzilla.gnome.org/show_bug.cgi?id=729900
2199 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
2201 * tests/check/gst/media.c:
2202 tests: add and remove pads only once
2203 In this test we simulate a dynamic pad by watching the caps event.
2204 Because of renegotiation in the base payloader now, this caps is sent
2205 multiple times but we can only deal with 1 invocation, use a variable to
2206 only 'add and remove' the pad once.
2208 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
2210 * tests/check/gst/rtspserver.c:
2211 tests: add unit test for correct handling of Require headers
2212 https://bugzilla.gnome.org/show_bug.cgi?id=729426
2214 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
2216 * gst/rtsp-server/rtsp-client.c:
2217 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
2218 Servers must handle Require headers and must report a failure
2219 if they don't handle any of the Required options, see RFC 2326,
2220 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
2221 https://bugzilla.gnome.org/show_bug.cgi?id=729426
2223 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
2228 === release 1.3.1 ===
2230 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
2236 * gst-rtsp-server.doap:
2239 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
2242 Automatic update of common submodule
2243 From bcb1518 to 211fa5f
2245 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
2250 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
2252 * tests/check/gst/sessionmedia.c:
2253 tests: fix memory leak in sessionmedia unit test
2255 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
2257 * gst/rtsp-server/rtsp-client.c:
2258 client: emit a signal before sending a message
2259 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
2261 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
2263 * gst/rtsp-server/rtsp-client.c:
2264 client: pass context to send_message
2265 Pass the current context to send_message, we will need it later.
2267 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
2269 * gst/rtsp-server/rtsp-client.c:
2270 client: fix typo in comment
2272 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
2274 * gst/rtsp-server/rtsp-media.c:
2275 media: Do not stop thread twice if default_prepare() fails
2277 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
2279 * gst/rtsp-server/rtsp-client.c:
2280 client: set the watch to flushing before going to NULL
2281 First set the watch to flushing so that we unblock any current and
2282 future attempt to send data on the watch, Then set the pipeline to
2284 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
2286 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
2288 * gst/rtsp-server/rtsp-session-pool.c:
2289 * tests/check/gst/sessionpool.c:
2290 rtsp-session-pool: Fixes annotation
2291 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
2292 in the sessionpool test.
2293 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
2295 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
2297 * gst/rtsp-server/rtsp-media.c:
2298 * gst/rtsp-server/rtsp-media.h:
2299 media: make media_prepare virtual
2300 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
2302 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
2304 * gst/rtsp-server/rtsp-media.c:
2305 * tests/check/gst/media.c:
2306 media: stop the thread in more error cases
2308 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
2310 * gst/rtsp-server/rtsp-media.c:
2311 * tests/check/gst/media.c:
2312 media: allow NULL as the thread
2313 Use the default context whan passing a NULL thread.
2315 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
2317 * gst/rtsp-server/rtsp-client.c:
2318 rtsp-client: indent cleanup
2319 Coverity was moaning about unreachable code, and I think it was just
2320 confused by { being before the label. We'll see if it pops up again.
2323 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
2325 * gst/rtsp-server/rtsp-client.c:
2326 * gst/rtsp-server/rtsp-media.c:
2327 client: Add drop-backlog property
2328 When we have too many messages queued for a client (currently hardcoded
2329 to 100) we overflow and drop the messages. Add a drop-backlog property
2330 to control this behaviour. Setting this property to FALSE will retry
2331 to send the messages to the client by waiting for more room in the
2333 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
2335 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
2337 * gst/rtsp-server/rtsp-client.c:
2338 client: support for POST before GET when setting up a tunnel
2340 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
2342 * gst/rtsp-server/rtsp-client.c:
2343 client: remove watch of the second client after http tunnel setup
2344 The second client will be freed after the HTTP tunnel has been set up.
2345 Make sure it's RTSP watch is never dispatched again.
2346 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
2348 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
2350 * gst/rtsp-server/rtsp-media.c:
2351 * tests/check/gst/media.c:
2352 media: Make media_prepare() fail if port allocation fails
2353 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
2355 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
2357 * tests/check/gst/media.c:
2358 media test: cleanup the thread pool in tests
2360 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
2362 * gst/rtsp-server/rtsp-media.c:
2363 * tests/check/gst/media.c:
2364 rtsp-media: Unblock blocked streams in unprepare
2365 The streams will be blocked when a live media is prepared.
2366 The streams should be unblocked in gst_rtsp_media_unprepare.
2367 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
2369 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
2371 * gst/rtsp-server/rtsp-media.c:
2372 media: release the state lock when going to NULL
2373 Set our state to UNPREPARING and release the state-lock before
2374 setting the pipeline to the NULL state. This way, any pad-added
2375 callback will be able to take the state-lock and check that we are now
2376 unpreparing instead of deadlocking.
2377 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
2379 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
2381 * gst/rtsp-server/rtsp-media.c:
2382 media: protect status with lock
2383 Make sure we only update the status with the lock.
2385 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
2387 * gst/rtsp-server/rtsp-client.c:
2388 * gst/rtsp-server/rtsp-sdp.c:
2389 rtsp: update for MIKEY API changes
2391 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
2393 * gst/rtsp-server/rtsp-client.c:
2394 client: parse the mikey response from the client
2395 Parse the mikey response from the client and update the policy for
2398 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
2400 * gst/rtsp-server/rtsp-stream.c:
2401 * gst/rtsp-server/rtsp-stream.h:
2402 stream: add method to set crypto info
2403 Make a method to configure the crypto information of a stream.
2404 Set udpsrc in READY instead of PAUSED so that we can configure caps
2407 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
2409 * gst/rtsp-server/rtsp-client.c:
2410 client: cleanup error paths
2412 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
2414 * gst/rtsp-server/rtsp-media.c:
2417 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
2419 * examples/test-video.c:
2420 test: enable SRTP only on RTSPS
2421 We only want to enable SRTP when doing rtsp over TLS so that we can
2422 exchange the keys in a secure way.
2424 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
2426 * examples/test-video.c:
2427 test: print an error on failure
2429 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
2432 * examples/test-video.c:
2433 * gst/rtsp-server/rtsp-sdp.c:
2434 * gst/rtsp-server/rtsp-stream.c:
2435 * tests/check/Makefile.am:
2436 stream: add SRTP support
2437 Install srtp encoder and decoder elements in rtpbin
2440 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2442 * tests/check/Makefile.am:
2443 * tests/check/gst/sessionpool.c:
2444 tests: Add unit tests for sessionpool
2445 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
2447 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2449 * tests/check/gst/threadpool.c:
2450 tests: Improve code coverage of rtsp-threadpool tests
2451 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
2453 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2455 * tests/check/gst/sessionmedia.c:
2456 tests: Improve code coverage for rtsp-session-media
2457 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
2459 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2461 gobject-introspection: Add annotations to support language bindings
2462 In addition a few cosmetic changes:
2463 * Adjust the order of arguments
2464 * Fix typo: occured -> occurred
2465 * Fix indentation after Return:-clauses
2466 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
2468 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2470 * gst/rtsp-server/rtsp-stream.c:
2471 rtsp-stream: Don't mix IPv4 and IPv6 addresses
2472 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
2474 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
2476 * gst/rtsp-server/rtsp-stream.c:
2477 stream: take caps after the session manager
2478 Take the caps for the SDP after they leave the rtpbin so that we can
2479 also get the properties added by rtpbin elements.
2481 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
2483 * gst/rtsp-server/rtsp-stream.c:
2484 stream: release lock while pushing out packets
2485 Keep a cache of the transports and use this to iterate the transport
2486 while pushing packets. This allows us to release the lock early.
2487 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
2489 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
2491 * gst/rtsp-server/rtsp-client.c:
2492 * gst/rtsp-server/rtsp-client.h:
2493 rtsp-client: vmethod for modifying tunnel GET response
2494 Add a vmethod tunnel_http_response where the response to the HTTP GET
2495 for tunneled connections can be modified.
2496 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
2498 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
2500 * gst/rtsp-server/rtsp-sdp.c:
2501 sdp: make 1 media line per profile
2502 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
2503 line in the SDP for each profile. The client is then supposed to pick
2504 one of the profiles in the SETUP request. Because the m= lines have the
2505 same pt, the client also knows that only 1 option is possible.
2507 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
2509 * gst/rtsp-server/rtsp-media-factory.c:
2510 * gst/rtsp-server/rtsp-media-factory.h:
2511 * gst/rtsp-server/rtsp-media.c:
2512 factory: add profile property and pass to media and streams
2514 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
2516 * examples/test-multicast.c:
2517 * gst/rtsp-server/rtsp-sdp.c:
2518 sdp: pass multicast connection for multicast-only stream
2519 Pass the multicast address of the stream in the connection info in the
2520 SDP so that clients try a multicast connection first.
2521 Only allow multicast connections in the test-multicast example. Also
2522 increase the TTL a little.
2524 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2527 .gitignore: Ignore gcov intermediate files
2528 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
2530 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
2532 * gst/rtsp-server/rtsp-stream.c:
2533 stream: release some locks in error cases
2535 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2537 docs: Enable and fix gtk-doc warnings
2538 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
2539 * addresspool/mediafactory: Add missing annotation colon
2540 * stream: Annotate return value
2541 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
2543 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
2546 Automatic update of common submodule
2547 From fe1672e to bcb1518
2549 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
2552 Automatic update of common submodule
2553 From 1a07da9 to fe1672e
2555 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
2557 * examples/Makefile.am:
2558 examples: use LDADD for libs instead of LDFLAGS
2560 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
2563 configure: make sure releases are in .doap file
2565 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
2567 * examples/test-cgroups.c:
2568 examples: test-cgroups: don't put code with side effects into g_assert()
2569 The g_assert() might get compiled out with the right
2570 compiler/preprocessor flags.
2572 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
2574 * examples/.gitignore:
2575 examples: add cgroup test binary to .gitignore
2577 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
2579 * examples/test-cgroups.c:
2580 examples: fix cgroup test build
2581 Fixes build failure caused by compiler warning:
2582 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
2584 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
2587 .gitignore: ignore temp files created in the course of 'make check'
2589 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
2591 * gst/rtsp-server/rtsp-media.c:
2592 rtsp-media: don't loose frames handling new PLAY request
2593 If client supplied a range check if the range specifies the start point.
2594 If not, then do an accurate seek to the current position. If a start
2595 point was specified do do a key unit seek to make sure the streaming
2596 starts with decodeable frames.
2597 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
2599 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
2601 * gst/rtsp-server/rtsp-media.c:
2602 Revert "media: only flush when setting a new start position"
2603 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
2604 We need to do the flush in all cases, demuxer block currently for
2607 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
2609 * gst/rtsp-server/rtsp-media.c:
2610 media: only flush when setting a new start position
2611 Only flush the pipeline when we change the start position with
2613 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
2615 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
2617 * gst/rtsp-server/rtsp-stream.c:
2618 stream: set ttl-mc before adding the socket
2619 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
2620 never be set on socket.
2621 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
2623 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
2625 * gst/rtsp-server/rtsp-media.c:
2626 media: stop thread if media is already prepared
2627 in gst_rtsp_media_prepare() the thread is not used if media is already
2628 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
2630 https://bugzilla.gnome.org/show_bug.cgi?id=724182
2632 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
2635 build: Ship gst-rtsp-server.doap file
2637 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
2639 * tests/check/gst/rtspserver.c:
2640 tests: Fix another compiler warning with gcc
2642 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
2644 * gst/rtsp-server/rtsp-client.c:
2645 * gst/rtsp-server/rtsp-mount-points.c:
2646 * gst/rtsp-server/rtsp-stream.c:
2647 * tests/check/gst/client.c:
2648 rtsp-server: Fix lots of compiler warnings with clang
2650 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
2653 * gst-rtsp-server.doap:
2654 * tests/Makefile.am:
2655 configure: Synchronise with the configure scripts of the other modules
2657 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
2660 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
2662 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
2664 * gst/rtsp-server/rtsp-media.c:
2665 * gst/rtsp-server/rtsp-stream.c:
2666 Revert "rtsp-server: support build against last stable release"
2667 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
2668 Let us require 1.2.3 now, which is going to be released in a few
2671 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
2673 * gst/rtsp-server/rtsp-session-media.c:
2674 * gst/rtsp-server/rtsp-stream-transport.c:
2675 session: improve RTP-Info
2676 Ignore streams that can't generate RTP-Info instead of failing.
2677 Don't return the empty string when all streams are unconfigured but
2678 return NULL so that we don't generate and empty RTP-Info header.
2679 Improve docs a little.
2681 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
2683 * gst/rtsp-server/rtsp-session-media.c:
2684 Don't free rtpinfo GString when it is NULL
2685 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
2687 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
2689 * gst/rtsp-server/rtsp-media.c:
2690 media: only set keyframe flag when modifying start
2691 Only set the keyframe flag when we modify the start position. The
2692 keyframe flag should probably be ignored when no change is requested but
2693 until we can claim this is all documented properly and all demuxer
2694 implement this, avoid setting the flag.
2695 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
2697 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
2699 * gst/rtsp-server/rtsp-thread-pool.c:
2700 thread-pool: Unref source after mainloop has quit to avoid races in GLib
2701 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
2703 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
2705 * gst/rtsp-server/rtsp-stream.c:
2706 stream: handle NULL seqnum and rtptime arguments
2708 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
2710 * gst/rtsp-server/rtsp-thread-pool.c:
2711 * tests/check/gst/threadpool.c:
2712 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
2713 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
2715 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
2717 * gst/rtsp-server/rtsp-stream.c:
2718 stream: add fallback for missing stats property
2719 Use a fallback when the payloader does not have a stats property
2720 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
2722 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
2725 Automatic update of common submodule
2726 From f7bc1c3 to 1a07da9
2728 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
2730 * gst/rtsp-server/rtsp-stream.c:
2731 stream: don't leak stats structure
2732 Don't leak the stats structure and deal with NULL stats.
2734 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
2736 * gst/rtsp-server/rtsp-stream.c:
2737 stream: Get rtpinfo properties atomically from payloader
2738 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
2740 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
2742 * gst/rtsp-server/rtsp-media.c:
2743 media: refactor state change functions and signals
2744 Make functions to set the target state and the pipeline state and emit
2745 the signals from those functions.
2747 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
2749 * gst/rtsp-server/rtsp-media.c:
2750 * gst/rtsp-server/rtsp-media.h:
2751 media: add signal to notify of pending state changes
2753 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
2755 * gst/rtsp-server/rtsp-media.c:
2756 * gst/rtsp-server/rtsp-stream.c:
2757 rtsp-server: support build against last stable release
2758 Until 1.2.3 is out with the new get_type function and we
2761 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
2763 * gst/rtsp-server/rtsp-stream.c:
2764 stream: fix compilation
2766 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
2768 * gst/rtsp-server/rtsp-media.c:
2769 * gst/rtsp-server/rtsp-media.h:
2770 * gst/rtsp-server/rtsp-stream.c:
2771 * gst/rtsp-server/rtsp-stream.h:
2772 stream: add property to configure profiles
2774 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
2776 * gst/rtsp-server/rtsp-client.c:
2777 client: let stream check supported transport
2778 Delegate the check if a transport is allowed to the stream.
2779 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
2781 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
2783 * gst/rtsp-server/rtsp-stream.c:
2784 * gst/rtsp-server/rtsp-stream.h:
2785 stream: add method to check supported transport
2786 Add a method to check if a transport is supported
2788 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
2791 configure.ac: Only check for gstreamer-check, not check
2792 We include check in gstreamer-check since quite some time now.
2794 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
2796 * gst/rtsp-server/rtsp-session-media.c:
2797 * gst/rtsp-server/rtsp-stream-transport.c:
2798 * gst/rtsp-server/rtsp-stream.c:
2799 * gst/rtsp-server/rtsp-stream.h:
2800 stream: return clock-rate from get_rtpinfo
2801 And use it to correct the rtptime to the requested start-time.
2802 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
2804 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
2806 * gst/rtsp-server/rtsp-session-media.c:
2807 * gst/rtsp-server/rtsp-stream-transport.c:
2808 * gst/rtsp-server/rtsp-stream-transport.h:
2809 session-media: calculate start-time
2811 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
2813 * gst/rtsp-server/rtsp-stream-transport.c:
2814 * gst/rtsp-server/rtsp-stream.c:
2815 * gst/rtsp-server/rtsp-stream.h:
2816 stream: also return the running-time
2817 Return the running-time in the rtpinfo as well.
2819 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
2821 * gst/rtsp-server/rtsp-client.c:
2822 * gst/rtsp-server/rtsp-session-media.c:
2823 * gst/rtsp-server/rtsp-session-media.h:
2824 * gst/rtsp-server/rtsp-stream-transport.c:
2825 * gst/rtsp-server/rtsp-stream-transport.h:
2826 session-media: let the session-media make the RTPInfo
2827 Add method to create the RTPInfo for a stream-transport.
2828 Add method to create the RTPInfo for all stream-transports in a
2830 Use the session-media RTPInfo code in client. This allows us to refactor
2831 another method to link the TCP callbacks.
2833 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
2835 mount-points: sort sequence before g_sequence_lookup
2836 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
2837 sort sequence if dirty, otherwise lookup will fail.
2838 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
2840 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
2843 configure: rename package from gst-rtsp to gst-rtsp-server
2844 To match git module name and avoid confusion with the
2845 rtsp lib in gst-plugins-base and rtsp plugin in -good.
2847 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
2850 configure: bump core/base/good requirement to 1.2.0
2851 Bump to released stable version and make implicit
2852 requirements explicit.
2854 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
2859 Fix broken gettext setup which is not used anyway
2861 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
2864 Automatic update of common submodule
2865 From dbedaa0 to d48bed3
2867 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
2869 * gst/rtsp-server/rtsp-client.c:
2870 * gst/rtsp-server/rtsp-media.c:
2871 * gst/rtsp-server/rtsp-media.h:
2872 media: add setup_sdp vmethod
2873 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
2874 gst_rtsp_media_setup_sdp.
2875 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
2877 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
2879 * gst/rtsp-server/rtsp-stream.c:
2880 rtsp-stream: Check return value of sscanf
2881 streamid is only valid if sscanf matched something.
2883 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
2885 * gst/rtsp-server/rtsp-client.c:
2886 rtsp-client: Fix iteration
2887 Wouldn't even enter the code block otherwise (i++ was used as the check
2888 and not the postfix).
2890 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
2892 * gst/rtsp-server/rtsp-client.c:
2893 * gst/rtsp-server/rtsp-client.h:
2894 client: add vmethod to configure media and streams
2895 Implement a vmethod that can be used to configure the media and the
2896 streams based on the current context. Handle the blocksize handling in
2897 the default handler.
2898 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
2900 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
2903 Make git ignore more unit test binaries
2905 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
2907 * gst/rtsp-server/rtsp-address-pool.h:
2908 * gst/rtsp-server/rtsp-auth.h:
2909 * gst/rtsp-server/rtsp-client.h:
2910 * gst/rtsp-server/rtsp-context.h:
2911 * gst/rtsp-server/rtsp-media-factory-uri.h:
2912 * gst/rtsp-server/rtsp-media-factory.h:
2913 * gst/rtsp-server/rtsp-media.h:
2914 * gst/rtsp-server/rtsp-mount-points.h:
2915 * gst/rtsp-server/rtsp-server.h:
2916 * gst/rtsp-server/rtsp-session-media.h:
2917 * gst/rtsp-server/rtsp-session-pool.h:
2918 * gst/rtsp-server/rtsp-session.h:
2919 * gst/rtsp-server/rtsp-stream-transport.h:
2920 * gst/rtsp-server/rtsp-stream.h:
2921 * gst/rtsp-server/rtsp-thread-pool.h:
2922 * gst/rtsp-server/rtsp-token.h:
2923 rtsp-server: add padding to many public structures
2924 Not mini objects though, since they are not subclassable
2925 anyway, nor kept on the stack or inlined in a structure.
2927 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
2929 media: add new create_rtpbin vmethod
2930 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
2931 https://bugzilla.gnome.org/show_bug.cgi?id=719734
2933 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
2935 * tests/check/gst/media.c:
2936 tests: fix memory leak, free test's thread pool
2937 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
2939 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
2941 * gst/rtsp-server/rtsp-stream-transport.c:
2942 stream-transport: free url in finalize
2944 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
2946 * gst/rtsp-server/rtsp-media.c:
2947 media: also do state change in suspended state
2949 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
2951 * gst/rtsp-server/rtsp-client.c:
2952 * gst/rtsp-server/rtsp-media.c:
2953 media: also handle prepare and range in suspended state
2954 When we are suspended, we are already prepared.
2955 We can get the range in the suspended state.
2957 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
2959 * tests/check/Makefile.am:
2960 * tests/check/gst/sessionmedia.c:
2961 check: add test for uri in setup
2962 Added unit tests for the new functionality in GstRTSPStreamTransport.
2963 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
2965 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
2967 * gst/rtsp-server/rtsp-client.c:
2968 client: store setup uri and use in PLAY response
2969 Store the uri used when doing the setup and use that in the PLAY
2971 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
2973 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
2975 * gst/rtsp-server/rtsp-stream-transport.c:
2976 * gst/rtsp-server/rtsp-stream-transport.h:
2977 stream-transport: add method to get/set url
2979 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
2981 * gst/rtsp-server/rtsp-client.c:
2982 client: suspend after SDP and unsuspend before PLAYING
2983 Based on patches by Ognyan Tonchev <ognyan@axis.com>
2984 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
2986 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
2988 * gst/rtsp-server/rtsp-media-factory.c:
2989 * gst/rtsp-server/rtsp-media-factory.h:
2990 * gst/rtsp-server/rtsp-media.c:
2991 * gst/rtsp-server/rtsp-media.h:
2992 * gst/rtsp-server/rtsp-session-media.c:
2993 * gst/rtsp-server/rtsp-session.c:
2994 * tests/check/gst/media.c:
2995 * tests/check/gst/mediafactory.c:
2996 media: add suspend modes
2997 Add support for different suspend modes. The stream is suspended right after
2998 producing the SDP and after PAUSE. Different suspend modes are available that
2999 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
3000 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
3001 state and RESET will bring the pipeline to the NULL state.
3002 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
3003 this means that the pipeline needs to be prerolled again.
3004 Base on patches by Ognyan Tonchev <ognyan@axis.com>
3005 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3007 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
3009 * gst/rtsp-server/rtsp-media.c:
3010 media: start live streams in blocked state
3011 Start live streams in the blocked state and make them preroll using the
3012 messages. This ensure that no data is played by the sink until we explicitly
3013 unblock the stream right before going to PLAYING.
3014 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3016 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
3018 * gst/rtsp-server/rtsp-media.c:
3019 media: refactor starting and waiting for preroll
3020 Based on patches from Ognyan Tonchev <ognyan@axis.com>
3021 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3023 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
3025 * gst/rtsp-server/rtsp-stream.c:
3026 * gst/rtsp-server/rtsp-stream.h:
3027 stream: add API to block streams
3028 Add an API to block on the streams and make it post a message.
3029 Based on patch by Ognyan Tonchev <ognyan@axis.com>
3030 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3032 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
3034 * docs/libs/Makefile.am:
3035 docs: Specify the override file
3036 Even if it's empty (for now) it avoids make distcheck complaining
3038 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
3040 * gst/rtsp-server/rtsp-media.c:
3041 media: move default implementations to where they are used
3043 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
3045 * gst/rtsp-server/rtsp-media.c:
3046 media: take the right lock in gst_rtsp_media_set_pipeline_state()
3047 We need to take the state_lock when calling this method.
3049 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
3051 * gst/rtsp-server/rtsp-media.c:
3052 media: handle add-added on non-bins too
3053 Handle dynamic payloaders that are not bins, as used in the unit-test.
3055 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3057 * gst/rtsp-server/rtsp-media-factory.c:
3058 * gst/rtsp-server/rtsp-media-factory.h:
3059 * gst/rtsp-server/rtsp-media.c:
3060 rtsp-media/-factory: Fix request pad name comments
3061 These must be escaped for gtk-doc to parse the comments without warnings.
3063 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
3065 rtsp-media: remove transports if media is in error status
3066 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
3067 trying to change to GST_STATE_NULL and media is in error status, we
3068 remove all transports.
3069 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
3071 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
3073 * gst/rtsp-server/rtsp-media.c:
3074 rtsp-media: use element metadata to find payloader
3075 Use the element metadata to find the payloader instead of checking
3077 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
3079 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
3081 rtsp-stream: add getter for payload type
3082 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
3083 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
3084 element and create the stream with this one instead of the dynpay%d
3086 https://bugzilla.gnome.org/show_bug.cgi?id=712396
3088 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3090 * gst/rtsp-server/rtsp-client.c:
3091 * gst/rtsp-server/rtsp-context.h:
3092 * gst/rtsp-server/rtsp-media.c:
3093 * gst/rtsp-server/rtsp-mount-points.c:
3094 * gst/rtsp-server/rtsp-server.c:
3095 * gst/rtsp-server/rtsp-token.c:
3096 rtsp-*: Refer to NULL as a constant in comments
3098 https://bugzilla.gnome.org/show_bug.cgi?id=714988
3100 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3102 rtsp-*: Fix type name typos in comments
3103 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
3104 * rtsp-auth: Refer to part of constant name as text
3105 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
3106 * rtsp-session-media: Fix GstRTSPSessionMedia typo
3107 * rtsp-stream: Fix typo when refering to GstBin
3108 https://bugzilla.gnome.org/show_bug.cgi?id=714988
3110 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3113 * docs/libs/gst-rtsp-server-docs.sgml:
3114 * docs/libs/gst-rtsp-server-sections.txt:
3115 docs: Improve documentation
3116 * Include annotation-glossary to quiet gtk-doc
3117 * Rename remaining ClientState -> Context
3118 * Rename object hierarchy file
3119 * Remove stale chapter references
3120 * Add missing function and object references
3121 * Include missing GstRTSPAddressPoolResult
3122 https://bugzilla.gnome.org/show_bug.cgi?id=714988
3124 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
3126 * gst/rtsp-server/rtsp-client.c:
3127 * gst/rtsp-server/rtsp-server.c:
3128 * gst/rtsp-server/rtsp-session-pool.c:
3129 * gst/rtsp-server/rtsp-session.c:
3130 * gst/rtsp-server/rtsp-stream.c:
3131 rtsp-server: sprinkle some allow-none annotations for g-i
3133 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
3135 * gst/rtsp-server/rtsp-stream.c:
3136 * gst/rtsp-server/rtsp-stream.h:
3137 stream: add method to filter transports
3138 Add a method to safely iterate and collect the stream transports
3139 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
3141 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
3143 * gst/rtsp-server/rtsp-client.c:
3144 * gst/rtsp-server/rtsp-server.c:
3145 * gst/rtsp-server/rtsp-session-pool.c:
3146 * gst/rtsp-server/rtsp-session.c:
3147 rtsp: allow NULL func in filters
3148 Passing a null function make the filters return a list of
3151 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
3153 * gst/rtsp-server/rtsp-address-pool.c:
3154 * tests/check/gst/addresspool.c:
3155 address-pool: fix address increment
3156 Use a guint instead of guint8 to increment the address. It's still not
3157 completely correct because a guint might not be able to hold the complete
3158 address range, but that's an enhacement for later.
3159 Add unit test to test improved behaviour.
3160 https://bugzilla.gnome.org/show_bug.cgi?id=708237
3162 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
3164 * gst/rtsp-server/rtsp-client.c:
3165 * tests/check/gst/client.c:
3166 client: allow absolute path in requests
3167 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
3169 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
3171 * gst/rtsp-server/rtsp-client.c:
3172 * gst/rtsp-server/rtsp-client.h:
3173 client: make make_path_from_uri a vmethod
3175 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
3177 * docs/libs/gst-rtsp-server-sections.txt:
3178 * gst/rtsp-server/rtsp-stream.c:
3179 * gst/rtsp-server/rtsp-stream.h:
3180 * tests/check/Makefile.am:
3181 * tests/check/gst/stream.c:
3182 stream: Add functions to get rtp and rtcp sockets
3183 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
3185 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
3187 * gst/rtsp-server/rtsp-context.c:
3188 * gst/rtsp-server/rtsp-context.h:
3189 context: defing a GType for the context
3190 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
3192 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
3194 * gst/rtsp-server/Makefile.am:
3195 * gst/rtsp-server/rtsp-auth.c:
3196 * gst/rtsp-server/rtsp-context.c:
3197 * gst/rtsp-server/rtsp-media.c:
3198 * gst/rtsp-server/rtsp-mount-points.c:
3199 * gst/rtsp-server/rtsp-server.h:
3200 * gst/rtsp-server/rtsp-session-media.c:
3201 * gst/rtsp-server/rtsp-session.c:
3202 * gst/rtsp-server/rtsp-stream.c:
3203 Fixed several GIR warnings
3205 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
3207 * gst/rtsp-server/rtsp-auth.c:
3210 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3212 * tests/check/Makefile.am:
3213 * tests/check/gst/token.c:
3214 tests: Add unit tests for token
3215 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
3217 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3219 * gst/rtsp-server/rtsp-token.c:
3220 token: Validate args for gst_rtsp_token_is_allowed
3221 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
3223 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3225 * gst/rtsp-server/rtsp-token.c:
3226 token: Fix bug when creating empty token
3227 We always want to have a valid GstStructure in the token.
3228 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
3230 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
3232 * gst/rtsp-server/rtsp-thread-pool.c:
3233 thread-pool: avoid race in shutdown
3234 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
3235 don't actually stop the mainloop ever. Solve this race by adding an idle source
3236 to the mainloop that calls the _quit. This way we immediately exit the mainloop
3237 if quit was called before we started it.
3239 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3241 * tests/check/Makefile.am:
3242 * tests/check/gst/permissions.c:
3243 tests: Add unit tests for permissions
3244 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
3246 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3248 * tests/check/gst/mediafactory.c:
3249 tests: Test mediafactory permissions
3250 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
3252 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3254 * gst/rtsp-server/rtsp-permissions.c:
3255 permissions: Fix refcounting when adding/removing roles
3256 Previously a role that was removed was unreffed twice, and when
3257 replacing an existing role the replaced role was freed while still being
3258 referenced. Both bugs are now fixed.
3259 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
3261 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3263 * tests/check/gst/media.c:
3264 * tests/check/gst/mediafactory.c:
3265 * tests/check/gst/rtspserver.c:
3266 tests: Check gst_rtsp_url_parse return value
3267 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
3269 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
3272 Automatic update of common submodule
3273 From 865aa20 to dbedaa0
3275 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
3277 * gst/rtsp-server/rtsp-server.c:
3278 rtsp-server: Fix socket leak
3279 https://bugzilla.gnome.org/show_bug.cgi?id=710088
3281 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
3283 * gst/rtsp-server/rtsp-session-pool.c:
3284 rtsp-session-pool: Make sure session IDs are properly URI-escaped
3285 https://bugzilla.gnome.org/show_bug.cgi?id=643812
3287 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
3289 * examples/.gitignore:
3290 * examples/test-video.c:
3291 examples: fix compilation when WITH_AUTH is defined
3292 https://bugzilla.gnome.org/show_bug.cgi?id=710228
3294 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
3297 gitignore: Add new test binary
3299 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
3301 * tests/check/Makefile.am:
3302 * tests/check/gst/threadpool.c:
3303 thread-pool: Add unit test for the thread pools
3304 https://bugzilla.gnome.org/show_bug.cgi?id=710228
3306 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
3308 * gst/rtsp-server/rtsp-thread-pool.c:
3309 thread-pool: Fix thread leak when reusing threads
3310 https://bugzilla.gnome.org/show_bug.cgi?id=709730
3312 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
3314 * gst/rtsp-server/rtsp-server.c:
3315 * tests/check/gst/rtspserver.c:
3316 tests: fixed racy behavior in rtspserver tests
3317 https://bugzilla.gnome.org/show_bug.cgi?id=710078
3319 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3321 * tests/check/gst/addresspool.c:
3322 tests: Improve address pool unit tests
3323 Add a range with mixed IPV4 and IPV6 addresses to pool.
3324 Get an IPV4 address from an IPV6-only pool.
3325 Get an IPV6 address from an IPV4-only pool.
3326 Reserve a IPV6 address from an IPV4-only pool.
3327 Check for unicast addresses in multicast-only pool.
3328 Check for unicast addresses in uni-/multicast-mixed pool.
3329 https://bugzilla.gnome.org/show_bug.cgi?id=710128
3331 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3333 * gst/rtsp-server/rtsp-client.c:
3334 client: append query string in PAUSE/PLAY/TEARDOWN as well
3336 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
3338 * gst/rtsp-server/rtsp-client.c:
3339 client: Add query to control path
3340 If the SETUP url contains a query it must be appended to the control
3341 path so that it matches any already created stream in the media. The
3342 query will also be appended to the session media path.
3344 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3346 * gst/rtsp-server/rtsp-media.c:
3347 rtsp-media: remove old line
3349 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
3351 * gst/rtsp-server/rtsp-stream.c:
3352 stream: Correct control comparison
3353 https://bugzilla.gnome.org/show_bug.cgi?id=709176
3355 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3357 * gst/rtsp-server/rtsp-media.c:
3358 media: Check dynamically if the pipeline supports seeking
3359 We should not depend on whether or not the pipeline state change
3360 returned NO_PREROLL or not. A media could dynamically change its
3361 element and switch from seekable to non seekable so it's best to test
3362 the seekable nature of the pipeline dynamically when we try to do a seek.
3364 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3366 * gst/rtsp-server/rtsp-media.c:
3367 media: Return FALSE if seeking is not supported
3369 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3371 * gst/rtsp-server/rtsp-media.c:
3372 rtsp-media: don't seek accurate by default
3373 Accurate seeking is perhaps a little overkill in the most common situation and
3374 causes some formats (mp3) over slow media to seek extremely slowly.
3376 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
3378 * tests/check/gst/rtspserver.c:
3379 tests: fix unit test
3380 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
3382 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
3384 * gst/rtsp-server/rtsp-client.c:
3385 client: Reply 400 if media cannot be constructed
3386 Reply 400 Bad Request instead of 503 Service Unavailable if media
3387 cannot be constructed in SETUP.
3388 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
3390 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
3392 * gst/rtsp-server/rtsp-client.c:
3393 client: Send setup reply once only
3394 If find_media() failed in handle_setup_request() two replies was sent.
3395 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
3397 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
3400 Automatic update of common submodule
3401 From 6b03ba7 to 865aa20
3403 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
3405 * gst/rtsp-server/rtsp-server.c:
3406 server: Emit client-connected signal earlier
3407 Emit client-connected before the client ref is given to a GSource,
3408 otherwise client-connected can be emitted after the client object has
3411 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
3413 * gst/rtsp-server/rtsp-address-pool.c:
3414 * gst/rtsp-server/rtsp-address-pool.h:
3415 * gst/rtsp-server/rtsp-stream.c:
3416 * tests/check/gst/addresspool.c:
3417 addresspool: return reason of failure
3418 Let gst_rtsp_address_pool_reserve_address() return the reason why
3419 the address could not be reserved.
3420 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
3422 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
3425 autogen.sh: Sync behaviour with other GStreamer modules
3426 Allows building from outside of tree amongst other things
3428 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
3431 Automatic update of common submodule
3432 From b613661 to 6b03ba7
3434 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
3437 Automatic update of common submodule
3438 From 74a6857 to b613661
3440 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
3443 Automatic update of common submodule
3444 From 01a7a46 to 74a6857
3446 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
3448 * gst/rtsp-server/rtsp-client.c:
3449 client: Do not read beyond end of path string
3450 If the setup was done without a control url, make sure we don't try to read the
3451 non-existing control string and crash.
3453 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3455 * gst/rtsp-server/rtsp-client.c:
3456 client: Fix RTPInfo header
3457 Refactor the method to make the content_base.
3458 Use the content-base and the control url to construct the RTPInfo
3461 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3463 * gst/rtsp-server/rtsp-client.c:
3464 client: map url to path only in describe
3465 Only map the request url to a path in the DESCRIBE method. The SDP then
3466 contains the base and control urls that should be used to SETUP/PAUSE/
3467 PLAY/TEARDOWN the media.
3469 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3471 * gst/rtsp-server/rtsp-client.c:
3472 Revert "client: map URL to path in requests"
3473 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
3474 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
3475 contains the base and control urls which are used in the SETUP, PLAY,
3476 PAUSE and TEARDOWN requests.
3478 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3480 * gst/rtsp-server/rtsp-client.c:
3481 client: map URL to path in requests
3483 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3485 * gst/rtsp-server/rtsp-client.c:
3486 * gst/rtsp-server/rtsp-mount-points.c:
3487 * gst/rtsp-server/rtsp-mount-points.h:
3488 mount-points: make vmethod to make path from uri
3489 Make a vmethod to transform an url into a path. The path is then used to lookup
3490 the factory. This makes it possible to also use other bits of the url, such as
3491 the query parameters, to locate the factory.
3493 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
3495 * gst/rtsp-server/rtsp-thread-pool.c:
3496 * gst/rtsp-server/rtsp-thread-pool.h:
3497 thread-pool: Add cleanup to wait for the threadpool to finish
3498 Also fix race condition if two threads are asking for the first
3499 thread from the thread pool at once. This would case two internal
3500 GThreadPools to be created.
3501 https://bugzilla.gnome.org/show_bug.cgi?id=707753
3503 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
3505 * gst/rtsp-server/rtsp-client.c:
3506 * tests/check/gst/client.c:
3507 client: free threadpool
3508 https://bugzilla.gnome.org/show_bug.cgi?id=707638
3510 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
3512 * tests/check/gst/mountpoints.c:
3513 mountpoints tests: unref matched factories
3514 https://bugzilla.gnome.org/show_bug.cgi?id=707638
3516 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
3518 * tests/check/gst/media.c:
3519 media tests: unref thread pool and caps
3520 https://bugzilla.gnome.org/show_bug.cgi?id=707638
3522 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
3524 * gst/rtsp-server/rtsp-auth.c:
3525 * gst/rtsp-server/rtsp-media-factory.c:
3526 * gst/rtsp-server/rtsp-media.c:
3527 auth, media, media-factory: unref permissions
3528 https://bugzilla.gnome.org/show_bug.cgi?id=707638
3530 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3532 * examples/Makefile.am:
3533 Makefile: add rule for appsrc example
3535 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3537 * examples/test-appsrc.c:
3538 tests: add appsrc example
3539 Add an example on how to use appsrc to feed the server pipeline with data.
3541 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
3543 * gst/rtsp-server/rtsp-client.c:
3544 rtsp-client: remove query part from content-base string
3545 Make sure that after the control url has been resolved, it's
3546 not a part of the query-string.
3547 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
3549 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3551 * gst/rtsp-server/rtsp-client.c:
3552 client: don't check url in response
3553 There is no url or method in the response to check
3555 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3557 * gst/rtsp-server/rtsp-client.c:
3558 * gst/rtsp-server/rtsp-client.h:
3559 Add handle-response signal for when we receive a GET_PARAMETER response
3561 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3563 * gst/rtsp-server/rtsp-server.c:
3564 Fix gst_rtsp_server_client_filter, using wrong variable type
3566 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
3568 * gst/rtsp-server/rtsp-media-factory-uri.c:
3569 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
3570 For AAC we need to check for framed=true instead of parsed=true.
3571 https://bugzilla.gnome.org/show_bug.cgi?id=701384
3573 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3575 * gst/rtsp-server/rtsp-stream.c:
3576 stream: optimize pipeline for protocols
3577 When TCP is not an allowed protocol for the stream, avoid creating the
3578 appsrc/appsink/queue and tee elements.
3580 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3582 * gst/rtsp-server/rtsp-media.c:
3583 media: set protocols on streams
3585 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3587 * gst/rtsp-server/rtsp-client.c:
3588 client: use protocols supported by stream
3590 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3592 * gst/rtsp-server/rtsp-media-factory.c:
3593 * gst/rtsp-server/rtsp-media.c:
3594 * gst/rtsp-server/rtsp-stream.c:
3595 media-factory: allow all protocols
3597 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3599 * gst/rtsp-server/rtsp-media.c:
3600 media: configure protocols in new streams
3602 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3604 * gst/rtsp-server/rtsp-stream.c:
3605 * gst/rtsp-server/rtsp-stream.h:
3606 stream: add protocols property
3608 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3610 * gst/rtsp-server/rtsp-media.c:
3611 rtsp-media: send state in "new-state" signal
3612 https://bugzilla.gnome.org/show_bug.cgi?id=705110
3614 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
3617 build: add subdir-objects to AM_INIT_AUTOMAKE
3618 Fixes warnings with automake 1.14
3619 https://bugzilla.gnome.org/show_bug.cgi?id=705350
3621 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3623 * docs/libs/gst-rtsp-server-sections.txt:
3624 * gst/rtsp-server/rtsp-client.c:
3625 * gst/rtsp-server/rtsp-server.c:
3626 * gst/rtsp-server/rtsp-server.h:
3627 server: add method to iterate clients of server
3629 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3631 * gst/rtsp-server/rtsp-media.c:
3632 * gst/rtsp-server/rtsp-media.h:
3633 Add vmethod for rtsp-media subclass to access rtpbin
3635 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3637 * gst/rtsp-server/rtsp-client.h:
3638 small documentation fix
3640 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3642 * gst/rtsp-server/rtsp-client.c:
3643 Do not take range header if range is invalid
3645 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3647 * docs/libs/gst-rtsp-server-sections.txt:
3648 * gst/rtsp-server/rtsp-media.c:
3649 media: add docs for new method
3651 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3653 * gst/rtsp-server/rtsp-media.c:
3654 * gst/rtsp-server/rtsp-media.h:
3655 Add API to rtsp-media set the pipeline's state
3657 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3659 * gst/rtsp-server/rtsp-media.c:
3660 Update current position/duration when gst_rtsp_media_get_range_string is called
3662 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3664 * examples/test-cgroups.c:
3665 tests: add some more docs
3667 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3669 * examples/test-cgroups.c:
3670 * gst/rtsp-server/Makefile.am:
3671 * gst/rtsp-server/rtsp-auth.c:
3672 * gst/rtsp-server/rtsp-auth.h:
3673 * gst/rtsp-server/rtsp-client.c:
3674 * gst/rtsp-server/rtsp-client.h:
3675 * gst/rtsp-server/rtsp-context.c:
3676 * gst/rtsp-server/rtsp-context.h:
3677 * gst/rtsp-server/rtsp-params.c:
3678 * gst/rtsp-server/rtsp-params.h:
3679 * gst/rtsp-server/rtsp-server.c:
3680 * gst/rtsp-server/rtsp-thread-pool.c:
3681 * gst/rtsp-server/rtsp-thread-pool.h:
3682 * tests/check/gst/client.c:
3683 ClientState -> Context
3684 Rename the clientstate to context and put the code in a separate file.
3686 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3688 * examples/test-auth.c:
3689 * gst/rtsp-server/rtsp-auth.c:
3690 * gst/rtsp-server/rtsp-auth.h:
3691 auth: add support for default token
3692 The default token is used when the user is not authenticated and can be used to
3693 give minimal permissions.
3695 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3697 * examples/test-auth.c:
3698 * gst/rtsp-server/rtsp-auth.c:
3699 auth: use defines when possible
3701 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3703 * gst/rtsp-server/rtsp-address-pool.c:
3704 address-pool: improve docs
3706 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3708 * gst/rtsp-server/rtsp-permissions.c:
3709 permissions: add the role to the copy
3711 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
3713 * gst/rtsp-server/rtsp-permissions.c:
3714 permissions: Also copy the roles
3716 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
3718 * gst/rtsp-server/rtsp-permissions.c:
3719 permissions: Make it build
3721 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3723 * gst/rtsp-server/rtsp-address-pool.h:
3726 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3728 * docs/libs/gst-rtsp-server-sections.txt:
3729 * gst/rtsp-server/rtsp-auth.c:
3730 * gst/rtsp-server/rtsp-auth.h:
3731 * gst/rtsp-server/rtsp-media.c:
3732 * gst/rtsp-server/rtsp-session-media.c:
3733 * gst/rtsp-server/rtsp-stream-transport.c:
3734 * gst/rtsp-server/rtsp-stream-transport.h:
3735 * gst/rtsp-server/rtsp-stream.c:
3736 * tests/check/gst/client.c:
3739 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3741 * docs/libs/gst-rtsp-server-sections.txt:
3742 * gst/rtsp-server/rtsp-address-pool.c:
3743 * gst/rtsp-server/rtsp-address-pool.h:
3744 * tests/check/gst/addresspool.c:
3745 * tests/check/gst/rtspserver.c:
3746 address-pool: cleanups
3747 Remove redundant method, improve docs.
3749 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3751 * docs/libs/gst-rtsp-server-sections.txt:
3752 * gst/rtsp-server/rtsp-auth.h:
3753 * gst/rtsp-server/rtsp-permissions.c:
3754 * gst/rtsp-server/rtsp-permissions.h:
3755 * gst/rtsp-server/rtsp-token.c:
3758 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3760 * gst/rtsp-server/rtsp-permissions.c:
3761 permissions: implement _remove_role
3763 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3765 * gst/rtsp-server/rtsp-permissions.c:
3766 permissions: update docs
3768 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3770 * tests/check/gst/client.c:
3771 tests: simplify tests
3772 Client settings are now disabled by default so we don't need an auth
3773 module to disable them.
3775 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3777 * gst/rtsp-server/rtsp-auth.c:
3778 auth: add default authorizations
3779 When no auth module is specified, use our table of defaults to look up the
3780 default value of the check instead of always allowing everything. This was
3781 we can disallow client settings by default.
3783 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3786 README: update readme
3788 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3790 * gst/rtsp-server/rtsp-thread-pool.c:
3791 * gst/rtsp-server/rtsp-thread-pool.h:
3792 thread-pool: add more docs
3794 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3796 * gst/rtsp-server/rtsp-thread-pool.c:
3797 * gst/rtsp-server/rtsp-thread-pool.h:
3798 thread-pool: fix race in thread reuse
3799 If we try to reuse a thread right after we made it stop, we end up using a
3800 stopped thread. Catch this case and only reuse threads that are not stopping.
3802 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3804 * gst/rtsp-server/rtsp-server.c:
3805 server: add small debug
3807 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3809 * tests/check/gst/client.c:
3811 Add some permissions to media so we can use the auth and enable
3814 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3816 * gst/rtsp-server/rtsp-client.c:
3817 client: support pushed context in handle_request
3818 If we already have a pushed state, reuse it and add our own things. This makes
3819 it easier to write tests.
3821 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3823 * gst/rtsp-server/rtsp-auth.c:
3824 auth: don't auth on methods
3825 Don't authorize on methods anymore but on the resources that we
3826 try to access, this is more flexible.
3827 Move the authorization checks to where they are needed and let the
3828 check return the response on error.
3830 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3832 * gst/rtsp-server/rtsp-mount-points.c:
3833 mount-points: add some debug
3835 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3837 * tests/check/gst/client.c:
3838 tests: almost fix test
3840 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3842 * gst/rtsp-server/rtsp-auth.c:
3843 * gst/rtsp-server/rtsp-auth.h:
3844 * gst/rtsp-server/rtsp-client.c:
3845 * gst/rtsp-server/rtsp-client.h:
3846 * gst/rtsp-server/rtsp-server.c:
3847 * gst/rtsp-server/rtsp-server.h:
3848 auth: let the auth module check client_settings
3849 Let the auth module decide if client settings are allowed for the
3852 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3854 * gst/rtsp-server/rtsp-token.c:
3855 * gst/rtsp-server/rtsp-token.h:
3856 token: add method to check boolean permission
3858 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3860 * examples/test-auth.c:
3861 * examples/test-cgroups.c:
3862 * gst/rtsp-server/rtsp-token.c:
3863 * gst/rtsp-server/rtsp-token.h:
3864 token: simplify token constructor
3865 Use variable arguments to make easier API.
3867 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3869 * examples/test-auth.c:
3870 * examples/test-cgroups.c:
3871 * gst/rtsp-server/rtsp-media-factory.c:
3872 * gst/rtsp-server/rtsp-media-factory.h:
3873 media-factory: add convenience API for factory
3875 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3877 * examples/test-auth.c:
3878 * examples/test-cgroups.c:
3879 * gst/rtsp-server/rtsp-permissions.c:
3880 * gst/rtsp-server/rtsp-permissions.h:
3881 permissions: simplify API a little
3882 Avoid passing GstStructure in the add_role method, use varargs instead
3883 to construct the structure behind the scenes. We can then also use the
3884 structure name as the role and simplify some more logic.
3886 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3888 * gst/rtsp-server/rtsp-auth.c:
3891 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3893 * gst/rtsp-server/rtsp-auth.c:
3894 * gst/rtsp-server/rtsp-auth.h:
3895 * gst/rtsp-server/rtsp-client.c:
3896 auth: handle unauthorized response
3897 Move handling of the unauthorized response to the auth module, it can add
3898 the appropriate headers to request authorization for the required method
3899 much better than the client.
3901 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3903 * gst/rtsp-server/rtsp-client.c:
3904 * gst/rtsp-server/rtsp-client.h:
3905 client: allow for sending any message, not only requests
3906 Change the _send_request() method to _send_message() so that we
3907 can both send requests and replies.
3909 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3911 * docs/libs/gst-rtsp-server-sections.txt:
3912 * gst/rtsp-server/rtsp-server.h:
3915 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3917 * examples/test-video.c:
3918 * gst/rtsp-server/rtsp-auth.c:
3919 * gst/rtsp-server/rtsp-auth.h:
3920 * gst/rtsp-server/rtsp-server.c:
3921 * gst/rtsp-server/rtsp-server.h:
3922 auth: move TLS handling to auth module
3923 Remove the TLS settings on the server and move it to the auth module because
3924 that is where security related bits go.
3926 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3928 * gst/rtsp-server/rtsp-client.c:
3929 * gst/rtsp-server/rtsp-client.h:
3930 client: add state push/pop
3932 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3934 * gst/rtsp-server/rtsp-client.c:
3935 * gst/rtsp-server/rtsp-client.h:
3936 client: add connection to state
3938 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3940 * gst/rtsp-server/rtsp-mount-points.c:
3941 mount-points: fix debug
3943 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3945 * tests/check/gst/media.c:
3946 tests: fix media test
3948 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3950 * gst/rtsp-server/rtsp-thread-pool.c:
3951 thread-pool: we don't require a state
3953 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3955 * gst/rtsp-server/rtsp-server.c:
3956 server: let context ref the server
3957 So that we don't risk losing the server object early anc crash.
3959 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3961 * tests/check/gst/client.c:
3962 tests: fix client test
3964 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3967 * docs/libs/gst-rtsp-server-docs.sgml:
3968 * docs/libs/gst-rtsp-server-sections.txt:
3969 * gst/rtsp-server/rtsp-address-pool.c:
3970 * gst/rtsp-server/rtsp-auth.c:
3971 * gst/rtsp-server/rtsp-client.c:
3972 * gst/rtsp-server/rtsp-client.h:
3973 * gst/rtsp-server/rtsp-media-factory-uri.c:
3974 * gst/rtsp-server/rtsp-media-factory.c:
3975 * gst/rtsp-server/rtsp-media-factory.h:
3976 * gst/rtsp-server/rtsp-media.c:
3977 * gst/rtsp-server/rtsp-mount-points.c:
3978 * gst/rtsp-server/rtsp-params.c:
3979 * gst/rtsp-server/rtsp-permissions.c:
3980 * gst/rtsp-server/rtsp-sdp.c:
3981 * gst/rtsp-server/rtsp-server.c:
3982 * gst/rtsp-server/rtsp-server.h:
3983 * gst/rtsp-server/rtsp-session-media.c:
3984 * gst/rtsp-server/rtsp-session-pool.c:
3985 * gst/rtsp-server/rtsp-session.c:
3986 * gst/rtsp-server/rtsp-stream-transport.c:
3987 * gst/rtsp-server/rtsp-stream.c:
3988 * gst/rtsp-server/rtsp-thread-pool.c:
3989 * gst/rtsp-server/rtsp-token.c:
3992 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3994 * gst/rtsp-server/rtsp-session-pool.c:
3995 * gst/rtsp-server/rtsp-session-pool.h:
3996 session-pool: make vmethod to create a session
3997 Make a vmethod to create a sessions so that subclasses can create
3998 custom session objects
4000 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4002 * gst/rtsp-server/rtsp-auth.c:
4003 * gst/rtsp-server/rtsp-media-factory.h:
4004 * gst/rtsp-server/rtsp-media.h:
4005 * gst/rtsp-server/rtsp-mount-points.h:
4006 * gst/rtsp-server/rtsp-session-pool.h:
4007 * gst/rtsp-server/rtsp-stream.h:
4010 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4012 * docs/libs/gst-rtsp-server-docs.sgml:
4013 * docs/libs/gst-rtsp-server-sections.txt:
4014 * gst/rtsp-server/rtsp-address-pool.c:
4015 * gst/rtsp-server/rtsp-address-pool.h:
4016 * gst/rtsp-server/rtsp-auth.c:
4017 * gst/rtsp-server/rtsp-client.h:
4018 * gst/rtsp-server/rtsp-media-factory.h:
4019 * gst/rtsp-server/rtsp-media.c:
4020 * gst/rtsp-server/rtsp-media.h:
4021 * gst/rtsp-server/rtsp-permissions.c:
4022 * gst/rtsp-server/rtsp-permissions.h:
4023 * gst/rtsp-server/rtsp-server.h:
4024 * gst/rtsp-server/rtsp-session-media.c:
4025 * gst/rtsp-server/rtsp-session-media.h:
4026 * gst/rtsp-server/rtsp-session-pool.h:
4027 * gst/rtsp-server/rtsp-session.h:
4028 * gst/rtsp-server/rtsp-stream-transport.h:
4029 * gst/rtsp-server/rtsp-stream.c:
4030 * gst/rtsp-server/rtsp-thread-pool.h:
4033 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4036 * examples/Makefile.am:
4037 configure: compile cgroup example conditionally
4038 Only compile the cgroup example when we have libcgroup
4040 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4043 * examples/Makefile.am:
4044 * examples/test-cgroups.c:
4045 examples: add cgroups example
4047 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4049 * tests/check/gst/rtspserver.c:
4050 tests: fix compilation
4052 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4054 * gst/rtsp-server/rtsp-thread-pool.c:
4055 thread-pool: fix vmethod invocation
4057 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4059 * gst/rtsp-server/rtsp-thread-pool.c:
4060 * gst/rtsp-server/rtsp-thread-pool.h:
4061 thread-pool: store thread type in thread
4063 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4065 * gst/rtsp-server/rtsp-client.c:
4066 client: pass thread from pool to media _prepare
4067 Get a thread from the configured threadpool and pass it to the prepare method of
4070 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4072 * gst/rtsp-server/rtsp-media.c:
4073 * gst/rtsp-server/rtsp-media.h:
4074 media: Accept a thread in _prepare
4075 Remove out own threadpool handling and use the provided thread and
4076 maincontext for the bus messages and the state changes.
4078 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4080 * gst/rtsp-server/rtsp-server.c:
4081 server: configure client thread pool
4083 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4085 * gst/rtsp-server/rtsp-client.c:
4086 * gst/rtsp-server/rtsp-client.h:
4087 client: add method to configure thread pool
4089 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4091 * gst/rtsp-server/rtsp-client.h:
4092 * gst/rtsp-server/rtsp-server.c:
4093 * gst/rtsp-server/rtsp-server.h:
4094 server: use thread pool
4095 Use the thread pool instead of doing our own thing.
4097 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4099 * gst/rtsp-server/Makefile.am:
4100 * gst/rtsp-server/rtsp-thread-pool.c:
4101 * gst/rtsp-server/rtsp-thread-pool.h:
4102 thread-pool: add object to manage threads
4103 Add an object to manage the client and media threads.
4105 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4107 * gst/rtsp-server/rtsp-auth.c:
4108 auth: debug authorization check
4110 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4112 * gst/rtsp-server/rtsp-media.c:
4113 media: start media pipeline in context
4114 Start the media pipeline in the provided context (or our default one
4115 when NULL). This makes sure that we run the bus thread in this context and that
4116 all media threads are children of this context.
4118 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4120 * gst/rtsp-server/rtsp-media-factory.c:
4121 factory: pass permissions to media by default
4123 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4125 * examples/test-auth.c:
4126 test: add permissions to auth test
4127 Ass some permissions to the media factory in the test.
4129 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4131 * gst/rtsp-server/rtsp-auth.c:
4132 * gst/rtsp-server/rtsp-auth.h:
4133 * gst/rtsp-server/rtsp-client.c:
4134 auth: simplify auth checks
4135 Remove client from methods, it's now in the state
4136 Perform the check specified by the string, use the information from the
4137 thread local context.
4139 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4141 * gst/rtsp-server/rtsp-client.c:
4142 * gst/rtsp-server/rtsp-client.h:
4143 client: add state to current thread
4144 Add the client to the ClientState object.
4145 Place the ClientState on the current thread.
4147 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4149 * gst/rtsp-server/rtsp-media-factory.c:
4150 * gst/rtsp-server/rtsp-media-factory.h:
4151 * gst/rtsp-server/rtsp-media.c:
4152 * gst/rtsp-server/rtsp-media.h:
4153 media: make it possible to set permissions
4154 Make it possible to set permissions on media and media factory objects
4156 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4158 * gst/rtsp-server/Makefile.am:
4159 * gst/rtsp-server/rtsp-permissions.c:
4160 * gst/rtsp-server/rtsp-permissions.h:
4161 permissions: add permissions object
4162 Add a mini object to store permissions based on a role.
4164 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4166 * examples/test-auth.c:
4167 * gst/rtsp-server/rtsp-auth.c:
4168 * gst/rtsp-server/rtsp-auth.h:
4169 * gst/rtsp-server/rtsp-client.c:
4170 auth: add auth checks
4171 Add an enum with auth checks and implement the checks in the auth object.
4172 Perform the checks from the client.
4174 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4176 * examples/test-auth.c:
4177 * gst/rtsp-server/rtsp-auth.c:
4178 * gst/rtsp-server/rtsp-auth.h:
4179 * gst/rtsp-server/rtsp-client.h:
4180 auth: use the token after authentication
4181 After we authenticated a user, keep the Token around in the state.
4183 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4185 * gst/rtsp-server/rtsp-client.c:
4186 * gst/rtsp-server/rtsp-media.c:
4187 * gst/rtsp-server/rtsp-media.h:
4188 * tests/check/gst/media.c:
4189 media: add optional context for bus messages
4190 Add an optional mainloop to _prepare that will handle the bus messages instead
4191 of always using the shared mainloop.
4193 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4195 * gst/rtsp-server/Makefile.am:
4196 * gst/rtsp-server/rtsp-token.c:
4197 * gst/rtsp-server/rtsp-token.h:
4198 token: add authorization token
4199 Add a simply miniobject that contains the authorizations. The object contains a
4200 GstStructure that hold all authorization fields. When a user is authenticated,
4201 the auth module will create a Token for the user. The token is then used to
4202 check what operations the user is allowed to do and various other configuration
4205 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4207 * examples/test-auth.c:
4208 * gst/rtsp-server/rtsp-auth.c:
4209 * gst/rtsp-server/rtsp-auth.h:
4210 * gst/rtsp-server/rtsp-client.c:
4211 * gst/rtsp-server/rtsp-client.h:
4212 * gst/rtsp-server/rtsp-media-factory.c:
4213 * gst/rtsp-server/rtsp-media-factory.h:
4214 * gst/rtsp-server/rtsp-media.c:
4215 * gst/rtsp-server/rtsp-media.h:
4216 auth: remove auth from media and factory
4217 Remove the auth object from media and factory. We want to have the RTSPClient
4218 authenticate and authorize resources, there is no need to place another auth
4219 manager on the media/factory.
4221 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4223 * examples/test-auth.c:
4224 * gst/rtsp-server/rtsp-auth.c:
4225 * gst/rtsp-server/rtsp-auth.h:
4226 * gst/rtsp-server/rtsp-client.h:
4227 auth: add support for multiple basic auth tokens
4228 Make it possible to add multiple basic authorisation tokens to one authorization
4229 object. Associate with each token an authorization group that will define what
4230 capabilities are allowed.
4232 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4234 * gst/rtsp-server/rtsp-client.c:
4235 client: error out on non-aggregate control
4236 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
4238 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4240 * gst/rtsp-server/rtsp-client.c:
4241 client: rework setup request a little
4242 Cache the media in DESCRIBE based on the longest matching path with the uri
4243 that we can find in the mount points.
4244 Rework the setup request a little to get the media from the session or from
4245 the longest matching path, this way we can derive the control string as
4246 everything after the path instead of hardcoding it.
4247 Find the stream based on the control string and only open a session when all
4250 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4252 * gst/rtsp-server/rtsp-media.c:
4253 * gst/rtsp-server/rtsp-media.h:
4254 media: add method to find a stream by control url
4256 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4258 * gst/rtsp-server/rtsp-stream.c:
4259 * gst/rtsp-server/rtsp-stream.h:
4260 stream: add method to check control url of stream
4262 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4264 * gst/rtsp-server/rtsp-client.c:
4265 * gst/rtsp-server/rtsp-session-media.c:
4266 * gst/rtsp-server/rtsp-session-media.h:
4267 * gst/rtsp-server/rtsp-session.c:
4268 * gst/rtsp-server/rtsp-session.h:
4269 session: use path matching for session media
4270 Use a path string instead of a uri to lookup session media in the sessions. Also
4271 use path matching to find the largest possible path that matches.
4273 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4275 * gst/rtsp-server/rtsp-client.c:
4276 * gst/rtsp-server/rtsp-mount-points.c:
4277 * gst/rtsp-server/rtsp-mount-points.h:
4278 * tests/check/gst/mountpoints.c:
4279 mount-points: remove useless vmethod
4280 Making lookups in the mount points should not be done with a URL, if there is a
4281 mapping to be done from URL to mount points, we'll need to do it somewhere
4284 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4286 * gst/rtsp-server/rtsp-mount-points.c:
4287 * gst/rtsp-server/rtsp-mount-points.h:
4288 * tests/check/gst/mountpoints.c:
4289 mount-points: improve mount point searching
4290 Use a GSequence to keep track of the mount points.
4291 Match a URL to the longest matching registered mount point. This should be the
4292 URL to perform aggreagate control and the remainder is the stream specific
4294 Add some unit tests for this.
4296 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
4298 * gst/rtsp-server/Makefile.am:
4299 rtsp-server: Allow building of static library
4301 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4303 * tests/check/gst/mediafactory.c:
4304 tests: fix compilation
4306 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4308 * gst/rtsp-server/rtsp-sdp.c:
4309 sdp: get control string from stream
4310 Use the control string as configured in the stream.
4312 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4314 * gst/rtsp-server/rtsp-stream.c:
4315 * gst/rtsp-server/rtsp-stream.h:
4316 stream: add methods and property to set control string
4318 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4320 * gst/rtsp-server/rtsp-client.c:
4322 Rename variables for clarity
4323 Keep media in state when we can
4325 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4327 * gst/rtsp-server/rtsp-client.c:
4328 * gst/rtsp-server/rtsp-stream.c:
4329 * gst/rtsp-server/rtsp-stream.h:
4330 stream: add more support for IPv6
4331 Rename _get_address to _get_multicast_address in GstRTSPStream to
4332 make it clear that this function only deals with multicast.
4333 Make it possible to have both an IPv4 and IPv6 multicast address on
4334 a stream. Give the client an IPv4 or IPv6 address depending on the
4335 address it used to connect to the server.
4336 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
4338 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4340 * gst/rtsp-server/rtsp-client.c:
4343 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4345 * gst/rtsp-server/rtsp-stream.c:
4346 stream: handle failed port allocation
4347 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
4348 can't allocate any family at all. Also keep track of what port families we
4350 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
4352 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4354 * gst/rtsp-server/rtsp-stream.c:
4355 stream: improve docs
4357 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4359 * gst/rtsp-server/rtsp-stream-transport.c:
4360 stream-transport: remove old if 0 block
4362 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
4364 * tests/check/gst/client.c:
4366 gst_rtsp_client_get_uri() has been removed
4367 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
4369 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4371 * gst/rtsp-server/rtsp-client.c:
4372 * gst/rtsp-server/rtsp-client.h:
4373 client: add method to filter managed sessions
4374 Add a method to filter the sessions managed by this client connection.
4375 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
4377 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4379 * gst/rtsp-server/rtsp-client.c:
4380 * gst/rtsp-server/rtsp-client.h:
4381 client: remove _get_uri() method
4382 Remove the get_uri() method on the client. A client has no uri, the uri
4383 property is an internal property to manage the last cached media for
4386 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4388 * gst/rtsp-server/rtsp-media-factory.h:
4389 media-factory: fix typo
4391 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
4393 * gst/rtsp-server/rtsp-media.c:
4394 rtsp-media: Do not leak the query in default_query_stop
4395 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
4397 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4399 * gst/rtsp-server/rtsp-media.c:
4400 media: don't unlock when conversion fails
4401 Don't unlock the state lock when conversion fails because it was not locked.
4403 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4405 * gst/rtsp-server/rtsp-media.c:
4406 * gst/rtsp-server/rtsp-media.h:
4407 Add query_position and query_stop vmethods to rtsp-media
4409 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4411 * gst/rtsp-server/rtsp-media.c:
4412 Fix typo in property install for rtsp-media's time-provider
4414 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4416 * gst/rtsp-server/rtsp-client.c:
4417 * gst/rtsp-server/rtsp-client.h:
4418 client: clean some variables
4419 Clean some variables and add some guards to _send_request()
4421 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4423 * gst/rtsp-server/rtsp-client.c:
4424 * gst/rtsp-server/rtsp-client.h:
4425 Add gst_rtsp_client_send_request API
4426 This makes it possible to send arbitrary messages to a client, such as
4427 SET_PARAMETER or GET_PARAMETER
4429 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4431 * gst/rtsp-server/rtsp-media.c:
4432 * gst/rtsp-server/rtsp-media.h:
4433 media: add _get_element() method
4434 Add method to get the element used when creating the media.
4435 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
4437 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4439 * gst/rtsp-server/rtsp-media.c:
4442 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4444 * gst/rtsp-server/rtsp-stream.c:
4445 * gst/rtsp-server/rtsp-stream.h:
4446 stream: allow access to the rtp session
4447 https://bugzilla.gnome.org/show_bug.cgi?id=703004
4449 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
4451 * gst/rtsp-server/rtsp-stream.c:
4452 * gst/rtsp-server/rtsp-stream.h:
4453 dscp qos support in gst-rtsp-stream
4454 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
4456 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4458 * tests/check/gst/rtspserver.c:
4460 Actually do what the comment says. Also keep the old code around, not sure what
4461 should happen when you get a 454 from a TEARDOWN, does it close the connection?
4462 it currently doesn't.
4464 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4466 * gst/rtsp-server/rtsp-client.c:
4467 client: also watch newly created session
4468 When we newly created a session, start watching it immediately instead of
4469 on the next request.
4471 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
4473 * tests/check/gst/client.c:
4474 tests: add unit test for new-session
4475 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
4477 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4479 * gst/rtsp-server/rtsp-client.c:
4480 client: emit new-session when new session is created
4481 Only emit new-session when we created a new session for a client, not when a
4482 client picked up a previous session.
4483 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
4485 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
4487 * gst/rtsp-server/rtsp-client.c:
4488 client: handle asterisk as path in requests
4489 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
4491 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4493 * gst/rtsp-server/rtsp-media.c:
4494 media: handle segment query format mismatch
4495 It's possible that the segment query returns with a different format than what
4496 we asked for, handle this case also.
4498 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
4500 * gst/rtsp-server/rtsp-media.c:
4501 media: use segment stop in collect_media_stats
4502 Use segment stop instead of duration as range end point.
4503 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
4505 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
4507 * gst/rtsp-server/rtsp-media.c:
4508 * tests/check/gst/media.c:
4509 rtsp-media: Do not leak the element in take_pipeline
4510 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
4512 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
4514 * gst/rtsp-server/rtsp-client.c:
4515 * gst/rtsp-server/rtsp-client.h:
4516 rtsp-client: Make configure_client_transport virtual
4517 This patch makes configure_client_transport virtual. The functionality is
4518 needed to handle some weird clients sending multicast transport settings as url
4520 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
4522 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
4524 * gst/rtsp-server/rtsp-client.c:
4525 * gst/rtsp-server/rtsp-client.h:
4526 rtsp-client: Make param_set and param_get virtual
4527 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
4529 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
4531 * gst/rtsp-server/rtsp-client.c:
4532 * gst/rtsp-server/rtsp-media.c:
4533 * gst/rtsp-server/rtsp-media.h:
4534 media: convert_range replaces get_range_times
4535 get_range_times worked for handling UTC ranges for seeks, but we also
4536 need to convert back from NPT to the requested unit in
4537 get_range_string. convert_range is now used for both.
4538 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
4540 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4542 * gst/rtsp-server/rtsp-client.c:
4543 * gst/rtsp-server/rtsp-sdp.c:
4544 * gst/rtsp-server/rtsp-sdp.h:
4545 sdp: cleanup sdp info
4546 We don't need to pass the proto, we can more easily check a boolean.
4547 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
4549 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
4551 * gst/rtsp-server/rtsp-sdp.c:
4552 use 0.0.0.0 or :: for c= line instead of server address
4554 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
4556 * gst/rtsp-server/rtsp-client.c:
4557 use local address, not remote, in SDP
4558 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
4560 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4563 Automatic update of common submodule
4564 From 098c0d7 to 01a7a46
4566 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
4568 * gst/rtsp-server/rtsp-media.c:
4569 * gst/rtsp-server/rtsp-media.h:
4570 media: possibility to override range time conversion
4571 Make it possible to override the conversion from GstRTSPTimeRange to
4572 GstClockTimes, that is done before seeking on the media
4573 pipeline. Overriding can be useful for UTC ranges, where the default
4574 conversion gives nanoseconds since 1900.
4575 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
4577 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
4579 * gst/rtsp-server/rtsp-server.c:
4580 * gst/rtsp-server/rtsp-server.h:
4581 rtsp-server: Expose the use_client_settings API
4582 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
4584 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
4586 * gst/rtsp-server/rtsp-client.c:
4587 * gst/rtsp-server/rtsp-stream.c:
4588 * gst/rtsp-server/rtsp-stream.h:
4589 rtspstream: handle both ipv4 and ipv6 clients
4590 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
4592 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4594 * gst/rtsp-server/rtsp-sdp.c:
4595 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
4596 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
4597 We already have a way to place extra attributes in the SDP by using a string
4598 property with prefix x- or a- in the caps.
4600 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4602 * gst/rtsp-server/rtsp-sdp.c:
4603 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
4604 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
4605 We already have a way to place extra attributes in the SDP, just make a string
4606 property in the payloader with a- or x- prefix.
4608 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4610 * gst/rtsp-server/rtsp-sdp.c:
4611 rtsp: place a- and x- properties as attributes
4612 application/x-rtp has properties with a- and x- prefixes that should be
4613 placed as attributes in the SDP for the media instead of being added to the
4616 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4618 * examples/Makefile.am:
4619 * examples/test-video.c:
4620 example: add TLS example
4622 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4624 * gst/rtsp-server/rtsp-server.c:
4625 * gst/rtsp-server/rtsp-server.h:
4626 server: add support for TLS
4627 Add methods to set and get a TLS certificate.
4628 Add vmethod to configure a new connection. By default, configure the TLS
4629 certificate in a new connection if needed.
4631 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4633 * gst/rtsp-server/rtsp-server.c:
4634 * gst/rtsp-server/rtsp-server.h:
4635 server: remove accept_client vmethod
4636 This vmethod is not very useful so remove it.
4638 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4640 * gst/rtsp-server/rtsp-server.c:
4641 server: don't crash on NULL GError
4643 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
4645 * gst/rtsp-server/rtsp-session-pool.c:
4646 rtsp-session-pool: corrected session timeout detection
4647 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
4649 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4651 * gst/rtsp-server/rtsp-client.c:
4652 client: improve debug
4654 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4656 * gst/rtsp-server/rtsp-client.c:
4657 * gst/rtsp-server/rtsp-client.h:
4658 * gst/rtsp-server/rtsp-server.c:
4659 server: refactor connection setup
4660 Let the server accept the socket connection and construct a GstRTSPConnection
4661 from it. Remove the code from the client and let the client only deal with
4662 a fully configure GstRTSPConnection object.
4663 We will need this later when the server will configure the connection for
4666 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4668 * gst/rtsp-server/rtsp-stream.c:
4669 stream: keep the transport object alive
4670 Keep the transport object alive while we have it as qdata on the
4673 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
4675 * gst/rtsp-server/rtsp-client.c:
4676 * gst/rtsp-server/rtsp-server.c:
4677 rtsp-server: Do not crash on nmapping of server
4678 * generate error when gst_rtsp_connection_accept fails
4679 * do not stop accepting incoming connections because
4680 accepting a client fails
4681 https://bugzilla.gnome.org/show_bug.cgi?id=701072
4683 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
4685 * gst/rtsp-server/rtsp-client.c:
4686 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
4687 https://bugzilla.gnome.org/show_bug.cgi?id=700953
4689 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
4691 * gst/rtsp-server/rtsp-sdp.c:
4692 rtsp-sdp: Parse framerate caps field and set SDP attribute
4693 The SDP attribute and its format is described in RFC4566.
4694 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
4696 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
4698 * gst/rtsp-server/rtsp-sdp.c:
4699 rtsp-sdp: Parse width/height from caps and set SDP attribute
4700 The SDP attribute and its format is described in RFC6064.
4701 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
4703 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
4705 * gst/rtsp-server/rtsp-sdp.c:
4706 * tests/check/gst/client.c:
4707 rtsp-sdp: add bandwidth line
4708 https://bugzilla.gnome.org/show_bug.cgi?id=699220
4710 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4713 Automatic update of common submodule
4714 From 5edcd85 to 098c0d7
4716 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
4718 * tests/check/gst/media.c:
4719 tests: add dynamic payloader prepare/unprepare check
4721 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4723 * gst/rtsp-server/rtsp-media.c:
4724 media: release lock when removing fakesink
4726 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4728 * gst/rtsp-server/rtsp-stream.c:
4729 stream: set elements to NULL before removing
4730 When removing a stream, set the elements to NULL first. This avoids
4731 element-is-not-in-NULL-state errors when we dispose the elements.
4733 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
4736 Automatic update of common submodule
4737 From 3cb3d3c to 5edcd85
4739 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4741 * gst/rtsp-server/rtsp-media.c:
4742 * gst/rtsp-server/rtsp-media.h:
4743 media: listen to pad-removed signals
4744 Listen to the pad-removed signal and remove the stream associated with the
4746 Add signal to be notified of the removed pad.
4747 Remove the fakesink in unprepare()
4748 Fix signatures of the signal methods
4750 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4752 * examples/test-sdp.c:
4753 tests: add example of reusable pipelines
4755 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
4757 * gst/rtsp-server/rtsp-stream.c:
4758 * gst/rtsp-server/rtsp-stream.h:
4759 stream: add method to get the srcpad
4761 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
4763 * tests/check/gst/media.c:
4764 check: add media prepare/unprepare test
4765 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4767 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
4769 * gst/rtsp-server/rtsp-media.c:
4770 media: disconnect from signal handlers in unprepare()
4771 We connected to the pad-added and no-more-pads signals in prepare() so
4772 we need to disconnect from them in unprepare().
4773 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4775 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
4777 * gst/rtsp-server/rtsp-media.c:
4778 media: don't free streams array
4779 Don't free the streams array in the unprepare() method, they were not
4781 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4783 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
4785 * gst/rtsp-server/rtsp-media.c:
4786 media: don't unref the pipeline in unprepare
4787 Unprepare() should undo what prepare() does. Because the pipeline is
4788 not created in prepare(), we should not unref it in unprepare()
4790 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
4792 * gst/rtsp-server/rtsp-stream.c:
4793 stream: clear session and caps for reuse
4794 Set the session and caps to NULL after unref otherwise we might unref
4796 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4798 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
4800 * gst/rtsp-server/rtsp-client.c:
4801 client: send out teardown signal before tearing down
4802 The advantage is that in the signal handler you get direct access to
4803 information about what streams are about to get torn down (in the
4804 GstRTSPClientState).
4805 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
4807 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
4809 * gst/rtsp-server/rtsp-client.c:
4810 * gst/rtsp-server/rtsp-client.h:
4811 client: expose connection
4812 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
4814 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
4817 Automatic update of common submodule
4818 From aed87ae to 3cb3d3c
4820 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4822 * gst/rtsp-server/rtsp-media.c:
4823 * gst/rtsp-server/rtsp-media.h:
4824 * gst/rtsp-server/rtsp-session-media.c:
4825 * gst/rtsp-server/rtsp-session-media.h:
4826 media: add method to get the base_time of the pipeline
4827 Together with a shared clock, this base-time could eventually be sent to
4828 the client so that it can reconstruct the exact running-time of the clock
4831 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4833 * gst/rtsp-server/Makefile.am:
4834 * gst/rtsp-server/rtsp-media.c:
4835 * gst/rtsp-server/rtsp-media.h:
4836 * gst/rtsp-server/rtsp-sdp.c:
4837 media: add GstNetTimeProvider support
4838 Add a property to let the media provide a GstNetTimeProvider for its clock.
4839 Make methods to get the clock and nettimeprovider
4840 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
4841 provider and also the current time of the clock. This should make it possible
4842 for (GStreamer) clients to slave their clock to the server clock.
4844 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
4847 Automatic update of common submodule
4848 From 04c7a1e to aed87ae
4850 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4852 * gst/rtsp-server/rtsp-media.c:
4853 media: wait for buffering to complete
4854 Wait for buffering to complete before changing the state to the target state.
4856 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4858 * gst/rtsp-server/rtsp-media.c:
4859 media: small cleanup
4861 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
4863 * tests/check/gst/rtspserver.c:
4864 tests: remove extra unref in test_setup_non_existing_stream
4865 The unref is not needed anymore, teardown runs without it.
4866 https://bugzilla.gnome.org/show_bug.cgi?id=696542
4868 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
4870 * tests/check/gst/rtspserver.c:
4871 tests: GSocketService cleanup in test_bind_already_in_use
4872 Use g_socket_service_stop so the rtspserver test stops listening for
4873 incoming connections in test_bind_already_in_use.
4874 https://bugzilla.gnome.org/show_bug.cgi?id=696541
4876 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
4878 * gst/rtsp-server/rtsp-media-factory.c:
4879 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
4880 Instead use a GWeakRef which is safe to use
4881 This is a known GLib bug, see:
4882 https://bugzilla.gnome.org/show_bug.cgi?id=667145
4884 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
4886 * gst/rtsp-server/rtsp-client.c:
4887 * gst/rtsp-server/rtsp-media.c:
4888 * gst/rtsp-server/rtsp-media.h:
4889 * gst/rtsp-server/rtsp-sdp.c:
4890 * tests/check/gst/media.c:
4891 * tests/check/gst/rtspserver.c:
4892 rtsp-media/client: Reply to PLAY request with same type of Range
4893 Remember the type of Range from the PLAY request and use the same type for
4896 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
4898 * gst/rtsp-server/rtsp-client.c:
4899 * gst/rtsp-server/rtsp-client.h:
4900 * tests/check/gst/client.c:
4901 rtsp-client: expose uri
4903 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
4905 * tests/check/gst/mediafactory.c:
4906 tests: Hold ref while creating second media
4907 To test if the media aren't shared, make sure we keep the first one while creating a second
4908 otherwise the same memory address may be reused.
4910 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
4913 configure: remove out-of-date comment
4915 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
4918 .gitignore: ignore more build files
4920 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
4922 * tests/check/Makefile.am:
4923 tests: use right _LIBS variable for gst-plugins-base libs
4925 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4927 * tests/check/Makefile.am:
4928 check: add librtp to libs
4930 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
4932 * tests/check/gst/rtspserver.c:
4933 tests: Add test to check selecting a port the server will send from
4935 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
4937 * tests/check/gst/rtspserver.c:
4938 tests: Make sure packets are actually received
4940 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4942 * gst/rtsp-server/rtsp-stream.c:
4943 stream: Select unicast address from pool if appropriate
4945 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
4947 * gst/rtsp-server/rtsp-stream.c:
4948 stream: Properties are always there in Gst 1.0
4950 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4952 * tests/check/gst/addresspool.c:
4953 tests: Add tests for unicast addresses in pool
4955 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
4957 * gst/rtsp-server/rtsp-address-pool.c:
4958 * tests/check/gst/addresspool.c:
4959 address-pool: Verify that multicast addresses are used for multicast and vice-versa
4961 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
4963 * docs/libs/gst-rtsp-server-sections.txt:
4964 * gst/rtsp-server/rtsp-address-pool.c:
4965 * gst/rtsp-server/rtsp-address-pool.h:
4966 * gst/rtsp-server/rtsp-stream.c:
4967 * tests/check/gst/addresspool.c:
4968 address-pool: Add unicast addresses
4970 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
4973 * gst/rtsp-server/rtsp-server.c:
4974 * tests/check/gst/rtspserver.c:
4975 rtsp-server: Limit the number of threads per server instance
4976 If we exceed the maximum, just round robin the clients over the existing
4979 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
4981 * gst/rtsp-server/rtsp-server.c:
4982 rtsp-server: No need to store the GMainContext in the client context
4984 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
4986 * tests/check/gst/rtspserver.c:
4987 tests: Add test for client disconnection
4989 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
4991 * tests/check/gst/rtspserver.c:
4992 tests: Test client and session timeouts with multiple threads
4994 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
4996 * gst/rtsp-server/rtsp-address-pool.c:
4997 * gst/rtsp-server/rtsp-auth.c:
4998 * gst/rtsp-server/rtsp-client.c:
4999 * gst/rtsp-server/rtsp-media-factory-uri.c:
5000 * gst/rtsp-server/rtsp-media-factory.c:
5001 * gst/rtsp-server/rtsp-media.c:
5002 * gst/rtsp-server/rtsp-mount-points.c:
5003 * gst/rtsp-server/rtsp-server.c:
5004 * gst/rtsp-server/rtsp-session-media.c:
5005 * gst/rtsp-server/rtsp-session-pool.c:
5006 * gst/rtsp-server/rtsp-session.c:
5007 Document locking and its order
5009 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
5011 * tests/check/gst/rtspserver.c:
5012 tests: Test that slow DESCRIBE don't block other clients
5014 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
5016 * tests/check/gst/client.c:
5017 tests: Add tests for client-requested multicast address
5019 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
5021 * docs/libs/gst-rtsp-server-sections.txt:
5022 docs: Put the various functions in the right sections
5024 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
5026 * docs/libs/gst-rtsp-server-docs.sgml:
5027 * docs/libs/gst-rtsp-server-sections.txt:
5028 * gst/rtsp-server/rtsp-address-pool.c:
5029 * gst/rtsp-server/rtsp-address-pool.h:
5030 docs: Generate docs for GstRTSPAddressPool
5032 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
5034 * gst/rtsp-server/rtsp-client.c:
5035 * gst/rtsp-server/rtsp-stream.c:
5036 * gst/rtsp-server/rtsp-stream.h:
5037 client: Check client provided addresses against the address pool
5039 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
5041 * gst/rtsp-server/rtsp-address-pool.c:
5042 * gst/rtsp-server/rtsp-address-pool.h:
5043 * tests/check/gst/addresspool.c:
5044 address-pool: Add API to request a specific address from the pool
5045 Also add relevant unit tests.
5047 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
5049 * tests/check/gst/mediafactory.c:
5050 tests: Check the passing around of a RTSPAddressPool
5051 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
5052 way down to the stream.
5054 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
5056 * tests/check/gst/addresspool.c:
5057 tests: Add more tests for the address pool
5059 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
5061 * gst/rtsp-server/rtsp-address-pool.c:
5062 address-pool: Fix off by one error
5063 When splitting a port range, the port after a skip is not part of range.
5065 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
5068 Automatic update of common submodule
5069 From 2de221c to 04c7a1e
5071 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
5074 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
5075 AM_CONFIG_HEADER was removed in automake 1.13
5076 https://bugzilla.gnome.org/show_bug.cgi?id=693368
5078 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
5081 Automatic update of common submodule
5082 From a942293 to 2de221c
5084 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5086 * gst/rtsp-server/rtsp-client.c:
5087 client: make sure the watch exists while sending data
5088 Protect the send_func with a lock. This allows us to wait for sending
5089 to complete before changing the send_func and user_data. We add an
5090 extra ref to the watch to make sure that it remains valid during
5092 When closing the connection, set the send_func to NULL
5093 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
5095 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5097 * tests/check/Makefile.am:
5098 tests: use GST_*_1_0 environment variables everywhere
5099 The _1_0 suffixed environment variables override the
5100 non-suffixed ones, so if we're in an environment that
5101 sets the _1_0 suffixed ones, such as jhbuild, we need
5102 to set those to make sure ours actually always get
5105 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5108 Automatic update of common submodule
5109 From acb04d9 to a942293
5111 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5113 * gst/rtsp-server/rtsp-client.c:
5114 rtsp-client: set the client backlog
5115 Set the client backlog to a reasonable default
5117 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
5119 * gst/rtsp-server/rtsp-media.c:
5120 rtsp-media: Make the element a constructor parameter
5121 https://bugzilla.gnome.org/show_bug.cgi?id=689594
5123 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5125 * docs/libs/Makefile.am:
5126 docs: Link with gcov library when gcov is enabled
5127 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
5129 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5131 * gst/rtsp-server/rtsp-media.c:
5132 media: match prepare with unprepare
5133 Really unprepare when there were an equal amount of prepare calls.
5135 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5137 * gst/rtsp-server/rtsp-media.c:
5138 media: media has to be unprepared in finalize
5139 Because unprepare takes away the last ref on the media.
5141 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5143 * gst/rtsp-server/rtsp-client.c:
5144 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
5145 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
5146 We can't use the refcount to trigger unprepare because it is the unprepare call
5147 that removes the last refcount after all messages are consumed. What we should
5148 probably do is make a prepared refcount and only unprepare when the refcount
5151 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5153 * gst/rtsp-server/rtsp-media.c:
5154 media: let the source unref the last media ref
5155 the last ref to the media is held by the source so we don't need to add more ref
5156 and unrefs, we simply destroy the media when the source is gone.
5158 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5160 * gst/rtsp-server/rtsp-media.c:
5161 media: improve debug
5163 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5165 * gst/rtsp-server/rtsp-media.c:
5167 Make sure we are in the right state when collecting the position and duration.
5168 Only make ourselves PREPARED when we were previously PREPARING.
5170 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5172 * gst/rtsp-server/rtsp-media.c:
5173 media: use g_object_ref/unref for GObjects
5175 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
5177 * gst/rtsp-server/rtsp-client.c:
5178 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
5179 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
5180 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
5181 isn't being used anymore.
5183 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
5185 * gst/rtsp-server/rtsp-media.c:
5186 Fix compiler warning
5188 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
5190 * gst/rtsp-server/rtsp-media-factory-uri.c:
5191 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
5193 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5195 * gst/rtsp-server/rtsp-session-media.h:
5198 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5200 * gst/rtsp-server/rtsp-media.c:
5201 * tests/check/gst/media.c:
5202 media: avoid element leak
5204 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5206 * gst/rtsp-server/rtsp-media.c:
5207 media: require an element in media constructor
5209 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5211 * gst/rtsp-server/rtsp-client.c:
5212 Revert "client: TEARDOWN brings that state to Init again"
5213 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
5214 The object is already disposed, there is no point in setting the state.
5216 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5218 * gst/rtsp-server/rtsp-client.c:
5219 client: TEARDOWN brings that state to Init again
5221 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5223 * docs/libs/gst-rtsp-server-sections.txt:
5224 * examples/test-auth.c:
5225 * gst/rtsp-server/rtsp-auth.c:
5226 * gst/rtsp-server/rtsp-auth.h:
5227 * gst/rtsp-server/rtsp-client.c:
5228 * gst/rtsp-server/rtsp-client.h:
5229 * gst/rtsp-server/rtsp-media-factory-uri.c:
5230 * gst/rtsp-server/rtsp-media-factory-uri.h:
5231 * gst/rtsp-server/rtsp-media-factory.c:
5232 * gst/rtsp-server/rtsp-media-factory.h:
5233 * gst/rtsp-server/rtsp-media.c:
5234 * gst/rtsp-server/rtsp-media.h:
5235 * gst/rtsp-server/rtsp-mount-points.c:
5236 * gst/rtsp-server/rtsp-mount-points.h:
5237 * gst/rtsp-server/rtsp-sdp.c:
5238 * gst/rtsp-server/rtsp-server.c:
5239 * gst/rtsp-server/rtsp-server.h:
5240 * gst/rtsp-server/rtsp-session-media.c:
5241 * gst/rtsp-server/rtsp-session-media.h:
5242 * gst/rtsp-server/rtsp-session-pool.c:
5243 * gst/rtsp-server/rtsp-session-pool.h:
5244 * gst/rtsp-server/rtsp-session.c:
5245 * gst/rtsp-server/rtsp-session.h:
5246 * gst/rtsp-server/rtsp-stream-transport.c:
5247 * gst/rtsp-server/rtsp-stream-transport.h:
5248 * gst/rtsp-server/rtsp-stream.c:
5249 * gst/rtsp-server/rtsp-stream.h:
5250 * tests/check/gst/media.c:
5251 rtsp: make object details private
5252 Make all object details private
5253 Add methods to access private bits
5255 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5257 * tests/check/Makefile.am:
5258 * tests/check/gst/media.c:
5259 tests: add media tests
5261 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5263 * gst/rtsp-server/rtsp-media.c:
5264 media: check if prepared for some methods
5265 Check that the media object is prepared before doing seek and getting the
5266 current position etc.
5267 Add some g_return checks.
5269 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5271 * tests/check/Makefile.am:
5272 * tests/check/gst/mediafactory.c:
5273 tests: add mediafactory test
5275 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5277 * gst/rtsp-server/rtsp-stream.c:
5278 stream: improve debug
5280 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5282 * gst/rtsp-server/rtsp-media.c:
5283 * gst/rtsp-server/rtsp-media.h:
5284 media: unref pipeline in finalize to avoid leaking it
5286 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5288 * gst/rtsp-server/rtsp-media-factory-uri.c:
5289 * gst/rtsp-server/rtsp-media.c:
5290 rtsp: use gst_object_unref on GstObjects
5292 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5294 * gst/rtsp-server/rtsp-media-factory.c:
5295 media-factory: require an url
5297 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5299 * examples/test-uri.c:
5300 examples: fix include
5302 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5304 * gst/rtsp-server/rtsp-server.h:
5305 server: remove unused include
5307 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5309 * tests/check/Makefile.am:
5310 * tests/check/gst/mountpoints.c:
5311 tests: add test for mountpoints
5313 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5315 * gst/rtsp-server/rtsp-client.c:
5316 client: fix factory leak
5317 Keep the factory in the state object only for authorization checks and make
5318 sure we unref it on failure. Also don't keep invalid objects in the state
5321 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5323 * gst/rtsp-server/rtsp-mount-points.c:
5324 mounts: add g_return_if guards
5326 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5328 * tests/check/gst/client.c:
5329 tests: add more tests
5331 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5333 * gst/rtsp-server/rtsp-client.c:
5334 client: improve debug
5336 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5338 * gst/rtsp-server/rtsp-client.c:
5339 client: improve debug and fix leaks
5340 Cleanup the uri and session when there is a bad request.
5342 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5347 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5349 * tests/check/gst/client.c:
5350 test: add test for session in options request
5352 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5354 * gst/rtsp-server/rtsp-client.c:
5355 client: use 454 when session can't be found
5356 We should use 454 when a session can't be found because there was no session
5357 pool configured in the server. This is not a server configuration problem
5358 because the server on which the request is done might not be the same one that
5359 will keep the sessions for us and so it does not need to support sessions.
5361 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5363 * gst/rtsp-server/rtsp-client.c:
5364 client: only free connection when there is one
5365 It's possible that the client doesn't have a connection when we try to free it.
5367 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5369 * tests/check/Makefile.am:
5370 * tests/check/gst/client.c:
5371 tests: add unit test for the client object
5373 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5375 * gst/rtsp-server/rtsp-client.c:
5376 client: small cleanup
5378 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5380 * gst/rtsp-server/rtsp-client.h:
5381 client: remove unused include
5383 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5385 * gst/rtsp-server/rtsp-client.c:
5386 client: fix compilation
5388 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5390 * gst/rtsp-server/rtsp-client.c:
5391 client: call destroy without the lock
5393 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5395 * gst/rtsp-server/rtsp-client.c:
5396 * gst/rtsp-server/rtsp-client.h:
5397 client: make the client usable without a socket
5398 Make a method to let the client handle a message and a callback when the client
5399 wants us to send a response message back. This makes it possible to also use the
5400 client object without the sockets, which should make it easier to test.
5402 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5404 * gst/rtsp-server/rtsp-client.c:
5405 * gst/rtsp-server/rtsp-client.h:
5406 client: small cleanup
5408 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5410 * docs/libs/gst-rtsp-server-sections.txt:
5411 * gst/rtsp-server/rtsp-client.c:
5412 * gst/rtsp-server/rtsp-client.h:
5413 * gst/rtsp-server/rtsp-server.c:
5414 client: remove reference to server
5415 We don't need to keep a ref to the server
5417 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5419 * gst/rtsp-server/rtsp-client.c:
5420 * gst/rtsp-server/rtsp-client.h:
5422 Also add some g_return_if()
5424 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5426 * gst/rtsp-server/rtsp-client.c:
5427 client: log more errors
5429 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5431 * gst/rtsp-server/rtsp-client.c:
5432 client: fix compilation
5434 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5436 * gst/rtsp-server/rtsp-client.c:
5437 * gst/rtsp-server/rtsp-client.h:
5438 client: add generic close-after-send support
5439 Add a property to send_response() to close the connection after the response has
5440 been sent to the client.
5442 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5445 * docs/libs/gst-rtsp-server-docs.sgml:
5446 * docs/libs/gst-rtsp-server-sections.txt:
5447 * docs/libs/gst-rtsp-server.types:
5448 * examples/test-auth.c:
5449 * examples/test-launch.c:
5450 * examples/test-mp4.c:
5451 * examples/test-multicast.c:
5452 * examples/test-multicast2.c:
5453 * examples/test-ogg.c:
5454 * examples/test-readme.c:
5455 * examples/test-sdp.c:
5456 * examples/test-uri.c:
5457 * examples/test-video.c:
5458 * gst/rtsp-server/Makefile.am:
5459 * gst/rtsp-server/rtsp-auth.h:
5460 * gst/rtsp-server/rtsp-client.c:
5461 * gst/rtsp-server/rtsp-client.h:
5462 * gst/rtsp-server/rtsp-media-mapping.c:
5463 * gst/rtsp-server/rtsp-media-mapping.h:
5464 * gst/rtsp-server/rtsp-mount-points.c:
5465 * gst/rtsp-server/rtsp-mount-points.h:
5466 * gst/rtsp-server/rtsp-server.c:
5467 * gst/rtsp-server/rtsp-server.h:
5468 * gst/rtsp-server/rtsp-session-media.c:
5469 * gst/rtsp-server/rtsp-session-pool.c:
5470 * gst/rtsp-server/rtsp-session-pool.h:
5471 * tests/check/gst/rtspserver.c:
5472 MediaMapping -> MountPoints
5473 Describes better what the object manages.
5475 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5478 configure: bump required version of -base
5480 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5482 * gst/rtsp-server/rtsp-media.c:
5485 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5487 * gst/rtsp-server/rtsp-media.c:
5488 * gst/rtsp-server/rtsp-media.h:
5489 media: support more Range formats
5490 Use the new -base methods to convert the Range string into a seek start and stop
5493 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5495 * examples/test-launch.c:
5496 examples: fix whitespace
5498 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5500 * examples/test-auth.c:
5501 test-auth: add example of how to remove sessions
5502 Add an example of the session filter api.
5504 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5506 * examples/test-uri.c:
5507 test-uri: remove mapping example
5509 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5511 * examples/test-uri.c:
5512 test-uri: fix callback signature
5514 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5516 * gst/rtsp-server/rtsp-media-factory.c:
5517 factory: keep ref to factory while media active
5518 While the media from a factory is alive, keep a ref to the factory.
5519 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
5521 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5523 * gst/rtsp-server/rtsp-media-factory-uri.c:
5524 factory-uri: add some debug
5526 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5528 * gst/rtsp-server/rtsp-stream.c:
5529 stream: set udp sources to PLAYING
5530 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
5531 so that it doesn't cause our pipeline to produce ASYNC-DONE.
5533 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5535 * gst/rtsp-server/rtsp-media-factory-uri.c:
5536 factory-uri: take ref to factory
5537 Take a ref to the factory that we place in our list.
5539 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5541 * tests/Makefile.am:
5542 * tests/test-reuse.c:
5543 test: add test for server reuse
5544 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
5546 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
5548 * gst/rtsp-server/rtsp-server.c:
5549 server: start and stop multiple times
5550 Stop listening on the RTSP port when the GSource is removed, so clients
5551 can't connect and the server can be started again.
5552 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
5554 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5556 * gst/rtsp-server/rtsp-server.c:
5557 server: fix small leak
5559 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5561 * gst/rtsp-server/rtsp-media.c:
5562 media: unref source in finish_unprepare
5563 The source is created in prepare, unref it in finish_unprepare.
5564 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
5566 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
5568 * gst/rtsp-server/rtsp-client.c:
5569 * gst/rtsp-server/rtsp-media.c:
5570 rtsp-media: remove bus watch before finalizing
5571 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
5572 * An extra media ref is added for the bus watch. This extra ref is unreffed by
5573 the GDestroyNotify function.
5574 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
5575 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
5576 gst_rtsp_media_unprepare before unreffing the media.
5577 This way, the bus watch will be removed before the media is finalized.
5578 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
5580 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
5582 * gst/rtsp-server/rtsp-client.c:
5583 * gst/rtsp-server/rtsp-client.h:
5584 client: wait until the TEARDOWN response is sent to close the connection
5585 Responses can be sent async so we need to wait until the TEARDOWN response has
5586 been written before we close the connection to the client. This avoids the risk
5587 of writing/polling closed sockets.
5588 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
5590 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
5592 * gst/rtsp-server/rtsp-stream.c:
5593 rtsp-stream: plug socket leak
5594 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
5596 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
5599 Automatic update of common submodule
5600 From 6bb6951 to a72faea
5602 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
5604 * gst/rtsp-server/rtsp-media-factory-uri.c:
5605 rtsp-server: don't use deprecated API
5607 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
5609 * gst/rtsp-server/rtsp-client.c:
5610 rtsp-client: fix unused-but-set-variable compiler warning
5611 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
5613 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5616 * docs/libs/gst-rtsp-server-sections.txt:
5617 * gst/rtsp-server/rtsp-client.c:
5620 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5622 * examples/Makefile.am:
5623 * examples/test-multicast2.c:
5624 examples: add another multicast example
5625 Add an example for how to configure separate multicast ranges for each media
5628 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5630 * examples/test-multicast.c:
5633 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5635 * gst/rtsp-server/rtsp-client.c:
5636 * gst/rtsp-server/rtsp-media.c:
5637 * gst/rtsp-server/rtsp-session-media.c:
5638 * gst/rtsp-server/rtsp-session-media.h:
5639 * gst/rtsp-server/rtsp-stream-transport.c:
5640 * gst/rtsp-server/rtsp-stream-transport.h:
5641 stream: use the address managed by the stream
5642 Use the address managed by the stream for multicast. This allows us to have 1
5643 multicast address for each stream.
5644 Because the address is now managed by the stream we don't have to pass it around
5646 Set the address pool on the streams.
5648 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5650 * gst/rtsp-server/rtsp-client.c:
5651 * gst/rtsp-server/rtsp-media.c:
5652 * gst/rtsp-server/rtsp-stream.c:
5655 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5657 * gst/rtsp-server/rtsp-media.c:
5658 * gst/rtsp-server/rtsp-media.h:
5659 media: add signal for new streams
5660 This allows applications to listen for new streams and configure properties on
5661 them, like the address pool.
5663 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5665 * gst/rtsp-server/rtsp-media.c:
5666 media: configure address pool in new streams
5668 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5670 * gst/rtsp-server/rtsp-stream.c:
5671 * gst/rtsp-server/rtsp-stream.h:
5672 stream: add methods to deal with address pool
5673 Add methods to get and set the address pool for the stream
5674 Add method to allocate and get the multicast addresses for this stream.
5676 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5678 * docs/libs/gst-rtsp-server-sections.txt:
5679 * gst/rtsp-server/rtsp-media.c:
5680 * gst/rtsp-server/rtsp-media.h:
5681 media: remove MTU property
5682 It is a stream property
5684 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5686 * gst/rtsp-server/rtsp-client.c:
5687 client: set blocksize only on stream
5688 Set the blocksize only on the current stream.
5690 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5692 * gst/rtsp-server/rtsp-stream.c:
5693 stream: share src and sink sockets
5694 the allocated socket is in the used-socket property, not socket.
5696 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5698 * gst/rtsp-server/rtsp-address-pool.c:
5699 * gst/rtsp-server/rtsp-address-pool.h:
5700 * gst/rtsp-server/rtsp-client.c:
5701 * gst/rtsp-server/rtsp-session-media.c:
5702 * gst/rtsp-server/rtsp-session-media.h:
5703 * gst/rtsp-server/rtsp-stream-transport.c:
5704 * gst/rtsp-server/rtsp-stream-transport.h:
5705 * tests/check/gst/addresspool.c:
5706 rtsp: make address-pool return an address object
5707 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
5708 store more info in the structure and allows us to more easily return the address
5709 to the right pool when no longer needed.
5710 Pass the address to the StreamTransport so that we can return it to the pool
5711 when the stream transport is freed or changed.
5713 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5715 * examples/Makefile.am:
5716 * examples/test-multicast.c:
5717 examples: add multicast example
5718 Show how to set up the multicast address pool so that media can be
5719 server with multicast.
5721 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5723 * gst/rtsp-server/rtsp-client.c:
5724 * gst/rtsp-server/rtsp-media-factory.c:
5725 * gst/rtsp-server/rtsp-media-factory.h:
5726 * gst/rtsp-server/rtsp-media.c:
5727 * gst/rtsp-server/rtsp-media.h:
5728 rtsp: use AddressPool
5729 Remove the multicast_group property.
5730 Use the configured addresspool to allocate multicast addresses.
5732 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5734 * gst/rtsp-server/rtsp-address-pool.c:
5735 * gst/rtsp-server/rtsp-address-pool.h:
5736 address-pool: add clear method
5738 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5740 * gst/rtsp-server/rtsp-address-pool.c:
5741 address-pool: small cleanups
5743 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5745 * tests/check/Makefile.am:
5746 * tests/check/gst/addresspool.c:
5747 tests: add addresspool unit test
5749 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5751 * gst/rtsp-server/Makefile.am:
5752 * gst/rtsp-server/rtsp-address-pool.c:
5753 * gst/rtsp-server/rtsp-address-pool.h:
5754 address-pool: add object to manage multicast addresses
5755 Make an object that can manage a rage of multicast addresses and ports.
5757 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5759 * gst/rtsp-server/rtsp-server.c:
5760 server: set default max-threads property
5762 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5764 * gst/rtsp-server/rtsp-media.c:
5765 media: wait for concurrent _prepare
5766 If a prepare is busy, wait for the result.
5768 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5770 * gst/rtsp-server/rtsp-media.c:
5771 media: add lock around message handler
5772 We don't want to dispatch messages while we are still processing the result of
5775 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5777 * gst/rtsp-server/rtsp-media.c:
5778 * gst/rtsp-server/rtsp-media.h:
5779 media: add lock to protect state changes
5781 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5783 * gst/rtsp-server/rtsp-stream.c:
5784 * gst/rtsp-server/rtsp-stream.h:
5787 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5789 * gst/rtsp-server/rtsp-stream-transport.c:
5790 * gst/rtsp-server/rtsp-stream-transport.h:
5791 * gst/rtsp-server/rtsp-stream.c:
5792 stream-transport: add keep-alive method
5794 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5796 * gst/rtsp-server/rtsp-stream-transport.c:
5797 * gst/rtsp-server/rtsp-stream-transport.h:
5798 * gst/rtsp-server/rtsp-stream.c:
5799 stream-transport: add method to handle RTP/RTCP
5800 Call new methods instead of poking into the structures directly.
5802 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5804 * gst/rtsp-server/rtsp-session-media.c:
5805 * gst/rtsp-server/rtsp-session-media.h:
5806 session-media: add locking
5808 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5810 * gst/rtsp-server/rtsp-session.c:
5811 * gst/rtsp-server/rtsp-session.h:
5812 session: add locking
5814 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5816 * gst/rtsp-server/rtsp-server.c:
5817 server: free old socket
5819 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5821 * gst/rtsp-server/rtsp-media-mapping.c:
5822 * gst/rtsp-server/rtsp-media-mapping.h:
5823 mapping: add locking
5825 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5827 * gst/rtsp-server/rtsp-media-factory.c:
5828 media-factory: add locking
5830 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5832 * gst/rtsp-server/rtsp-auth.c:
5833 * gst/rtsp-server/rtsp-auth.h:
5836 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5838 * gst/rtsp-server/rtsp-server.c:
5839 * gst/rtsp-server/rtsp-server.h:
5840 server: add max-thread property
5842 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5844 * gst/rtsp-server/rtsp-server.c:
5845 * gst/rtsp-server/rtsp-server.h:
5846 server: use a threadpool for the mainloops
5848 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5850 * gst/rtsp-server/rtsp-client.c:
5851 * gst/rtsp-server/rtsp-client.h:
5852 client: rename method
5853 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
5854 don't really create the client from the socket, we use the socket for the
5857 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5859 * gst/rtsp-server/rtsp-client.c:
5860 * gst/rtsp-server/rtsp-client.h:
5861 * gst/rtsp-server/rtsp-server.c:
5862 server: rework maincontext handling in clients
5863 Make a separate method to attach a client to a MainContext.
5864 Let the server decide in what GMainContext the client will operate and give this
5865 context to the client in attach. Then the server can later decide to use a
5866 separate thread for each client or just use the mainthread.
5868 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5870 * gst/rtsp-server/rtsp-client.c:
5871 * gst/rtsp-server/rtsp-session.c:
5872 * gst/rtsp-server/rtsp-session.h:
5873 session: move session header code in session object
5875 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
5879 * examples/test-auth.c:
5880 * examples/test-launch.c:
5881 * examples/test-mp4.c:
5882 * examples/test-ogg.c:
5883 * examples/test-readme.c:
5884 * examples/test-sdp.c:
5885 * examples/test-uri.c:
5886 * examples/test-video.c:
5887 * gst/rtsp-server/rtsp-auth.c:
5888 * gst/rtsp-server/rtsp-auth.h:
5889 * gst/rtsp-server/rtsp-client.c:
5890 * gst/rtsp-server/rtsp-client.h:
5891 * gst/rtsp-server/rtsp-media-factory-uri.c:
5892 * gst/rtsp-server/rtsp-media-factory-uri.h:
5893 * gst/rtsp-server/rtsp-media-factory.c:
5894 * gst/rtsp-server/rtsp-media-factory.h:
5895 * gst/rtsp-server/rtsp-media-mapping.c:
5896 * gst/rtsp-server/rtsp-media-mapping.h:
5897 * gst/rtsp-server/rtsp-media.c:
5898 * gst/rtsp-server/rtsp-media.h:
5899 * gst/rtsp-server/rtsp-params.c:
5900 * gst/rtsp-server/rtsp-params.h:
5901 * gst/rtsp-server/rtsp-sdp.c:
5902 * gst/rtsp-server/rtsp-sdp.h:
5903 * gst/rtsp-server/rtsp-server.c:
5904 * gst/rtsp-server/rtsp-server.h:
5905 * gst/rtsp-server/rtsp-session-media.c:
5906 * gst/rtsp-server/rtsp-session-media.h:
5907 * gst/rtsp-server/rtsp-session-pool.c:
5908 * gst/rtsp-server/rtsp-session-pool.h:
5909 * gst/rtsp-server/rtsp-session.c:
5910 * gst/rtsp-server/rtsp-session.h:
5911 * gst/rtsp-server/rtsp-stream-transport.c:
5912 * gst/rtsp-server/rtsp-stream-transport.h:
5913 * gst/rtsp-server/rtsp-stream.c:
5914 * gst/rtsp-server/rtsp-stream.h:
5915 * tests/check/gst/rtspserver.c:
5916 * tests/test-cleanup.c:
5919 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
5921 * gst/rtsp-server/rtsp-media.c:
5922 * gst/rtsp-server/rtsp-session-media.c:
5923 * gst/rtsp-server/rtsp-session.c:
5924 rtsp-server: added annotations to indicate type of ownership transfer of return values
5925 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5927 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
5930 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
5932 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
5935 * bindings/Makefile.am:
5936 * bindings/vala/Makefile.am:
5937 * bindings/vala/gst-rtsp-server-0.10.deps:
5938 * bindings/vala/gst-rtsp-server-0.10.vapi:
5939 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
5940 * bindings/vala/packages/gst-rtsp-server-0.10.files:
5941 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
5942 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
5943 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
5945 bindings: remove vala bindings
5946 They'll be reunited with the other GStreamer bindings
5947 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5949 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5951 * gst/rtsp-server/rtsp-client.c:
5952 * gst/rtsp-server/rtsp-session-media.c:
5953 * gst/rtsp-server/rtsp-session-media.h:
5954 * gst/rtsp-server/rtsp-stream-transport.c:
5955 * gst/rtsp-server/rtsp-stream-transport.h:
5956 rtsp: only create transport when needed
5957 Only create the StreamTransport when configured.
5959 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5961 * gst/rtsp-server/rtsp-client.c:
5962 client: small cleanup
5964 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5966 * gst/rtsp-server/rtsp-client.c:
5967 * gst/rtsp-server/rtsp-client.h:
5968 * gst/rtsp-server/rtsp-stream-transport.c:
5969 * gst/rtsp-server/rtsp-stream-transport.h:
5970 rtsp: refactor configuration of transport
5971 Move the configuration of the transport to a place where it makes
5974 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5976 * gst/rtsp-server/rtsp-client.c:
5977 client: refactor transport parsing
5979 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5981 * gst/rtsp-server/rtsp-client.c:
5982 client: refuse to change the MTU on shared media
5983 If we change the MTU of chared media, it changes for all clients.
5984 We don't want to set the MTU to something large for clients that
5987 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5989 * examples/test-mp4.c:
5990 * gst/rtsp-server/rtsp-media.c:
5991 small fixes to docs and debug
5993 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5995 * gst/rtsp-server/rtsp-stream.c:
5996 stream: transports must already have been removed
5998 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6000 * gst/rtsp-server/rtsp-media.c:
6001 * gst/rtsp-server/rtsp-stream.c:
6002 * gst/rtsp-server/rtsp-stream.h:
6003 stream: improve join and leave of the pipeline
6005 Do the cleanup properly
6008 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6010 * gst/rtsp-server/rtsp-media.c:
6011 media: move unprepare below default implementation
6012 Makes it easier to find the default implementation
6014 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6016 * gst/rtsp-server/rtsp-media.c:
6017 media: signal unprepared when we actually finish
6019 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6021 * gst/rtsp-server/rtsp-media.c:
6022 media: no need to unlock, unprepare does that when needed
6024 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6026 * docs/libs/gst-rtsp-server-sections.txt:
6027 * gst/rtsp-server/rtsp-media-factory.h:
6028 * gst/rtsp-server/rtsp-media-mapping.c:
6029 * gst/rtsp-server/rtsp-media.h:
6030 * gst/rtsp-server/rtsp-params.c:
6031 * gst/rtsp-server/rtsp-server.c:
6032 * gst/rtsp-server/rtsp-session-pool.h:
6033 * gst/rtsp-server/rtsp-session.c:
6034 * gst/rtsp-server/rtsp-session.h:
6035 * gst/rtsp-server/rtsp-stream-transport.h:
6036 * gst/rtsp-server/rtsp-stream.h:
6039 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6041 * gst/rtsp-server/rtsp-client.c:
6042 * gst/rtsp-server/rtsp-media-mapping.h:
6043 * gst/rtsp-server/rtsp-media.c:
6044 * gst/rtsp-server/rtsp-media.h:
6045 * gst/rtsp-server/rtsp-server.h:
6046 * gst/rtsp-server/rtsp-stream.c:
6047 * gst/rtsp-server/rtsp-stream.h:
6048 rtsp: fix MTU setting
6049 Fix setting of the MTU. There is no need for a vmethod.
6051 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6056 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6059 configure: bump version number after refactoring
6061 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6063 * gst/rtsp-server/Makefile.am:
6064 * gst/rtsp-server/rtsp-client.c:
6065 * gst/rtsp-server/rtsp-client.h:
6066 * gst/rtsp-server/rtsp-media-factory-uri.c:
6067 * gst/rtsp-server/rtsp-media-factory.c:
6068 * gst/rtsp-server/rtsp-media-factory.h:
6069 * gst/rtsp-server/rtsp-media.c:
6070 * gst/rtsp-server/rtsp-media.h:
6071 * gst/rtsp-server/rtsp-sdp.c:
6072 * gst/rtsp-server/rtsp-session-media.c:
6073 * gst/rtsp-server/rtsp-session-media.h:
6074 * gst/rtsp-server/rtsp-session.c:
6075 * gst/rtsp-server/rtsp-session.h:
6076 * gst/rtsp-server/rtsp-stream-transport.c:
6077 * gst/rtsp-server/rtsp-stream-transport.h:
6078 * gst/rtsp-server/rtsp-stream.c:
6079 * gst/rtsp-server/rtsp-stream.h:
6080 rtsp: massive refactoring
6081 Make GObjects from the remaining simple structures.
6082 Remove GstRTSPSessionStream, it's not needed.
6083 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
6084 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
6085 a GstRTSPStream should be transported to a client.
6086 Rename GstRTSPMediaFactory::get_element -> create_element because that
6087 more accurately describes what it does.
6088 Make nice methods instead of poking in the structures.
6089 Move some methods inside the relevant object source code.
6090 Use GPtrArray to store objects instead of plain arrays, it is more
6091 natural and allows us to more easily clean up.
6092 Move the allocation of udp ports to the Stream object. The Stream object
6093 contains the elements needed to stream the media to a client.
6094 Improve the prepare and unprepare methods. Unprepare should now undo
6095 everything prepare did. Improve also async unprepare when doing EOS on
6096 shutdown. Make sure we always unprepare correctly.
6098 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
6100 * gst/rtsp-server/rtsp-client.c:
6101 rtsp-client: Unref server address clients connected to
6102 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
6104 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
6106 * gst/rtsp-server/rtsp-server.c:
6107 rtsp-server: don't ref server socket if it is NULL
6108 Fixes test_bind_already_in_use unit test again after commit 6a497440.
6109 https://bugzilla.gnome.org/show_bug.cgi?id=686644
6111 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
6113 * tests/check/Makefile.am:
6114 tests: Add libgio link dependency
6115 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
6117 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6119 * gst/rtsp-server/rtsp-media-mapping.c:
6120 * gst/rtsp-server/rtsp-media-mapping.h:
6121 rtsp-media-mapping: rename find_media vfunc to find_factory
6122 The virtual method and class method should have the same name
6123 so it is correctly represented in GIR file
6124 https://bugzilla.gnome.org/show_bug.cgi?id=680777
6126 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6128 * gst/rtsp-server/rtsp-auth.c:
6129 * gst/rtsp-server/rtsp-client.c:
6130 * gst/rtsp-server/rtsp-media-factory-uri.c:
6131 * gst/rtsp-server/rtsp-media-factory.c:
6132 * gst/rtsp-server/rtsp-media-mapping.c:
6133 * gst/rtsp-server/rtsp-media.c:
6134 * gst/rtsp-server/rtsp-server.c:
6135 * gst/rtsp-server/rtsp-session-pool.c:
6136 * gst/rtsp-server/rtsp-session.c:
6137 rtsp-server: fixed comments and GIR annotations
6138 https://bugzilla.gnome.org/show_bug.cgi?id=680777
6140 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
6142 * gst/rtsp-server/rtsp-media-mapping.c:
6143 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
6145 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
6147 * gst/rtsp-server/rtsp-server.c:
6148 rtsp-server: allow binding on port 0 (binds on a random port)
6150 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
6152 * gst/rtsp-server/rtsp-server.c:
6153 * gst/rtsp-server/rtsp-server.h:
6154 rtsp-server: add bound-port property
6155 bound-port can be used to retrieve the port number when the server is bound on
6156 port 0, which binds on a random port.
6158 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
6160 * gst/rtsp-server/rtsp-media-factory.c:
6161 * gst/rtsp-server/rtsp-media-factory.h:
6162 rtsp-media-factory: make ::get_element overridable by GI bindings
6163 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
6164 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
6165 as the invoker for ::get_element(), making it overridable by GI generated
6168 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
6170 * gst/rtsp-server/rtsp-media-factory-uri.c:
6171 rtsp-media-factory-uri: don't autoplug parsers in a loop
6172 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
6175 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
6177 * gst/rtsp-server/Makefile.am:
6178 Explicitly link against gio. Fix link error on mac.
6180 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
6182 * gst/rtsp-server/rtsp-session.c:
6183 session: add ttl to the transport header in SETUP
6184 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
6186 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
6188 * gst/rtsp-server/rtsp-client.c:
6189 * gst/rtsp-server/rtsp-client.h:
6190 * gst/rtsp-server/rtsp-media.c:
6191 client: Use client transport settings for multicast if allowed.
6192 This patch makes it possible for the client to send transport settings for
6193 multicast (destination && ttl). Client settings must be explicitly allowed or
6194 the server will use its own settings.
6195 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
6197 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
6200 Automatic update of common submodule
6201 From 6c0b52c to 6bb6951
6203 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
6205 * gst/rtsp-server/rtsp-client.c:
6206 rtsp-client: do not destroy the rtsp watch
6207 Don't destroy the client watch while dispatching. The rtsp watch is
6208 automatically destroyed after the rtsp watch function closed() has
6210 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
6212 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
6215 Automatic update of common submodule
6216 From 4f962f7 to 6c0b52c
6218 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
6220 * gst/rtsp-server/rtsp-media.c:
6221 media: fix check for seekability
6223 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6225 * gst/rtsp-server/rtsp-client.c:
6226 client: use more GIO
6227 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
6229 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6231 * gst/rtsp-server/rtsp-server.c:
6232 server: remove obsolete includes
6234 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
6236 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
6237 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
6238 be available in "on_new_ssrc". The transports are added in
6239 gst_rtsp_media_set_state when going to PLAYING state. However,
6240 "on_new_ssrc" might be called before this happens.
6241 https://bugzilla.gnome.org/show_bug.cgi?id=683304
6243 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
6245 * gst/rtsp-server/rtsp-client.c:
6246 * gst/rtsp-server/rtsp-client.h:
6247 rtsp-client: add signals for rtsp requests (fixes #683287)
6249 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
6251 * gst/rtsp-server/rtsp-client.c:
6252 * gst/rtsp-server/rtsp-client.h:
6253 add new-session signal to rtsp-client (fixes #683058)
6255 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
6258 Automatic update of common submodule
6259 From 668acee to 4f962f7
6261 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
6263 * gst/rtsp-server/rtsp-server.c:
6264 * tests/check/gst/rtspserver.c:
6265 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
6266 Do not assume that *error is set in g_socket_address_enumerator_next.
6267 Added test_bind_already_in_use unit-test.
6268 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
6270 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
6273 Automatic update of common submodule
6274 From 94ccf4c to 668acee
6276 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
6278 * gst/rtsp-server/rtsp-client.c:
6279 * gst/rtsp-server/rtsp-client.h:
6280 rtsp-client: make create_sdp virtual method
6281 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
6283 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6286 Automatic update of common submodule
6287 From 98e386f to 94ccf4c
6289 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6291 * gst/rtsp-server/rtsp-client.c:
6294 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
6296 * gst/rtsp-server/rtsp-client.c:
6297 * gst/rtsp-server/rtsp-client.h:
6298 * gst/rtsp-server/rtsp-server.c:
6299 * gst/rtsp-server/rtsp-server.h:
6300 rtsp-server: use an existing socket to establish HTTP tunnel
6301 Make it possible to transfer a socket from an HTTP server to be used as
6302 an RTSP over HTTP tunnel.
6304 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
6306 * gst/rtsp-server/rtsp-client.c:
6307 * gst/rtsp-server/rtsp-media.c:
6308 * gst/rtsp-server/rtsp-media.h:
6309 rtsp: Handle the blocksize parameter
6310 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
6312 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
6314 * tests/check/Makefile.am:
6315 * tests/check/gst/rtspserver.c:
6316 Have unit test get header from source dir, not installed dir
6317 This makes compilation of unit tests work in a build directory other
6318 than the source directory.
6319 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
6321 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
6323 * gst/rtsp-server/rtsp-media.c:
6324 rtsp-media: update for gst_element_make_from_uri() changes
6326 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
6329 * tests/Makefile.am:
6330 * tests/check/Makefile.am:
6331 * tests/check/gst/rtspserver.c:
6333 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
6335 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
6337 * gst/rtsp-server/rtsp-media.c:
6338 rtsp-media: don't collect media stats when going to NULL
6339 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
6341 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6343 * gst/rtsp-server/rtsp-client.c:
6344 client: don't leak transports
6346 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
6348 * gst/rtsp-server/rtsp-client.c:
6349 rtsp-client: free transport on no_stream in SETUP handler
6351 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
6353 * gst/rtsp-server/rtsp-client.c:
6354 rtsp-client: changed session media iteration
6355 In client_unlink_session: now don't iterate in session->medias
6356 list where items are removed by gst_rtsp_session_release_media.
6357 Instead, repeatedly remove the first item.
6359 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
6361 * gst/rtsp-server/rtsp-client.c:
6362 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
6363 GstRTSPSessionMedia is not a GObject type. When the
6364 GstRTSPSession is freed, it will free the media.
6366 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
6368 * gst/rtsp-server/rtsp-media-factory.c:
6369 factory: plug pad leak in collect_streams
6370 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
6371 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
6372 will take one reference, and the other reference will otherwise
6375 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
6378 configure: suppress some warnings when debug is disabled
6379 Warnings about unused variables should be suppressed if core has the
6380 debug system disabled.
6381 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
6383 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6385 * docs/libs/Makefile.am:
6386 docs: fix build in uninstalled setup
6387 Include gst-plugins-base libs properly.
6389 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
6391 * docs/libs/gst-rtsp-server.types:
6392 docs: include headers defining rtsp-server object types
6393 Fixes compiler warnings during docs build.
6394 https://bugzilla.gnome.org/show_bug.cgi?id=676824
6396 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
6399 configure: Add warning flags for compiler when configuring
6400 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
6402 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6405 Automatic update of common submodule
6406 From 03a0e57 to 98e386f
6408 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6411 Automatic update of common submodule
6412 From 1fab359 to 03a0e57
6414 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
6416 * gst/rtsp-server/rtsp-client.c:
6417 client: fix GSocketAddress leak in gst_rtsp_client_accept
6418 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
6420 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6423 Automatic update of common submodule
6424 From f1b5a96 to 1fab359
6426 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6429 Automatic update of common submodule
6430 From 92b7266 to f1b5a96
6432 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6435 Automatic update of common submodule
6436 From ec1c4a8 to 92b7266
6438 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6441 Automatic update of common submodule
6442 From 3429ba6 to ec1c4a8
6444 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
6446 * gst/rtsp-server/rtsp-auth.c:
6447 * gst/rtsp-server/rtsp-client.c:
6448 * gst/rtsp-server/rtsp-media-factory-uri.c:
6449 * gst/rtsp-server/rtsp-server.c:
6450 rtsp: fix compiler warnings
6451 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
6453 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6456 Automatic update of common submodule
6457 From dc70203 to 3429ba6
6459 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6461 * gst/rtsp-server/rtsp-client.c:
6462 * gst/rtsp-server/rtsp-media-factory.c:
6463 * gst/rtsp-server/rtsp-media-factory.h:
6464 * gst/rtsp-server/rtsp-media.c:
6465 * gst/rtsp-server/rtsp-media.h:
6466 * gst/rtsp-server/rtsp-server.c:
6467 * gst/rtsp-server/rtsp-server.h:
6468 * gst/rtsp-server/rtsp-session-pool.c:
6469 * gst/rtsp-server/rtsp-session-pool.h:
6470 rtsp-server: port to new thread API
6472 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6475 Automatic update of common submodule
6476 From 6db25be to dc70203
6478 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6480 * gst/rtsp-server/rtsp-auth.c:
6481 * gst/rtsp-server/rtsp-auth.h:
6482 * gst/rtsp-server/rtsp-client.c:
6483 rtsp-server: Fix compilation and compiler warnings
6485 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6489 * gst/rtsp-server/Makefile.am:
6490 configure: Modernize autotools setup a bit
6491 Also we now only create tar.bz2 and tar.xz tarballs.
6493 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6496 Automatic update of common submodule
6497 From 464fe15 to 6db25be
6499 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6502 Automatic update of common submodule
6503 From 7fda524 to 464fe15
6505 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6508 * docs/libs/Makefile.am:
6509 * docs/version.entities.in:
6511 * gst/rtsp-server/Makefile.am:
6512 * pkgconfig/Makefile.am:
6513 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
6514 * pkgconfig/gstreamer-rtsp-server.pc.in:
6515 * tests/Makefile.am:
6516 rtsp-server: Update versioning
6518 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6520 Merge remote-tracking branch 'origin/0.10'
6522 gst/rtsp-server/rtsp-session-pool.c
6524 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6526 * gst/rtsp-server/rtsp-session-pool.c:
6527 rtsp-server: Don't use deprecated GLib API
6529 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6531 Replace master with 0.11
6533 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6535 Merge branch 'master' into 0.11
6537 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6539 Merge branch 'master' into 0.11
6541 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
6544 A couple minor typo fixes
6546 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6548 * gst/rtsp-server/rtsp-media.c:
6549 media: fix state of the appqueue
6551 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6553 * gst/rtsp-server/rtsp-media-factory-uri.c:
6554 factory: use videoconvert
6556 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6558 * gst/rtsp-server/rtsp-media-factory-uri.c:
6559 factory: change to new style caps
6561 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6563 * gst/rtsp-server/rtsp-client.c:
6564 * gst/rtsp-server/rtsp-client.h:
6565 * gst/rtsp-server/rtsp-media-factory-uri.c:
6566 * gst/rtsp-server/rtsp-media.c:
6567 * gst/rtsp-server/rtsp-server.c:
6568 * gst/rtsp-server/rtsp-server.h:
6569 * gst/rtsp-server/rtsp-session-pool.c:
6570 rtsp-server: port to GIO
6573 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6576 configure: fix build
6578 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6581 docs: fix for gst_rtsp_server_set_port() -> _set_service()
6582 https://bugzilla.gnome.org/show_bug.cgi?id=666548
6584 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6587 * examples/Makefile.am:
6588 First rule of gst-rtsp-server club: don't talk about gst-phonon
6590 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6593 * pkgconfig/Makefile.am:
6594 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
6595 * pkgconfig/gst-rtsp-server.pc.in:
6596 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
6597 * pkgconfig/gstreamer-rtsp-server.pc.in:
6598 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
6599 For consistency with all other modules.
6601 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6603 * gst/rtsp-server/rtsp-client.c:
6604 rtsp-client: update for new map API
6606 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6609 * bindings/Makefile.am:
6610 * bindings/python/Makefile.am:
6611 * bindings/python/arg-types.py:
6612 * bindings/python/codegen/Makefile.am:
6613 * bindings/python/codegen/__init__.py:
6614 * bindings/python/codegen/argtypes.py:
6615 * bindings/python/codegen/code-coverage.py:
6616 * bindings/python/codegen/codegen.py:
6617 * bindings/python/codegen/definitions.py:
6618 * bindings/python/codegen/defsparser.py:
6619 * bindings/python/codegen/docextract.py:
6620 * bindings/python/codegen/docgen.py:
6621 * bindings/python/codegen/fileprefix.override:
6622 * bindings/python/codegen/fileprefixmodule.c:
6623 * bindings/python/codegen/h2def.py:
6624 * bindings/python/codegen/mergedefs.py:
6625 * bindings/python/codegen/mkskel.py:
6626 * bindings/python/codegen/override.py:
6627 * bindings/python/codegen/reversewrapper.py:
6628 * bindings/python/codegen/scmexpr.py:
6629 * bindings/python/rtspserver-types.defs:
6630 * bindings/python/rtspserver.defs:
6631 * bindings/python/rtspserver.override:
6632 * bindings/python/rtspservermodule.c:
6633 * bindings/python/test.py:
6635 python: remove pygst-based python bindings
6636 pygi is the future, apparently.
6638 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
6641 Automatic update of common submodule
6642 From c463bc0 to 7fda524
6644 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6647 Automatic update of common submodule
6648 From 2a59016 to c463bc0
6650 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6653 Automatic update of common submodule
6654 From 0807187 to 2a59016
6656 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6659 Automatic update of common submodule
6660 From 11f0cd5 to 0807187
6662 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6664 * examples/test-auth.c:
6665 example: update for new caps
6667 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6669 * examples/test-video.c:
6670 * gst/rtsp-server/rtsp-client.c:
6671 * gst/rtsp-server/rtsp-media-factory-uri.c:
6672 * gst/rtsp-server/rtsp-media.c:
6673 * gst/rtsp-server/rtsp-media.h:
6674 * gst/rtsp-server/rtsp-session.c:
6675 * gst/rtsp-server/rtsp-session.h:
6676 rtsp-server: port some more to 0.11
6678 Remove bufferlist stuff
6680 Add queue before appsink now that preroll-queue-len is gone.
6681 Update for request pad changes.
6683 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6685 Merge branch 'master' into 0.11
6687 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
6689 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6690 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
6691 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6693 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
6695 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6696 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
6697 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6699 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6701 Merge branch 'master' into 0.11
6703 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6705 * gst/rtsp-server/rtsp-media.c:
6706 * gst/rtsp-server/rtsp-media.h:
6707 media: add a seekable boolean
6708 Maintain the seekable state with a new variable instead of reusing the
6711 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
6713 * gst/rtsp-server/rtsp-media.c:
6714 Disallow seek in live media
6716 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6718 Merge branch 'master' into 0.11
6720 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
6722 * gst/rtsp-server/rtsp-server.c:
6723 #ifdef statements for windows socket creation were missing
6725 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
6728 Automatic update of common submodule
6729 From a39eb83 to 11f0cd5
6731 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
6734 Automatic update of common submodule
6735 From 605cd9a to a39eb83
6737 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6739 Merge branch 'master' into 0.11
6741 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6743 * gst/rtsp-server/rtsp-client.c:
6744 client: use method to access property
6746 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6748 * gst/rtsp-server/rtsp-media-factory.c:
6749 * gst/rtsp-server/rtsp-media-factory.h:
6750 media-factory: add protocols property
6751 Add a property to configure the allowed protocols in the media created from the
6754 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6756 * gst/rtsp-server/rtsp-media-factory.c:
6757 * gst/rtsp-server/rtsp-media-factory.h:
6758 media-factory: add media-configure signal
6759 Add signal to allow the application to configure the media after it was created
6762 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6764 * gst/rtsp-server/rtsp-client.c:
6765 client: use method to access property
6767 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6769 * gst/rtsp-server/rtsp-media-factory.c:
6770 * gst/rtsp-server/rtsp-media-factory.h:
6771 media-factory: add protocols property
6772 Add a property to configure the allowed protocols in the media created from the
6775 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6777 * gst/rtsp-server/rtsp-media-factory.c:
6778 * gst/rtsp-server/rtsp-media-factory.h:
6779 media-factory: add media-configure signal
6780 Add signal to allow the application to configure the media after it was created
6783 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6785 Merge branch 'master' into 0.11
6787 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6789 * gst/rtsp-server/rtsp-client.c:
6790 client: use media multicast group
6792 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6794 * gst/rtsp-server/rtsp-media-factory.h:
6795 * gst/rtsp-server/rtsp-server.h:
6796 * gst/rtsp-server/rtsp-session-pool.h:
6797 * gst/rtsp-server/rtsp-session.h:
6800 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6802 * gst/rtsp-server/rtsp-client.c:
6803 * gst/rtsp-server/rtsp-sdp.h:
6804 sdp: copy and free the server ip address
6805 Copy and free the server ip address to make memory management easier later.
6807 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6809 * gst/rtsp-server/rtsp-media-factory.c:
6810 media-factory: configure multicast in media
6812 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6814 * gst/rtsp-server/rtsp-media.c:
6815 * gst/rtsp-server/rtsp-media.h:
6816 media: add property for multicast group
6817 Add a property to configure the multicast group in the media.
6818 Based on patches from Marc Leeman and Robert Krakora.
6820 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6822 * gst/rtsp-server/rtsp-media-factory.c:
6823 * gst/rtsp-server/rtsp-media-factory.h:
6824 media-factory: add property for multicast group
6825 Add a property to configure the multicast group in the media factory.
6826 Based on patches from Marc Leeman and Robert Krakora.
6828 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6830 * gst/rtsp-server/rtsp-client.c:
6831 client: do configuration of transport in one place
6832 Move the configuration of the transport destination address to where we also
6833 configure the other bits.
6835 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6837 * gst/rtsp-server/rtsp-client.c:
6838 client: use media multicast group
6840 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6842 * gst/rtsp-server/rtsp-media-factory.h:
6843 * gst/rtsp-server/rtsp-server.h:
6844 * gst/rtsp-server/rtsp-session-pool.h:
6845 * gst/rtsp-server/rtsp-session.h:
6848 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6850 * gst/rtsp-server/rtsp-client.c:
6851 * gst/rtsp-server/rtsp-sdp.h:
6852 sdp: copy and free the server ip address
6853 Copy and free the server ip address to make memory management easier later.
6855 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6857 * gst/rtsp-server/rtsp-media-factory.c:
6858 media-factory: configure multicast in media
6860 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6862 * gst/rtsp-server/rtsp-media.c:
6863 * gst/rtsp-server/rtsp-media.h:
6864 media: add property for multicast group
6865 Add a property to configure the multicast group in the media.
6866 Based on patches from Marc Leeman and Robert Krakora.
6868 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6870 * gst/rtsp-server/rtsp-media-factory.c:
6871 * gst/rtsp-server/rtsp-media-factory.h:
6872 media-factory: add property for multicast group
6873 Add a property to configure the multicast group in the media factory.
6874 Based on patches from Marc Leeman and Robert Krakora.
6876 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6878 * gst/rtsp-server/rtsp-client.c:
6879 client: do configuration of transport in one place
6880 Move the configuration of the transport destination address to where we also
6881 configure the other bits.
6883 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6885 Merge branch 'master' into 0.11
6887 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6889 * gst/rtsp-server/rtsp-client.c:
6890 client: destroy pipeline on client disconnect with no prior TEARDOWN.
6891 The problem occurs when the client abruptly closes the connection without
6892 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
6893 server is where the pipeline gets torn down. Since this handler is not called,
6894 the pipeline remains and is up and running. Subsequent clients get their own
6895 pipelines and if the do not issue TEARDOWNs then those pipelines will also
6896 remain up and running. This is a resource leak.
6898 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6900 Merge branch 'master' into 0.11
6902 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
6904 * gst/rtsp-server/rtsp-media-factory.c:
6905 * gst/rtsp-server/rtsp-media-factory.h:
6906 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
6907 For example, it can be used to retrieve source elements like appsrc, in a more
6908 convenient way than subclassing get_element.
6910 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6912 Merge branch 'master' into 0.11
6914 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
6916 * gst/rtsp-server/rtsp-server.c:
6917 rtsp-server: hold on to reference while using object
6919 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6921 * gst/rtsp-server/rtsp-media.c:
6924 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6927 configure: use unstable api
6929 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
6931 * gst/rtsp-server/rtsp-client.c:
6932 client: fix reference counting
6934 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
6936 * gst/rtsp-server/rtsp-client.c:
6937 * gst/rtsp-server/rtsp-media.c:
6938 fix compiler warnings about unused variables
6940 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
6942 * examples/test-launch.c:
6943 * examples/test-readme.c:
6944 * examples/test-uri.c:
6945 * examples/test-video.c:
6946 examples: tell rtsp uri when ready
6948 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
6951 Automatic update of common submodule
6952 From 69b981f to 605cd9a
6954 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6956 * gst/rtsp-server/rtsp-client.c:
6957 client: update for buffer API change
6959 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6961 * gst/rtsp-server/Makefile.am:
6962 Makefile.am: 0.10 => @GST_MAJORMINOR@
6964 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6966 * gst/rtsp-server/rtsp-media-factory-uri.c:
6967 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
6969 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6971 * gst/rtsp-server/.gitignore:
6972 .gitignore: 0.10 => 0.11
6974 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6976 * gst/rtsp-server/Makefile.am:
6977 Makefile.am: 0.10 => @GST_MAJORMINOR@
6979 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6981 Merge branch 'master' into 0.11
6983 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
6986 Automatic update of common submodule
6987 From 9e5bbd5 to 69b981f
6989 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
6992 Automatic update of common submodule
6993 From fd35073 to 9e5bbd5
6995 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
6998 Automatic update of common submodule
6999 From 46dfcea to fd35073
7001 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7003 * gst/rtsp-server/rtsp-media-factory-uri.c:
7004 * gst/rtsp-server/rtsp-media.c:
7005 media: port to new caps API
7007 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7009 Merge branch 'master' into 0.11
7011 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
7013 * bindings/vala/gst-rtsp-server-0.10.vapi:
7014 Updated Vala bindings.
7015 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
7017 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
7019 * gst/rtsp-server/rtsp-server.c:
7020 * gst/rtsp-server/rtsp-server.h:
7021 Add a signal for newly connected clients.
7022 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
7024 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
7026 * bindings/python/rtspserver.override:
7027 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
7029 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7031 * gst/rtsp-server/Makefile.am:
7032 * gst/rtsp-server/rtsp-client.c:
7033 * gst/rtsp-server/rtsp-funnel.c:
7034 * gst/rtsp-server/rtsp-funnel.h:
7035 * gst/rtsp-server/rtsp-media.c:
7036 rtsp-server: port to 0.11
7038 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7043 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7045 Merge branch 'master' into 0.11
7050 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7053 Automatic update of common submodule
7054 From c3cafe1 to 46dfcea
7056 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
7058 * bindings/python/Makefile.am:
7059 * bindings/python/rtspserver.defs:
7060 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
7062 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
7064 * bindings/python/arg-types.py:
7065 python bindings: add GstRTSPUrlParam
7066 Needed to implement MediaFactory virtual proxies
7068 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
7070 * bindings/python/arg-types.py:
7071 python bindings: fix returning GstRTSPUrl types
7073 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
7075 * bindings/python/arg-types.py:
7076 python bindings: add arg type for GstRTSPUrl
7078 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
7080 * bindings/python/rtspserver.defs:
7081 python bindings: fix the definition of MediaFactory.collect_stream
7083 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
7086 Automatic update of common submodule
7087 From 1ccbe09 to c3cafe1
7089 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7092 Automatic update of common submodule
7093 From 193b717 to 1ccbe09
7095 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
7098 Automatic update of common submodule
7099 From b77e2bf to 193b717
7101 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7104 build: Include lcov.mak to allow test coverage report generation
7106 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7109 Automatic update of common submodule
7110 From d8814b6 to b77e2bf
7112 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7115 Automatic update of common submodule
7116 From 6aaa286 to d8814b6
7118 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
7121 Automatic update of common submodule
7122 From 6aec6b9 to 6aaa286
7124 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
7127 autogen: wingo signed comment
7129 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
7131 * gst/rtsp-server/rtsp-session-pool.c:
7132 session: use full charset for RTSP session ID
7133 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
7134 session ID more difficult.
7135 https://bugzilla.gnome.org/show_bug.cgi?id=643812
7137 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7139 * gst/rtsp-server/Makefile.am:
7140 rtsp-server: Don't install the funnel header
7142 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
7145 Automatic update of common submodule
7146 From 1de7f6a to 6aec6b9
7148 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7151 configure: require core/base 0.10.31
7152 Needed at least for gst_plugin_feature_rank_compare_func().
7154 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
7157 Automatic update of common submodule
7158 From f94d739 to 1de7f6a
7160 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7162 * gst/rtsp-server/rtsp-media.c:
7163 media: remove more unused code
7165 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7167 * gst/rtsp-server/rtsp-media.c:
7168 * gst/rtsp-server/rtsp-media.h:
7169 media: remove duplicate filtering
7170 Remove the duplicate filtering code now that we have a released -good version.
7171 Give a warning instead.
7173 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7175 * gst/rtsp-server/rtsp-media-factory.c:
7176 * gst/rtsp-server/rtsp-media.c:
7177 media: fix default buffer size
7179 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7181 * gst/rtsp-server/rtsp-media-factory.c:
7182 * gst/rtsp-server/rtsp-media-factory.h:
7183 media-factory: add property to configure the buffer-size
7184 Add a property to configure the kernel UDP buffer size.
7186 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7188 * gst/rtsp-server/rtsp-media.c:
7189 * gst/rtsp-server/rtsp-media.h:
7190 media: add property to configure kernel buffer sizes
7191 Add a property to configure the kernel UDP buffer size.
7193 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7196 configure: set PYGOBJECT_REQ before using it
7197 https://bugzilla.gnome.org/show_bug.cgi?id=640641
7199 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7202 docs: recursive into sub-directories on 'make upload'
7204 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7206 * docs/libs/gst-rtsp-server-docs.sgml:
7207 * docs/version.entities.in:
7208 docs: mention full version these docs are for, not just major-minor
7210 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7215 === release 0.10.8 ===
7217 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7222 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7224 * gst/rtsp-server/rtsp-server.c:
7225 rtsp-server: clarify docs a little
7227 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7229 * gst/rtsp-server/rtsp-media.c:
7230 media: init debug category before starting thread
7232 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7234 * gst/rtsp-server/rtsp-auth.c:
7235 auth: add realm to make it more spec compliant
7237 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7239 * gst/rtsp-server/rtsp-server.c:
7240 * gst/rtsp-server/rtsp-server.h:
7243 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7245 * examples/test-video.c:
7246 example: improve example docs a little
7248 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7250 * gst/rtsp-server/rtsp-server.c:
7251 server: ensure the watch has a ref to the server
7253 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7255 * gst/rtsp-server/rtsp-server.c:
7256 server: simpify channel function
7258 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7260 * gst/rtsp-server/rtsp-server.c:
7261 * gst/rtsp-server/rtsp-server.h:
7262 server: simplify management of channel and source
7263 We don't need to keep around the channel and source objects. Let the mainloop
7264 and the source manage the source and channel respectively.
7266 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7272 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7275 * tests/Makefile.am:
7276 * tests/test-cleanup.c:
7277 tests: add tests directory and cleanup test
7279 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7281 * gst/rtsp-server/rtsp-media-factory-uri.c:
7282 * gst/rtsp-server/rtsp-media-factory.c:
7283 * gst/rtsp-server/rtsp-media-mapping.c:
7284 * gst/rtsp-server/rtsp-media.c:
7285 * gst/rtsp-server/rtsp-session-pool.c:
7286 * gst/rtsp-server/rtsp-session.c:
7287 server: improve debugging in various objects
7289 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7291 * gst/rtsp-server/rtsp-server.c:
7292 server: chain up to the parent finalize
7294 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
7296 * bindings/python/rtspserver-types.defs:
7297 * bindings/python/rtspserver.defs:
7298 * bindings/python/rtspserver.override:
7299 * bindings/python/test.py:
7300 gst-rtsp-server: update python bindings
7302 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7304 * gst/rtsp-server/rtsp-client.c:
7305 client: use the response from the clientstate
7306 Create the response object only once and store in the client state.
7307 Make all methods use the state response,
7309 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7311 * gst/rtsp-server/rtsp-server.c:
7312 server: use signal to keep track of clients
7313 Keep track of all the clients that the server creates and remove them when they
7314 fire the 'closed' signal.
7316 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7318 * gst/rtsp-server/rtsp-client.c:
7319 * gst/rtsp-server/rtsp-client.h:
7320 client: emit signal when closing
7322 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7324 * examples/.gitignore:
7325 * examples/Makefile.am:
7326 * examples/test-auth.c:
7327 * examples/test-video.c:
7328 * gst/rtsp-server/rtsp-auth.c:
7329 * gst/rtsp-server/rtsp-auth.h:
7330 * gst/rtsp-server/rtsp-client.c:
7331 * gst/rtsp-server/rtsp-media-factory.c:
7332 * gst/rtsp-server/rtsp-media.c:
7333 * gst/rtsp-server/rtsp-media.h:
7334 * gst/rtsp-server/rtsp-session-pool.h:
7335 * gst/rtsp-server/rtsp-session.h:
7336 media: enable per factory authorisations
7337 Allow for adding a GstRTSPAuth on the factory and media level and check
7338 permissions when accessing the factory.
7339 Add hints to the auth methods for future more fine grained authorisation.
7340 Add example application for per factory authentication.
7342 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7344 * gst/rtsp-server/rtsp-auth.c:
7345 * gst/rtsp-server/rtsp-auth.h:
7346 * gst/rtsp-server/rtsp-client.c:
7347 * gst/rtsp-server/rtsp-client.h:
7348 * gst/rtsp-server/rtsp-params.c:
7349 * gst/rtsp-server/rtsp-params.h:
7350 rtsp-server: Pass ClientState structure arround
7351 Pass the collected information for the ongoing request in a GstRTSPClientState
7352 structure that we can then pass around to simplify the method arguments. This
7353 will also be handy when we implement logging functionality.
7355 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7357 * gst/rtsp-server/rtsp-media-factory.c:
7358 * gst/rtsp-server/rtsp-media-factory.h:
7359 media-factory: add methods to configure authorisation
7361 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7363 * gst/rtsp-server/rtsp-client.c:
7364 client: unref auth in finalize
7366 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7368 * gst/rtsp-server/rtsp-server.c:
7369 server: unref auth in finalize
7371 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7373 * docs/libs/gst-rtsp-server-docs.sgml:
7374 * docs/libs/gst-rtsp-server-sections.txt:
7375 * docs/libs/gst-rtsp-server.types:
7378 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7380 * gst/rtsp-server/rtsp-server.c:
7381 * gst/rtsp-server/rtsp-server.h:
7382 server: separate create and accept
7383 Create separate create and accept methods so that subclasses can create custom
7385 Configure the server in the client object and prepare for keeping track of
7388 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7390 * gst/rtsp-server/rtsp-client.c:
7391 * gst/rtsp-server/rtsp-client.h:
7392 client: add support for setting the server.
7393 Add support for keeping a ref to the server that started this client
7396 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7398 * gst/rtsp-server/rtsp-auth.c:
7399 auth: fix memleak and add some docs
7400 Fix a memleak of the basic auth token.
7401 Add docs for the helper function
7403 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7405 * gst/rtsp-server/rtsp-auth.c:
7406 * gst/rtsp-server/rtsp-auth.h:
7407 * gst/rtsp-server/rtsp-client.c:
7408 client: delegate setup of auth to the manager
7409 Delegate the configuration of the authentication tokens to the manager object
7412 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7414 * examples/test-video.c:
7415 * gst/rtsp-server/Makefile.am:
7416 * gst/rtsp-server/rtsp-auth.c:
7417 * gst/rtsp-server/rtsp-auth.h:
7418 * gst/rtsp-server/rtsp-client.c:
7419 * gst/rtsp-server/rtsp-client.h:
7420 * gst/rtsp-server/rtsp-server.c:
7421 * gst/rtsp-server/rtsp-server.h:
7422 auth: add authentication object
7423 Add an object that can check the authorization of requests.
7424 Implement basic authentication.
7425 Add example authentication to test-video
7427 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7429 * gst/rtsp-server/rtsp-server.c:
7430 * gst/rtsp-server/rtsp-server.h:
7431 server: move includes back
7432 the includes are needed for sockaddr_in.
7434 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7436 * gst/rtsp-server/rtsp-client.c:
7437 * gst/rtsp-server/rtsp-client.h:
7438 * gst/rtsp-server/rtsp-server.c:
7439 * gst/rtsp-server/rtsp-server.h:
7440 rtsp: move network includes where they are needed
7442 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
7444 * gst/rtsp-server/rtsp-media.h:
7445 rtsp-media.h: Minor corrections in comments.
7448 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
7451 Automatic update of common submodule
7452 From e572c87 to f94d739
7454 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7458 * docs/libs/.gitignore:
7459 * examples/.gitignore:
7460 * gst/rtsp-server/.gitignore:
7463 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7465 * docs/libs/Makefile.am:
7466 docs: We don't build ps/pdf for API reference docs
7468 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7471 Automatic update of common submodule
7472 From ccbaa85 to e572c87
7474 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7477 Automatic update of common submodule
7478 From 46445ad to ccbaa85
7480 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7482 * gst/rtsp-server/Makefile.am:
7483 * gst/rtsp-server/fs-funnel.c:
7484 * gst/rtsp-server/fs-funnel.h:
7485 * gst/rtsp-server/rtsp-funnel.c:
7486 * gst/rtsp-server/rtsp-funnel.h:
7487 * gst/rtsp-server/rtsp-media.c:
7488 funnel: rename fsfunnel to rtspfunnel
7489 Rename the funnel to avoid conflicts with the farsight one.
7491 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7493 * gst/rtsp-server/Makefile.am:
7494 * gst/rtsp-server/fs-funnel.c:
7495 * gst/rtsp-server/fs-funnel.h:
7496 * gst/rtsp-server/rtsp-media.c:
7497 rtsp-media: add and use fsfunnel
7498 Add a copy of fsfunnel to the build because input-selector removed the (broken)
7499 select-all property that we need.
7501 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7503 * gst/rtsp-server/Makefile.am:
7504 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
7505 Use PKG_CONFIG_PATH specified at configure time (if any) as well
7506 for the g-ir-compiler, rather than just assuming the env var has
7509 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7516 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
7518 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7521 * gst/rtsp-server/Makefile.am:
7522 gobject-introspection: fix g-i build for uninstalled setup
7523 Requires gst-plugins-base git (> 0.10.31.2).
7525 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7527 * examples/test-uri.c:
7528 examples: add some more options and comments
7530 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7532 * gst/rtsp-server/rtsp-media-factory-uri.c:
7533 factory-uri: use right property type
7535 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7537 * gst/rtsp-server/rtsp-media-factory-uri.c:
7538 factory-uri: attempt to configure buffer-lists
7539 Attempt to configure buffer lists in the payloader for improved performance.
7541 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7543 * gst/rtsp-server/rtsp-media.c:
7544 media: attempt to configure bigger UDP buffers
7545 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
7546 send buffers with high bitrate streams.
7548 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
7550 * gst/rtsp-server/rtsp-client.c:
7551 client: use the socket length from getsockname
7552 Use the length returned by getsockname to perform the getnameinfo call because
7553 the size can depend on the socket type and platform.
7556 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7558 * docs/libs/gst-rtsp-server-docs.sgml:
7559 * docs/libs/gst-rtsp-server-sections.txt:
7560 docs: add uri factory to the docs
7562 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7564 * gst/rtsp-server/rtsp-client.c:
7565 * gst/rtsp-server/rtsp-media.h:
7568 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7570 * gst/rtsp-server/rtsp-client.c:
7571 * gst/rtsp-server/rtsp-media.c:
7572 * gst/rtsp-server/rtsp-media.h:
7573 * gst/rtsp-server/rtsp-session.c:
7574 * gst/rtsp-server/rtsp-session.h:
7575 rtsp-server: add support for buffer lists
7576 Add support for sending bufferlists received from appsink.
7579 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7581 * gst/rtsp-server/rtsp-client.c:
7582 * gst/rtsp-server/rtsp-media.c:
7583 * gst/rtsp-server/rtsp-media.h:
7584 * gst/rtsp-server/rtsp-sdp.c:
7585 media: make method to retrieve the play range
7586 Make a method to retrieve the playback range so that we can conditionally create
7587 a different range for the SDP and the PLAY requests.
7589 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7591 * gst/rtsp-server/rtsp-media.c:
7592 * gst/rtsp-server/rtsp-media.h:
7593 media: add signal to notify of state changes
7595 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7597 * gst/rtsp-server/rtsp-client.h:
7598 client: cleanup headers
7600 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7602 * gst/rtsp-server/rtsp-client.c:
7605 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7607 * gst/rtsp-server/rtsp-media-factory-uri.c:
7608 * gst/rtsp-server/rtsp-media-factory-uri.h:
7609 factory-uri: add support for gstpay
7610 Add an option to prefer gstpay over decoder + raw payloader.
7612 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7614 * gst/rtsp-server/rtsp-media-factory-uri.c:
7615 * gst/rtsp-server/rtsp-media-factory-uri.h:
7616 factory-uri: rework the autoplugger.
7617 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
7620 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7622 * gst/rtsp-server/rtsp-media-factory-uri.c:
7623 factory-uri: use better factory filter
7624 Make better payloader filter based on autoplug rank and RTP use case.
7626 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7629 Automatic update of common submodule
7630 From 169462a to 46445ad
7632 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7634 * gst/rtsp-server/rtsp-server.c:
7635 server: set SO_REUSEADDR before bind
7636 Set the SO_REUSEADDR _before_ bind() to make it actually work.
7638 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7640 * gst/rtsp-server/rtsp-media.c:
7641 * gst/rtsp-server/rtsp-media.h:
7642 media: emit prepared signal when prepared
7643 Make a 'prepared' signal and emit it when we successfully prepared the element.
7644 This signal can be used to configure the media object after it has been prepared
7647 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
7650 Automatic update of common submodule
7651 From 011bcc8 to 169462a
7653 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
7655 python an optional dependency
7656 * configure.ac: Move up valgrind and g-i checks. Make the python
7657 dependency optional, as it was before.
7659 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7661 Merge branch 'master' into 0.11
7666 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7668 * gst/rtsp-server/rtsp-media.c:
7669 media: update range when active clients changed
7670 When we changed the number of active clients, update the current range
7671 information because we want the second client connecting to a shared resource
7672 continue from where the stream currently.
7674 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7676 * gst/rtsp-server/rtsp-media-factory-uri.c:
7677 * gst/rtsp-server/rtsp-media-factory-uri.h:
7678 factory-uri: add colorspace and fix pt
7679 Rework the way we pass data to the autoplugger.
7680 When we have raw caps, plug a converter element to make pluggin to raw
7681 payloaders more successful.
7682 Make sure all dynamically plugged payloaders have a unique payload types.
7684 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7686 * examples/Makefile.am:
7687 * examples/test-uri.c:
7688 example: add example of the uri factory
7690 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7692 * gst/rtsp-server/Makefile.am:
7693 * gst/rtsp-server/rtsp-media-factory-uri.c:
7694 * gst/rtsp-server/rtsp-media-factory-uri.h:
7695 * gst/rtsp-server/rtsp-server.h:
7696 factory-uri: add a factory to stream any URI
7697 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
7700 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7702 * gst/rtsp-server/rtsp-media.c:
7703 * gst/rtsp-server/rtsp-media.h:
7704 media: ignore spurious ASYNC_DONE messages
7705 When we are dynamically adding pads, the addition of the udpsrc elements will
7706 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
7707 the real ASYNC_DONE when everything is prerolled.
7709 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7711 * gst/rtsp-server/rtsp-media-factory.c:
7712 * gst/rtsp-server/rtsp-media-factory.h:
7713 media-factory: make lock macro
7715 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
7717 * gst/rtsp-server/rtsp-client.c:
7718 rtsp-server: Remove unused variable and dead assignment
7720 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
7722 * examples/test-launch.c:
7723 * examples/test-mp4.c:
7724 * examples/test-ogg.c:
7725 * examples/test-readme.c:
7726 * examples/test-sdp.c:
7727 * examples/test-video.c:
7728 examples: Run gst-indent
7730 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
7732 * gst/rtsp-server/rtsp-client.c:
7733 * gst/rtsp-server/rtsp-media-factory.c:
7734 * gst/rtsp-server/rtsp-media-mapping.c:
7735 * gst/rtsp-server/rtsp-media.c:
7736 * gst/rtsp-server/rtsp-params.c:
7737 * gst/rtsp-server/rtsp-sdp.c:
7738 * gst/rtsp-server/rtsp-server.c:
7739 * gst/rtsp-server/rtsp-session-pool.c:
7740 * gst/rtsp-server/rtsp-session.c:
7741 rtsp-server: Run gst-indent
7742 Since it wasn't using the upstream common previously, there was no
7743 indentation check before commiting.
7745 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
7747 * gst/rtsp-server/rtsp-media-mapping.h:
7748 * gst/rtsp-server/rtsp-media.c:
7749 * gst/rtsp-server/rtsp-media.h:
7750 * gst/rtsp-server/rtsp-sdp.c:
7751 * gst/rtsp-server/rtsp-session-pool.h:
7752 * gst/rtsp-server/rtsp-session.c:
7753 * gst/rtsp-server/rtsp-session.h:
7754 rtsp-server: Some more doc fixups
7756 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7759 Makefile: Add cruft-cleaning support
7761 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7766 * docs/libs/Makefile.am:
7767 * docs/libs/gst-rtsp-server-docs.sgml:
7768 * docs/libs/gst-rtsp-server-sections.txt:
7769 * docs/libs/gst-rtsp-server.types:
7770 * docs/version.entities.in:
7771 docs: Add gtk-doc build system
7773 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7775 * gst/rtsp-server/Makefile.am:
7776 Makefile.am: Use standard GIR make behaviour
7778 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7782 autogen/configure: Bring more in sync to standard gst module behaviour
7784 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7786 * gst/rtsp-server/rtsp-media.c:
7787 media: warn and fail when gstrtpbin is not found
7789 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7792 configure: open 0.11 branch
7794 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
7798 Add common submodule
7800 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
7803 * common/Makefile.am:
7804 * common/c-to-xml.py:
7806 * common/coverage/coverage-report-entry.pl:
7807 * common/coverage/coverage-report.pl:
7808 * common/coverage/coverage-report.xsl:
7809 * common/coverage/lcov.mak:
7810 * common/gettext.patch:
7811 * common/glib-gen.mak:
7812 * common/gst-autogen.sh:
7813 * common/gst-xmlinspect.py:
7815 * common/gstdoc-scangobj:
7816 * common/gtk-doc-plugins.mak:
7817 * common/gtk-doc.mak:
7818 * common/m4/.gitignore:
7819 * common/m4/Makefile.am:
7821 * common/m4/as-ac-expand.m4:
7822 * common/m4/as-auto-alt.m4:
7823 * common/m4/as-compiler-flag.m4:
7824 * common/m4/as-compiler.m4:
7825 * common/m4/as-docbook.m4:
7826 * common/m4/as-libtool-tags.m4:
7827 * common/m4/as-libtool.m4:
7828 * common/m4/as-python.m4:
7829 * common/m4/as-scrub-include.m4:
7830 * common/m4/as-version.m4:
7831 * common/m4/ax_create_stdint_h.m4:
7832 * common/m4/check.m4:
7833 * common/m4/glib-gettext.m4:
7834 * common/m4/gst-arch.m4:
7835 * common/m4/gst-args.m4:
7836 * common/m4/gst-check.m4:
7837 * common/m4/gst-debuginfo.m4:
7838 * common/m4/gst-default.m4:
7839 * common/m4/gst-doc.m4:
7840 * common/m4/gst-error.m4:
7841 * common/m4/gst-feature.m4:
7842 * common/m4/gst-function.m4:
7843 * common/m4/gst-gettext.m4:
7844 * common/m4/gst-glib2.m4:
7845 * common/m4/gst-libxml2.m4:
7846 * common/m4/gst-plugindir.m4:
7847 * common/m4/gst-valgrind.m4:
7848 * common/m4/gtk-doc.m4:
7849 * common/m4/introspection.m4:
7851 * common/mangle-tmpl.py:
7852 * common/plugins.xsl:
7854 * common/release.mak:
7855 * common/scangobj-merge.py:
7856 * common/upload.mak:
7857 common: Remove static version
7859 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
7861 * common/m4/introspection.m4:
7862 Update introspection.m4 to match usage
7864 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7868 Remove old stuff from the README
7870 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7875 === release 0.10.7 ===
7877 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7882 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7884 * examples/test-ogg.c:
7885 test-ogg: remove parsers
7886 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
7887 buffers with timestamps. Using the parsers also seems to break things.
7889 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7891 * bindings/vala/gst-rtsp-server-0.10.vapi:
7892 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7893 Updated Vala bindings
7895 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7897 * common/m4/introspection.m4:
7899 * gst/rtsp-server/Makefile.am:
7900 Added initial gobject-introspection support
7902 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7904 * gst/rtsp-server/rtsp-media-factory.c:
7905 media-factory: don't use host for shared hash key
7906 When we generate the key to share made between connections, don't include the
7907 host used to connect so that we can share media even if between clients that
7908 connected with localhost and ones with the ip address.
7910 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7912 * bindings/vala/Makefile.am:
7913 build: fix distcheck
7915 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7917 * bindings/vala/gst-rtsp-server-0.10.vapi:
7918 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7919 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7920 Update Vala bindings
7922 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7924 * bindings/vala/Makefile.am:
7926 Fix configure checks and installation location for Vala bindings
7929 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7934 === release 0.10.6 ===
7936 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7939 configure: release 0.10.6
7941 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7943 * gst/rtsp-server/rtsp-media.c:
7944 media: help the compiler a little
7946 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7948 * gst/rtsp-server/rtsp-media.c:
7949 * gst/rtsp-server/rtsp-media.h:
7950 * gst/rtsp-server/rtsp-session.c:
7951 media: cleanup media transport before freeing
7952 Cleanup the media transport data before freeing. In particular, remove the qdata
7953 from the rtpsource object.
7955 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7957 * gst/rtsp-server/rtsp-media-factory.c:
7958 * gst/rtsp-server/rtsp-media-factory.h:
7959 * gst/rtsp-server/rtsp-media.c:
7960 * gst/rtsp-server/rtsp-media.h:
7961 media-factory: add eos-shutdown property
7962 Add an eos-shutdown property that will send an EOS to the pipeline before
7963 shutting it down. This allows for nice cleanup in case of a muxer.
7966 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7968 * gst/rtsp-server/rtsp-media.c:
7969 * gst/rtsp-server/rtsp-media.h:
7970 media: use multiudpsink send-duplicates when we can
7971 If we have a new enough multiudpsink with the send-duplicates property, use this
7972 instead of doing our own filtering. Our custom filtering code should eventually
7973 be removed when we can depend on a released -good.
7975 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7977 * gst/rtsp-server/rtsp-media.c:
7978 media: don't leak destinations
7979 Refactor and cleanup the destinations array when the stream is destroyed.
7981 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7983 * gst/rtsp-server/rtsp-media.c:
7984 * gst/rtsp-server/rtsp-media.h:
7985 media: don't add udp addresses multiple times
7986 Keep track of the udp addresses we added to udpsink and never add the same udp
7987 destination twice. This avoids duplicate packets when using multicast.
7989 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7991 * gst/rtsp-server/rtsp-server.c:
7992 server: disable use of SO_LINGER
7993 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
7994 server close()s the connection.
7996 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7998 * gst/rtsp-server/rtsp-server.c:
7999 server: use 5 second linger period in SO_LINGER
8000 Wait 5 seconds before clearing the send buffers and reseting the connection with
8001 the client when we do a close. This should be enough time to get the message to
8005 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
8007 * gst/rtsp-server/rtsp-server.c:
8008 server: use SO_LINGER
8009 SO_LINGER on the socket will make sure that any pending data on the socket is
8010 flushed ASAP and that the socket connection is reset. This makes sure that the
8011 socket can be reused immediately.
8014 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8017 README: add blurb about shared media factories
8019 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
8021 * gst/rtsp-server/rtsp-media.c:
8022 Add stdlib.h for atoi()
8024 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8026 * bindings/python/Makefile.am:
8027 * bindings/vala/Makefile.am:
8028 build: distcheck fixes
8029 Fix 'make distcheck', somewhat (it still fails because it tries to
8030 install files into /usr/share/vala/vapi/ irrespective of the
8033 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8036 configure: bump core/base requirements to released version
8037 Makes things less confusing for people.
8039 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8042 configure: fail if GStreamer core/base requirements are not met
8044 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8046 * gst/rtsp-server/rtsp-client.c:
8047 client: improve client cleanups
8048 Make sure the session does not timeout when using TCP. We need to do this
8049 because quicktime player does not send RTCP for some reason in tunneled
8051 Refactor some cleanup code.
8054 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8056 * gst/rtsp-server/rtsp-session.c:
8057 * gst/rtsp-server/rtsp-session.h:
8058 session: add support for prevent session timeouts
8059 Add an atomix counter to prevent session timeouts when we are, for example,
8062 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8064 * gst/rtsp-server/rtsp-client.c:
8065 client: fix unlink on session timeouts
8066 When our session times out, make sure we unlink all streams in this
8068 Remove the tunnelid when closing the connection.
8070 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8072 * gst/rtsp-server/rtsp-session.c:
8073 session: small cleanups
8075 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8077 * gst/rtsp-server/rtsp-client.c:
8078 client: handle lost_tunnel callbacks
8079 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
8080 hashtable so that we can reuse it for when the client reopens the POST
8082 Close the connection after a TEARDOWN.
8083 Make sure or watchid is cleared when the watch is removed.
8086 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8088 * gst/rtsp-server/rtsp-client.c:
8089 * gst/rtsp-server/rtsp-media.c:
8090 * gst/rtsp-server/rtsp-sdp.c:
8091 rtsp-server: add more support for multicast
8093 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8096 * gst/rtsp-server/rtsp-media.c:
8097 * gst/rtsp-server/rtsp-media.h:
8098 media: allow configuration of allowed lower transport
8100 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8102 * gst/rtsp-server/rtsp-client.h:
8103 * gst/rtsp-server/rtsp-media.c:
8104 * gst/rtsp-server/rtsp-media.h:
8105 * gst/rtsp-server/rtsp-sdp.c:
8106 * gst/rtsp-server/rtsp-sdp.h:
8107 * gst/rtsp-server/rtsp-server.c:
8108 rtsp: keep track of server ip and ipv6
8109 Keep track of how the client connected to the server and setup the udp ports
8110 with the same protocol.
8111 Copy the server ip address in the SDP so that clients can send RTCP back to
8114 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8116 * gst/rtsp-server/rtsp-session.c:
8119 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8121 * gst/rtsp-server/rtsp-client.c:
8122 client: use right size for malloc
8124 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8126 * gst/rtsp-server/rtsp-server.c:
8127 server: comment ipv6 server listening address
8129 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8131 * gst/rtsp-server/rtsp-media.c:
8132 media: allow for ipv6 sockets
8134 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8136 * gst/rtsp-server/rtsp-server.c:
8137 * gst/rtsp-server/rtsp-server.h:
8138 server: rework server part
8139 Allow setting a bind address, make sure we can deal with ipv6.
8140 Remove the port property and change with the service property.
8142 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8144 * gst/rtsp-server/rtsp-media.h:
8145 media: update comments a little
8147 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8149 * gst/rtsp-server/rtsp-client.c:
8150 client: make content-base better
8151 Use the URI formatting functions to make a content-base. Also make sure that
8152 there is a trailing / at the end.
8154 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8156 * gst/rtsp-server/rtsp-client.c:
8157 client: guard against invalid paths
8159 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8161 * examples/test-video.c:
8162 test: catch server bind errors
8164 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
8166 * gst/rtsp-server/rtsp-media.c:
8167 rtspmedia: emit "unprepared" if _prepare fails.
8168 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
8169 media object is removed from its factory's cache.
8171 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8173 * gst/rtsp-server/rtsp-media.c:
8174 media: collect media position when seek completes
8176 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
8178 * gst/rtsp-server/rtsp-client.c:
8179 client: call unlink_streams in client finalize
8182 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8184 * gst/rtsp-server/rtsp-media.c:
8185 media: limit the time to wait to something huge
8186 Avoid waiting forever but limit the timeout to 20 seconds.
8188 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8190 * gst/rtsp-server/rtsp-sdp.c:
8191 sdp: reindent and check for prepared status
8193 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8195 * gst/rtsp-server/rtsp-media.c:
8196 * gst/rtsp-server/rtsp-media.h:
8197 * gst/rtsp-server/rtsp-session.c:
8198 media: avoid doing _get_state() for state changes
8199 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
8200 until the media is prerolled or in error. This avoids doing a blocking call of
8201 gst_element_get_state() that can cause lockups when there is an error.
8204 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8206 * gst/rtsp-server/rtsp-media.c:
8209 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8211 * gst/rtsp-server/rtsp-media-factory.c:
8212 media-factory: better error handling
8213 Improve the error handling a bit.
8215 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8217 * gst/rtsp-server/rtsp-client.c:
8218 client: rework transport parsing
8219 Rework the transport parsing code so that we can ignore transports we don't
8220 support instead of just picking the first one we can parse.
8221 Configure a (for now hardcoded) destination for multicast transports.
8223 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8225 * gst/rtsp-server/rtsp-media.c:
8226 media: set multicast sink parameters
8227 Disable loop and automatic multicast join on the udpsink elements.
8228 Add some more debug info.
8229 Reset some state variables in the right place.
8230 Use the right port numbers for multicast.
8232 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8234 * gst/rtsp-server/rtsp-session.c:
8235 session: handle transport setup correctly
8236 Handle UDP, MCAST and TCP transport negotiation more correctly.
8237 Store the server session SSRC in the transport.
8239 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8241 * gst/rtsp-server/rtsp-client.c:
8242 rtsp-client: implement error_full
8243 Implement error_full to avoid some segfaults when the rtspconnection calls it.
8246 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8249 * gst/rtsp-server/rtsp-client.c:
8250 * gst/rtsp-server/rtsp-server.c:
8251 docs: update docs and comments
8253 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
8255 * gst/rtsp-server/rtsp-sdp.c:
8256 sdp: make server work better when behind a proxy
8258 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8260 * gst/rtsp-server/rtsp-client.c:
8261 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
8263 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8265 * gst/rtsp-server/rtsp-client.c:
8266 * gst/rtsp-server/rtsp-media-factory.c:
8267 * gst/rtsp-server/rtsp-media-mapping.c:
8268 * gst/rtsp-server/rtsp-media.c:
8269 * gst/rtsp-server/rtsp-server.c:
8270 * gst/rtsp-server/rtsp-session-pool.c:
8271 * gst/rtsp-server/rtsp-session.c:
8272 Use GStreamer's debugging subsystem
8274 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8276 * gst/rtsp-server/rtsp-media-factory.c:
8277 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
8279 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8284 === release 0.10.5 ===
8286 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8291 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8294 configure: bump required versions
8296 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
8298 * gst/rtsp-server/rtsp-client.c:
8299 client: call weak-unref on client->sessions from finalize
8302 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8304 * gst/rtsp-server/rtsp-media.c:
8305 media: Fixed crasher where caps got unref'ed too often
8307 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8310 * pkgconfig/.gitignore:
8311 * pkgconfig/Makefile.am:
8312 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
8313 Added pkg-config file to use gst-rtsp-server uninstalled
8315 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8317 * gst/rtsp-server/rtsp-media.c:
8318 media: add some docs
8320 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
8322 * gst/rtsp-server/rtsp-client.c:
8323 rtsp: Use gst_rtsp_watch_send_message().
8324 Use gst_rtsp_watch_send_message() since the old API which used
8325 gst_rtsp_watch_queue_message() has been deprecated.
8327 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8332 === release 0.10.4 ===
8334 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8339 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8341 * gst/rtsp-server/rtsp-client.c:
8342 * gst/rtsp-server/rtsp-session.c:
8343 * gst/rtsp-server/rtsp-session.h:
8344 rtsp: allocate channels in TCP mode
8345 When the client does not provide us with channels in TCP mode, allocate channels
8348 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8350 * gst/rtsp-server/rtsp-client.c:
8351 client: don't crash when tunnelid is missing
8352 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
8353 don't crash but return an error response to the client.
8356 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8358 * bindings/vala/gst-rtsp-server-0.10.vapi:
8359 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8360 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8361 bindings: update vala bindings with new method
8363 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8365 * gst/rtsp-server/rtsp-session-pool.c:
8366 * gst/rtsp-server/rtsp-session-pool.h:
8367 sessionpool: add function to filter sessions
8368 Add generic function to retrieve/remove sessions.
8370 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8373 configure: bump core/base requirements to release
8375 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8377 * gst/rtsp-server/rtsp-media.c:
8378 media: fix indentation
8380 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8382 * gst/rtsp-server/rtsp-media.c:
8383 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
8385 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8387 * gst/rtsp-server/rtsp-media.c:
8388 set state and remove elements of media in for loop
8390 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
8392 * bindings/vala/gst-rtsp-server-0.10.vapi:
8393 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8394 Added gst_rtsp_media_remove_elements function to Vala bindings
8396 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
8398 * gst/rtsp-server/rtsp-media.c:
8399 * gst/rtsp-server/rtsp-media.h:
8400 Added gst_rtsp_media_remove_elements function
8402 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
8404 * gst/rtsp-server/rtsp-media.c:
8405 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
8407 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8409 * bindings/vala/gst-rtsp-server-0.10.vapi:
8410 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8411 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8412 Updated Vala bindings
8414 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8416 * gst/rtsp-server/rtsp-media.c:
8417 * gst/rtsp-server/rtsp-media.h:
8418 Added vmethod unprepare to GstRTSPMedia
8419 The default implementation sets the state of the pipeline to GST_STATE_NULL
8421 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8423 * gst/rtsp-server/rtsp-media-factory.c:
8424 * gst/rtsp-server/rtsp-media-factory.h:
8425 Made collect_streams function public
8427 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8429 * gst/rtsp-server/rtsp-media-factory.c:
8430 * gst/rtsp-server/rtsp-media-factory.h:
8431 * gst/rtsp-server/rtsp-media.c:
8432 Added vmethod create_pipeline to GstRTSPMediaFactory
8433 The pipeline is created in this method and the GstRTSPMedia's element is added to it
8435 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8437 * gst/rtsp-server/rtsp-client.c:
8438 client: use g_source_destroy()
8439 We need to use g_source_destroy() because we might have added the source to a
8440 different main context than the default one.
8442 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8444 * gst/rtsp-server/Makefile.am:
8445 * gst/rtsp-server/rtsp-client.c:
8446 * gst/rtsp-server/rtsp-params.c:
8447 * gst/rtsp-server/rtsp-params.h:
8448 rtsp: prepare for handling GET/SET_PARAMETER
8449 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
8451 Fix return codes of handlers.
8453 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8455 * gst/rtsp-server/rtsp-media.c:
8456 media: don't leak session pads
8458 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8460 * gst/rtsp-server/rtsp-media.c:
8461 media: clean up the messages a bit
8463 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8465 * gst/rtsp-server/rtsp-sdp.c:
8466 sdp: warn and skip streams without media
8468 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8470 * bindings/vala/gst-rtsp-server-0.10.vapi:
8471 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8472 vala: Fixed typo in header file of RTSPMediaStream
8474 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8476 * gst/rtsp-server/rtsp-media.c:
8479 Make dumping RTCP stats configurable
8481 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8483 * gst/rtsp-server/rtsp-media.c:
8484 media: be less verbose and leak less
8486 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8488 * gst/rtsp-server/rtsp-media.c:
8489 media: don't leak the destination address
8491 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8493 * gst/rtsp-server/rtsp-client.c:
8494 * gst/rtsp-server/rtsp-media.c:
8495 * gst/rtsp-server/rtsp-media.h:
8496 * gst/rtsp-server/rtsp-session.c:
8497 * gst/rtsp-server/rtsp-session.h:
8498 rtsp: use RTCP to keep the session alive
8499 Use the RTCP rtcp-from stats field to find the associated session and use this
8500 to keep the session alive.
8502 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8504 * gst/rtsp-server/rtsp-session.c:
8505 session: add 5sec to the real session timeout
8506 Allow the session to live 5sec longer before really timing out. This should give
8507 clients some extra time to keep the session active.
8509 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8511 * gst/rtsp-server/rtsp-client.c:
8512 client: replay OK to GET/SET_PARAMETER
8513 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
8514 so that we return OK for those requests.
8516 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8518 * gst/rtsp-server/rtsp-media.c:
8519 * gst/rtsp-server/rtsp-media.h:
8520 media: keep track of active transports
8521 Keep track of which transport is active to avoid closing the connection too
8523 Remove the destination transport also when going to NULL.
8524 Print some stats about the SDES and other RTCP messages we receive from the
8527 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8529 * examples/.gitignore:
8530 * examples/Makefile.am:
8531 * examples/test-sdp.c:
8532 example: add SDP relay example
8534 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8536 * gst/rtsp-server/rtsp-media.c:
8537 media: also count active TCP connections
8539 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8541 * gst/rtsp-server/rtsp-media-factory.c:
8542 * gst/rtsp-server/rtsp-media.c:
8543 * gst/rtsp-server/rtsp-media.h:
8544 rtsp: add support for dynamic elements
8545 Add support for dynamic elements.
8546 Don't set live pipelines back to paused.
8548 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8550 * gst/rtsp-server/rtsp-sdp.c:
8551 sdp: don't add encoding name when absent in caps
8553 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8555 * gst/rtsp-server/rtsp-client.c:
8556 client: warn when we can't do RTP-Info
8558 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8560 * gst/rtsp-server/rtsp-media-factory.c:
8561 factory: factor out the stream construction
8563 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8565 * gst/rtsp-server/rtsp-client.c:
8566 client: only add RTP-Info when we have the info
8567 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
8570 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8575 === release 0.10.3 ===
8577 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8581 - Fixes a bug where it put the wrong verion in pkgconfig
8582 - Link RTP and RTCP sources
8584 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8586 * gst/rtsp-server/rtsp-media.c:
8587 * gst/rtsp-server/rtsp-media.h:
8588 media: link the RTP udpsrc to the session manager
8589 Link the RTP udpsrc and the appsrc to the session manager so that they don't
8590 shut down when the client sends a packet to open firewalls.
8592 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8594 * pkgconfig/gst-rtsp-server.pc.in:
8595 Don't use hard-coded version number in pkg-config file
8597 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8602 === release 0.10.2 ===
8604 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8609 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8612 * common/m4/.gitignore:
8613 * examples/.gitignore:
8614 * pkgconfig/.gitignore:
8615 add some .gitignore files
8617 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8619 * gst/rtsp-server/rtsp-media.c:
8620 media: seek to key frames
8622 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8624 * gst/rtsp-server/rtsp-media.c:
8625 media: emit the unprepared signal by id
8626 Emit the unprepared signal by id instead of name and set the media as
8629 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8631 * gst/rtsp-server/rtsp-media.c:
8632 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
8634 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8636 * gst/rtsp-server/rtsp-server.c:
8637 Added finalize function to GstRTPSPServer to unref session pool and media mapping
8639 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8641 * bindings/vala/gst-rtsp-server-0.10.vapi:
8642 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8643 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8644 Updated vala bindings
8646 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8648 * gst/rtsp-server/Makefile.am:
8649 * gst/rtsp-server/rtsp-client.c:
8650 * gst/rtsp-server/rtsp-media.c:
8651 server: use appsink and appsrc with the API
8652 Use the appsink/appsrc API instead of the signals for higher
8655 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8657 * examples/test-ogg.c:
8658 tests: set the payload type correctly
8660 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8662 * gst/rtsp-server/rtsp-media-factory.c:
8663 factory: connect to the unprepare signal
8664 Connect to the unprepare signal for non-reusable media so that we can remove
8665 them from the cache.
8667 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8669 * gst/rtsp-server/rtsp-media.c:
8670 * gst/rtsp-server/rtsp-media.h:
8671 media: add signal to notify of unprepare
8673 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8675 * gst/rtsp-server/rtsp-media.c:
8676 * gst/rtsp-server/rtsp-media.h:
8677 media: more work on making the media shared
8678 Add a reusable flag to medias, indicating that they can be reused after a state
8682 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8684 * examples/test-readme.c:
8685 examples: mark the example as shared for testing
8687 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8689 * gst/rtsp-server/rtsp-media.c:
8690 * gst/rtsp-server/rtsp-media.h:
8691 client: support shared media
8692 Always perform the state actions even if the target state of the pipeline is
8693 already correct, we still want to add/remove the transports when we are dealing
8695 Keep a counter of the number of active transports for a media so that we can use
8696 this to perform a state change when needed.
8697 Perform a state change of the pipeline only when the first transport was added
8698 or when there are no active transports.
8700 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8702 * gst/rtsp-server/rtsp-client.c:
8703 client: fix refcounting crasher
8704 Don't need to remove the weak refs in the finalize methods, they are already
8705 removed in the dispose.
8706 Don't register the callback with a DestroyNofity.
8708 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8710 * gst/rtsp-server/rtsp-client.c:
8711 Fix rtsp client refcount management in TCP mode.
8712 Don't unref a client ref we never had. Fixes an unref
8713 of an already-free client object after a client
8714 teardown request for me.
8716 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8718 * gst/rtsp-server/rtsp-session.c:
8719 docs: fix typo in API docs
8721 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8723 * gst/rtsp-server/rtsp-media.c:
8725 Keep the udp sources in playing even if we go to paused. unlock the sources when
8727 Add some more debug info.
8728 Only seek when we need to.
8729 Keep track of the position when we go to paused.
8731 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8733 * gst/rtsp-server/rtsp-client.c:
8734 * gst/rtsp-server/rtsp-media.c:
8735 * gst/rtsp-server/rtsp-media.h:
8736 Add beginnings of seeking.
8737 Parse the Range header and perform a seek on the pipeline for the requested
8738 position. It's disabled currently until I figure out what's going wrong.
8740 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8742 * gst/rtsp-server/rtsp-client.c:
8743 allow pause requests for now.
8746 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8748 * gst/rtsp-server/rtsp-client.c:
8749 Remove weak ref on the session in teardown
8750 We need to remove our weakref from the session when we do a teardown because
8751 else we close the TCP connection prematurely.
8753 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8755 * gst/rtsp-server/rtsp-client.c:
8756 * gst/rtsp-server/rtsp-client.h:
8757 * gst/rtsp-server/rtsp-session-pool.c:
8758 Do some more session cleanup
8759 Make session timeout kill the TCP connection that currently watches the
8761 Remove the client timeout property.
8763 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8765 * gst/rtsp-server/rtsp-client.c:
8766 * gst/rtsp-server/rtsp-client.h:
8767 * gst/rtsp-server/rtsp-media.c:
8768 * gst/rtsp-server/rtsp-media.h:
8769 * gst/rtsp-server/rtsp-server.c:
8770 * gst/rtsp-server/rtsp-session.c:
8771 * gst/rtsp-server/rtsp-session.h:
8773 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
8776 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8778 * examples/Makefile.am:
8779 * examples/test-launch.c:
8780 Add example server that takes launch lines
8781 Add an example server that streams any -launch line.
8783 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8785 * examples/test-readme.c:
8786 * gst/rtsp-server/rtsp-client.c:
8787 * gst/rtsp-server/rtsp-media.c:
8788 * gst/rtsp-server/rtsp-media.h:
8789 Add support for live streams
8790 Add support for live streams and ranges
8791 Start on handling TCP data transfer.
8793 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8795 * gst/rtsp-server/rtsp-media.c:
8796 Free the pipeline before other things
8799 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8801 * gst/rtsp-server/rtsp-client.c:
8802 Only free the pending tunnel if there is one
8805 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8807 * gst/rtsp-server/rtsp-client.c:
8808 * gst/rtsp-server/rtsp-client.h:
8809 * gst/rtsp-server/rtsp-media.c:
8810 rtsp-server: Add support for tunneling
8811 Add support for tunneling over HTTP.
8812 Use new connection methods to retrieve the url.
8813 Dispatch messages based on the message type instead of blindly
8814 assuming it's always a request.
8815 Keep track of the watch id so that we can remove it later.
8816 Set the media pipeline to NULL before unreffing the pipeline.
8818 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8820 * gst/rtsp-server/rtsp-client.c:
8821 * gst/rtsp-server/rtsp-client.h:
8822 Fix for channel -> watch rename in gstreamer
8823 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
8825 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8827 * gst/rtsp-server/rtsp-client.c:
8828 * gst/rtsp-server/rtsp-client.h:
8830 Use the async RTSP channels instead of spawning a new thread for each client.
8831 If a sessionid is specified in a request, fail if we don't have the session.
8833 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8835 * gst/rtsp-server/rtsp-media.c:
8836 Add better debug info
8837 Add some better debug info.
8839 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8841 * examples/test-video.c:
8843 Add support for session timeouts in the example.
8845 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8847 * gst/rtsp-server/rtsp-session-pool.c:
8848 * gst/rtsp-server/rtsp-session-pool.h:
8849 Pass GTimeVal around for performance reasons
8850 Get the current time only once and pass it around so that sessions don't have to
8851 get the current time anymore.
8852 Add experimental support for a GSource that dispatches when the session needs to
8855 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8857 * gst/rtsp-server/rtsp-session.c:
8858 * gst/rtsp-server/rtsp-session.h:
8859 Add better support for session timeouts
8860 Add a method to request the number of milliseconds when a session will timeout.
8862 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8864 * gst/rtsp-server/rtsp-media.c:
8865 * gst/rtsp-server/rtsp-media.h:
8866 Add suport for RTP manager monitoring
8867 Add the first stage in monitoring the rtp manager.
8868 Make sure we don't update the state to something we don't want.
8870 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8872 * gst/rtsp-server/rtsp-client.c:
8873 Add support for session keepalive
8874 Get and update the session timeout for all requests. get the session as early as
8877 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8879 * gst/rtsp-server/rtsp-media-factory.h:
8880 * gst/rtsp-server/rtsp-media.c:
8881 * gst/rtsp-server/rtsp-media.h:
8882 Handle media bus messages
8883 Handle media bus messages in a custom mainloop and dispatch them to the
8884 RTSPMedia objects. Let the default implementation handle some common messages.
8886 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8888 * gst/rtsp-server/rtsp-client.c:
8889 * gst/rtsp-server/rtsp-session-pool.c:
8890 * gst/rtsp-server/rtsp-session.c:
8891 Some more session timeout handling
8892 Move the session header setting code to a central place so that we always add
8893 the timeout parameter too.
8894 Handle timeouts by running the session cleanup code.
8895 Stop media before cleaning up.
8897 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8899 * gst/rtsp-server/rtsp-client.c:
8900 * gst/rtsp-server/rtsp-client.h:
8901 Add timeout property
8902 Add a timeout property ot the client and make the other properties into GObject
8905 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8907 * gst/rtsp-server/rtsp-session-pool.c:
8908 Use getters and setters in property code
8909 Use the getters and setters for the timeout property instead of locking
8912 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8914 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
8916 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8918 * gst/rtsp-server/rtsp-session-pool.c:
8919 * gst/rtsp-server/rtsp-session-pool.h:
8920 * gst/rtsp-server/rtsp-session.c:
8921 * gst/rtsp-server/rtsp-session.h:
8922 Add more timeout stuff
8923 Add method to check if a session is expired.
8924 Add method to perform cleanup on a session pool.
8926 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8928 * gst/rtsp-server/rtsp-client.c:
8929 * gst/rtsp-server/rtsp-session-pool.c:
8930 * gst/rtsp-server/rtsp-session-pool.h:
8931 * gst/rtsp-server/rtsp-session.c:
8932 * gst/rtsp-server/rtsp-session.h:
8933 Add beginnings of session timeouts and limits
8934 Add the timeout value to the Session header for unusual timeout values.
8935 Allow us to configure a limit to the amount of active sessions in a pool. Set a
8936 limit on the amount of retry we do after a sessionid collision.
8937 Add properties to the sessionid and the timeout of a session. Keep track of
8938 creation time and last access time for sessions.
8940 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8942 * gst/rtsp-server/rtsp-client.c:
8943 * gst/rtsp-server/rtsp-media.c:
8944 * gst/rtsp-server/rtsp-media.h:
8945 * gst/rtsp-server/rtsp-sdp.c:
8946 * gst/rtsp-server/rtsp-session-pool.c:
8947 * gst/rtsp-server/rtsp-session.c:
8948 * gst/rtsp-server/rtsp-session.h:
8949 Cleanup of sessions and more
8950 Fix the refcounting of media and sessions in the client. Properly clean up the
8951 session data when the client performs a teardown.
8952 Add Server header to responses.
8953 Allow for multiple uri setups in one session.
8954 Add Range header to the PLAY response and add the range attribute to the SDP
8956 Fix the session pool remove method, it used the wrong key in the hashtable. Also
8957 give the ownership of the sessionid to the session object.
8959 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8961 * gst/rtsp-server/rtsp-server.c:
8962 * gst/rtsp-server/rtsp-server.h:
8964 Rename the 'server_port' variable to simply 'port'.
8966 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8969 * gst/rtsp-server/rtsp-client.c:
8970 * gst/rtsp-server/rtsp-media.c:
8971 * gst/rtsp-server/rtsp-media.h:
8972 * gst/rtsp-server/rtsp-session.c:
8973 * gst/rtsp-server/rtsp-session.h:
8974 Rework the way we handle transports for streams
8975 Make the media accept an array of transports for the streams that we have
8976 configured for the play/pause requests.
8977 Implement server states for a client and its media.
8978 Require 0.10.22.1 (git HEAD) of gstreamer.
8980 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8982 * gst/rtsp-server/rtsp-client.c:
8983 * gst/rtsp-server/rtsp-media-factory.c:
8984 Drop const from functions dealing with urls
8985 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
8986 have the right const in them.
8988 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8990 * gst/rtsp-server/rtsp-client.c:
8991 * gst/rtsp-server/rtsp-media.c:
8992 * gst/rtsp-server/rtsp-sdp.c:
8996 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8998 * gst/rtsp-server/rtsp-client.c:
8999 * gst/rtsp-server/rtsp-media-factory.c:
9000 * gst/rtsp-server/rtsp-media.c:
9001 * gst/rtsp-server/rtsp-media.h:
9003 Don't keep a reference to the GstRTSPMedia in the stream.
9004 Free more things when freeing the GstRTSPMedia.
9006 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9009 * gst/rtsp-server/rtsp-media-factory.c:
9010 * gst/rtsp-server/rtsp-media-factory.h:
9011 * gst/rtsp-server/rtsp-media.c:
9012 * gst/rtsp-server/rtsp-media.h:
9013 * gst/rtsp-server/rtsp-server.c:
9014 * gst/rtsp-server/rtsp-server.h:
9015 More docs and small cleanups
9016 Add some more docs and update the README
9017 Cleanup some method names.
9018 Remove an unneeded idx field in the GstRTSPMediaStream
9020 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9023 * examples/Makefile.am:
9024 * examples/test-readme.c:
9025 Add a README and more example code
9026 Add a README file that contains a small introduction on how to use the server
9027 along with the example code explained in the readme.
9029 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9031 * gst/rtsp-server/rtsp-media.c:
9032 * gst/rtsp-server/rtsp-server.c:
9033 Fix some leaks and change default port
9034 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
9035 we finished the initial preroll. If we keep them locked, setting the pipeline to
9036 NULL will not stop and clean up the sources correctly.
9037 Change the default RTSP port to 8554 aka the official alternative RTSP port.
9039 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9041 * gst/rtsp-server/rtsp-session.c:
9042 * gst/rtsp-server/rtsp-session.h:
9043 Cleanups to the session object
9044 Remove some unneeded variables in the session state of a stream such as the
9045 owner media and the server transport.
9046 Get the configuration of a media stream in a session based on the media_stream
9047 in the original object instead of our cached index.
9048 Free more data in the finalize method.
9050 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9052 * gst/rtsp-server/rtsp-client.c:
9053 * gst/rtsp-server/rtsp-client.h:
9054 Cleanups and reuse media from DESCRIBE
9055 Handle thread create errors.
9056 Rename some internal methods to better match what they actually do.
9057 Handle misconfiguration of session_pool and media_mapping gracefully.
9058 Cache the DESCRIBE media and uri in the client connection and reuse them when
9059 we receive a SETUP request in the same connection for the same uri.
9060 Cleanup the client connection object.
9062 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9064 * gst/rtsp-server/rtsp-media-factory.c:
9065 * gst/rtsp-server/rtsp-media-factory.h:
9066 * gst/rtsp-server/rtsp-media.c:
9067 * gst/rtsp-server/rtsp-media.h:
9068 Add shared properties to media and factory
9069 Add the shared property to media.
9070 Implement some simple caching in the factory depending on if the media is shared
9073 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9075 * gst/rtsp-server/rtsp-client.c:
9076 Add a little comment
9077 Add some comment about the content-base header.
9079 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9081 * examples/Makefile.am:
9083 * examples/test-mp4.c:
9084 * examples/test-ogg.c:
9085 * examples/test-video.c:
9086 * gst/rtsp-server/Makefile.am:
9087 * gst/rtsp-server/rtsp-client.c:
9088 * gst/rtsp-server/rtsp-client.h:
9089 * gst/rtsp-server/rtsp-media-factory.c:
9090 * gst/rtsp-server/rtsp-media-factory.h:
9091 * gst/rtsp-server/rtsp-media.c:
9092 * gst/rtsp-server/rtsp-media.h:
9093 * gst/rtsp-server/rtsp-sdp.c:
9094 * gst/rtsp-server/rtsp-sdp.h:
9095 * gst/rtsp-server/rtsp-server.c:
9096 * gst/rtsp-server/rtsp-server.h:
9097 * gst/rtsp-server/rtsp-session.c:
9098 * gst/rtsp-server/rtsp-session.h:
9099 Reorganize things, prepare for media sharing
9100 Added various other test server examples
9101 Move the SDP message generation to a separate helper.
9102 Refactor common code for finding the session.
9103 Add content-base for realplayer compatibility
9104 Clean up request uris before processing for better vlc compatibility.
9105 Move prerolling and pipeline construction to the RTSPMedia object.
9106 Use multiudpsink for future pipeline reuse.
9108 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9114 === release 0.10.1 ===
9116 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9122 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9124 * bindings/vala/Makefile.am:
9126 Add more directories and files to the dist.
9128 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9130 * bindings/python/Makefile.am:
9131 * bindings/python/rtspserver.override:
9132 Fixed compile error of python bindings
9134 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9136 * bindings/vala/gst-rtsp-server-0.10.vapi:
9137 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9138 Marked values as nullable accordingly
9140 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9142 * bindings/vala/gst-rtsp-server-0.10.vapi:
9143 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
9144 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9145 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9146 Updated Vala bindings
9148 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9150 * gst/rtsp-server/rtsp-client.c:
9151 * gst/rtsp-server/rtsp-media-mapping.c:
9152 * gst/rtsp-server/rtsp-media-mapping.h:
9153 * gst/rtsp-server/rtsp-media.h:
9154 * gst/rtsp-server/rtsp-session-pool.h:
9155 Cleanups and doc updates
9156 Add some more documentation and do some minor cleanups here and there.
9158 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9160 * gst/rtsp-server/rtsp-client.c:
9161 * gst/rtsp-server/rtsp-media-factory.c:
9162 * gst/rtsp-server/rtsp-media-factory.h:
9163 * gst/rtsp-server/rtsp-media.c:
9164 * gst/rtsp-server/rtsp-media.h:
9165 * gst/rtsp-server/rtsp-session.c:
9166 * gst/rtsp-server/rtsp-session.h:
9168 Rename GstRTSPMediaBin to GstRTSPMedia
9169 Parse the request url into a GstRTSPUri object and pass this object to the
9170 various handlers and methods that require the uri.
9172 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9176 Add some more docs and remove some old code from the example.
9178 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9180 * gst/rtsp-server/rtsp-client.c:
9181 Handle state change failures better
9182 Handle state change failures better when changing the state of the pipeline to
9185 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9187 * gst/rtsp-server/rtsp-media-factory.c:
9188 * gst/rtsp-server/rtsp-media-factory.h:
9189 Make element creation more extendible
9190 Add get_element vmethod to the default MediaFactory so that subclasses can just
9191 override that method and still use the default logic for making a MediaBin from
9194 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9197 * gst/rtsp-server/Makefile.am:
9198 * gst/rtsp-server/rtsp-client.c:
9199 * gst/rtsp-server/rtsp-client.h:
9200 * gst/rtsp-server/rtsp-media-factory.c:
9201 * gst/rtsp-server/rtsp-media-factory.h:
9202 * gst/rtsp-server/rtsp-media-mapping.c:
9203 * gst/rtsp-server/rtsp-media-mapping.h:
9204 * gst/rtsp-server/rtsp-media.c:
9205 * gst/rtsp-server/rtsp-media.h:
9206 * gst/rtsp-server/rtsp-server.c:
9207 * gst/rtsp-server/rtsp-server.h:
9208 * gst/rtsp-server/rtsp-session.c:
9209 * gst/rtsp-server/rtsp-session.h:
9210 Make the server handle arbitrary pipelines
9211 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
9212 The GstMediaBin object has a handle to a bin with elements and to a list of
9213 GstMediaStream objects that this bin produces.
9214 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
9215 with methods to register and remove those mappings.
9216 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
9217 used by the server instance.
9218 Modify the example application so that it shows how to create custom pipelines
9219 attached to a specific mount point.
9220 Various misc cleanps.
9222 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9224 * gst/rtsp-server/rtsp-server.c:
9225 * gst/rtsp-server/rtsp-server.h:
9226 Allow setting a custom media factory for a server
9228 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9230 * gst/rtsp-server/rtsp-client.c:
9231 * gst/rtsp-server/rtsp-client.h:
9232 Allow setting a custom media factory for a client.
9234 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9236 * gst/rtsp-server/Makefile.am:
9237 Add Makefile entry for the media factory
9239 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9241 * gst/rtsp-server/rtsp-media-factory.c:
9242 * gst/rtsp-server/rtsp-media-factory.h:
9243 Add media factory to map urls to media pipeline objects.
9245 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9247 * gst/rtsp-server/rtsp-media.c:
9248 * gst/rtsp-server/rtsp-media.h:
9249 Add comments. Remove unused field
9251 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9253 * gst/rtsp-server/rtsp-session-pool.c:
9254 * gst/rtsp-server/rtsp-session-pool.h:
9255 Allow custom session pools to override the session id allocation algorithms Add some comments.
9257 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9259 * gst/rtsp-server/rtsp-session.h:
9262 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9264 * gst/rtsp-server/rtsp-client.c:
9265 * gst/rtsp-server/rtsp-client.h:
9266 Move the connection code in one place Add some comments
9268 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9270 * gst/rtsp-server/rtsp-server.c:
9271 * gst/rtsp-server/rtsp-server.h:
9272 Make vmethod to create and accept new clients. Add some docs.
9274 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9276 * gst/rtsp-server/rtsp-server.c:
9277 * gst/rtsp-server/rtsp-server.h:
9278 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
9280 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9282 * gst/rtsp-server/rtsp-client.c:
9283 * gst/rtsp-server/rtsp-client.h:
9284 Name the parameters more appropriately.
9286 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9288 * gst/rtsp-server/rtsp-session-pool.c:
9289 Do some more cleanup of the session pool.
9291 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9293 * gst/rtsp-server/Makefile.am:
9294 * gst/rtsp-server/rtsp-client.c:
9295 Check if return value of gst_rtsp_session_get_media is not NULL
9297 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9299 * gst/rtsp-server/Makefile.am:
9300 Install rtsp-session and rtsp-session-pool headers
9302 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9307 * bindings/python/Makefile.am:
9308 * bindings/python/arg-types.py:
9309 * bindings/python/codegen/Makefile.am:
9310 * bindings/python/codegen/__init__.py:
9311 * bindings/python/codegen/argtypes.py:
9312 * bindings/python/codegen/code-coverage.py:
9313 * bindings/python/codegen/codegen.py:
9314 * bindings/python/codegen/definitions.py:
9315 * bindings/python/codegen/defsparser.py:
9316 * bindings/python/codegen/docextract.py:
9317 * bindings/python/codegen/docgen.py:
9318 * bindings/python/codegen/fileprefix.override:
9319 * bindings/python/codegen/fileprefixmodule.c:
9320 * bindings/python/codegen/h2def.py:
9321 * bindings/python/codegen/mergedefs.py:
9322 * bindings/python/codegen/mkskel.py:
9323 * bindings/python/codegen/override.py:
9324 * bindings/python/codegen/reversewrapper.py:
9325 * bindings/python/codegen/scmexpr.py:
9326 * bindings/python/rtspserver-types.defs:
9327 * bindings/python/rtspserver.defs:
9328 * bindings/python/rtspserver.override:
9329 * bindings/python/rtspservermodule.c:
9331 Add python bindings.
9333 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9335 * bindings/Makefile.am:
9337 Don't go into python dir when requirements for python bindings are missing
9339 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9341 * bindings/Makefile.am:
9342 * bindings/vala/Makefile.am:
9344 Install Vala bindings if vala is available
9346 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9348 * bindings/vala/gst-rtsp-server-0.10.deps:
9349 * bindings/vala/gst-rtsp-server-0.10.vapi:
9350 * bindings/vala/gst-rtsp-server.vapi:
9351 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
9352 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
9353 * bindings/vala/packages/gst-rtsp-server-0.10.files:
9354 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9355 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9356 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
9357 * bindings/vala/packages/gst-rtsp-server.deps:
9358 * bindings/vala/packages/gst-rtsp-server.excludes:
9359 * bindings/vala/packages/gst-rtsp-server.files:
9360 * bindings/vala/packages/gst-rtsp-server.gi:
9361 * bindings/vala/packages/gst-rtsp-server.metadata:
9362 * bindings/vala/packages/gst-rtsp-server.namespace:
9363 Regenerated Vala bindings
9365 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9367 * bindings/vala/gst-rtsp-server.vapi:
9368 * bindings/vala/packages/gst-rtsp-server.metadata:
9369 Fixed typo in included headers for vala bindings
9371 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9375 * pkgconfig/Makefile.am:
9376 * pkgconfig/gst-rtsp-server.pc.in:
9377 Added pkgconfig file
9379 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
9381 * bindings/vala/gst-rtsp-server.vapi:
9382 * bindings/vala/packages/gst-rtsp-server.excludes:
9383 * bindings/vala/packages/gst-rtsp-server.gi:
9384 * bindings/vala/packages/gst-rtsp-server.metadata:
9385 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
9387 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
9389 * bindings/vala/gst-rtsp-server.vapi:
9390 * bindings/vala/packages/gst-rtsp-server.deps:
9391 * bindings/vala/packages/gst-rtsp-server.files:
9392 * bindings/vala/packages/gst-rtsp-server.gi:
9393 * bindings/vala/packages/gst-rtsp-server.metadata:
9394 * bindings/vala/packages/gst-rtsp-server.namespace:
9397 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
9399 * gst/rtsp-server/rtsp-session.c:
9400 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
9402 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
9404 * examples/Makefile.am:
9405 * gst/rtsp-server/Makefile.am:
9406 Put GStreamer version in library name
9408 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9410 * examples/Makefile.am:
9411 * gst/rtsp-server/Makefile.am:
9412 Fix some issues to pass distcheck
9414 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9416 * gst/rtsp-server/rtsp-server.c:
9417 Added port property to GstRTSPServer class.
9419 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9424 * examples/Makefile.am:
9427 * gst/rtsp-server/Makefile.am:
9428 * gst/rtsp-server/rtsp-client.c:
9429 * gst/rtsp-server/rtsp-client.h:
9430 * gst/rtsp-server/rtsp-media.c:
9431 * gst/rtsp-server/rtsp-media.h:
9432 * gst/rtsp-server/rtsp-server.c:
9433 * gst/rtsp-server/rtsp-server.h:
9434 * gst/rtsp-server/rtsp-session-pool.c:
9435 * gst/rtsp-server/rtsp-session-pool.h:
9436 * gst/rtsp-server/rtsp-session.c:
9437 * gst/rtsp-server/rtsp-session.h:
9440 * src/rtsp-client.c:
9441 * src/rtsp-client.h:
9444 * src/rtsp-server.c:
9445 * src/rtsp-server.h:
9446 * src/rtsp-session-pool.c:
9447 * src/rtsp-session-pool.h:
9448 * src/rtsp-session.c:
9449 * src/rtsp-session.h:
9450 Split in library and example program
9452 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
9454 * src/rtsp-client.h:
9455 Removed obsolete variable
9457 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
9459 * src/rtsp-client.c:
9460 * src/rtsp-client.h:
9461 Removed pipeline variable GstRTSPClient, because it's only used in one function
9463 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9466 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
9468 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
9470 * src/rtsp-session.c:
9471 Initialize some more vars.
9473 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
9475 * src/rtsp-session.c:
9476 Initialize variable to avoid compiler warning.
9478 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
9481 Add a reasonable generic .gitignore