3 2016-03-01 Sebastian Dröge <slomo@coaxion.net>
8 2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
11 Automatic update of common submodule
12 From b64f03f to 6f2d209
14 2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
16 * gst/rtsp-sink/gstrtspclientsink.c:
17 * tests/check/gst/rtspclientsink.c:
18 rtspsink: Fix some leaks in rtspclientsink and the unit test.
19 https://bugzilla.gnome.org/show_bug.cgi?id=762525
21 2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
23 * tests/check/gst/media.c:
24 * tests/check/gst/rtspclientsink.c:
25 * tests/check/gst/rtspserver.c:
26 * tests/check/gst/stream.c:
27 tests: unit test fixes
28 Removed port allocation test from the media suite.
29 The port allocation failure is now in the stream suite.
31 Make sure that the media is suspended after the DESCRIBE request
32 before reconfiguring the UDP sinks.
34 In the RECORD case we have to set async property to false
35 for the appsink element in the test in order to make sure
36 that the media pipeline doesn't hang in start_preroll().
37 https://bugzilla.gnome.org/show_bug.cgi?id=757488
39 2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
41 * gst/rtsp-server/rtsp-client.c:
42 * gst/rtsp-server/rtsp-stream.c:
43 * gst/rtsp-server/rtsp-stream.h:
44 rtsp-stream: postpone UDP socket allocation until SETUP
45 Postpone the allocation of the UDP sockets until we know
46 what transport has been chosen by the client.
47 Both unicast and multicast UDP sources are created in one
49 https://bugzilla.gnome.org/show_bug.cgi?id=757488
51 2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
53 * gst/rtsp-server/rtsp-stream.c:
54 rtsp-stream: postpone the creation of the UDP sources
55 Code refactoring: allocate the UDP ports after the sender and
56 the reciver parts have been created.
57 We postpone the creation of the UDP sources until the UDP
58 ports have been allocated.
59 https://bugzilla.gnome.org/show_bug.cgi?id=757488
61 2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
63 * gst/rtsp-server/rtsp-stream.c:
64 rtsp-stream: added function for setting UDP sources to PLAYING state
65 Code refactoring: Introduced a function for setting UDP sources
67 https://bugzilla.gnome.org/show_bug.cgi?id=757488
69 2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
71 * gst/rtsp-server/rtsp-stream.c:
72 rtsp-stream: added function for creating and configuring UDP sources
73 Code refactoring: create and configure UDP sources in a separate function.
74 https://bugzilla.gnome.org/show_bug.cgi?id=757488
76 2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
78 * gst/rtsp-server/rtsp-stream.c:
79 rtsp-stream: added function for RTP/RTCP socket configuration
80 Code refactoring: configure RTP and RTCP sockets for UDP sinks
81 in a separate function.
82 https://bugzilla.gnome.org/show_bug.cgi?id=757488
84 2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
86 * gst/rtsp-server/rtsp-stream.c:
87 rtsp-stream: added function for creating and configuring UDP sinks
88 Code refactoring: create and configure UDP sinks in a separate function.
89 https://bugzilla.gnome.org/show_bug.cgi?id=757488
91 2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
93 * gst/rtsp-server/rtsp-stream.c:
94 rtsp-stream: added helper function for creating the sender/receiver parts
95 Code refactoring: introduced helper function for creating
96 the receiver and the sender parts of the streaming pipeline.
97 https://bugzilla.gnome.org/show_bug.cgi?id=757488
99 2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
104 === release 1.7.2 ===
106 2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
112 * gst-rtsp-server.doap:
115 2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
117 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
118 uninstalled.pc: add support for non libtool build systems
119 Currently the .la path is provided which requires to use libtool as
120 mentioned in the GStreamer manual section-helloworld-compilerun.html.
121 It is fine as long as the application is built using libtool.
122 So currently it is not possible to compile a GStreamer application
123 within gst-uninstalled with CMake or other build system different
125 This patch allows to do the following in gst-uninstalled env:
126 gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
127 gstreamer-rtsp-server-1.0)
128 Previously it required to prepend libtool --mode=link
129 https://bugzilla.gnome.org/show_bug.cgi?id=720778
131 2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
133 * gst/rtsp-sink/gstrtspclientsink.c:
134 rtspclientsink: remove check for impossible condition
135 Goto error label checks stream to see if it needs to be unreferenced before
136 returning, but this goto jumps happens before the stream is ever set, so it
137 will always be NULL in this error label.
140 2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
142 * gst/rtsp-sink/gstrtspclientsink.c:
143 rtspclientsink: clean switch statements
144 Coverity demands for fallthrough statements to be clearly commented,
145 to distinguish from accidental fall throughs. And it also needs all
146 cases to finish with a break, even if the break is never going to be
147 executed like in the case of a continue jump.
151 2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
153 * tests/check/Makefile.am:
154 tests: extend the AM_TESTS_ENVIRONMENT from check.mak
155 To get the CK_DEFAULT_TIMEOUT defined for all tests
156 Also removes a 120 seconds timeout that was set as default
157 explicitly in this module
158 https://bugzilla.gnome.org/show_bug.cgi?id=761472
160 2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
164 Automatic update of common submodule
165 From 86e4663 to b64f03f
167 2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
169 * gst/rtsp-server/rtsp-media.c:
170 rtsp-media: fix state_lock not locked again when preroll fails
171 https://bugzilla.gnome.org/show_bug.cgi?id=761399
173 2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
176 configure: Move plugin specific flags below all the others
177 They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
178 -no-undefined. And -no-undefined is required on Windows to build DLLs.
180 2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
182 * gst/rtsp-sink/gstrtspclientsink.c:
183 rtspclientsink: Simplify slightly using new -base API
184 Use the new Mikey and SDP API in the base plugins libs
185 to simplify some code.
186 https://bugzilla.gnome.org/show_bug.cgi?id=758180
188 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
193 * gst/rtsp-sink/Makefile.am:
194 * gst/rtsp-sink/gstrtspclientsink.c:
195 * gst/rtsp-sink/gstrtspclientsink.h:
196 * gst/rtsp-sink/plugin.c:
197 * tests/check/Makefile.am:
198 * tests/check/gst/rtspclientsink.c:
199 rtspsink: Add rtspclientsink element
200 Add an rtspclientsink element that accepts streams for which
201 there is a registered payloader and sends them to
202 an RTSP server using RECORD.
203 Sending is synchronised to the pipeline clock. Payload-types
204 are automatically selected. The 'new-payloader' signal is fired
205 for custom configuration of payloaders when they are created.
206 Can now stream a movie like this:
208 ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
209 decodebin name=depay1 ! audioconvert ! autoaudiosink )"
211 gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
212 queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
213 https://bugzilla.gnome.org/show_bug.cgi?id=758180
215 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
217 * gst/rtsp-server/rtsp-stream.c:
218 * gst/rtsp-server/rtsp-stream.h:
219 rtsp-stream: Add functions for using rtsp-stream from the client
220 Add a boolean to indicate that the rtsp-stream is running on the
221 'client' side of an RTSP connection, for sending streams via
222 RECORD. In that case, the roles of the client/server ports
223 in transport setup are swapped.
224 https://bugzilla.gnome.org/show_bug.cgi?id=758180
226 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
228 * gst/rtsp-server/rtsp-sdp.c:
229 * gst/rtsp-server/rtsp-sdp.h:
230 rtsp-sdp: Add gst_rtsp_sdp_from_stream()
231 A new function that adds info from a GstRTSPStream into an SDP message.
232 https://bugzilla.gnome.org/show_bug.cgi?id=758180
234 2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
236 * gst/rtsp-server/rtsp-media.c:
237 rtsp-media: Fix mutex beeing unlocked while they should be locked
238 https://bugzilla.gnome.org/show_bug.cgi?id=761226
240 2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
242 * gst/rtsp-server/rtsp-media-factory.c:
243 rtsp-media-factory: add missing break in "clock" property setter
246 2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
248 * gst/rtsp-server/rtsp-stream.c:
249 rtsp-stream: fixed assert during update transport
250 When RTSP server trying update transport during multicast, it throws an
251 assert. The assert is thrown because it is trying to get the parent of
252 an non-existing funnel element.
253 https://bugzilla.gnome.org/show_bug.cgi?id=760150
255 2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
257 * gst/rtsp-server/rtsp-permissions.h:
258 * gst/rtsp-server/rtsp-thread-pool.h:
259 * gst/rtsp-server/rtsp-token.h:
260 docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
261 gtk-doc can handle static inline functions just fine these days,
262 there's no need for this stuff any more.
264 2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
266 * gst/rtsp-server/rtsp-media.c:
267 * gst/rtsp-server/rtsp-sdp.c:
268 sdp: replace duplicated codes to call new base sdp apis
269 https://bugzilla.gnome.org/show_bug.cgi?id=745880
271 2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
273 * examples/test-netclock.c:
274 test-netclock: Use the new API to configure a clock directly
276 2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
278 * gst/rtsp-server/rtsp-media-factory.c:
279 * gst/rtsp-server/rtsp-media-factory.h:
280 * gst/rtsp-server/rtsp-media.c:
281 * gst/rtsp-server/rtsp-media.h:
282 rtsp-media: Add API to directly configure a clock on the media pipelines
284 2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
286 * gst/rtsp-server/rtsp-media.c:
287 rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
289 2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
291 * gst/rtsp-server/rtsp-media-factory.c:
292 rtsp-media-factory: Add FIXME for 2.0
294 2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
296 * gst/rtsp-server/rtsp-stream.c:
297 rtsp-stream: Fix indentation
299 2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
301 * gst/rtsp-server/rtsp-media.c:
302 rtsp-media: Do not prepare media after media times out
303 Deferred calls to start_prepare() can be deferred past the point until
304 which wait_preroll() and by proxy gst_rtsp_media_get_status() is
305 prepared to wait. Previously there was no lock and no check for this
306 situation. This meant that a media could be prepared and unprepared
307 simultaneously by two different threads. Now a lock is in place and a
308 suitable check is done.
309 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
311 2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
313 * gst/rtsp-server/rtsp-client.c:
314 * gst/rtsp-server/rtsp-media-factory.c:
315 * gst/rtsp-server/rtsp-media-factory.h:
316 * gst/rtsp-server/rtsp-media.c:
317 * gst/rtsp-server/rtsp-media.h:
318 rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
319 Without TEARDOWN it might be desireable to keep the media running and continue
320 sending data to the client, even if the RTSP connection itself is
322 Only do this for session medias that have only UDP transports. If there's at
323 least on TCP transport, it will stop working and cause problems when the
324 connection is disconnected.
325 https://bugzilla.gnome.org/show_bug.cgi?id=758999
327 2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
332 === release 1.7.1 ===
334 2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
340 * gst-rtsp-server.doap:
343 2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
346 configure: Make -Bsymbolic check work with clang.
347 Update the -Bsymbolic check with the version glib has. This version
349 https://bugzilla.gnome.org/show_bug.cgi?id=759713
351 2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
353 * gst/rtsp-server/rtsp-session-pool.c:
354 rtsp-session-pool: Avoid dollar sign ($) in session ids
355 Live555 in VLC strips off dollar signs and then gets very confused,
356 we don't loose too much entropy by just skipping it.
358 2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
360 * gst/rtsp-server/rtsp-address-pool.h:
361 * gst/rtsp-server/rtsp-auth.h:
362 * gst/rtsp-server/rtsp-client.h:
363 * gst/rtsp-server/rtsp-media-factory-uri.h:
364 * gst/rtsp-server/rtsp-media-factory.h:
365 * gst/rtsp-server/rtsp-media.h:
366 * gst/rtsp-server/rtsp-mount-points.h:
367 * gst/rtsp-server/rtsp-permissions.h:
368 * gst/rtsp-server/rtsp-server.h:
369 * gst/rtsp-server/rtsp-session-media.h:
370 * gst/rtsp-server/rtsp-session-pool.h:
371 * gst/rtsp-server/rtsp-session.h:
372 * gst/rtsp-server/rtsp-stream-transport.h:
373 * gst/rtsp-server/rtsp-stream.h:
374 * gst/rtsp-server/rtsp-thread-pool.h:
375 * gst/rtsp-server/rtsp-token.h:
376 rtsp-server: Add g_autoptr() support to all types
377 https://bugzilla.gnome.org/show_bug.cgi?id=754464
379 2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
381 * gst/rtsp-server/rtsp-stream.c:
382 rtsp-stream: fixed valgrind error
383 Fixed the valgrind error in unit test. The UDP source created during
384 gst_rtsp_stream_join_bin() was not released while destroying the rtp
386 https://bugzilla.gnome.org/show_bug.cgi?id=759010
388 2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
392 Automatic update of common submodule
393 From b319909 to 86e4663
395 2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
397 * gst/rtsp-server/rtsp-client.c:
398 rtsp-client: suspend media during setup request
399 SETUP request from clients needs to suspend the media to clear the
400 prerolled buffers. Otherwise it will not affect the prerolled buffer
401 and the prerolled buffers will be incorrect (for example block-size
402 from setup request will not affect the prerolled buffer unless the
404 https://bugzilla.gnome.org/show_bug.cgi?id=758268
406 2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
408 * gst/rtsp-server/rtsp-stream.c:
409 rtsp-stream: create stream pipeline based on transport
410 Based on the protocol, create the rtsp stream pipeline. If only TCP or
411 only UDP is set as the transport protocol, it will not add the extra tee
412 or queue element to the pipeline. Both these elements will be added, if
413 it supports both TCP and UDP protocols. This improves the pipeline
414 performance when one protocol is present.
415 https://bugzilla.gnome.org/show_bug.cgi?id=758179
417 2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
419 * gst/rtsp-server/rtsp-stream.c:
420 rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
421 Adding them when not needed will start some logic inside rtpbin that might be
422 problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
423 would start up a rtpjitterbuffer and behave in weird ways.
424 We still set up the UDP sources for RTP receiving for a sender media to be
425 able to receive any packets sent by the client for NAT traversal. They will
426 all go to a fakesink though.
427 Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
428 NO_PREROLL, which will cause deadlocks when seeking the media as it will never
429 receive ASYNC_DONE after a seek.
430 https://bugzilla.gnome.org/show_bug.cgi?id=758319
432 2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
434 * gst/rtsp-server/rtsp-stream.c:
435 rtsp-stream: Disable multicast loopback for the multicast udp sources too
436 On POSIX this setting is for sender sockets, on Windows for receiver sockets.
437 Previously we were only setting this for sender sockets, which caused looped
438 back packets to be received on Windows if a multicast transport was used.
440 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
442 * examples/test-record-auth.c:
443 * examples/test-record.c:
444 examples: Actually use the provided port in the record examples
446 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
448 * examples/test-record-auth.c:
449 test-record-auth: Add the option to build in TLS support
451 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
453 * examples/test-auth.c:
454 test-auth: Use an 'anonymous' user for unauthenticated default
455 There's a comment on one of the resources that 'user' and 'admin'
456 shouldn't even be able to see it, but they can if the default
457 token is 'admin2', since that gives them access anyway.
459 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
461 * examples/.gitignore:
462 * examples/Makefile.am:
463 * examples/test-record-auth.c:
464 Add test-record-auth example
466 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
468 * gst/rtsp-server/rtsp-client.c:
469 * tests/check/gst/client.c:
470 rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
472 2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
474 * gst/rtsp-server/rtsp-server.c:
475 rtsp-server: Change the logic so we don't pop a NULL context
476 When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
477 will sometimes fail. This call is made before any context is pushed
478 resulting in an attempt to pop a NULL context.
479 https://bugzilla.gnome.org/show_bug.cgi?id=757949
481 2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
483 * tests/check/gst/rtspserver.c:
484 rtspserver: Add udp-mcast transport SETUP test
485 Refactor utility functions in the test file so they can handle
486 more than UDP and TCP as lower transport.
487 https://bugzilla.gnome.org/show_bug.cgi?id=756969
489 2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
491 * gst/rtsp-server/rtsp-stream.c:
492 rtsp-stream: Always unref return value of gst_object_get_parent()
493 Fixes a leak of a GstBin in the udp-mcast case.
494 https://bugzilla.gnome.org/show_bug.cgi?id=756968
496 2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
499 Automatic update of common submodule
500 From b99800a to b319909
502 2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
505 Use new GST_ENABLE_EXTRA_CHECKS #define
506 https://bugzilla.gnome.org/show_bug.cgi?id=756870
508 2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
511 Automatic update of common submodule
512 From 6babecd to b99800a
514 2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
517 Update GLib dependency to 2.40.0
519 2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
521 * examples/test-mp4.c:
522 * gst/rtsp-server/rtsp-stream.c:
523 stream: listen to sender ssrc signals
524 https://bugzilla.gnome.org/show_bug.cgi?id=746747
526 2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
529 common: update for new suppression
530 Makes check-valgrind pass with glib 2.46
532 2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
534 * gst/rtsp-server/rtsp-media.c:
535 rtsp-media: Take reference to media that will be prepared
536 default_prepare() takes a transfer-none reference GstRTSPMedia object.
537 Later on a g_idle_source_new() is created and a pointer to the media
538 object is passed as user data. If the media is freed before the idle
539 source is dispatched the media object pointer is invalid, but the idle
540 source callback expects it to still be valid. To fix this a reference to
541 the media object is taken when registering the source callback function
542 and a corresponding release of the reference is done when the souce is
544 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
546 2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
548 * examples/test-launch.c:
549 * examples/test-mp4.c:
550 * examples/test-ogg.c:
551 * examples/test-record.c:
552 * examples/test-uri.c:
553 rtsp-server: Fix memory leaks when context parse fails
554 When g_option_context_parse fails, context and error variables are not getting free'd
555 which results in memory leaks. Free'ing the same.
556 And replacing g_error_free with g_clear_error, which checks if the error being passed
557 is not NULL and sets the variable to NULL on free'ing.
558 https://bugzilla.gnome.org/show_bug.cgi?id=753863
560 2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
565 === release 1.6.0 ===
567 2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
573 * gst-rtsp-server.doap:
576 === release 1.5.91 ===
578 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
584 * gst-rtsp-server.doap:
587 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
589 * docs/libs/gst-rtsp-server-sections.txt:
590 * gst/rtsp-server/rtsp-stream.c:
591 stream: fix docs for recently-added get/set_buffer_size API
592 https://bugzilla.gnome.org/show_bug.cgi?id=749095
594 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
596 * gst/rtsp-server/rtsp-media.c:
597 rtsp-media: Don't crash on encrypted RTX SDP
598 In parse_keymgmt(), don't mutate the input string that's been passed
599 as const, especially since we might need the original value again if
600 the same key info applies to multiple streams (RTX, for example).
601 https://bugzilla.gnome.org/show_bug.cgi?id=754753
603 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
605 * examples/test-mp4.c:
606 test-mp4: Support filenames with spaces in them. Error out on too few arguments
608 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
610 * examples/test-record.c:
611 test-record: Check parameter count and print out help
612 If no launch pipeline was supplied, print out some help
614 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
616 * gst/rtsp-server/rtsp-media.c:
617 * gst/rtsp-server/rtsp-stream.c:
618 * gst/rtsp-server/rtsp-stream.h:
619 rtsp-stream: Implement UDP buffer size setting.
620 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
622 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
623 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
625 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
627 * gst/rtsp-server/rtsp-media.h:
628 rtsp-media: Fix small typo causing gtk-doc to complain
630 === release 1.5.90 ===
632 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
638 * gst-rtsp-server.doap:
641 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
643 * gst/rtsp-server/rtsp-media-factory.c:
644 media-factory: get port number through gst_rtsp_url_get_port
645 https://bugzilla.gnome.org/show_bug.cgi?id=753473
647 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
649 * tests/check/gst/media.c:
650 media-test: Removing unnecessary assertion
651 https://bugzilla.gnome.org/show_bug.cgi?id=753385
653 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
655 * gst/rtsp-server/rtsp-server.c:
656 Document that source keeps a ref on server until it's destroyed
657 https://bugzilla.gnome.org/show_bug.cgi?id=749227
659 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
661 * tests/check/gst/media.c:
662 media-test: Test for multiple dynamic payload
663 https://bugzilla.gnome.org/show_bug.cgi?id=753385
665 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
667 * gst/rtsp-server/rtsp-media.c:
668 media: Only add fakesink once per pipeline
669 The intention is to prevent going PLAYING state before pads are created.
670 If there was mutilple dynamic payload, it would leak few fakesink and
671 actually prevent from ever reaching playing state.
672 https://bugzilla.gnome.org/show_bug.cgi?id=753385
674 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
676 * gst/rtsp-server/rtsp-media.c:
677 Revert "rtsp-media: Only add 1 fakesink per pipeline"
678 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
680 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
682 * gst/rtsp-server/rtsp-media.c:
683 rtsp-media: Only add 1 fakesink per pipeline
684 There should be only one fakesink per pipeline, not per dynpay. This
685 would lead to element naming clash.
687 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
689 * gst/rtsp-server/rtsp-media.c:
690 rtsp-media: assertion error due to wrong condition check
691 In media to caps function, reserved_keys array is being used for variable i,
692 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
693 changed it to variable j
694 https://bugzilla.gnome.org/show_bug.cgi?id=753009
696 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
698 * gst/rtsp-server/rtsp-media.c:
699 rtsp-media: Strip keys from the fmtp that we use internally in our caps
700 Skip keys from the fmtp, which we already use ourselves for the
701 caps. Some software is adding random things like clock-rate into
702 the fmtp, and we would otherwise here set a string-typed clock-rate
703 in the caps... and thus fail to create valid RTP caps
704 https://bugzilla.gnome.org/show_bug.cgi?id=753009
706 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
708 * gst/rtsp-server/rtsp-thread-pool.c:
709 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
710 https://bugzilla.gnome.org/show_bug.cgi?id=752640
712 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
715 Automatic update of common submodule
716 From f74b2df to 9aed1d7
718 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
723 === release 1.5.2 ===
725 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
731 * gst-rtsp-server.doap:
734 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
736 * gst/rtsp-server/rtsp-client.c:
737 * gst/rtsp-server/rtsp-client.h:
738 * tests/check/gst/client.c:
739 rtsp-client: allow application to decide what requirements are supported
740 Add "check-requirements" signal and vfunc to allow application
741 (and subclasses) to check the requirements.
742 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
743 https://bugzilla.gnome.org/show_bug.cgi?id=749417
745 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
748 Automatic update of common submodule
749 From 6015d26 to f74b2df
751 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
753 * gst/rtsp-server/rtsp-media.c:
754 rtsp-media: Always use real payloader when creating streams
755 A bin that contains the real payloader might be used as payloader. In this
756 case we have to get the real payloader for the various properties it provides.
757 Example use cases for this are bins that payload some media and then have
758 additional elements that add metadata or RTP extension headers to the stream.
759 https://bugzilla.gnome.org/show_bug.cgi?id=750800
761 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
763 * examples/test-netclock-client.c:
764 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
766 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
768 * examples/test-netclock-client.c:
769 * examples/test-netclock.c:
770 test-netclock: Use new ntp-time-source property on rtpbin
771 Select the clock time to be used as NTP time source. This allows proper
772 synchronization between receivers, independent of sharing base times, and just
773 requires them to use the same clock.
775 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
777 * examples/test-netclock-client.c:
778 * examples/test-netclock.c:
779 test-netclock: Setting the same base time on sender and receiver is not necessary
780 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
782 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
784 * gst/rtsp-server/rtsp-stream.c:
785 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
786 https://bugzilla.gnome.org/show_bug.cgi?id=750764
788 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
790 * docs/libs/gst-rtsp-server.types:
791 docs: add missing types
792 https://bugzilla.gnome.org/show_bug.cgi?id=750764
794 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
796 * docs/libs/gst-rtsp-server-sections.txt:
797 docs: add missing apis
798 https://bugzilla.gnome.org/show_bug.cgi?id=750764
800 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
802 * examples/test-netclock-client.c:
803 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
805 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
807 * docs/libs/gst-rtsp-server-sections.txt:
808 * gst/rtsp-server/rtsp-auth.c:
809 * gst/rtsp-server/rtsp-auth.h:
810 GstRTSPAuth: Add client certificate authentication support
811 https://bugzilla.gnome.org/show_bug.cgi?id=750471
813 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
815 * examples/test-netclock-client.c:
816 test-netclock-client: Use new GstClock API to wait for clock synchronization
818 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
820 * examples/test-netclock-client.c:
821 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
822 A mainloop is needed to get glimagesink to display something on OSX, and
823 the source-setup signal just makes things a little bit easier.
825 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
828 Automatic update of common submodule
829 From d9a3353 to 6015d26
831 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
834 Automatic update of common submodule
835 From d37af32 to d9a3353
837 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
840 Automatic update of common submodule
841 From 21ba2e5 to d37af32
843 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
846 Automatic update of common submodule
847 From c408583 to 21ba2e5
849 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
851 * docs/libs/Makefile.am:
852 docs: remove variables that we define in the snippet from common
853 This is syncing our Makefile.am with upstream gtkdoc.
855 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
858 Automatic update of common submodule
859 From 44a3517 to c408583
861 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
866 === release 1.5.1 ===
868 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
874 * gst-rtsp-server.doap:
877 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
879 * gst/rtsp-server/rtsp-client.c:
880 rtsp-client: No flush during Teardown.
881 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
882 backlog is empty it can happen that just a part of a message will be
883 sent and rest is in backlog queue. If then flush during teardown
884 just a part of message will be sent.This can lead to client miss
885 teardown response since it expect to get the last part of message.
886 The flushing during teardown was introduced to fix a deadlock that now
887 is fixed more generally in handle_request by temporary setting backlog
889 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
891 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
893 * tests/check/Makefile.am:
894 tests: Use AM_TESTS_ENVIRONMENT
895 Needed by the new automake test runner and the
896 current version of the common submodule.
898 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
900 * gst/rtsp-server/rtsp-media.h:
901 * gst/rtsp-server/rtsp-stream.h:
902 rtsp-server: Use single-include rtsp header to make sure we get all definitions
904 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
906 * gst/rtsp-server/rtsp-media.c:
907 rtsp-media: Mark some more functions static
909 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
911 * gst/rtsp-server/rtsp-media.c:
912 rtsp-media: Only unblock the media in suspend() when actually changing the state
913 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
915 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
917 * examples/test-video-rtx.c:
918 examples: Use AVPF profile for the RTX example
920 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
922 * gst/rtsp-server/rtsp-sdp.c:
923 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
925 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
927 * gst/rtsp-server/rtsp-stream.c:
928 rtsp-stream: get valid clock-rate from last-sample
929 clock-rate in last-sample's caps is integer, not unsigned.
930 To get this value properly, variable needs to be type-casted to int.
931 https://bugzilla.gnome.org/show_bug.cgi?id=747614
933 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
937 autogen.sh: only run autopoint if gettext requested in configure.ac
938 Not just because there happens to be a po directory.
939 https://bugzilla.gnome.org/show_bug.cgi?id=748058
941 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
944 Revert "configure.ac: uncomment gettext version setup"
945 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
946 We don't need a gettext setup here and there's no po
947 directory either, so no reason why autopoint would be
948 run in the first place.
949 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
951 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
953 * examples/test-multicast.c:
954 * examples/test-multicast2.c:
955 * examples/test-sdp.c:
956 * examples/test-video-rtx.c:
957 * examples/test-video.c:
958 * tests/test-cleanup.c:
959 * tests/test-reuse.c:
960 Fix timeout function signatures across tests and examples
962 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
964 * tests/check/Makefile.am:
965 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
966 Make sure the test environment is set up.
967 https://bugzilla.gnome.org//show_bug.cgi?id=747624
969 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
972 configure: bump automake requirement to 1.14 and autoconf to 2.69
973 This is only required for builds from git, people can still
974 build tarballs if they only have older autotools.
975 https://bugzilla.gnome.org//show_bug.cgi?id=747624
977 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
980 configure.ac: uncomment gettext version setup
981 Fixes autogen.sh. It would run autopoint, which would complain
982 that it could not find the gettext version in configure.ac.
983 https://bugzilla.gnome.org/show_bug.cgi?id=748058
985 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
987 * examples/test-video-rtx.c:
988 test-video-rtx: set exact payload type to PCMA payloader
989 Setting wrong payload type causes failure to do retransmission through audio stream
990 https://bugzilla.gnome.org/show_bug.cgi?id=747839
992 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
994 * gst/rtsp-server/rtsp-media.c:
995 * gst/rtsp-server/rtsp-stream.c:
996 * gst/rtsp-server/rtsp-stream.h:
997 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
998 Because of duplicated g_signal_connect for request-aux-sender signal,
999 wrong stream pointer is passed to the signal handler.
1000 Instead of passing each stream, pass stream array and get the relevant stream.
1001 https://bugzilla.gnome.org/show_bug.cgi?id=747839
1003 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
1007 Update autogen.sh to latest version from common
1008 Fixes build after aclocal_check etc. helpers have been removed.
1010 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
1013 Automatic update of common submodule
1014 From bc76a8b to c8fb372
1016 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
1018 * gst/rtsp-server/rtsp-stream.c:
1019 rtsp-stream: Limit the queues to 1 buffer
1020 We only need them to be able to pre-roll, queueing up more data here
1021 is only going to harm latency and memory usage.
1023 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
1025 * gst/rtsp-server/rtsp-stream.c:
1026 rtsp-stream: Update comment and ASCII art to the latest code
1027 We have a queue in front of the udpsink too to prevent the pipeline from
1030 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
1032 * gst/rtsp-server/rtsp-stream.c:
1033 rtsp-media: Properly return first rtptime
1034 Instead we where returning first GstBuffer timestamp. This would result
1035 in clock skew and unwanted behaviour in RTSP playback.
1036 https://bugzilla.gnome.org/show_bug.cgi?id=746479
1038 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
1040 * gst/rtsp-server/rtsp-stream.c:
1041 rtsp-stream: Don't leave buffer mapped
1042 If the seq is NULL, the RTP buffer was left mapped. We should always
1045 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
1050 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
1052 * gst/rtsp-server/rtsp-media-factory.c:
1053 * tests/check/gst/client.c:
1054 Fix double semicolons
1056 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
1058 * gst/rtsp-server/rtsp-stream.c:
1059 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
1060 This gives more accurate values than asking the payloader. There might be
1061 queueing happening between the payloader and the sink.
1062 https://bugzilla.gnome.org/show_bug.cgi?id=745704
1064 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
1066 * gst/rtsp-server/rtsp-media.c:
1067 rtsp-media: Don't seek for PLAY if the position will not change
1068 https://bugzilla.gnome.org/show_bug.cgi?id=745704
1070 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
1072 * gst/rtsp-server/rtsp-media.c:
1073 rtsp-media: Don't include payload type in the caps for framesize
1074 When the sdp media attribute framesize are converted to caps
1075 the <payload> should not be included.
1076 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
1077 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
1079 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
1081 * gst/rtsp-server/rtsp-sdp.c:
1082 rtsp-sdp: add payload type to the sdp framesize attribute
1083 The sdp framesize attribute is desribed in RFC6064. It is specified
1084 for payloading of H263 and has the following form
1085 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
1086 should be added to the caps in a payloader and the <payload type> should
1087 be added by the rtsp-server.
1088 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
1090 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
1092 * examples/test-uri.c:
1093 examples: test-uri: fix tainted variable
1094 Insignificant but this keeps Coverity happy.
1097 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
1099 * examples/.gitignore:
1100 * examples/Makefile.am:
1101 * examples/test-netclock-client.c:
1102 * examples/test-netclock.c:
1103 examples: Add a simple example of network synch for live streams.
1104 An example server and client that works for synchronising live streams
1105 only - as it can't support pause/play.
1107 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
1109 * gst/rtsp-server/rtsp-media-factory.c:
1110 * gst/rtsp-server/rtsp-media-factory.h:
1111 rtsp-media-factory: Add functions to set/get the media gtype
1112 Allow specifying the GType of a GstRtspMedia subclass to create
1113 as a simpler way to get the factory to create a custom
1114 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
1116 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
1118 * gst/rtsp-server/rtsp-media.c:
1119 rtsp-media: fix double unlock in _get_buffer_size()
1120 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
1121 because of double g_mutex_unlock () usage.
1122 https://bugzilla.gnome.org/show_bug.cgi?id=745434
1124 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
1126 * gst/rtsp-server/rtsp-session-pool.c:
1127 * gst/rtsp-server/rtsp-session.c:
1128 * gst/rtsp-server/rtsp-session.h:
1129 rtsp-session: Use monotonic time for RTSP session timeout
1130 Changed RTSP session timeout handling to monotonic time
1131 and deprecating the API for current system time.
1132 This fixes timeouts when the system time changes.
1133 https://bugzilla.gnome.org/show_bug.cgi?id=743346
1135 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
1137 * gst/rtsp-server/rtsp-client.c:
1138 * gst/rtsp-server/rtsp-media.c:
1139 rtsp-client: Only error out in PLAY if seeking actually failed
1140 If the media was just not seekable, we continue from whatever position we are
1141 and let the client decide if that is what is wanted or not.
1142 Only if the actual seek failed, we can't really recover and should error out.
1144 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
1146 * gst/rtsp-server/rtsp-stream.c:
1147 rtsp-stream: Add necessary queues between tee and multiudpsink
1148 https://bugzilla.gnome.org/show_bug.cgi?id=744379
1150 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
1152 * gst/rtsp-server/rtsp-client.c:
1153 * gst/rtsp-server/rtsp-media.c:
1154 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
1155 Instead error out properly the same way as if the SEEKING query already
1158 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
1160 * gst/rtsp-server/rtsp-stream.h:
1161 rtsp-stream: minor code formatting fix
1163 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
1165 * gst/rtsp-server/rtsp-media.c:
1166 rtsp-media: fix logic for collect_streams
1167 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
1168 all streams it knows if it got any, and can check if the transport mode is OK.
1171 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
1173 * gst/rtsp-server/rtsp-media.c:
1174 rtsp-media: Don't set the transport mode based on what elements we find
1175 Just print a warning if the one that was set before disagrees with what
1176 elements we found. It must already be set to something before as this
1177 function is called after we received the SDP from ANNOUNCE in RECORD mode,
1178 and we would reject ANNOUNCE if the RECORD flag was not set.
1180 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
1182 * tests/check/gst/rtspserver.c:
1183 tests: rtspserver: rename shadowed variable
1184 We have two different 'sink' variables here,
1185 rename one of them for clarity.
1187 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
1189 * gst/rtsp-server/rtsp-client.c:
1190 rtsp-client: fix awkward if clause
1192 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
1194 * examples/test-uri.c:
1195 examples: test-uri: improve uri argument handling and accept file names
1196 Print an error if the argument passed is not a URI and can't
1197 be converted into one, or no arguments have been provided.
1199 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
1201 * examples/test-uri.c:
1202 examples: test-uri: don't remove mount point after 10 seconds
1203 It's very irritating when trying to test stuff repeatedly
1204 and serves no real purpose other than showing that it can
1207 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
1209 * examples/.gitignore:
1210 examples: add new test-record to .gitignore
1212 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
1214 * examples/test-record.c:
1215 * gst/rtsp-server/rtsp-client.c:
1216 * gst/rtsp-server/rtsp-media-factory.c:
1217 * gst/rtsp-server/rtsp-media-factory.h:
1218 * gst/rtsp-server/rtsp-media.c:
1219 * gst/rtsp-server/rtsp-media.h:
1220 * tests/check/gst/rtspserver.c:
1221 rtsp-media: Use flags to distinguish between PLAY and RECORD media
1223 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
1225 * examples/test-record.c:
1226 test-record: Set latency for playback-style example to 2s instead of 200ms
1228 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
1230 * tests/check/gst/rtspserver.c:
1231 tests: add some unit tests for ANNOUNCE and RECORD
1232 https://bugzilla.gnome.org/show_bug.cgi?id=743175
1234 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
1236 * gst/rtsp-server/rtsp-client.c:
1237 rtsp-client: fix a couple of leaks in handle_announce
1239 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
1241 * gst/rtsp-server/rtsp-media-factory.c:
1242 * gst/rtsp-server/rtsp-media-factory.h:
1243 * gst/rtsp-server/rtsp-media.c:
1244 * gst/rtsp-server/rtsp-media.h:
1245 rtsp-media: Expose latency setting for setting the rtpbin latency
1247 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
1249 * examples/test-record.c:
1250 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
1252 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
1254 * gst/rtsp-server/rtsp-stream.c:
1255 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
1257 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
1259 * examples/Makefile.am:
1260 * examples/test-record.c:
1261 * gst/rtsp-server/rtsp-client.c:
1262 * gst/rtsp-server/rtsp-client.h:
1263 * gst/rtsp-server/rtsp-media-factory.c:
1264 * gst/rtsp-server/rtsp-media-factory.h:
1265 * gst/rtsp-server/rtsp-media.c:
1266 * gst/rtsp-server/rtsp-media.h:
1267 * gst/rtsp-server/rtsp-session-media.c:
1268 * gst/rtsp-server/rtsp-stream.c:
1269 * gst/rtsp-server/rtsp-stream.h:
1270 Add initial support for RECORD
1271 We currently only support media that is RECORD or PLAY only, not both at once.
1272 https://bugzilla.gnome.org/show_bug.cgi?id=743175
1274 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
1276 * gst/rtsp-server/rtsp-stream.c:
1277 rtsp-stream: RTCP and RTP transport cache cookies seperated
1278 RTCP packets were not sent because the same tr_cache_cookie was used for
1279 both RTP and RTCP. So only one of the tr_cache lists were populated
1280 depending on which one was sent first. If the tr_cache list is not
1281 populated then no packets can be sent. Most often this happened to be
1282 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
1283 resulted in both the tr_cache_lists to be populated regardless of which
1285 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
1287 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
1289 * gst/rtsp-server/rtsp-stream.c:
1290 rtsp-stream: fix false compiler warning
1291 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
1293 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
1295 * gst/rtsp-server/rtsp-client.c:
1296 rtsp-client: log interleaved data received
1298 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
1300 * gst/rtsp-server/rtsp-client.c:
1301 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
1303 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
1305 * gst/rtsp-server/rtsp-client.c:
1306 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
1308 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
1310 * gst/rtsp-server/rtsp-client.c:
1311 rtsp-client: Use a random session ID in the SDP
1312 RFC4566 Section 5.2 says that it should make the username, session id,
1313 nettype, addrtype and unicast address tuple globally unique. Always using
1314 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
1315 Instead let's create a 64 bit random number, which at least brings us
1316 closer to the goal of global uniqueness.
1317 https://tools.ietf.org/html/rfc4566#section-5.2
1319 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
1321 * examples/test-launch.c:
1322 * examples/test-mp4.c:
1323 * examples/test-ogg.c:
1324 * examples/test-uri.c:
1325 examples: Don't call gst_init() and gst_get_option_group()
1326 The latter calls the former at the appropriate time.
1328 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
1330 * gst/rtsp-server/rtsp-client.c:
1331 rtsp-client: Drop trailing \0 of RTSP DATA messages
1332 We add a trailing \0 in GstRTSPConnection to make parsing of
1333 string message bodies easier (e.g. the SDP from DESCRIBE) but
1334 for actual data this means we have to drop it or otherwise
1335 create invalid data.
1337 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
1339 * gst/rtsp-server/rtsp-stream.c:
1340 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
1341 Fixes crash when two threads access handle_new_sample() at the same
1342 time, one for RTP, one for RTCP.
1343 Otherwise, when iterating over the transports cache, it might be modified by
1344 another thread at the same time if the transports cookie has changed.
1345 https://bugzilla.gnome.org/show_bug.cgi?id=742954
1347 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
1349 * gst/rtsp-server/rtsp-stream.c:
1350 rtsp-stream: Set format=TIME on our app sources for TCP
1352 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
1354 * gst/rtsp-server/rtsp-session-pool.c:
1355 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
1356 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
1357 RFC 2326 states that session IDs may consist of alphanumeric as well as
1358 the safe characters $-_.+ -- N.B. the percent character is not allowed.
1359 Previously the session ID was URI-escaped, this meant that any character
1360 which was not alphanumeric or any of the characters +-._~ would be
1361 percent encoded. While the RFC (surprisingly) mentions that linear white
1362 space in session IDs should be URI-escaped, it does not say anything
1363 about other characters. Moreover no white space is allowed in the
1364 session ID. Finally the percent character which is the result of
1365 URI-escaping is not allowed in a session ID.
1366 So there is no reason to do any URI-escaping, and now it is removed.
1367 https://bugzilla.gnome.org/show_bug.cgi?id=742869
1369 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
1372 Automatic update of common submodule
1373 From f2c6b95 to bc76a8b
1375 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
1378 Fix 'make check' from top-level directory
1380 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
1382 * examples/test-launch.c:
1383 * examples/test-mp4.c:
1384 * examples/test-ogg.c:
1385 * examples/test-uri.c:
1386 examples: Add command-line parsing and take a 'port' argument
1387 This allows users to run multiple servers on different ports for testing.
1388 Only done for examples that actually take arguments and hence are capable of
1389 outputting different streams for each instance on each port.
1390 https://bugzilla.gnome.org/show_bug.cgi?id=742115
1392 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
1394 * gst/rtsp-server/rtsp-client.c:
1395 * gst/rtsp-server/rtsp-client.h:
1396 rtsp-client: Add a send_message default signal handler
1397 This allows subclasses to easily hook into the response sending
1398 mechanism without doing everything from a signal, which seems
1399 awkward from subclasses.
1401 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
1404 Automatic update of common submodule
1405 From ef1ffdc to f2c6b95
1407 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1411 configure: add --disable-examples switch
1412 https://bugzilla.gnome.org/show_bug.cgi?id=741678
1414 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
1416 * examples/.gitignore:
1417 * examples/Makefile.am:
1418 * examples/test-video-rtx.c:
1419 examples: add a retransmisison example implementing RFC4588
1420 Currently only SSRC-multiplexed rtx streams are supported
1422 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
1424 * gst/rtsp-server/rtsp-stream.c:
1425 rtsp-stream: Fix some minor memory leaks
1427 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
1429 * gst/rtsp-server/rtsp-media.c:
1430 rtsp-media: Some minor cleanup
1432 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
1434 * gst/rtsp-server/rtsp-stream.c:
1435 rtsp-stream: Fix compiler warnings
1436 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
1437 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1439 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
1440 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1443 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
1445 * docs/libs/gst-rtsp-server-sections.txt:
1446 * gst/rtsp-server/rtsp-media-factory.c:
1447 * gst/rtsp-server/rtsp-media-factory.h:
1448 * gst/rtsp-server/rtsp-media.c:
1449 * gst/rtsp-server/rtsp-media.h:
1450 * gst/rtsp-server/rtsp-sdp.c:
1451 * gst/rtsp-server/rtsp-stream.c:
1452 * gst/rtsp-server/rtsp-stream.h:
1453 media: implement ssrc-multiplexed retransmission support
1454 based off RFC 4588 and the server-rtpaux example in -good
1456 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
1458 * gst/rtsp-server/rtsp-client.c:
1459 * gst/rtsp-server/rtsp-stream-transport.c:
1460 * gst/rtsp-server/rtsp-stream.c:
1461 rtsp: Ref transports in hash table.
1462 Also ref streams for transports.
1463 This solves a crash when reciving a rtcp after teardown but before
1465 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
1467 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
1470 Automatic update of common submodule
1471 From 7bb2bce to ef1ffdc
1473 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
1475 * gst/rtsp-server/rtsp-client.c:
1476 client: refactor cleanup of cached media
1478 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
1480 * tests/check/gst/client.c:
1482 The session leak is now fixed, lets remove those FIXME comments.
1484 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
1486 * tests/check/gst/rtspserver.c:
1487 tests: Test to setup two sessions on one connection
1488 https://bugzilla.gnome.org/show_bug.cgi?id=739112
1490 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
1492 * tests/check/gst/rtspserver.c:
1493 tests: Test setup with tcp transport
1494 https://bugzilla.gnome.org/show_bug.cgi?id=739112
1496 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
1498 * gst/rtsp-server/rtsp-client.c:
1499 client: Configure transport after creating session media
1500 The default implementation of configure_client_transport() in
1501 rtsp-client uses the session media when it chooses channels for
1502 interleaved traffic.
1503 https://bugzilla.gnome.org/show_bug.cgi?id=739112
1505 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
1507 * gst/rtsp-server/rtsp-client.c:
1508 * gst/rtsp-server/rtsp-session-media.c:
1509 client: Stop caching media in client when doing setup
1510 If the media has been managed by a session media, it should not be
1511 cached in the client any longer. The GstRTSPSessionMedia object is now
1512 responsible for unpreparing the GstRTSPMedia object using
1513 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
1515 https://bugzilla.gnome.org/show_bug.cgi?id=739112
1517 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1519 * gst/rtsp-server/rtsp-stream.c:
1520 rtsp-stream: unref srtp decoder when leaving bin
1521 https://bugzilla.gnome.org/show_bug.cgi?id=739481
1523 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1525 * gst/rtsp-server/rtsp-client.c:
1526 rtsp-client: mikey memory leaks
1527 https://bugzilla.gnome.org/show_bug.cgi?id=739383
1529 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
1532 Automatic update of common submodule
1533 From 84d06cd to 7bb2bce
1535 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
1538 Parallelise 'make check-valgrind'
1540 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
1543 Automatic update of common submodule
1544 From a8c8939 to 84d06cd
1546 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
1549 Automatic update of common submodule
1550 From 36388a1 to a8c8939
1552 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
1554 * gst/rtsp-server/rtsp-media.c:
1555 rtsp-media: deactivate media when shutting down from paused
1556 This was only done when going directly from playing.
1557 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
1559 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1561 * gst/rtsp-server/rtsp-client.c:
1562 * gst/rtsp-server/rtsp-context.h:
1563 rtsp-client: add stream transport to context
1564 We add the stream transport to the context so we can get the configured
1565 client stream transport in the setup request signal.
1566 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
1568 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1570 * gst/rtsp-server/rtsp-stream.c:
1571 stream: release lock even not all transports have been removed
1572 We don't want to keep the lock even we return FALSE because not all the
1573 transports have been removed. This could lead into a deadlock.
1574 https://bugzilla.gnome.org/show_bug.cgi?id=737797
1576 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
1578 * gst/rtsp-server/rtsp-sdp.c:
1579 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
1580 These were renamed in GstRTPBasePayload in 1.0
1582 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1584 * gst/rtsp-server/rtsp-client.c:
1585 client: set session media to NULL without the lock
1586 We need to set session medias to NULL without the client lock otherwise
1587 we can end up in a deadlock if another thread is waiting for the lock
1588 and media unprepare is also waiting for that thread to end.
1589 https://bugzilla.gnome.org/show_bug.cgi?id=737690
1591 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
1593 * gst/rtsp-server/rtsp-media.c:
1594 rtsp-media: Set state to UNPREPARING in all cases
1596 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
1598 * gst/rtsp-server/rtsp-media.c:
1599 media: set state to unpreparing when unprepare is initiated
1600 https://bugzilla.gnome.org/show_bug.cgi?id=737675
1602 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
1604 * gst/rtsp-server/rtsp-client.c:
1605 rtsp-client: Remove backlog limit while processings requests
1606 If the backlog limit is kept two cases of deadlocks may be
1607 encountered when streaming over TCP. Without the backlog
1608 limit this deadlocks can not happen, at the expence of
1610 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
1612 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
1614 * gst/rtsp-server/rtsp-client.c:
1615 rtsp-client: do not free main context before rtsp watch
1616 https://bugzilla.gnome.org/show_bug.cgi?id=737110
1618 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
1620 * tests/check/gst/rtspserver.c:
1621 tests: Extend unit test timeout to accomodate for valgrind
1622 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1624 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
1626 * gst/rtsp-server/rtsp-client.c:
1627 * gst/rtsp-server/rtsp-session.c:
1628 * gst/rtsp-server/rtsp-stream-transport.c:
1629 rtsp-*: Treat sending packets to clients as keepalive
1630 As long as gst-rtsp-server can successfully send RTP/RTCP data to
1631 clients then the client must be reading. This change makes the server
1632 timeout the connection if the client stops reading.
1633 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1635 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
1637 * gst/rtsp-server/rtsp-client.c:
1638 rtsp-client: Allow backlog to grow while expiring session
1639 Allow the send backlog in the RTSP watch to grow to unlimited size while
1640 attempting to bring the media pipeline to NULL due to a session
1641 expiring. Without this change the appsink element cannot change state
1642 because it is blocked while rendering data in the new_sample callback.
1643 This callback will block until it has successfully put the data into the
1644 send backlog. There is a chance that the send backlog is full at this
1645 point which means that the callback may block for a long time, possibly
1646 forever. Therefore the media pipeline may also be prevented from
1647 changing state for a long time.
1648 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1650 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
1652 * gst/rtsp-server/rtsp-client.c:
1653 rtsp-client: Make old compilers happy
1654 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
1655 Just in case that guint8 doesn't fit in a pointer. Just in case ...
1657 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
1659 * gst/rtsp-server/rtsp-client.c:
1660 client: raise the backlog limits before pausing
1661 We need to raise the backlog limits before pausing the pipeline or else
1662 the appsink might be blocking in the render method in wait_backlog() and
1663 we would deadlock waiting for paused.
1664 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
1666 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
1668 * gst/rtsp-server/rtsp-client.c:
1669 client: make define for the WATCH_BACKLOG
1670 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
1672 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
1674 * gst/rtsp-server/rtsp-client.c:
1675 client: simplify session transport handling
1676 link/unlink of the transport in a session was done to keep track of all
1677 TCP transports and to send RTP/RTCP data to the streams. We can simplify
1678 that by putting all the TCP transports in a hashtable indexed with the
1680 We also don't need to link/unlink the transports when we pause/resume
1681 the streams. The same effect is already achieved when we pause/play the
1682 media. Indeed, when we pause the media, the transport is removed from
1683 the media and the callbacks will not be called anymore.
1684 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
1686 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
1688 * gst/rtsp-server/rtsp-stream-transport.c:
1689 * gst/rtsp-server/rtsp-stream-transport.h:
1690 stream-transport: make method to handle received data
1691 Make a method to handle the data received on a channel. It sends the
1692 data to the stream of the transport on the RTP or RTCP pads based on
1695 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
1697 * examples/test-mp4.c:
1698 test: add example of dumping RTCP reports
1700 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
1702 * gst/rtsp-server/rtsp-media.c:
1703 * gst/rtsp-server/rtsp-stream.c:
1704 * gst/rtsp-server/rtsp-stream.h:
1705 rtsp-media: Make sure that sequence numbers are monotonic after pause
1706 The sequence number is not monotonic for RTP packets after pause. The
1707 reason is basepayloader generates a randon sequence number when the
1708 pipeline goes from ready to pause. With this fix generation of sequence
1709 number will be monotonic when going from pause to play request.
1710 https://bugzilla.gnome.org/show_bug.cgi?id=736017
1712 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
1714 * gst/rtsp-server/rtsp-client.c:
1715 rtsp-client: Protect saved clients watch with a mutex
1716 Fixes a crash when close() is called while merging clients
1717 in handle_tunnel(). In that case close() would destroy the
1718 watch while it is still being used in handle_tunnel().
1719 https://bugzilla.gnome.org/show_bug.cgi?id=735570
1721 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
1723 * gst/rtsp-server/rtsp-stream.c:
1724 rtsp-stream: Remove the multicast group udp sources when removing from the bin
1726 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
1728 * gst/rtsp-server/rtsp-media.c:
1729 * gst/rtsp-server/rtsp-stream.c:
1730 * gst/rtsp-server/rtsp-stream.h:
1731 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
1732 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
1733 seeking and will always continue counting the time. This leads to
1734 the NPT after a backwards seek to be something completely different
1735 to the actual seek position.
1736 https://bugzilla.gnome.org/show_bug.cgi?id=732644
1738 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
1740 * examples/test-appsrc.c:
1741 examples: fix another reference leak
1742 gst_rtsp_media_get_element() returns a new ref.
1744 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1746 * examples/test-appsrc.c:
1747 examples: unref element after usage
1748 gst_bin_get_by_name_recurse_up() returns an element
1749 reference that must be unreffed after usage.
1750 https://bugzilla.gnome.org/show_bug.cgi?id=734546
1752 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
1754 * gst/rtsp-server/rtsp-media.c:
1755 signals: Fix copy-pasto in target-state signal offset
1757 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
1761 Makefile: Add usage of build-checks step
1762 Allows building checks without running them
1764 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
1766 * gst/rtsp-server/rtsp-stream.c:
1767 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
1768 When a UDP multicast transport is used it is expected that the server listens
1769 for RTP and RTCP packets on the multicast group with the corresponding port.
1770 Without this we will never get RTCP packets from clients in multicast mode.
1771 https://bugzilla.gnome.org/show_bug.cgi?id=732238
1773 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
1778 === release 1.4.0 ===
1780 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
1786 * gst-rtsp-server.doap:
1789 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
1791 * gst/rtsp-server/rtsp-media.h:
1792 media: correct misspelled words in description
1793 https://bugzilla.gnome.org/show_bug.cgi?id=733244
1795 === release 1.3.91 ===
1797 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
1803 * gst-rtsp-server.doap:
1806 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
1808 * docs/libs/gst-rtsp-server-sections.txt:
1811 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
1813 * gst/rtsp-server/rtsp-server.c:
1814 server: implement client REMOVE filter
1816 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
1818 * gst/rtsp-server/rtsp-client.c:
1819 * gst/rtsp-server/rtsp-client.h:
1820 client: expose _close() method
1821 Expose a previously internal close method to close the client
1824 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
1826 * gst/rtsp-server/rtsp-session-pool.c:
1827 session-pool: signal session-removed outside of the lock
1828 Release the lock before emiting the session-removed signal.
1830 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
1832 * gst/rtsp-server/rtsp-client.c:
1833 * gst/rtsp-server/rtsp-server.c:
1834 * gst/rtsp-server/rtsp-session-pool.c:
1835 * gst/rtsp-server/rtsp-session.c:
1836 * gst/rtsp-server/rtsp-stream.c:
1837 filter: Release lock in filter functions
1838 Release the object lock before calling the filter functions. We need to
1839 keep a cookie to detect when the list changed during the filter
1840 callback. We also keep a hashtable to make sure we only call the filter
1841 function once for each object in case of concurrent modification.
1842 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
1844 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
1846 * gst/rtsp-server/rtsp-client.c:
1847 client: check if watch is set in handle_teardown()
1848 The unit tests run without a watch
1850 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
1852 * tests/check/gst/client.c:
1853 client tests: send teardown to cleanup session
1855 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
1857 * tests/check/gst/rtspserver.c:
1858 server tests: send teardown to cleanup session
1860 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
1862 * gst/rtsp-server/rtsp-client.c:
1863 client: keep ref to client for the session removed handler
1864 This extra ref will be dropped when all client sessions have been
1865 removed. A session is removed when a client sends teardown, closes its
1866 endpoint of the TCP connection or the sessions expires.
1867 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
1869 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
1871 * gst/rtsp-server/rtsp-client.c:
1872 * gst/rtsp-server/rtsp-session.c:
1873 * tests/check/gst/client.c:
1874 client: manage media in session as a last step
1875 Once we manage a media in a session, we can't unmanage it anymore
1876 without destroying it. Therefore, first check everything before we
1877 manage the media, otherwise if something is wrong we have no way to
1879 If we created a new session and something went wrong, remove the session
1880 again. Fixes a leak in the unit test.
1882 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
1884 * examples/test-mp4.c:
1885 * examples/test-ogg.c:
1886 examples: print 'stream ready at url' for mp4 and ogg example
1888 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
1890 * gst/rtsp-server/rtsp-client.c:
1891 * gst/rtsp-server/rtsp-sdp.c:
1892 rtsp: fix for MIKEY api change
1894 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
1896 * gst/rtsp-server/rtsp-client.c:
1897 client: free watch context only once
1898 The watch context is freed when the source is destroyed. Avoids
1899 a CRITICAL when we try to unref the context twice.
1901 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
1903 * gst/rtsp-server/rtsp-client.c:
1906 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
1908 * gst/rtsp-server/rtsp-client.c:
1909 client: protect sessions with lock
1910 Protect the list of sessions with the lock.
1911 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
1913 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
1915 * gst/rtsp-server/rtsp-client.c:
1916 Client: keep a ref to the session
1917 Don't just keep a weak ref to the session objects but use a hard ref. We
1918 will be notified when a session is removed from the pool (expired) with
1919 the new session-removed signal.
1920 Don't automatically close the RTSP connection when all the sessions of
1921 a client are removed, a client can continue to operate and it can create
1922 a new session if it wants. If you want to remove the client from the
1923 server, you have to use gst_rtsp_server_client_filter() now.
1924 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
1925 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
1927 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
1929 * gst/rtsp-server/rtsp-session-pool.c:
1930 * gst/rtsp-server/rtsp-session-pool.h:
1931 session-pool: add session-removed signal
1932 Add a signal to be notified when a session is removed from the pool.
1934 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
1936 * gst/rtsp-server/Makefile.am:
1937 * gst/rtsp-server/rtsp-server.h:
1938 Make rtsp-server.h a single-include header, use it for G-I
1939 https://bugzilla.gnome.org/show_bug.cgi?id=732411
1941 === release 1.3.90 ===
1943 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
1949 * gst-rtsp-server.doap:
1952 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
1954 * gst/rtsp-server/rtsp-stream.c:
1955 stream: crypto can be NULL
1957 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
1959 * gst/rtsp-server/rtsp-client.c:
1960 * gst/rtsp-server/rtsp-media.c:
1961 * gst/rtsp-server/rtsp-mount-points.c:
1962 introspection: add missing allow-none annotations
1963 https://bugzilla.gnome.org/show_bug.cgi?id=730952
1965 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
1967 * gst/rtsp-server/rtsp-address-pool.c:
1968 * gst/rtsp-server/rtsp-media.c:
1969 * gst/rtsp-server/rtsp-session-media.c:
1970 * gst/rtsp-server/rtsp-session-pool.c:
1971 * gst/rtsp-server/rtsp-stream-transport.c:
1972 * gst/rtsp-server/rtsp-stream.c:
1973 * gst/rtsp-server/rtsp-token.c:
1974 introspection: add (nullable) annotations to return values
1975 https://bugzilla.gnome.org/show_bug.cgi?id=730952
1977 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
1979 * gst/rtsp-server/rtsp-client.c:
1980 * gst/rtsp-server/rtsp-stream.c:
1981 gi: improve annotations
1982 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
1984 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
1986 * gst/rtsp-server/rtsp-client.c:
1987 * gst/rtsp-server/rtsp-media-factory.c:
1988 * gst/rtsp-server/rtsp-media.c:
1989 * gst/rtsp-server/rtsp-server.c:
1990 signals: use generic marshal function
1991 Use the generic C marshal function.
1992 Use more explicit type instead of G_TYPE_POINTER
1994 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
1996 * gst/rtsp-server/rtsp-context.h:
1997 context: add type macro
1999 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
2001 * gst/rtsp-server/rtsp-client.c:
2002 * gst/rtsp-server/rtsp-sdp.c:
2003 * gst/rtsp-server/rtsp-sdp.h:
2004 sdp: hide key length defines
2005 They don't have a namespace.
2007 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
2012 === release 1.3.3 ===
2014 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
2020 * gst-rtsp-server.doap:
2023 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2025 * gst/rtsp-server/rtsp-client.c:
2026 * gst/rtsp-server/rtsp-sdp.c:
2027 * gst/rtsp-server/rtsp-sdp.h:
2028 mikey: add different key length parameters
2029 Add encryption and authentication key length parameters to MIKEY. For
2030 the encoders, the key lengths are obtained from the cipher and auth
2031 algorithms set in the caps. For the decoders, they are obtained while
2032 parsing the key management from the client.
2033 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
2035 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
2037 * tests/check/gst/stream.c:
2038 stream tests: Make sure we get right multicast address from stream
2039 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
2041 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
2043 * gst/rtsp-server/rtsp-client.c:
2044 client: ref the context until rtsp watch is alive
2045 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
2047 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
2049 * gst/rtsp-server/rtsp-client.c:
2050 client: Destroy the rtsp watch after connection close
2052 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
2054 * gst/rtsp-server/rtsp-media.c:
2055 media: fix confusing comment
2057 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
2059 * gst/rtsp-server/rtsp-session.c:
2060 rtsp-session: Timeout in header.
2061 Adding the possbilty to always have timout in header.
2062 This is configurabe with setting "timeout-always-visible".
2063 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
2065 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
2070 === release 1.3.2 ===
2072 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
2079 * gst-rtsp-server.doap:
2082 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
2085 Automatic update of common submodule
2086 From 211fa5f to 1f5d3c3
2088 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
2090 * gst/rtsp-server/rtsp-client.c:
2091 client: store TCP ports in transport
2092 Store the TCP ports in the transport when we are doing RTSP over TCP.
2093 This way, we can easily get to the ports from the transport.
2094 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
2096 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2098 * gst/rtsp-server/rtsp-stream.c:
2099 stream: add signals for new RTP/RTCP encoders
2100 New signals to allow the user to configure the dynamically created
2102 https://bugzilla.gnome.org/show_bug.cgi?id=730228
2104 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
2106 * gst/rtsp-server/rtsp-media.c:
2107 * gst/rtsp-server/rtsp-media.h:
2108 media: Make suspend()/unsuspend() virtual
2109 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
2111 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2113 * gst/rtsp-server/rtsp-client.c:
2114 client: fix send-message signal marshaller
2115 Use generic marshalling for the send-message signal. It has
2116 two POINTER arguments, not just one.
2117 https://bugzilla.gnome.org/show_bug.cgi?id=729900
2119 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
2121 * tests/check/gst/media.c:
2122 tests: add and remove pads only once
2123 In this test we simulate a dynamic pad by watching the caps event.
2124 Because of renegotiation in the base payloader now, this caps is sent
2125 multiple times but we can only deal with 1 invocation, use a variable to
2126 only 'add and remove' the pad once.
2128 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
2130 * tests/check/gst/rtspserver.c:
2131 tests: add unit test for correct handling of Require headers
2132 https://bugzilla.gnome.org/show_bug.cgi?id=729426
2134 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
2136 * gst/rtsp-server/rtsp-client.c:
2137 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
2138 Servers must handle Require headers and must report a failure
2139 if they don't handle any of the Required options, see RFC 2326,
2140 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
2141 https://bugzilla.gnome.org/show_bug.cgi?id=729426
2143 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
2148 === release 1.3.1 ===
2150 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
2156 * gst-rtsp-server.doap:
2159 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
2162 Automatic update of common submodule
2163 From bcb1518 to 211fa5f
2165 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
2170 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
2172 * tests/check/gst/sessionmedia.c:
2173 tests: fix memory leak in sessionmedia unit test
2175 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
2177 * gst/rtsp-server/rtsp-client.c:
2178 client: emit a signal before sending a message
2179 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
2181 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
2183 * gst/rtsp-server/rtsp-client.c:
2184 client: pass context to send_message
2185 Pass the current context to send_message, we will need it later.
2187 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
2189 * gst/rtsp-server/rtsp-client.c:
2190 client: fix typo in comment
2192 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
2194 * gst/rtsp-server/rtsp-media.c:
2195 media: Do not stop thread twice if default_prepare() fails
2197 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
2199 * gst/rtsp-server/rtsp-client.c:
2200 client: set the watch to flushing before going to NULL
2201 First set the watch to flushing so that we unblock any current and
2202 future attempt to send data on the watch, Then set the pipeline to
2204 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
2206 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
2208 * gst/rtsp-server/rtsp-session-pool.c:
2209 * tests/check/gst/sessionpool.c:
2210 rtsp-session-pool: Fixes annotation
2211 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
2212 in the sessionpool test.
2213 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
2215 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
2217 * gst/rtsp-server/rtsp-media.c:
2218 * gst/rtsp-server/rtsp-media.h:
2219 media: make media_prepare virtual
2220 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
2222 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
2224 * gst/rtsp-server/rtsp-media.c:
2225 * tests/check/gst/media.c:
2226 media: stop the thread in more error cases
2228 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
2230 * gst/rtsp-server/rtsp-media.c:
2231 * tests/check/gst/media.c:
2232 media: allow NULL as the thread
2233 Use the default context whan passing a NULL thread.
2235 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
2237 * gst/rtsp-server/rtsp-client.c:
2238 rtsp-client: indent cleanup
2239 Coverity was moaning about unreachable code, and I think it was just
2240 confused by { being before the label. We'll see if it pops up again.
2243 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
2245 * gst/rtsp-server/rtsp-client.c:
2246 * gst/rtsp-server/rtsp-media.c:
2247 client: Add drop-backlog property
2248 When we have too many messages queued for a client (currently hardcoded
2249 to 100) we overflow and drop the messages. Add a drop-backlog property
2250 to control this behaviour. Setting this property to FALSE will retry
2251 to send the messages to the client by waiting for more room in the
2253 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
2255 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
2257 * gst/rtsp-server/rtsp-client.c:
2258 client: support for POST before GET when setting up a tunnel
2260 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
2262 * gst/rtsp-server/rtsp-client.c:
2263 client: remove watch of the second client after http tunnel setup
2264 The second client will be freed after the HTTP tunnel has been set up.
2265 Make sure it's RTSP watch is never dispatched again.
2266 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
2268 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
2270 * gst/rtsp-server/rtsp-media.c:
2271 * tests/check/gst/media.c:
2272 media: Make media_prepare() fail if port allocation fails
2273 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
2275 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
2277 * tests/check/gst/media.c:
2278 media test: cleanup the thread pool in tests
2280 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
2282 * gst/rtsp-server/rtsp-media.c:
2283 * tests/check/gst/media.c:
2284 rtsp-media: Unblock blocked streams in unprepare
2285 The streams will be blocked when a live media is prepared.
2286 The streams should be unblocked in gst_rtsp_media_unprepare.
2287 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
2289 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
2291 * gst/rtsp-server/rtsp-media.c:
2292 media: release the state lock when going to NULL
2293 Set our state to UNPREPARING and release the state-lock before
2294 setting the pipeline to the NULL state. This way, any pad-added
2295 callback will be able to take the state-lock and check that we are now
2296 unpreparing instead of deadlocking.
2297 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
2299 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
2301 * gst/rtsp-server/rtsp-media.c:
2302 media: protect status with lock
2303 Make sure we only update the status with the lock.
2305 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
2307 * gst/rtsp-server/rtsp-client.c:
2308 * gst/rtsp-server/rtsp-sdp.c:
2309 rtsp: update for MIKEY API changes
2311 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
2313 * gst/rtsp-server/rtsp-client.c:
2314 client: parse the mikey response from the client
2315 Parse the mikey response from the client and update the policy for
2318 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
2320 * gst/rtsp-server/rtsp-stream.c:
2321 * gst/rtsp-server/rtsp-stream.h:
2322 stream: add method to set crypto info
2323 Make a method to configure the crypto information of a stream.
2324 Set udpsrc in READY instead of PAUSED so that we can configure caps
2327 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
2329 * gst/rtsp-server/rtsp-client.c:
2330 client: cleanup error paths
2332 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
2334 * gst/rtsp-server/rtsp-media.c:
2337 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
2339 * examples/test-video.c:
2340 test: enable SRTP only on RTSPS
2341 We only want to enable SRTP when doing rtsp over TLS so that we can
2342 exchange the keys in a secure way.
2344 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
2346 * examples/test-video.c:
2347 test: print an error on failure
2349 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
2352 * examples/test-video.c:
2353 * gst/rtsp-server/rtsp-sdp.c:
2354 * gst/rtsp-server/rtsp-stream.c:
2355 * tests/check/Makefile.am:
2356 stream: add SRTP support
2357 Install srtp encoder and decoder elements in rtpbin
2360 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2362 * tests/check/Makefile.am:
2363 * tests/check/gst/sessionpool.c:
2364 tests: Add unit tests for sessionpool
2365 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
2367 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2369 * tests/check/gst/threadpool.c:
2370 tests: Improve code coverage of rtsp-threadpool tests
2371 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
2373 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2375 * tests/check/gst/sessionmedia.c:
2376 tests: Improve code coverage for rtsp-session-media
2377 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
2379 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2381 gobject-introspection: Add annotations to support language bindings
2382 In addition a few cosmetic changes:
2383 * Adjust the order of arguments
2384 * Fix typo: occured -> occurred
2385 * Fix indentation after Return:-clauses
2386 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
2388 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2390 * gst/rtsp-server/rtsp-stream.c:
2391 rtsp-stream: Don't mix IPv4 and IPv6 addresses
2392 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
2394 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
2396 * gst/rtsp-server/rtsp-stream.c:
2397 stream: take caps after the session manager
2398 Take the caps for the SDP after they leave the rtpbin so that we can
2399 also get the properties added by rtpbin elements.
2401 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
2403 * gst/rtsp-server/rtsp-stream.c:
2404 stream: release lock while pushing out packets
2405 Keep a cache of the transports and use this to iterate the transport
2406 while pushing packets. This allows us to release the lock early.
2407 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
2409 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
2411 * gst/rtsp-server/rtsp-client.c:
2412 * gst/rtsp-server/rtsp-client.h:
2413 rtsp-client: vmethod for modifying tunnel GET response
2414 Add a vmethod tunnel_http_response where the response to the HTTP GET
2415 for tunneled connections can be modified.
2416 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
2418 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
2420 * gst/rtsp-server/rtsp-sdp.c:
2421 sdp: make 1 media line per profile
2422 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
2423 line in the SDP for each profile. The client is then supposed to pick
2424 one of the profiles in the SETUP request. Because the m= lines have the
2425 same pt, the client also knows that only 1 option is possible.
2427 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
2429 * gst/rtsp-server/rtsp-media-factory.c:
2430 * gst/rtsp-server/rtsp-media-factory.h:
2431 * gst/rtsp-server/rtsp-media.c:
2432 factory: add profile property and pass to media and streams
2434 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
2436 * examples/test-multicast.c:
2437 * gst/rtsp-server/rtsp-sdp.c:
2438 sdp: pass multicast connection for multicast-only stream
2439 Pass the multicast address of the stream in the connection info in the
2440 SDP so that clients try a multicast connection first.
2441 Only allow multicast connections in the test-multicast example. Also
2442 increase the TTL a little.
2444 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2447 .gitignore: Ignore gcov intermediate files
2448 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
2450 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
2452 * gst/rtsp-server/rtsp-stream.c:
2453 stream: release some locks in error cases
2455 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2457 docs: Enable and fix gtk-doc warnings
2458 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
2459 * addresspool/mediafactory: Add missing annotation colon
2460 * stream: Annotate return value
2461 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
2463 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
2466 Automatic update of common submodule
2467 From fe1672e to bcb1518
2469 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
2472 Automatic update of common submodule
2473 From 1a07da9 to fe1672e
2475 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
2477 * examples/Makefile.am:
2478 examples: use LDADD for libs instead of LDFLAGS
2480 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
2483 configure: make sure releases are in .doap file
2485 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
2487 * examples/test-cgroups.c:
2488 examples: test-cgroups: don't put code with side effects into g_assert()
2489 The g_assert() might get compiled out with the right
2490 compiler/preprocessor flags.
2492 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
2494 * examples/.gitignore:
2495 examples: add cgroup test binary to .gitignore
2497 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
2499 * examples/test-cgroups.c:
2500 examples: fix cgroup test build
2501 Fixes build failure caused by compiler warning:
2502 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
2504 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
2507 .gitignore: ignore temp files created in the course of 'make check'
2509 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
2511 * gst/rtsp-server/rtsp-media.c:
2512 rtsp-media: don't loose frames handling new PLAY request
2513 If client supplied a range check if the range specifies the start point.
2514 If not, then do an accurate seek to the current position. If a start
2515 point was specified do do a key unit seek to make sure the streaming
2516 starts with decodeable frames.
2517 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
2519 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
2521 * gst/rtsp-server/rtsp-media.c:
2522 Revert "media: only flush when setting a new start position"
2523 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
2524 We need to do the flush in all cases, demuxer block currently for
2527 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
2529 * gst/rtsp-server/rtsp-media.c:
2530 media: only flush when setting a new start position
2531 Only flush the pipeline when we change the start position with
2533 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
2535 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
2537 * gst/rtsp-server/rtsp-stream.c:
2538 stream: set ttl-mc before adding the socket
2539 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
2540 never be set on socket.
2541 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
2543 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
2545 * gst/rtsp-server/rtsp-media.c:
2546 media: stop thread if media is already prepared
2547 in gst_rtsp_media_prepare() the thread is not used if media is already
2548 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
2550 https://bugzilla.gnome.org/show_bug.cgi?id=724182
2552 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
2555 build: Ship gst-rtsp-server.doap file
2557 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
2559 * tests/check/gst/rtspserver.c:
2560 tests: Fix another compiler warning with gcc
2562 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
2564 * gst/rtsp-server/rtsp-client.c:
2565 * gst/rtsp-server/rtsp-mount-points.c:
2566 * gst/rtsp-server/rtsp-stream.c:
2567 * tests/check/gst/client.c:
2568 rtsp-server: Fix lots of compiler warnings with clang
2570 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
2573 * gst-rtsp-server.doap:
2574 * tests/Makefile.am:
2575 configure: Synchronise with the configure scripts of the other modules
2577 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
2580 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
2582 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
2584 * gst/rtsp-server/rtsp-media.c:
2585 * gst/rtsp-server/rtsp-stream.c:
2586 Revert "rtsp-server: support build against last stable release"
2587 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
2588 Let us require 1.2.3 now, which is going to be released in a few
2591 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
2593 * gst/rtsp-server/rtsp-session-media.c:
2594 * gst/rtsp-server/rtsp-stream-transport.c:
2595 session: improve RTP-Info
2596 Ignore streams that can't generate RTP-Info instead of failing.
2597 Don't return the empty string when all streams are unconfigured but
2598 return NULL so that we don't generate and empty RTP-Info header.
2599 Improve docs a little.
2601 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
2603 * gst/rtsp-server/rtsp-session-media.c:
2604 Don't free rtpinfo GString when it is NULL
2605 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
2607 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
2609 * gst/rtsp-server/rtsp-media.c:
2610 media: only set keyframe flag when modifying start
2611 Only set the keyframe flag when we modify the start position. The
2612 keyframe flag should probably be ignored when no change is requested but
2613 until we can claim this is all documented properly and all demuxer
2614 implement this, avoid setting the flag.
2615 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
2617 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
2619 * gst/rtsp-server/rtsp-thread-pool.c:
2620 thread-pool: Unref source after mainloop has quit to avoid races in GLib
2621 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
2623 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
2625 * gst/rtsp-server/rtsp-stream.c:
2626 stream: handle NULL seqnum and rtptime arguments
2628 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
2630 * gst/rtsp-server/rtsp-thread-pool.c:
2631 * tests/check/gst/threadpool.c:
2632 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
2633 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
2635 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
2637 * gst/rtsp-server/rtsp-stream.c:
2638 stream: add fallback for missing stats property
2639 Use a fallback when the payloader does not have a stats property
2640 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
2642 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
2645 Automatic update of common submodule
2646 From f7bc1c3 to 1a07da9
2648 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
2650 * gst/rtsp-server/rtsp-stream.c:
2651 stream: don't leak stats structure
2652 Don't leak the stats structure and deal with NULL stats.
2654 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
2656 * gst/rtsp-server/rtsp-stream.c:
2657 stream: Get rtpinfo properties atomically from payloader
2658 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
2660 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
2662 * gst/rtsp-server/rtsp-media.c:
2663 media: refactor state change functions and signals
2664 Make functions to set the target state and the pipeline state and emit
2665 the signals from those functions.
2667 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
2669 * gst/rtsp-server/rtsp-media.c:
2670 * gst/rtsp-server/rtsp-media.h:
2671 media: add signal to notify of pending state changes
2673 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
2675 * gst/rtsp-server/rtsp-media.c:
2676 * gst/rtsp-server/rtsp-stream.c:
2677 rtsp-server: support build against last stable release
2678 Until 1.2.3 is out with the new get_type function and we
2681 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
2683 * gst/rtsp-server/rtsp-stream.c:
2684 stream: fix compilation
2686 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
2688 * gst/rtsp-server/rtsp-media.c:
2689 * gst/rtsp-server/rtsp-media.h:
2690 * gst/rtsp-server/rtsp-stream.c:
2691 * gst/rtsp-server/rtsp-stream.h:
2692 stream: add property to configure profiles
2694 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
2696 * gst/rtsp-server/rtsp-client.c:
2697 client: let stream check supported transport
2698 Delegate the check if a transport is allowed to the stream.
2699 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
2701 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
2703 * gst/rtsp-server/rtsp-stream.c:
2704 * gst/rtsp-server/rtsp-stream.h:
2705 stream: add method to check supported transport
2706 Add a method to check if a transport is supported
2708 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
2711 configure.ac: Only check for gstreamer-check, not check
2712 We include check in gstreamer-check since quite some time now.
2714 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
2716 * gst/rtsp-server/rtsp-session-media.c:
2717 * gst/rtsp-server/rtsp-stream-transport.c:
2718 * gst/rtsp-server/rtsp-stream.c:
2719 * gst/rtsp-server/rtsp-stream.h:
2720 stream: return clock-rate from get_rtpinfo
2721 And use it to correct the rtptime to the requested start-time.
2722 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
2724 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
2726 * gst/rtsp-server/rtsp-session-media.c:
2727 * gst/rtsp-server/rtsp-stream-transport.c:
2728 * gst/rtsp-server/rtsp-stream-transport.h:
2729 session-media: calculate start-time
2731 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
2733 * gst/rtsp-server/rtsp-stream-transport.c:
2734 * gst/rtsp-server/rtsp-stream.c:
2735 * gst/rtsp-server/rtsp-stream.h:
2736 stream: also return the running-time
2737 Return the running-time in the rtpinfo as well.
2739 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
2741 * gst/rtsp-server/rtsp-client.c:
2742 * gst/rtsp-server/rtsp-session-media.c:
2743 * gst/rtsp-server/rtsp-session-media.h:
2744 * gst/rtsp-server/rtsp-stream-transport.c:
2745 * gst/rtsp-server/rtsp-stream-transport.h:
2746 session-media: let the session-media make the RTPInfo
2747 Add method to create the RTPInfo for a stream-transport.
2748 Add method to create the RTPInfo for all stream-transports in a
2750 Use the session-media RTPInfo code in client. This allows us to refactor
2751 another method to link the TCP callbacks.
2753 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
2755 mount-points: sort sequence before g_sequence_lookup
2756 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
2757 sort sequence if dirty, otherwise lookup will fail.
2758 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
2760 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
2763 configure: rename package from gst-rtsp to gst-rtsp-server
2764 To match git module name and avoid confusion with the
2765 rtsp lib in gst-plugins-base and rtsp plugin in -good.
2767 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
2770 configure: bump core/base/good requirement to 1.2.0
2771 Bump to released stable version and make implicit
2772 requirements explicit.
2774 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
2779 Fix broken gettext setup which is not used anyway
2781 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
2784 Automatic update of common submodule
2785 From dbedaa0 to d48bed3
2787 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
2789 * gst/rtsp-server/rtsp-client.c:
2790 * gst/rtsp-server/rtsp-media.c:
2791 * gst/rtsp-server/rtsp-media.h:
2792 media: add setup_sdp vmethod
2793 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
2794 gst_rtsp_media_setup_sdp.
2795 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
2797 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
2799 * gst/rtsp-server/rtsp-stream.c:
2800 rtsp-stream: Check return value of sscanf
2801 streamid is only valid if sscanf matched something.
2803 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
2805 * gst/rtsp-server/rtsp-client.c:
2806 rtsp-client: Fix iteration
2807 Wouldn't even enter the code block otherwise (i++ was used as the check
2808 and not the postfix).
2810 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
2812 * gst/rtsp-server/rtsp-client.c:
2813 * gst/rtsp-server/rtsp-client.h:
2814 client: add vmethod to configure media and streams
2815 Implement a vmethod that can be used to configure the media and the
2816 streams based on the current context. Handle the blocksize handling in
2817 the default handler.
2818 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
2820 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
2823 Make git ignore more unit test binaries
2825 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
2827 * gst/rtsp-server/rtsp-address-pool.h:
2828 * gst/rtsp-server/rtsp-auth.h:
2829 * gst/rtsp-server/rtsp-client.h:
2830 * gst/rtsp-server/rtsp-context.h:
2831 * gst/rtsp-server/rtsp-media-factory-uri.h:
2832 * gst/rtsp-server/rtsp-media-factory.h:
2833 * gst/rtsp-server/rtsp-media.h:
2834 * gst/rtsp-server/rtsp-mount-points.h:
2835 * gst/rtsp-server/rtsp-server.h:
2836 * gst/rtsp-server/rtsp-session-media.h:
2837 * gst/rtsp-server/rtsp-session-pool.h:
2838 * gst/rtsp-server/rtsp-session.h:
2839 * gst/rtsp-server/rtsp-stream-transport.h:
2840 * gst/rtsp-server/rtsp-stream.h:
2841 * gst/rtsp-server/rtsp-thread-pool.h:
2842 * gst/rtsp-server/rtsp-token.h:
2843 rtsp-server: add padding to many public structures
2844 Not mini objects though, since they are not subclassable
2845 anyway, nor kept on the stack or inlined in a structure.
2847 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
2849 media: add new create_rtpbin vmethod
2850 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
2851 https://bugzilla.gnome.org/show_bug.cgi?id=719734
2853 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
2855 * tests/check/gst/media.c:
2856 tests: fix memory leak, free test's thread pool
2857 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
2859 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
2861 * gst/rtsp-server/rtsp-stream-transport.c:
2862 stream-transport: free url in finalize
2864 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
2866 * gst/rtsp-server/rtsp-media.c:
2867 media: also do state change in suspended state
2869 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
2871 * gst/rtsp-server/rtsp-client.c:
2872 * gst/rtsp-server/rtsp-media.c:
2873 media: also handle prepare and range in suspended state
2874 When we are suspended, we are already prepared.
2875 We can get the range in the suspended state.
2877 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
2879 * tests/check/Makefile.am:
2880 * tests/check/gst/sessionmedia.c:
2881 check: add test for uri in setup
2882 Added unit tests for the new functionality in GstRTSPStreamTransport.
2883 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
2885 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
2887 * gst/rtsp-server/rtsp-client.c:
2888 client: store setup uri and use in PLAY response
2889 Store the uri used when doing the setup and use that in the PLAY
2891 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
2893 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
2895 * gst/rtsp-server/rtsp-stream-transport.c:
2896 * gst/rtsp-server/rtsp-stream-transport.h:
2897 stream-transport: add method to get/set url
2899 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
2901 * gst/rtsp-server/rtsp-client.c:
2902 client: suspend after SDP and unsuspend before PLAYING
2903 Based on patches by Ognyan Tonchev <ognyan@axis.com>
2904 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
2906 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
2908 * gst/rtsp-server/rtsp-media-factory.c:
2909 * gst/rtsp-server/rtsp-media-factory.h:
2910 * gst/rtsp-server/rtsp-media.c:
2911 * gst/rtsp-server/rtsp-media.h:
2912 * gst/rtsp-server/rtsp-session-media.c:
2913 * gst/rtsp-server/rtsp-session.c:
2914 * tests/check/gst/media.c:
2915 * tests/check/gst/mediafactory.c:
2916 media: add suspend modes
2917 Add support for different suspend modes. The stream is suspended right after
2918 producing the SDP and after PAUSE. Different suspend modes are available that
2919 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
2920 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
2921 state and RESET will bring the pipeline to the NULL state.
2922 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
2923 this means that the pipeline needs to be prerolled again.
2924 Base on patches by Ognyan Tonchev <ognyan@axis.com>
2925 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2927 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
2929 * gst/rtsp-server/rtsp-media.c:
2930 media: start live streams in blocked state
2931 Start live streams in the blocked state and make them preroll using the
2932 messages. This ensure that no data is played by the sink until we explicitly
2933 unblock the stream right before going to PLAYING.
2934 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2936 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
2938 * gst/rtsp-server/rtsp-media.c:
2939 media: refactor starting and waiting for preroll
2940 Based on patches from Ognyan Tonchev <ognyan@axis.com>
2941 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2943 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
2945 * gst/rtsp-server/rtsp-stream.c:
2946 * gst/rtsp-server/rtsp-stream.h:
2947 stream: add API to block streams
2948 Add an API to block on the streams and make it post a message.
2949 Based on patch by Ognyan Tonchev <ognyan@axis.com>
2950 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2952 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
2954 * docs/libs/Makefile.am:
2955 docs: Specify the override file
2956 Even if it's empty (for now) it avoids make distcheck complaining
2958 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
2960 * gst/rtsp-server/rtsp-media.c:
2961 media: move default implementations to where they are used
2963 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
2965 * gst/rtsp-server/rtsp-media.c:
2966 media: take the right lock in gst_rtsp_media_set_pipeline_state()
2967 We need to take the state_lock when calling this method.
2969 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
2971 * gst/rtsp-server/rtsp-media.c:
2972 media: handle add-added on non-bins too
2973 Handle dynamic payloaders that are not bins, as used in the unit-test.
2975 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2977 * gst/rtsp-server/rtsp-media-factory.c:
2978 * gst/rtsp-server/rtsp-media-factory.h:
2979 * gst/rtsp-server/rtsp-media.c:
2980 rtsp-media/-factory: Fix request pad name comments
2981 These must be escaped for gtk-doc to parse the comments without warnings.
2983 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
2985 rtsp-media: remove transports if media is in error status
2986 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
2987 trying to change to GST_STATE_NULL and media is in error status, we
2988 remove all transports.
2989 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
2991 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
2993 * gst/rtsp-server/rtsp-media.c:
2994 rtsp-media: use element metadata to find payloader
2995 Use the element metadata to find the payloader instead of checking
2997 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
2999 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
3001 rtsp-stream: add getter for payload type
3002 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
3003 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
3004 element and create the stream with this one instead of the dynpay%d
3006 https://bugzilla.gnome.org/show_bug.cgi?id=712396
3008 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3010 * gst/rtsp-server/rtsp-client.c:
3011 * gst/rtsp-server/rtsp-context.h:
3012 * gst/rtsp-server/rtsp-media.c:
3013 * gst/rtsp-server/rtsp-mount-points.c:
3014 * gst/rtsp-server/rtsp-server.c:
3015 * gst/rtsp-server/rtsp-token.c:
3016 rtsp-*: Refer to NULL as a constant in comments
3018 https://bugzilla.gnome.org/show_bug.cgi?id=714988
3020 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3022 rtsp-*: Fix type name typos in comments
3023 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
3024 * rtsp-auth: Refer to part of constant name as text
3025 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
3026 * rtsp-session-media: Fix GstRTSPSessionMedia typo
3027 * rtsp-stream: Fix typo when refering to GstBin
3028 https://bugzilla.gnome.org/show_bug.cgi?id=714988
3030 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3033 * docs/libs/gst-rtsp-server-docs.sgml:
3034 * docs/libs/gst-rtsp-server-sections.txt:
3035 docs: Improve documentation
3036 * Include annotation-glossary to quiet gtk-doc
3037 * Rename remaining ClientState -> Context
3038 * Rename object hierarchy file
3039 * Remove stale chapter references
3040 * Add missing function and object references
3041 * Include missing GstRTSPAddressPoolResult
3042 https://bugzilla.gnome.org/show_bug.cgi?id=714988
3044 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
3046 * gst/rtsp-server/rtsp-client.c:
3047 * gst/rtsp-server/rtsp-server.c:
3048 * gst/rtsp-server/rtsp-session-pool.c:
3049 * gst/rtsp-server/rtsp-session.c:
3050 * gst/rtsp-server/rtsp-stream.c:
3051 rtsp-server: sprinkle some allow-none annotations for g-i
3053 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
3055 * gst/rtsp-server/rtsp-stream.c:
3056 * gst/rtsp-server/rtsp-stream.h:
3057 stream: add method to filter transports
3058 Add a method to safely iterate and collect the stream transports
3059 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
3061 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
3063 * gst/rtsp-server/rtsp-client.c:
3064 * gst/rtsp-server/rtsp-server.c:
3065 * gst/rtsp-server/rtsp-session-pool.c:
3066 * gst/rtsp-server/rtsp-session.c:
3067 rtsp: allow NULL func in filters
3068 Passing a null function make the filters return a list of
3071 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
3073 * gst/rtsp-server/rtsp-address-pool.c:
3074 * tests/check/gst/addresspool.c:
3075 address-pool: fix address increment
3076 Use a guint instead of guint8 to increment the address. It's still not
3077 completely correct because a guint might not be able to hold the complete
3078 address range, but that's an enhacement for later.
3079 Add unit test to test improved behaviour.
3080 https://bugzilla.gnome.org/show_bug.cgi?id=708237
3082 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
3084 * gst/rtsp-server/rtsp-client.c:
3085 * tests/check/gst/client.c:
3086 client: allow absolute path in requests
3087 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
3089 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
3091 * gst/rtsp-server/rtsp-client.c:
3092 * gst/rtsp-server/rtsp-client.h:
3093 client: make make_path_from_uri a vmethod
3095 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
3097 * docs/libs/gst-rtsp-server-sections.txt:
3098 * gst/rtsp-server/rtsp-stream.c:
3099 * gst/rtsp-server/rtsp-stream.h:
3100 * tests/check/Makefile.am:
3101 * tests/check/gst/stream.c:
3102 stream: Add functions to get rtp and rtcp sockets
3103 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
3105 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
3107 * gst/rtsp-server/rtsp-context.c:
3108 * gst/rtsp-server/rtsp-context.h:
3109 context: defing a GType for the context
3110 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
3112 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
3114 * gst/rtsp-server/Makefile.am:
3115 * gst/rtsp-server/rtsp-auth.c:
3116 * gst/rtsp-server/rtsp-context.c:
3117 * gst/rtsp-server/rtsp-media.c:
3118 * gst/rtsp-server/rtsp-mount-points.c:
3119 * gst/rtsp-server/rtsp-server.h:
3120 * gst/rtsp-server/rtsp-session-media.c:
3121 * gst/rtsp-server/rtsp-session.c:
3122 * gst/rtsp-server/rtsp-stream.c:
3123 Fixed several GIR warnings
3125 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
3127 * gst/rtsp-server/rtsp-auth.c:
3130 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3132 * tests/check/Makefile.am:
3133 * tests/check/gst/token.c:
3134 tests: Add unit tests for token
3135 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
3137 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3139 * gst/rtsp-server/rtsp-token.c:
3140 token: Validate args for gst_rtsp_token_is_allowed
3141 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
3143 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3145 * gst/rtsp-server/rtsp-token.c:
3146 token: Fix bug when creating empty token
3147 We always want to have a valid GstStructure in the token.
3148 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
3150 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
3152 * gst/rtsp-server/rtsp-thread-pool.c:
3153 thread-pool: avoid race in shutdown
3154 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
3155 don't actually stop the mainloop ever. Solve this race by adding an idle source
3156 to the mainloop that calls the _quit. This way we immediately exit the mainloop
3157 if quit was called before we started it.
3159 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3161 * tests/check/Makefile.am:
3162 * tests/check/gst/permissions.c:
3163 tests: Add unit tests for permissions
3164 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
3166 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3168 * tests/check/gst/mediafactory.c:
3169 tests: Test mediafactory permissions
3170 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
3172 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3174 * gst/rtsp-server/rtsp-permissions.c:
3175 permissions: Fix refcounting when adding/removing roles
3176 Previously a role that was removed was unreffed twice, and when
3177 replacing an existing role the replaced role was freed while still being
3178 referenced. Both bugs are now fixed.
3179 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
3181 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3183 * tests/check/gst/media.c:
3184 * tests/check/gst/mediafactory.c:
3185 * tests/check/gst/rtspserver.c:
3186 tests: Check gst_rtsp_url_parse return value
3187 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
3189 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
3192 Automatic update of common submodule
3193 From 865aa20 to dbedaa0
3195 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
3197 * gst/rtsp-server/rtsp-server.c:
3198 rtsp-server: Fix socket leak
3199 https://bugzilla.gnome.org/show_bug.cgi?id=710088
3201 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
3203 * gst/rtsp-server/rtsp-session-pool.c:
3204 rtsp-session-pool: Make sure session IDs are properly URI-escaped
3205 https://bugzilla.gnome.org/show_bug.cgi?id=643812
3207 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
3209 * examples/.gitignore:
3210 * examples/test-video.c:
3211 examples: fix compilation when WITH_AUTH is defined
3212 https://bugzilla.gnome.org/show_bug.cgi?id=710228
3214 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
3217 gitignore: Add new test binary
3219 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
3221 * tests/check/Makefile.am:
3222 * tests/check/gst/threadpool.c:
3223 thread-pool: Add unit test for the thread pools
3224 https://bugzilla.gnome.org/show_bug.cgi?id=710228
3226 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
3228 * gst/rtsp-server/rtsp-thread-pool.c:
3229 thread-pool: Fix thread leak when reusing threads
3230 https://bugzilla.gnome.org/show_bug.cgi?id=709730
3232 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
3234 * gst/rtsp-server/rtsp-server.c:
3235 * tests/check/gst/rtspserver.c:
3236 tests: fixed racy behavior in rtspserver tests
3237 https://bugzilla.gnome.org/show_bug.cgi?id=710078
3239 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3241 * tests/check/gst/addresspool.c:
3242 tests: Improve address pool unit tests
3243 Add a range with mixed IPV4 and IPV6 addresses to pool.
3244 Get an IPV4 address from an IPV6-only pool.
3245 Get an IPV6 address from an IPV4-only pool.
3246 Reserve a IPV6 address from an IPV4-only pool.
3247 Check for unicast addresses in multicast-only pool.
3248 Check for unicast addresses in uni-/multicast-mixed pool.
3249 https://bugzilla.gnome.org/show_bug.cgi?id=710128
3251 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3253 * gst/rtsp-server/rtsp-client.c:
3254 client: append query string in PAUSE/PLAY/TEARDOWN as well
3256 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
3258 * gst/rtsp-server/rtsp-client.c:
3259 client: Add query to control path
3260 If the SETUP url contains a query it must be appended to the control
3261 path so that it matches any already created stream in the media. The
3262 query will also be appended to the session media path.
3264 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3266 * gst/rtsp-server/rtsp-media.c:
3267 rtsp-media: remove old line
3269 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
3271 * gst/rtsp-server/rtsp-stream.c:
3272 stream: Correct control comparison
3273 https://bugzilla.gnome.org/show_bug.cgi?id=709176
3275 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3277 * gst/rtsp-server/rtsp-media.c:
3278 media: Check dynamically if the pipeline supports seeking
3279 We should not depend on whether or not the pipeline state change
3280 returned NO_PREROLL or not. A media could dynamically change its
3281 element and switch from seekable to non seekable so it's best to test
3282 the seekable nature of the pipeline dynamically when we try to do a seek.
3284 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3286 * gst/rtsp-server/rtsp-media.c:
3287 media: Return FALSE if seeking is not supported
3289 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3291 * gst/rtsp-server/rtsp-media.c:
3292 rtsp-media: don't seek accurate by default
3293 Accurate seeking is perhaps a little overkill in the most common situation and
3294 causes some formats (mp3) over slow media to seek extremely slowly.
3296 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
3298 * tests/check/gst/rtspserver.c:
3299 tests: fix unit test
3300 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
3302 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
3304 * gst/rtsp-server/rtsp-client.c:
3305 client: Reply 400 if media cannot be constructed
3306 Reply 400 Bad Request instead of 503 Service Unavailable if media
3307 cannot be constructed in SETUP.
3308 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
3310 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
3312 * gst/rtsp-server/rtsp-client.c:
3313 client: Send setup reply once only
3314 If find_media() failed in handle_setup_request() two replies was sent.
3315 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
3317 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
3320 Automatic update of common submodule
3321 From 6b03ba7 to 865aa20
3323 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
3325 * gst/rtsp-server/rtsp-server.c:
3326 server: Emit client-connected signal earlier
3327 Emit client-connected before the client ref is given to a GSource,
3328 otherwise client-connected can be emitted after the client object has
3331 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
3333 * gst/rtsp-server/rtsp-address-pool.c:
3334 * gst/rtsp-server/rtsp-address-pool.h:
3335 * gst/rtsp-server/rtsp-stream.c:
3336 * tests/check/gst/addresspool.c:
3337 addresspool: return reason of failure
3338 Let gst_rtsp_address_pool_reserve_address() return the reason why
3339 the address could not be reserved.
3340 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
3342 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
3345 autogen.sh: Sync behaviour with other GStreamer modules
3346 Allows building from outside of tree amongst other things
3348 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
3351 Automatic update of common submodule
3352 From b613661 to 6b03ba7
3354 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
3357 Automatic update of common submodule
3358 From 74a6857 to b613661
3360 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
3363 Automatic update of common submodule
3364 From 01a7a46 to 74a6857
3366 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
3368 * gst/rtsp-server/rtsp-client.c:
3369 client: Do not read beyond end of path string
3370 If the setup was done without a control url, make sure we don't try to read the
3371 non-existing control string and crash.
3373 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3375 * gst/rtsp-server/rtsp-client.c:
3376 client: Fix RTPInfo header
3377 Refactor the method to make the content_base.
3378 Use the content-base and the control url to construct the RTPInfo
3381 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3383 * gst/rtsp-server/rtsp-client.c:
3384 client: map url to path only in describe
3385 Only map the request url to a path in the DESCRIBE method. The SDP then
3386 contains the base and control urls that should be used to SETUP/PAUSE/
3387 PLAY/TEARDOWN the media.
3389 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3391 * gst/rtsp-server/rtsp-client.c:
3392 Revert "client: map URL to path in requests"
3393 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
3394 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
3395 contains the base and control urls which are used in the SETUP, PLAY,
3396 PAUSE and TEARDOWN requests.
3398 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3400 * gst/rtsp-server/rtsp-client.c:
3401 client: map URL to path in requests
3403 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3405 * gst/rtsp-server/rtsp-client.c:
3406 * gst/rtsp-server/rtsp-mount-points.c:
3407 * gst/rtsp-server/rtsp-mount-points.h:
3408 mount-points: make vmethod to make path from uri
3409 Make a vmethod to transform an url into a path. The path is then used to lookup
3410 the factory. This makes it possible to also use other bits of the url, such as
3411 the query parameters, to locate the factory.
3413 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
3415 * gst/rtsp-server/rtsp-thread-pool.c:
3416 * gst/rtsp-server/rtsp-thread-pool.h:
3417 thread-pool: Add cleanup to wait for the threadpool to finish
3418 Also fix race condition if two threads are asking for the first
3419 thread from the thread pool at once. This would case two internal
3420 GThreadPools to be created.
3421 https://bugzilla.gnome.org/show_bug.cgi?id=707753
3423 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
3425 * gst/rtsp-server/rtsp-client.c:
3426 * tests/check/gst/client.c:
3427 client: free threadpool
3428 https://bugzilla.gnome.org/show_bug.cgi?id=707638
3430 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
3432 * tests/check/gst/mountpoints.c:
3433 mountpoints tests: unref matched factories
3434 https://bugzilla.gnome.org/show_bug.cgi?id=707638
3436 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
3438 * tests/check/gst/media.c:
3439 media tests: unref thread pool and caps
3440 https://bugzilla.gnome.org/show_bug.cgi?id=707638
3442 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
3444 * gst/rtsp-server/rtsp-auth.c:
3445 * gst/rtsp-server/rtsp-media-factory.c:
3446 * gst/rtsp-server/rtsp-media.c:
3447 auth, media, media-factory: unref permissions
3448 https://bugzilla.gnome.org/show_bug.cgi?id=707638
3450 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3452 * examples/Makefile.am:
3453 Makefile: add rule for appsrc example
3455 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3457 * examples/test-appsrc.c:
3458 tests: add appsrc example
3459 Add an example on how to use appsrc to feed the server pipeline with data.
3461 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
3463 * gst/rtsp-server/rtsp-client.c:
3464 rtsp-client: remove query part from content-base string
3465 Make sure that after the control url has been resolved, it's
3466 not a part of the query-string.
3467 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
3469 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3471 * gst/rtsp-server/rtsp-client.c:
3472 client: don't check url in response
3473 There is no url or method in the response to check
3475 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3477 * gst/rtsp-server/rtsp-client.c:
3478 * gst/rtsp-server/rtsp-client.h:
3479 Add handle-response signal for when we receive a GET_PARAMETER response
3481 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3483 * gst/rtsp-server/rtsp-server.c:
3484 Fix gst_rtsp_server_client_filter, using wrong variable type
3486 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
3488 * gst/rtsp-server/rtsp-media-factory-uri.c:
3489 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
3490 For AAC we need to check for framed=true instead of parsed=true.
3491 https://bugzilla.gnome.org/show_bug.cgi?id=701384
3493 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3495 * gst/rtsp-server/rtsp-stream.c:
3496 stream: optimize pipeline for protocols
3497 When TCP is not an allowed protocol for the stream, avoid creating the
3498 appsrc/appsink/queue and tee elements.
3500 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3502 * gst/rtsp-server/rtsp-media.c:
3503 media: set protocols on streams
3505 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3507 * gst/rtsp-server/rtsp-client.c:
3508 client: use protocols supported by stream
3510 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3512 * gst/rtsp-server/rtsp-media-factory.c:
3513 * gst/rtsp-server/rtsp-media.c:
3514 * gst/rtsp-server/rtsp-stream.c:
3515 media-factory: allow all protocols
3517 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3519 * gst/rtsp-server/rtsp-media.c:
3520 media: configure protocols in new streams
3522 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3524 * gst/rtsp-server/rtsp-stream.c:
3525 * gst/rtsp-server/rtsp-stream.h:
3526 stream: add protocols property
3528 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3530 * gst/rtsp-server/rtsp-media.c:
3531 rtsp-media: send state in "new-state" signal
3532 https://bugzilla.gnome.org/show_bug.cgi?id=705110
3534 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
3537 build: add subdir-objects to AM_INIT_AUTOMAKE
3538 Fixes warnings with automake 1.14
3539 https://bugzilla.gnome.org/show_bug.cgi?id=705350
3541 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3543 * docs/libs/gst-rtsp-server-sections.txt:
3544 * gst/rtsp-server/rtsp-client.c:
3545 * gst/rtsp-server/rtsp-server.c:
3546 * gst/rtsp-server/rtsp-server.h:
3547 server: add method to iterate clients of server
3549 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3551 * gst/rtsp-server/rtsp-media.c:
3552 * gst/rtsp-server/rtsp-media.h:
3553 Add vmethod for rtsp-media subclass to access rtpbin
3555 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3557 * gst/rtsp-server/rtsp-client.h:
3558 small documentation fix
3560 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3562 * gst/rtsp-server/rtsp-client.c:
3563 Do not take range header if range is invalid
3565 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3567 * docs/libs/gst-rtsp-server-sections.txt:
3568 * gst/rtsp-server/rtsp-media.c:
3569 media: add docs for new method
3571 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3573 * gst/rtsp-server/rtsp-media.c:
3574 * gst/rtsp-server/rtsp-media.h:
3575 Add API to rtsp-media set the pipeline's state
3577 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3579 * gst/rtsp-server/rtsp-media.c:
3580 Update current position/duration when gst_rtsp_media_get_range_string is called
3582 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3584 * examples/test-cgroups.c:
3585 tests: add some more docs
3587 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3589 * examples/test-cgroups.c:
3590 * gst/rtsp-server/Makefile.am:
3591 * gst/rtsp-server/rtsp-auth.c:
3592 * gst/rtsp-server/rtsp-auth.h:
3593 * gst/rtsp-server/rtsp-client.c:
3594 * gst/rtsp-server/rtsp-client.h:
3595 * gst/rtsp-server/rtsp-context.c:
3596 * gst/rtsp-server/rtsp-context.h:
3597 * gst/rtsp-server/rtsp-params.c:
3598 * gst/rtsp-server/rtsp-params.h:
3599 * gst/rtsp-server/rtsp-server.c:
3600 * gst/rtsp-server/rtsp-thread-pool.c:
3601 * gst/rtsp-server/rtsp-thread-pool.h:
3602 * tests/check/gst/client.c:
3603 ClientState -> Context
3604 Rename the clientstate to context and put the code in a separate file.
3606 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3608 * examples/test-auth.c:
3609 * gst/rtsp-server/rtsp-auth.c:
3610 * gst/rtsp-server/rtsp-auth.h:
3611 auth: add support for default token
3612 The default token is used when the user is not authenticated and can be used to
3613 give minimal permissions.
3615 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3617 * examples/test-auth.c:
3618 * gst/rtsp-server/rtsp-auth.c:
3619 auth: use defines when possible
3621 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3623 * gst/rtsp-server/rtsp-address-pool.c:
3624 address-pool: improve docs
3626 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3628 * gst/rtsp-server/rtsp-permissions.c:
3629 permissions: add the role to the copy
3631 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
3633 * gst/rtsp-server/rtsp-permissions.c:
3634 permissions: Also copy the roles
3636 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
3638 * gst/rtsp-server/rtsp-permissions.c:
3639 permissions: Make it build
3641 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3643 * gst/rtsp-server/rtsp-address-pool.h:
3646 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3648 * docs/libs/gst-rtsp-server-sections.txt:
3649 * gst/rtsp-server/rtsp-auth.c:
3650 * gst/rtsp-server/rtsp-auth.h:
3651 * gst/rtsp-server/rtsp-media.c:
3652 * gst/rtsp-server/rtsp-session-media.c:
3653 * gst/rtsp-server/rtsp-stream-transport.c:
3654 * gst/rtsp-server/rtsp-stream-transport.h:
3655 * gst/rtsp-server/rtsp-stream.c:
3656 * tests/check/gst/client.c:
3659 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3661 * docs/libs/gst-rtsp-server-sections.txt:
3662 * gst/rtsp-server/rtsp-address-pool.c:
3663 * gst/rtsp-server/rtsp-address-pool.h:
3664 * tests/check/gst/addresspool.c:
3665 * tests/check/gst/rtspserver.c:
3666 address-pool: cleanups
3667 Remove redundant method, improve docs.
3669 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3671 * docs/libs/gst-rtsp-server-sections.txt:
3672 * gst/rtsp-server/rtsp-auth.h:
3673 * gst/rtsp-server/rtsp-permissions.c:
3674 * gst/rtsp-server/rtsp-permissions.h:
3675 * gst/rtsp-server/rtsp-token.c:
3678 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3680 * gst/rtsp-server/rtsp-permissions.c:
3681 permissions: implement _remove_role
3683 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3685 * gst/rtsp-server/rtsp-permissions.c:
3686 permissions: update docs
3688 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3690 * tests/check/gst/client.c:
3691 tests: simplify tests
3692 Client settings are now disabled by default so we don't need an auth
3693 module to disable them.
3695 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3697 * gst/rtsp-server/rtsp-auth.c:
3698 auth: add default authorizations
3699 When no auth module is specified, use our table of defaults to look up the
3700 default value of the check instead of always allowing everything. This was
3701 we can disallow client settings by default.
3703 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3706 README: update readme
3708 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3710 * gst/rtsp-server/rtsp-thread-pool.c:
3711 * gst/rtsp-server/rtsp-thread-pool.h:
3712 thread-pool: add more docs
3714 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3716 * gst/rtsp-server/rtsp-thread-pool.c:
3717 * gst/rtsp-server/rtsp-thread-pool.h:
3718 thread-pool: fix race in thread reuse
3719 If we try to reuse a thread right after we made it stop, we end up using a
3720 stopped thread. Catch this case and only reuse threads that are not stopping.
3722 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3724 * gst/rtsp-server/rtsp-server.c:
3725 server: add small debug
3727 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3729 * tests/check/gst/client.c:
3731 Add some permissions to media so we can use the auth and enable
3734 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3736 * gst/rtsp-server/rtsp-client.c:
3737 client: support pushed context in handle_request
3738 If we already have a pushed state, reuse it and add our own things. This makes
3739 it easier to write tests.
3741 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3743 * gst/rtsp-server/rtsp-auth.c:
3744 auth: don't auth on methods
3745 Don't authorize on methods anymore but on the resources that we
3746 try to access, this is more flexible.
3747 Move the authorization checks to where they are needed and let the
3748 check return the response on error.
3750 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3752 * gst/rtsp-server/rtsp-mount-points.c:
3753 mount-points: add some debug
3755 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3757 * tests/check/gst/client.c:
3758 tests: almost fix test
3760 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3762 * gst/rtsp-server/rtsp-auth.c:
3763 * gst/rtsp-server/rtsp-auth.h:
3764 * gst/rtsp-server/rtsp-client.c:
3765 * gst/rtsp-server/rtsp-client.h:
3766 * gst/rtsp-server/rtsp-server.c:
3767 * gst/rtsp-server/rtsp-server.h:
3768 auth: let the auth module check client_settings
3769 Let the auth module decide if client settings are allowed for the
3772 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3774 * gst/rtsp-server/rtsp-token.c:
3775 * gst/rtsp-server/rtsp-token.h:
3776 token: add method to check boolean permission
3778 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3780 * examples/test-auth.c:
3781 * examples/test-cgroups.c:
3782 * gst/rtsp-server/rtsp-token.c:
3783 * gst/rtsp-server/rtsp-token.h:
3784 token: simplify token constructor
3785 Use variable arguments to make easier API.
3787 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3789 * examples/test-auth.c:
3790 * examples/test-cgroups.c:
3791 * gst/rtsp-server/rtsp-media-factory.c:
3792 * gst/rtsp-server/rtsp-media-factory.h:
3793 media-factory: add convenience API for factory
3795 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3797 * examples/test-auth.c:
3798 * examples/test-cgroups.c:
3799 * gst/rtsp-server/rtsp-permissions.c:
3800 * gst/rtsp-server/rtsp-permissions.h:
3801 permissions: simplify API a little
3802 Avoid passing GstStructure in the add_role method, use varargs instead
3803 to construct the structure behind the scenes. We can then also use the
3804 structure name as the role and simplify some more logic.
3806 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3808 * gst/rtsp-server/rtsp-auth.c:
3811 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3813 * gst/rtsp-server/rtsp-auth.c:
3814 * gst/rtsp-server/rtsp-auth.h:
3815 * gst/rtsp-server/rtsp-client.c:
3816 auth: handle unauthorized response
3817 Move handling of the unauthorized response to the auth module, it can add
3818 the appropriate headers to request authorization for the required method
3819 much better than the client.
3821 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3823 * gst/rtsp-server/rtsp-client.c:
3824 * gst/rtsp-server/rtsp-client.h:
3825 client: allow for sending any message, not only requests
3826 Change the _send_request() method to _send_message() so that we
3827 can both send requests and replies.
3829 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3831 * docs/libs/gst-rtsp-server-sections.txt:
3832 * gst/rtsp-server/rtsp-server.h:
3835 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3837 * examples/test-video.c:
3838 * gst/rtsp-server/rtsp-auth.c:
3839 * gst/rtsp-server/rtsp-auth.h:
3840 * gst/rtsp-server/rtsp-server.c:
3841 * gst/rtsp-server/rtsp-server.h:
3842 auth: move TLS handling to auth module
3843 Remove the TLS settings on the server and move it to the auth module because
3844 that is where security related bits go.
3846 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3848 * gst/rtsp-server/rtsp-client.c:
3849 * gst/rtsp-server/rtsp-client.h:
3850 client: add state push/pop
3852 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3854 * gst/rtsp-server/rtsp-client.c:
3855 * gst/rtsp-server/rtsp-client.h:
3856 client: add connection to state
3858 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3860 * gst/rtsp-server/rtsp-mount-points.c:
3861 mount-points: fix debug
3863 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3865 * tests/check/gst/media.c:
3866 tests: fix media test
3868 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3870 * gst/rtsp-server/rtsp-thread-pool.c:
3871 thread-pool: we don't require a state
3873 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3875 * gst/rtsp-server/rtsp-server.c:
3876 server: let context ref the server
3877 So that we don't risk losing the server object early anc crash.
3879 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3881 * tests/check/gst/client.c:
3882 tests: fix client test
3884 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3887 * docs/libs/gst-rtsp-server-docs.sgml:
3888 * docs/libs/gst-rtsp-server-sections.txt:
3889 * gst/rtsp-server/rtsp-address-pool.c:
3890 * gst/rtsp-server/rtsp-auth.c:
3891 * gst/rtsp-server/rtsp-client.c:
3892 * gst/rtsp-server/rtsp-client.h:
3893 * gst/rtsp-server/rtsp-media-factory-uri.c:
3894 * gst/rtsp-server/rtsp-media-factory.c:
3895 * gst/rtsp-server/rtsp-media-factory.h:
3896 * gst/rtsp-server/rtsp-media.c:
3897 * gst/rtsp-server/rtsp-mount-points.c:
3898 * gst/rtsp-server/rtsp-params.c:
3899 * gst/rtsp-server/rtsp-permissions.c:
3900 * gst/rtsp-server/rtsp-sdp.c:
3901 * gst/rtsp-server/rtsp-server.c:
3902 * gst/rtsp-server/rtsp-server.h:
3903 * gst/rtsp-server/rtsp-session-media.c:
3904 * gst/rtsp-server/rtsp-session-pool.c:
3905 * gst/rtsp-server/rtsp-session.c:
3906 * gst/rtsp-server/rtsp-stream-transport.c:
3907 * gst/rtsp-server/rtsp-stream.c:
3908 * gst/rtsp-server/rtsp-thread-pool.c:
3909 * gst/rtsp-server/rtsp-token.c:
3912 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3914 * gst/rtsp-server/rtsp-session-pool.c:
3915 * gst/rtsp-server/rtsp-session-pool.h:
3916 session-pool: make vmethod to create a session
3917 Make a vmethod to create a sessions so that subclasses can create
3918 custom session objects
3920 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3922 * gst/rtsp-server/rtsp-auth.c:
3923 * gst/rtsp-server/rtsp-media-factory.h:
3924 * gst/rtsp-server/rtsp-media.h:
3925 * gst/rtsp-server/rtsp-mount-points.h:
3926 * gst/rtsp-server/rtsp-session-pool.h:
3927 * gst/rtsp-server/rtsp-stream.h:
3930 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3932 * docs/libs/gst-rtsp-server-docs.sgml:
3933 * docs/libs/gst-rtsp-server-sections.txt:
3934 * gst/rtsp-server/rtsp-address-pool.c:
3935 * gst/rtsp-server/rtsp-address-pool.h:
3936 * gst/rtsp-server/rtsp-auth.c:
3937 * gst/rtsp-server/rtsp-client.h:
3938 * gst/rtsp-server/rtsp-media-factory.h:
3939 * gst/rtsp-server/rtsp-media.c:
3940 * gst/rtsp-server/rtsp-media.h:
3941 * gst/rtsp-server/rtsp-permissions.c:
3942 * gst/rtsp-server/rtsp-permissions.h:
3943 * gst/rtsp-server/rtsp-server.h:
3944 * gst/rtsp-server/rtsp-session-media.c:
3945 * gst/rtsp-server/rtsp-session-media.h:
3946 * gst/rtsp-server/rtsp-session-pool.h:
3947 * gst/rtsp-server/rtsp-session.h:
3948 * gst/rtsp-server/rtsp-stream-transport.h:
3949 * gst/rtsp-server/rtsp-stream.c:
3950 * gst/rtsp-server/rtsp-thread-pool.h:
3953 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3956 * examples/Makefile.am:
3957 configure: compile cgroup example conditionally
3958 Only compile the cgroup example when we have libcgroup
3960 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3963 * examples/Makefile.am:
3964 * examples/test-cgroups.c:
3965 examples: add cgroups example
3967 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3969 * tests/check/gst/rtspserver.c:
3970 tests: fix compilation
3972 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3974 * gst/rtsp-server/rtsp-thread-pool.c:
3975 thread-pool: fix vmethod invocation
3977 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3979 * gst/rtsp-server/rtsp-thread-pool.c:
3980 * gst/rtsp-server/rtsp-thread-pool.h:
3981 thread-pool: store thread type in thread
3983 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3985 * gst/rtsp-server/rtsp-client.c:
3986 client: pass thread from pool to media _prepare
3987 Get a thread from the configured threadpool and pass it to the prepare method of
3990 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3992 * gst/rtsp-server/rtsp-media.c:
3993 * gst/rtsp-server/rtsp-media.h:
3994 media: Accept a thread in _prepare
3995 Remove out own threadpool handling and use the provided thread and
3996 maincontext for the bus messages and the state changes.
3998 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4000 * gst/rtsp-server/rtsp-server.c:
4001 server: configure client thread pool
4003 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4005 * gst/rtsp-server/rtsp-client.c:
4006 * gst/rtsp-server/rtsp-client.h:
4007 client: add method to configure thread pool
4009 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4011 * gst/rtsp-server/rtsp-client.h:
4012 * gst/rtsp-server/rtsp-server.c:
4013 * gst/rtsp-server/rtsp-server.h:
4014 server: use thread pool
4015 Use the thread pool instead of doing our own thing.
4017 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4019 * gst/rtsp-server/Makefile.am:
4020 * gst/rtsp-server/rtsp-thread-pool.c:
4021 * gst/rtsp-server/rtsp-thread-pool.h:
4022 thread-pool: add object to manage threads
4023 Add an object to manage the client and media threads.
4025 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4027 * gst/rtsp-server/rtsp-auth.c:
4028 auth: debug authorization check
4030 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4032 * gst/rtsp-server/rtsp-media.c:
4033 media: start media pipeline in context
4034 Start the media pipeline in the provided context (or our default one
4035 when NULL). This makes sure that we run the bus thread in this context and that
4036 all media threads are children of this context.
4038 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4040 * gst/rtsp-server/rtsp-media-factory.c:
4041 factory: pass permissions to media by default
4043 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4045 * examples/test-auth.c:
4046 test: add permissions to auth test
4047 Ass some permissions to the media factory in the test.
4049 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4051 * gst/rtsp-server/rtsp-auth.c:
4052 * gst/rtsp-server/rtsp-auth.h:
4053 * gst/rtsp-server/rtsp-client.c:
4054 auth: simplify auth checks
4055 Remove client from methods, it's now in the state
4056 Perform the check specified by the string, use the information from the
4057 thread local context.
4059 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4061 * gst/rtsp-server/rtsp-client.c:
4062 * gst/rtsp-server/rtsp-client.h:
4063 client: add state to current thread
4064 Add the client to the ClientState object.
4065 Place the ClientState on the current thread.
4067 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4069 * gst/rtsp-server/rtsp-media-factory.c:
4070 * gst/rtsp-server/rtsp-media-factory.h:
4071 * gst/rtsp-server/rtsp-media.c:
4072 * gst/rtsp-server/rtsp-media.h:
4073 media: make it possible to set permissions
4074 Make it possible to set permissions on media and media factory objects
4076 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4078 * gst/rtsp-server/Makefile.am:
4079 * gst/rtsp-server/rtsp-permissions.c:
4080 * gst/rtsp-server/rtsp-permissions.h:
4081 permissions: add permissions object
4082 Add a mini object to store permissions based on a role.
4084 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4086 * examples/test-auth.c:
4087 * gst/rtsp-server/rtsp-auth.c:
4088 * gst/rtsp-server/rtsp-auth.h:
4089 * gst/rtsp-server/rtsp-client.c:
4090 auth: add auth checks
4091 Add an enum with auth checks and implement the checks in the auth object.
4092 Perform the checks from the client.
4094 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4096 * examples/test-auth.c:
4097 * gst/rtsp-server/rtsp-auth.c:
4098 * gst/rtsp-server/rtsp-auth.h:
4099 * gst/rtsp-server/rtsp-client.h:
4100 auth: use the token after authentication
4101 After we authenticated a user, keep the Token around in the state.
4103 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4105 * gst/rtsp-server/rtsp-client.c:
4106 * gst/rtsp-server/rtsp-media.c:
4107 * gst/rtsp-server/rtsp-media.h:
4108 * tests/check/gst/media.c:
4109 media: add optional context for bus messages
4110 Add an optional mainloop to _prepare that will handle the bus messages instead
4111 of always using the shared mainloop.
4113 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4115 * gst/rtsp-server/Makefile.am:
4116 * gst/rtsp-server/rtsp-token.c:
4117 * gst/rtsp-server/rtsp-token.h:
4118 token: add authorization token
4119 Add a simply miniobject that contains the authorizations. The object contains a
4120 GstStructure that hold all authorization fields. When a user is authenticated,
4121 the auth module will create a Token for the user. The token is then used to
4122 check what operations the user is allowed to do and various other configuration
4125 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4127 * examples/test-auth.c:
4128 * gst/rtsp-server/rtsp-auth.c:
4129 * gst/rtsp-server/rtsp-auth.h:
4130 * gst/rtsp-server/rtsp-client.c:
4131 * gst/rtsp-server/rtsp-client.h:
4132 * gst/rtsp-server/rtsp-media-factory.c:
4133 * gst/rtsp-server/rtsp-media-factory.h:
4134 * gst/rtsp-server/rtsp-media.c:
4135 * gst/rtsp-server/rtsp-media.h:
4136 auth: remove auth from media and factory
4137 Remove the auth object from media and factory. We want to have the RTSPClient
4138 authenticate and authorize resources, there is no need to place another auth
4139 manager on the media/factory.
4141 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4143 * examples/test-auth.c:
4144 * gst/rtsp-server/rtsp-auth.c:
4145 * gst/rtsp-server/rtsp-auth.h:
4146 * gst/rtsp-server/rtsp-client.h:
4147 auth: add support for multiple basic auth tokens
4148 Make it possible to add multiple basic authorisation tokens to one authorization
4149 object. Associate with each token an authorization group that will define what
4150 capabilities are allowed.
4152 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4154 * gst/rtsp-server/rtsp-client.c:
4155 client: error out on non-aggregate control
4156 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
4158 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4160 * gst/rtsp-server/rtsp-client.c:
4161 client: rework setup request a little
4162 Cache the media in DESCRIBE based on the longest matching path with the uri
4163 that we can find in the mount points.
4164 Rework the setup request a little to get the media from the session or from
4165 the longest matching path, this way we can derive the control string as
4166 everything after the path instead of hardcoding it.
4167 Find the stream based on the control string and only open a session when all
4170 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4172 * gst/rtsp-server/rtsp-media.c:
4173 * gst/rtsp-server/rtsp-media.h:
4174 media: add method to find a stream by control url
4176 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4178 * gst/rtsp-server/rtsp-stream.c:
4179 * gst/rtsp-server/rtsp-stream.h:
4180 stream: add method to check control url of stream
4182 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4184 * gst/rtsp-server/rtsp-client.c:
4185 * gst/rtsp-server/rtsp-session-media.c:
4186 * gst/rtsp-server/rtsp-session-media.h:
4187 * gst/rtsp-server/rtsp-session.c:
4188 * gst/rtsp-server/rtsp-session.h:
4189 session: use path matching for session media
4190 Use a path string instead of a uri to lookup session media in the sessions. Also
4191 use path matching to find the largest possible path that matches.
4193 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4195 * gst/rtsp-server/rtsp-client.c:
4196 * gst/rtsp-server/rtsp-mount-points.c:
4197 * gst/rtsp-server/rtsp-mount-points.h:
4198 * tests/check/gst/mountpoints.c:
4199 mount-points: remove useless vmethod
4200 Making lookups in the mount points should not be done with a URL, if there is a
4201 mapping to be done from URL to mount points, we'll need to do it somewhere
4204 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4206 * gst/rtsp-server/rtsp-mount-points.c:
4207 * gst/rtsp-server/rtsp-mount-points.h:
4208 * tests/check/gst/mountpoints.c:
4209 mount-points: improve mount point searching
4210 Use a GSequence to keep track of the mount points.
4211 Match a URL to the longest matching registered mount point. This should be the
4212 URL to perform aggreagate control and the remainder is the stream specific
4214 Add some unit tests for this.
4216 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
4218 * gst/rtsp-server/Makefile.am:
4219 rtsp-server: Allow building of static library
4221 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4223 * tests/check/gst/mediafactory.c:
4224 tests: fix compilation
4226 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4228 * gst/rtsp-server/rtsp-sdp.c:
4229 sdp: get control string from stream
4230 Use the control string as configured in the stream.
4232 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4234 * gst/rtsp-server/rtsp-stream.c:
4235 * gst/rtsp-server/rtsp-stream.h:
4236 stream: add methods and property to set control string
4238 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4240 * gst/rtsp-server/rtsp-client.c:
4242 Rename variables for clarity
4243 Keep media in state when we can
4245 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4247 * gst/rtsp-server/rtsp-client.c:
4248 * gst/rtsp-server/rtsp-stream.c:
4249 * gst/rtsp-server/rtsp-stream.h:
4250 stream: add more support for IPv6
4251 Rename _get_address to _get_multicast_address in GstRTSPStream to
4252 make it clear that this function only deals with multicast.
4253 Make it possible to have both an IPv4 and IPv6 multicast address on
4254 a stream. Give the client an IPv4 or IPv6 address depending on the
4255 address it used to connect to the server.
4256 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
4258 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4260 * gst/rtsp-server/rtsp-client.c:
4263 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4265 * gst/rtsp-server/rtsp-stream.c:
4266 stream: handle failed port allocation
4267 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
4268 can't allocate any family at all. Also keep track of what port families we
4270 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
4272 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4274 * gst/rtsp-server/rtsp-stream.c:
4275 stream: improve docs
4277 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4279 * gst/rtsp-server/rtsp-stream-transport.c:
4280 stream-transport: remove old if 0 block
4282 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
4284 * tests/check/gst/client.c:
4286 gst_rtsp_client_get_uri() has been removed
4287 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
4289 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4291 * gst/rtsp-server/rtsp-client.c:
4292 * gst/rtsp-server/rtsp-client.h:
4293 client: add method to filter managed sessions
4294 Add a method to filter the sessions managed by this client connection.
4295 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
4297 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4299 * gst/rtsp-server/rtsp-client.c:
4300 * gst/rtsp-server/rtsp-client.h:
4301 client: remove _get_uri() method
4302 Remove the get_uri() method on the client. A client has no uri, the uri
4303 property is an internal property to manage the last cached media for
4306 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4308 * gst/rtsp-server/rtsp-media-factory.h:
4309 media-factory: fix typo
4311 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
4313 * gst/rtsp-server/rtsp-media.c:
4314 rtsp-media: Do not leak the query in default_query_stop
4315 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
4317 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4319 * gst/rtsp-server/rtsp-media.c:
4320 media: don't unlock when conversion fails
4321 Don't unlock the state lock when conversion fails because it was not locked.
4323 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4325 * gst/rtsp-server/rtsp-media.c:
4326 * gst/rtsp-server/rtsp-media.h:
4327 Add query_position and query_stop vmethods to rtsp-media
4329 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4331 * gst/rtsp-server/rtsp-media.c:
4332 Fix typo in property install for rtsp-media's time-provider
4334 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4336 * gst/rtsp-server/rtsp-client.c:
4337 * gst/rtsp-server/rtsp-client.h:
4338 client: clean some variables
4339 Clean some variables and add some guards to _send_request()
4341 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4343 * gst/rtsp-server/rtsp-client.c:
4344 * gst/rtsp-server/rtsp-client.h:
4345 Add gst_rtsp_client_send_request API
4346 This makes it possible to send arbitrary messages to a client, such as
4347 SET_PARAMETER or GET_PARAMETER
4349 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4351 * gst/rtsp-server/rtsp-media.c:
4352 * gst/rtsp-server/rtsp-media.h:
4353 media: add _get_element() method
4354 Add method to get the element used when creating the media.
4355 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
4357 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4359 * gst/rtsp-server/rtsp-media.c:
4362 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4364 * gst/rtsp-server/rtsp-stream.c:
4365 * gst/rtsp-server/rtsp-stream.h:
4366 stream: allow access to the rtp session
4367 https://bugzilla.gnome.org/show_bug.cgi?id=703004
4369 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
4371 * gst/rtsp-server/rtsp-stream.c:
4372 * gst/rtsp-server/rtsp-stream.h:
4373 dscp qos support in gst-rtsp-stream
4374 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
4376 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4378 * tests/check/gst/rtspserver.c:
4380 Actually do what the comment says. Also keep the old code around, not sure what
4381 should happen when you get a 454 from a TEARDOWN, does it close the connection?
4382 it currently doesn't.
4384 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4386 * gst/rtsp-server/rtsp-client.c:
4387 client: also watch newly created session
4388 When we newly created a session, start watching it immediately instead of
4389 on the next request.
4391 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
4393 * tests/check/gst/client.c:
4394 tests: add unit test for new-session
4395 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
4397 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4399 * gst/rtsp-server/rtsp-client.c:
4400 client: emit new-session when new session is created
4401 Only emit new-session when we created a new session for a client, not when a
4402 client picked up a previous session.
4403 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
4405 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
4407 * gst/rtsp-server/rtsp-client.c:
4408 client: handle asterisk as path in requests
4409 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
4411 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4413 * gst/rtsp-server/rtsp-media.c:
4414 media: handle segment query format mismatch
4415 It's possible that the segment query returns with a different format than what
4416 we asked for, handle this case also.
4418 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
4420 * gst/rtsp-server/rtsp-media.c:
4421 media: use segment stop in collect_media_stats
4422 Use segment stop instead of duration as range end point.
4423 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
4425 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
4427 * gst/rtsp-server/rtsp-media.c:
4428 * tests/check/gst/media.c:
4429 rtsp-media: Do not leak the element in take_pipeline
4430 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
4432 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
4434 * gst/rtsp-server/rtsp-client.c:
4435 * gst/rtsp-server/rtsp-client.h:
4436 rtsp-client: Make configure_client_transport virtual
4437 This patch makes configure_client_transport virtual. The functionality is
4438 needed to handle some weird clients sending multicast transport settings as url
4440 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
4442 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
4444 * gst/rtsp-server/rtsp-client.c:
4445 * gst/rtsp-server/rtsp-client.h:
4446 rtsp-client: Make param_set and param_get virtual
4447 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
4449 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
4451 * gst/rtsp-server/rtsp-client.c:
4452 * gst/rtsp-server/rtsp-media.c:
4453 * gst/rtsp-server/rtsp-media.h:
4454 media: convert_range replaces get_range_times
4455 get_range_times worked for handling UTC ranges for seeks, but we also
4456 need to convert back from NPT to the requested unit in
4457 get_range_string. convert_range is now used for both.
4458 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
4460 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4462 * gst/rtsp-server/rtsp-client.c:
4463 * gst/rtsp-server/rtsp-sdp.c:
4464 * gst/rtsp-server/rtsp-sdp.h:
4465 sdp: cleanup sdp info
4466 We don't need to pass the proto, we can more easily check a boolean.
4467 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
4469 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
4471 * gst/rtsp-server/rtsp-sdp.c:
4472 use 0.0.0.0 or :: for c= line instead of server address
4474 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
4476 * gst/rtsp-server/rtsp-client.c:
4477 use local address, not remote, in SDP
4478 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
4480 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4483 Automatic update of common submodule
4484 From 098c0d7 to 01a7a46
4486 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
4488 * gst/rtsp-server/rtsp-media.c:
4489 * gst/rtsp-server/rtsp-media.h:
4490 media: possibility to override range time conversion
4491 Make it possible to override the conversion from GstRTSPTimeRange to
4492 GstClockTimes, that is done before seeking on the media
4493 pipeline. Overriding can be useful for UTC ranges, where the default
4494 conversion gives nanoseconds since 1900.
4495 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
4497 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
4499 * gst/rtsp-server/rtsp-server.c:
4500 * gst/rtsp-server/rtsp-server.h:
4501 rtsp-server: Expose the use_client_settings API
4502 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
4504 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
4506 * gst/rtsp-server/rtsp-client.c:
4507 * gst/rtsp-server/rtsp-stream.c:
4508 * gst/rtsp-server/rtsp-stream.h:
4509 rtspstream: handle both ipv4 and ipv6 clients
4510 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
4512 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4514 * gst/rtsp-server/rtsp-sdp.c:
4515 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
4516 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
4517 We already have a way to place extra attributes in the SDP by using a string
4518 property with prefix x- or a- in the caps.
4520 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4522 * gst/rtsp-server/rtsp-sdp.c:
4523 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
4524 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
4525 We already have a way to place extra attributes in the SDP, just make a string
4526 property in the payloader with a- or x- prefix.
4528 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4530 * gst/rtsp-server/rtsp-sdp.c:
4531 rtsp: place a- and x- properties as attributes
4532 application/x-rtp has properties with a- and x- prefixes that should be
4533 placed as attributes in the SDP for the media instead of being added to the
4536 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4538 * examples/Makefile.am:
4539 * examples/test-video.c:
4540 example: add TLS example
4542 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4544 * gst/rtsp-server/rtsp-server.c:
4545 * gst/rtsp-server/rtsp-server.h:
4546 server: add support for TLS
4547 Add methods to set and get a TLS certificate.
4548 Add vmethod to configure a new connection. By default, configure the TLS
4549 certificate in a new connection if needed.
4551 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4553 * gst/rtsp-server/rtsp-server.c:
4554 * gst/rtsp-server/rtsp-server.h:
4555 server: remove accept_client vmethod
4556 This vmethod is not very useful so remove it.
4558 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4560 * gst/rtsp-server/rtsp-server.c:
4561 server: don't crash on NULL GError
4563 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
4565 * gst/rtsp-server/rtsp-session-pool.c:
4566 rtsp-session-pool: corrected session timeout detection
4567 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
4569 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4571 * gst/rtsp-server/rtsp-client.c:
4572 client: improve debug
4574 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4576 * gst/rtsp-server/rtsp-client.c:
4577 * gst/rtsp-server/rtsp-client.h:
4578 * gst/rtsp-server/rtsp-server.c:
4579 server: refactor connection setup
4580 Let the server accept the socket connection and construct a GstRTSPConnection
4581 from it. Remove the code from the client and let the client only deal with
4582 a fully configure GstRTSPConnection object.
4583 We will need this later when the server will configure the connection for
4586 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4588 * gst/rtsp-server/rtsp-stream.c:
4589 stream: keep the transport object alive
4590 Keep the transport object alive while we have it as qdata on the
4593 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
4595 * gst/rtsp-server/rtsp-client.c:
4596 * gst/rtsp-server/rtsp-server.c:
4597 rtsp-server: Do not crash on nmapping of server
4598 * generate error when gst_rtsp_connection_accept fails
4599 * do not stop accepting incoming connections because
4600 accepting a client fails
4601 https://bugzilla.gnome.org/show_bug.cgi?id=701072
4603 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
4605 * gst/rtsp-server/rtsp-client.c:
4606 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
4607 https://bugzilla.gnome.org/show_bug.cgi?id=700953
4609 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
4611 * gst/rtsp-server/rtsp-sdp.c:
4612 rtsp-sdp: Parse framerate caps field and set SDP attribute
4613 The SDP attribute and its format is described in RFC4566.
4614 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
4616 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
4618 * gst/rtsp-server/rtsp-sdp.c:
4619 rtsp-sdp: Parse width/height from caps and set SDP attribute
4620 The SDP attribute and its format is described in RFC6064.
4621 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
4623 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
4625 * gst/rtsp-server/rtsp-sdp.c:
4626 * tests/check/gst/client.c:
4627 rtsp-sdp: add bandwidth line
4628 https://bugzilla.gnome.org/show_bug.cgi?id=699220
4630 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4633 Automatic update of common submodule
4634 From 5edcd85 to 098c0d7
4636 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
4638 * tests/check/gst/media.c:
4639 tests: add dynamic payloader prepare/unprepare check
4641 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4643 * gst/rtsp-server/rtsp-media.c:
4644 media: release lock when removing fakesink
4646 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4648 * gst/rtsp-server/rtsp-stream.c:
4649 stream: set elements to NULL before removing
4650 When removing a stream, set the elements to NULL first. This avoids
4651 element-is-not-in-NULL-state errors when we dispose the elements.
4653 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
4656 Automatic update of common submodule
4657 From 3cb3d3c to 5edcd85
4659 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4661 * gst/rtsp-server/rtsp-media.c:
4662 * gst/rtsp-server/rtsp-media.h:
4663 media: listen to pad-removed signals
4664 Listen to the pad-removed signal and remove the stream associated with the
4666 Add signal to be notified of the removed pad.
4667 Remove the fakesink in unprepare()
4668 Fix signatures of the signal methods
4670 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4672 * examples/test-sdp.c:
4673 tests: add example of reusable pipelines
4675 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
4677 * gst/rtsp-server/rtsp-stream.c:
4678 * gst/rtsp-server/rtsp-stream.h:
4679 stream: add method to get the srcpad
4681 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
4683 * tests/check/gst/media.c:
4684 check: add media prepare/unprepare test
4685 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4687 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
4689 * gst/rtsp-server/rtsp-media.c:
4690 media: disconnect from signal handlers in unprepare()
4691 We connected to the pad-added and no-more-pads signals in prepare() so
4692 we need to disconnect from them in unprepare().
4693 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4695 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
4697 * gst/rtsp-server/rtsp-media.c:
4698 media: don't free streams array
4699 Don't free the streams array in the unprepare() method, they were not
4701 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4703 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
4705 * gst/rtsp-server/rtsp-media.c:
4706 media: don't unref the pipeline in unprepare
4707 Unprepare() should undo what prepare() does. Because the pipeline is
4708 not created in prepare(), we should not unref it in unprepare()
4710 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
4712 * gst/rtsp-server/rtsp-stream.c:
4713 stream: clear session and caps for reuse
4714 Set the session and caps to NULL after unref otherwise we might unref
4716 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4718 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
4720 * gst/rtsp-server/rtsp-client.c:
4721 client: send out teardown signal before tearing down
4722 The advantage is that in the signal handler you get direct access to
4723 information about what streams are about to get torn down (in the
4724 GstRTSPClientState).
4725 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
4727 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
4729 * gst/rtsp-server/rtsp-client.c:
4730 * gst/rtsp-server/rtsp-client.h:
4731 client: expose connection
4732 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
4734 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
4737 Automatic update of common submodule
4738 From aed87ae to 3cb3d3c
4740 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4742 * gst/rtsp-server/rtsp-media.c:
4743 * gst/rtsp-server/rtsp-media.h:
4744 * gst/rtsp-server/rtsp-session-media.c:
4745 * gst/rtsp-server/rtsp-session-media.h:
4746 media: add method to get the base_time of the pipeline
4747 Together with a shared clock, this base-time could eventually be sent to
4748 the client so that it can reconstruct the exact running-time of the clock
4751 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4753 * gst/rtsp-server/Makefile.am:
4754 * gst/rtsp-server/rtsp-media.c:
4755 * gst/rtsp-server/rtsp-media.h:
4756 * gst/rtsp-server/rtsp-sdp.c:
4757 media: add GstNetTimeProvider support
4758 Add a property to let the media provide a GstNetTimeProvider for its clock.
4759 Make methods to get the clock and nettimeprovider
4760 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
4761 provider and also the current time of the clock. This should make it possible
4762 for (GStreamer) clients to slave their clock to the server clock.
4764 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
4767 Automatic update of common submodule
4768 From 04c7a1e to aed87ae
4770 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4772 * gst/rtsp-server/rtsp-media.c:
4773 media: wait for buffering to complete
4774 Wait for buffering to complete before changing the state to the target state.
4776 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4778 * gst/rtsp-server/rtsp-media.c:
4779 media: small cleanup
4781 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
4783 * tests/check/gst/rtspserver.c:
4784 tests: remove extra unref in test_setup_non_existing_stream
4785 The unref is not needed anymore, teardown runs without it.
4786 https://bugzilla.gnome.org/show_bug.cgi?id=696542
4788 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
4790 * tests/check/gst/rtspserver.c:
4791 tests: GSocketService cleanup in test_bind_already_in_use
4792 Use g_socket_service_stop so the rtspserver test stops listening for
4793 incoming connections in test_bind_already_in_use.
4794 https://bugzilla.gnome.org/show_bug.cgi?id=696541
4796 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
4798 * gst/rtsp-server/rtsp-media-factory.c:
4799 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
4800 Instead use a GWeakRef which is safe to use
4801 This is a known GLib bug, see:
4802 https://bugzilla.gnome.org/show_bug.cgi?id=667145
4804 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
4806 * gst/rtsp-server/rtsp-client.c:
4807 * gst/rtsp-server/rtsp-media.c:
4808 * gst/rtsp-server/rtsp-media.h:
4809 * gst/rtsp-server/rtsp-sdp.c:
4810 * tests/check/gst/media.c:
4811 * tests/check/gst/rtspserver.c:
4812 rtsp-media/client: Reply to PLAY request with same type of Range
4813 Remember the type of Range from the PLAY request and use the same type for
4816 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
4818 * gst/rtsp-server/rtsp-client.c:
4819 * gst/rtsp-server/rtsp-client.h:
4820 * tests/check/gst/client.c:
4821 rtsp-client: expose uri
4823 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
4825 * tests/check/gst/mediafactory.c:
4826 tests: Hold ref while creating second media
4827 To test if the media aren't shared, make sure we keep the first one while creating a second
4828 otherwise the same memory address may be reused.
4830 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
4833 configure: remove out-of-date comment
4835 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
4838 .gitignore: ignore more build files
4840 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
4842 * tests/check/Makefile.am:
4843 tests: use right _LIBS variable for gst-plugins-base libs
4845 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4847 * tests/check/Makefile.am:
4848 check: add librtp to libs
4850 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
4852 * tests/check/gst/rtspserver.c:
4853 tests: Add test to check selecting a port the server will send from
4855 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
4857 * tests/check/gst/rtspserver.c:
4858 tests: Make sure packets are actually received
4860 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4862 * gst/rtsp-server/rtsp-stream.c:
4863 stream: Select unicast address from pool if appropriate
4865 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
4867 * gst/rtsp-server/rtsp-stream.c:
4868 stream: Properties are always there in Gst 1.0
4870 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4872 * tests/check/gst/addresspool.c:
4873 tests: Add tests for unicast addresses in pool
4875 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
4877 * gst/rtsp-server/rtsp-address-pool.c:
4878 * tests/check/gst/addresspool.c:
4879 address-pool: Verify that multicast addresses are used for multicast and vice-versa
4881 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
4883 * docs/libs/gst-rtsp-server-sections.txt:
4884 * gst/rtsp-server/rtsp-address-pool.c:
4885 * gst/rtsp-server/rtsp-address-pool.h:
4886 * gst/rtsp-server/rtsp-stream.c:
4887 * tests/check/gst/addresspool.c:
4888 address-pool: Add unicast addresses
4890 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
4893 * gst/rtsp-server/rtsp-server.c:
4894 * tests/check/gst/rtspserver.c:
4895 rtsp-server: Limit the number of threads per server instance
4896 If we exceed the maximum, just round robin the clients over the existing
4899 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
4901 * gst/rtsp-server/rtsp-server.c:
4902 rtsp-server: No need to store the GMainContext in the client context
4904 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
4906 * tests/check/gst/rtspserver.c:
4907 tests: Add test for client disconnection
4909 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
4911 * tests/check/gst/rtspserver.c:
4912 tests: Test client and session timeouts with multiple threads
4914 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
4916 * gst/rtsp-server/rtsp-address-pool.c:
4917 * gst/rtsp-server/rtsp-auth.c:
4918 * gst/rtsp-server/rtsp-client.c:
4919 * gst/rtsp-server/rtsp-media-factory-uri.c:
4920 * gst/rtsp-server/rtsp-media-factory.c:
4921 * gst/rtsp-server/rtsp-media.c:
4922 * gst/rtsp-server/rtsp-mount-points.c:
4923 * gst/rtsp-server/rtsp-server.c:
4924 * gst/rtsp-server/rtsp-session-media.c:
4925 * gst/rtsp-server/rtsp-session-pool.c:
4926 * gst/rtsp-server/rtsp-session.c:
4927 Document locking and its order
4929 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
4931 * tests/check/gst/rtspserver.c:
4932 tests: Test that slow DESCRIBE don't block other clients
4934 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
4936 * tests/check/gst/client.c:
4937 tests: Add tests for client-requested multicast address
4939 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
4941 * docs/libs/gst-rtsp-server-sections.txt:
4942 docs: Put the various functions in the right sections
4944 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
4946 * docs/libs/gst-rtsp-server-docs.sgml:
4947 * docs/libs/gst-rtsp-server-sections.txt:
4948 * gst/rtsp-server/rtsp-address-pool.c:
4949 * gst/rtsp-server/rtsp-address-pool.h:
4950 docs: Generate docs for GstRTSPAddressPool
4952 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4954 * gst/rtsp-server/rtsp-client.c:
4955 * gst/rtsp-server/rtsp-stream.c:
4956 * gst/rtsp-server/rtsp-stream.h:
4957 client: Check client provided addresses against the address pool
4959 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
4961 * gst/rtsp-server/rtsp-address-pool.c:
4962 * gst/rtsp-server/rtsp-address-pool.h:
4963 * tests/check/gst/addresspool.c:
4964 address-pool: Add API to request a specific address from the pool
4965 Also add relevant unit tests.
4967 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
4969 * tests/check/gst/mediafactory.c:
4970 tests: Check the passing around of a RTSPAddressPool
4971 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
4972 way down to the stream.
4974 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
4976 * tests/check/gst/addresspool.c:
4977 tests: Add more tests for the address pool
4979 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
4981 * gst/rtsp-server/rtsp-address-pool.c:
4982 address-pool: Fix off by one error
4983 When splitting a port range, the port after a skip is not part of range.
4985 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
4988 Automatic update of common submodule
4989 From 2de221c to 04c7a1e
4991 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
4994 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
4995 AM_CONFIG_HEADER was removed in automake 1.13
4996 https://bugzilla.gnome.org/show_bug.cgi?id=693368
4998 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
5001 Automatic update of common submodule
5002 From a942293 to 2de221c
5004 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5006 * gst/rtsp-server/rtsp-client.c:
5007 client: make sure the watch exists while sending data
5008 Protect the send_func with a lock. This allows us to wait for sending
5009 to complete before changing the send_func and user_data. We add an
5010 extra ref to the watch to make sure that it remains valid during
5012 When closing the connection, set the send_func to NULL
5013 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
5015 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5017 * tests/check/Makefile.am:
5018 tests: use GST_*_1_0 environment variables everywhere
5019 The _1_0 suffixed environment variables override the
5020 non-suffixed ones, so if we're in an environment that
5021 sets the _1_0 suffixed ones, such as jhbuild, we need
5022 to set those to make sure ours actually always get
5025 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5028 Automatic update of common submodule
5029 From acb04d9 to a942293
5031 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5033 * gst/rtsp-server/rtsp-client.c:
5034 rtsp-client: set the client backlog
5035 Set the client backlog to a reasonable default
5037 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
5039 * gst/rtsp-server/rtsp-media.c:
5040 rtsp-media: Make the element a constructor parameter
5041 https://bugzilla.gnome.org/show_bug.cgi?id=689594
5043 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5045 * docs/libs/Makefile.am:
5046 docs: Link with gcov library when gcov is enabled
5047 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
5049 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5051 * gst/rtsp-server/rtsp-media.c:
5052 media: match prepare with unprepare
5053 Really unprepare when there were an equal amount of prepare calls.
5055 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5057 * gst/rtsp-server/rtsp-media.c:
5058 media: media has to be unprepared in finalize
5059 Because unprepare takes away the last ref on the media.
5061 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5063 * gst/rtsp-server/rtsp-client.c:
5064 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
5065 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
5066 We can't use the refcount to trigger unprepare because it is the unprepare call
5067 that removes the last refcount after all messages are consumed. What we should
5068 probably do is make a prepared refcount and only unprepare when the refcount
5071 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5073 * gst/rtsp-server/rtsp-media.c:
5074 media: let the source unref the last media ref
5075 the last ref to the media is held by the source so we don't need to add more ref
5076 and unrefs, we simply destroy the media when the source is gone.
5078 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5080 * gst/rtsp-server/rtsp-media.c:
5081 media: improve debug
5083 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5085 * gst/rtsp-server/rtsp-media.c:
5087 Make sure we are in the right state when collecting the position and duration.
5088 Only make ourselves PREPARED when we were previously PREPARING.
5090 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5092 * gst/rtsp-server/rtsp-media.c:
5093 media: use g_object_ref/unref for GObjects
5095 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
5097 * gst/rtsp-server/rtsp-client.c:
5098 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
5099 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
5100 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
5101 isn't being used anymore.
5103 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
5105 * gst/rtsp-server/rtsp-media.c:
5106 Fix compiler warning
5108 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
5110 * gst/rtsp-server/rtsp-media-factory-uri.c:
5111 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
5113 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5115 * gst/rtsp-server/rtsp-session-media.h:
5118 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5120 * gst/rtsp-server/rtsp-media.c:
5121 * tests/check/gst/media.c:
5122 media: avoid element leak
5124 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5126 * gst/rtsp-server/rtsp-media.c:
5127 media: require an element in media constructor
5129 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5131 * gst/rtsp-server/rtsp-client.c:
5132 Revert "client: TEARDOWN brings that state to Init again"
5133 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
5134 The object is already disposed, there is no point in setting the state.
5136 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5138 * gst/rtsp-server/rtsp-client.c:
5139 client: TEARDOWN brings that state to Init again
5141 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5143 * docs/libs/gst-rtsp-server-sections.txt:
5144 * examples/test-auth.c:
5145 * gst/rtsp-server/rtsp-auth.c:
5146 * gst/rtsp-server/rtsp-auth.h:
5147 * gst/rtsp-server/rtsp-client.c:
5148 * gst/rtsp-server/rtsp-client.h:
5149 * gst/rtsp-server/rtsp-media-factory-uri.c:
5150 * gst/rtsp-server/rtsp-media-factory-uri.h:
5151 * gst/rtsp-server/rtsp-media-factory.c:
5152 * gst/rtsp-server/rtsp-media-factory.h:
5153 * gst/rtsp-server/rtsp-media.c:
5154 * gst/rtsp-server/rtsp-media.h:
5155 * gst/rtsp-server/rtsp-mount-points.c:
5156 * gst/rtsp-server/rtsp-mount-points.h:
5157 * gst/rtsp-server/rtsp-sdp.c:
5158 * gst/rtsp-server/rtsp-server.c:
5159 * gst/rtsp-server/rtsp-server.h:
5160 * gst/rtsp-server/rtsp-session-media.c:
5161 * gst/rtsp-server/rtsp-session-media.h:
5162 * gst/rtsp-server/rtsp-session-pool.c:
5163 * gst/rtsp-server/rtsp-session-pool.h:
5164 * gst/rtsp-server/rtsp-session.c:
5165 * gst/rtsp-server/rtsp-session.h:
5166 * gst/rtsp-server/rtsp-stream-transport.c:
5167 * gst/rtsp-server/rtsp-stream-transport.h:
5168 * gst/rtsp-server/rtsp-stream.c:
5169 * gst/rtsp-server/rtsp-stream.h:
5170 * tests/check/gst/media.c:
5171 rtsp: make object details private
5172 Make all object details private
5173 Add methods to access private bits
5175 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5177 * tests/check/Makefile.am:
5178 * tests/check/gst/media.c:
5179 tests: add media tests
5181 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5183 * gst/rtsp-server/rtsp-media.c:
5184 media: check if prepared for some methods
5185 Check that the media object is prepared before doing seek and getting the
5186 current position etc.
5187 Add some g_return checks.
5189 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5191 * tests/check/Makefile.am:
5192 * tests/check/gst/mediafactory.c:
5193 tests: add mediafactory test
5195 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5197 * gst/rtsp-server/rtsp-stream.c:
5198 stream: improve debug
5200 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5202 * gst/rtsp-server/rtsp-media.c:
5203 * gst/rtsp-server/rtsp-media.h:
5204 media: unref pipeline in finalize to avoid leaking it
5206 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5208 * gst/rtsp-server/rtsp-media-factory-uri.c:
5209 * gst/rtsp-server/rtsp-media.c:
5210 rtsp: use gst_object_unref on GstObjects
5212 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5214 * gst/rtsp-server/rtsp-media-factory.c:
5215 media-factory: require an url
5217 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5219 * examples/test-uri.c:
5220 examples: fix include
5222 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5224 * gst/rtsp-server/rtsp-server.h:
5225 server: remove unused include
5227 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5229 * tests/check/Makefile.am:
5230 * tests/check/gst/mountpoints.c:
5231 tests: add test for mountpoints
5233 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5235 * gst/rtsp-server/rtsp-client.c:
5236 client: fix factory leak
5237 Keep the factory in the state object only for authorization checks and make
5238 sure we unref it on failure. Also don't keep invalid objects in the state
5241 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5243 * gst/rtsp-server/rtsp-mount-points.c:
5244 mounts: add g_return_if guards
5246 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5248 * tests/check/gst/client.c:
5249 tests: add more tests
5251 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5253 * gst/rtsp-server/rtsp-client.c:
5254 client: improve debug
5256 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5258 * gst/rtsp-server/rtsp-client.c:
5259 client: improve debug and fix leaks
5260 Cleanup the uri and session when there is a bad request.
5262 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5267 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5269 * tests/check/gst/client.c:
5270 test: add test for session in options request
5272 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5274 * gst/rtsp-server/rtsp-client.c:
5275 client: use 454 when session can't be found
5276 We should use 454 when a session can't be found because there was no session
5277 pool configured in the server. This is not a server configuration problem
5278 because the server on which the request is done might not be the same one that
5279 will keep the sessions for us and so it does not need to support sessions.
5281 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5283 * gst/rtsp-server/rtsp-client.c:
5284 client: only free connection when there is one
5285 It's possible that the client doesn't have a connection when we try to free it.
5287 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5289 * tests/check/Makefile.am:
5290 * tests/check/gst/client.c:
5291 tests: add unit test for the client object
5293 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5295 * gst/rtsp-server/rtsp-client.c:
5296 client: small cleanup
5298 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5300 * gst/rtsp-server/rtsp-client.h:
5301 client: remove unused include
5303 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5305 * gst/rtsp-server/rtsp-client.c:
5306 client: fix compilation
5308 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5310 * gst/rtsp-server/rtsp-client.c:
5311 client: call destroy without the lock
5313 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5315 * gst/rtsp-server/rtsp-client.c:
5316 * gst/rtsp-server/rtsp-client.h:
5317 client: make the client usable without a socket
5318 Make a method to let the client handle a message and a callback when the client
5319 wants us to send a response message back. This makes it possible to also use the
5320 client object without the sockets, which should make it easier to test.
5322 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5324 * gst/rtsp-server/rtsp-client.c:
5325 * gst/rtsp-server/rtsp-client.h:
5326 client: small cleanup
5328 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5330 * docs/libs/gst-rtsp-server-sections.txt:
5331 * gst/rtsp-server/rtsp-client.c:
5332 * gst/rtsp-server/rtsp-client.h:
5333 * gst/rtsp-server/rtsp-server.c:
5334 client: remove reference to server
5335 We don't need to keep a ref to the server
5337 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5339 * gst/rtsp-server/rtsp-client.c:
5340 * gst/rtsp-server/rtsp-client.h:
5342 Also add some g_return_if()
5344 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5346 * gst/rtsp-server/rtsp-client.c:
5347 client: log more errors
5349 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5351 * gst/rtsp-server/rtsp-client.c:
5352 client: fix compilation
5354 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5356 * gst/rtsp-server/rtsp-client.c:
5357 * gst/rtsp-server/rtsp-client.h:
5358 client: add generic close-after-send support
5359 Add a property to send_response() to close the connection after the response has
5360 been sent to the client.
5362 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5365 * docs/libs/gst-rtsp-server-docs.sgml:
5366 * docs/libs/gst-rtsp-server-sections.txt:
5367 * docs/libs/gst-rtsp-server.types:
5368 * examples/test-auth.c:
5369 * examples/test-launch.c:
5370 * examples/test-mp4.c:
5371 * examples/test-multicast.c:
5372 * examples/test-multicast2.c:
5373 * examples/test-ogg.c:
5374 * examples/test-readme.c:
5375 * examples/test-sdp.c:
5376 * examples/test-uri.c:
5377 * examples/test-video.c:
5378 * gst/rtsp-server/Makefile.am:
5379 * gst/rtsp-server/rtsp-auth.h:
5380 * gst/rtsp-server/rtsp-client.c:
5381 * gst/rtsp-server/rtsp-client.h:
5382 * gst/rtsp-server/rtsp-media-mapping.c:
5383 * gst/rtsp-server/rtsp-media-mapping.h:
5384 * gst/rtsp-server/rtsp-mount-points.c:
5385 * gst/rtsp-server/rtsp-mount-points.h:
5386 * gst/rtsp-server/rtsp-server.c:
5387 * gst/rtsp-server/rtsp-server.h:
5388 * gst/rtsp-server/rtsp-session-media.c:
5389 * gst/rtsp-server/rtsp-session-pool.c:
5390 * gst/rtsp-server/rtsp-session-pool.h:
5391 * tests/check/gst/rtspserver.c:
5392 MediaMapping -> MountPoints
5393 Describes better what the object manages.
5395 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5398 configure: bump required version of -base
5400 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5402 * gst/rtsp-server/rtsp-media.c:
5405 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5407 * gst/rtsp-server/rtsp-media.c:
5408 * gst/rtsp-server/rtsp-media.h:
5409 media: support more Range formats
5410 Use the new -base methods to convert the Range string into a seek start and stop
5413 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5415 * examples/test-launch.c:
5416 examples: fix whitespace
5418 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5420 * examples/test-auth.c:
5421 test-auth: add example of how to remove sessions
5422 Add an example of the session filter api.
5424 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5426 * examples/test-uri.c:
5427 test-uri: remove mapping example
5429 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5431 * examples/test-uri.c:
5432 test-uri: fix callback signature
5434 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5436 * gst/rtsp-server/rtsp-media-factory.c:
5437 factory: keep ref to factory while media active
5438 While the media from a factory is alive, keep a ref to the factory.
5439 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
5441 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5443 * gst/rtsp-server/rtsp-media-factory-uri.c:
5444 factory-uri: add some debug
5446 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5448 * gst/rtsp-server/rtsp-stream.c:
5449 stream: set udp sources to PLAYING
5450 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
5451 so that it doesn't cause our pipeline to produce ASYNC-DONE.
5453 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5455 * gst/rtsp-server/rtsp-media-factory-uri.c:
5456 factory-uri: take ref to factory
5457 Take a ref to the factory that we place in our list.
5459 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5461 * tests/Makefile.am:
5462 * tests/test-reuse.c:
5463 test: add test for server reuse
5464 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
5466 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
5468 * gst/rtsp-server/rtsp-server.c:
5469 server: start and stop multiple times
5470 Stop listening on the RTSP port when the GSource is removed, so clients
5471 can't connect and the server can be started again.
5472 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
5474 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5476 * gst/rtsp-server/rtsp-server.c:
5477 server: fix small leak
5479 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5481 * gst/rtsp-server/rtsp-media.c:
5482 media: unref source in finish_unprepare
5483 The source is created in prepare, unref it in finish_unprepare.
5484 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
5486 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
5488 * gst/rtsp-server/rtsp-client.c:
5489 * gst/rtsp-server/rtsp-media.c:
5490 rtsp-media: remove bus watch before finalizing
5491 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
5492 * An extra media ref is added for the bus watch. This extra ref is unreffed by
5493 the GDestroyNotify function.
5494 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
5495 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
5496 gst_rtsp_media_unprepare before unreffing the media.
5497 This way, the bus watch will be removed before the media is finalized.
5498 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
5500 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
5502 * gst/rtsp-server/rtsp-client.c:
5503 * gst/rtsp-server/rtsp-client.h:
5504 client: wait until the TEARDOWN response is sent to close the connection
5505 Responses can be sent async so we need to wait until the TEARDOWN response has
5506 been written before we close the connection to the client. This avoids the risk
5507 of writing/polling closed sockets.
5508 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
5510 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
5512 * gst/rtsp-server/rtsp-stream.c:
5513 rtsp-stream: plug socket leak
5514 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
5516 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
5519 Automatic update of common submodule
5520 From 6bb6951 to a72faea
5522 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
5524 * gst/rtsp-server/rtsp-media-factory-uri.c:
5525 rtsp-server: don't use deprecated API
5527 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
5529 * gst/rtsp-server/rtsp-client.c:
5530 rtsp-client: fix unused-but-set-variable compiler warning
5531 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
5533 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5536 * docs/libs/gst-rtsp-server-sections.txt:
5537 * gst/rtsp-server/rtsp-client.c:
5540 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5542 * examples/Makefile.am:
5543 * examples/test-multicast2.c:
5544 examples: add another multicast example
5545 Add an example for how to configure separate multicast ranges for each media
5548 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5550 * examples/test-multicast.c:
5553 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5555 * gst/rtsp-server/rtsp-client.c:
5556 * gst/rtsp-server/rtsp-media.c:
5557 * gst/rtsp-server/rtsp-session-media.c:
5558 * gst/rtsp-server/rtsp-session-media.h:
5559 * gst/rtsp-server/rtsp-stream-transport.c:
5560 * gst/rtsp-server/rtsp-stream-transport.h:
5561 stream: use the address managed by the stream
5562 Use the address managed by the stream for multicast. This allows us to have 1
5563 multicast address for each stream.
5564 Because the address is now managed by the stream we don't have to pass it around
5566 Set the address pool on the streams.
5568 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5570 * gst/rtsp-server/rtsp-client.c:
5571 * gst/rtsp-server/rtsp-media.c:
5572 * gst/rtsp-server/rtsp-stream.c:
5575 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5577 * gst/rtsp-server/rtsp-media.c:
5578 * gst/rtsp-server/rtsp-media.h:
5579 media: add signal for new streams
5580 This allows applications to listen for new streams and configure properties on
5581 them, like the address pool.
5583 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5585 * gst/rtsp-server/rtsp-media.c:
5586 media: configure address pool in new streams
5588 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5590 * gst/rtsp-server/rtsp-stream.c:
5591 * gst/rtsp-server/rtsp-stream.h:
5592 stream: add methods to deal with address pool
5593 Add methods to get and set the address pool for the stream
5594 Add method to allocate and get the multicast addresses for this stream.
5596 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5598 * docs/libs/gst-rtsp-server-sections.txt:
5599 * gst/rtsp-server/rtsp-media.c:
5600 * gst/rtsp-server/rtsp-media.h:
5601 media: remove MTU property
5602 It is a stream property
5604 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5606 * gst/rtsp-server/rtsp-client.c:
5607 client: set blocksize only on stream
5608 Set the blocksize only on the current stream.
5610 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5612 * gst/rtsp-server/rtsp-stream.c:
5613 stream: share src and sink sockets
5614 the allocated socket is in the used-socket property, not socket.
5616 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5618 * gst/rtsp-server/rtsp-address-pool.c:
5619 * gst/rtsp-server/rtsp-address-pool.h:
5620 * gst/rtsp-server/rtsp-client.c:
5621 * gst/rtsp-server/rtsp-session-media.c:
5622 * gst/rtsp-server/rtsp-session-media.h:
5623 * gst/rtsp-server/rtsp-stream-transport.c:
5624 * gst/rtsp-server/rtsp-stream-transport.h:
5625 * tests/check/gst/addresspool.c:
5626 rtsp: make address-pool return an address object
5627 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
5628 store more info in the structure and allows us to more easily return the address
5629 to the right pool when no longer needed.
5630 Pass the address to the StreamTransport so that we can return it to the pool
5631 when the stream transport is freed or changed.
5633 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5635 * examples/Makefile.am:
5636 * examples/test-multicast.c:
5637 examples: add multicast example
5638 Show how to set up the multicast address pool so that media can be
5639 server with multicast.
5641 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5643 * gst/rtsp-server/rtsp-client.c:
5644 * gst/rtsp-server/rtsp-media-factory.c:
5645 * gst/rtsp-server/rtsp-media-factory.h:
5646 * gst/rtsp-server/rtsp-media.c:
5647 * gst/rtsp-server/rtsp-media.h:
5648 rtsp: use AddressPool
5649 Remove the multicast_group property.
5650 Use the configured addresspool to allocate multicast addresses.
5652 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5654 * gst/rtsp-server/rtsp-address-pool.c:
5655 * gst/rtsp-server/rtsp-address-pool.h:
5656 address-pool: add clear method
5658 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5660 * gst/rtsp-server/rtsp-address-pool.c:
5661 address-pool: small cleanups
5663 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5665 * tests/check/Makefile.am:
5666 * tests/check/gst/addresspool.c:
5667 tests: add addresspool unit test
5669 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5671 * gst/rtsp-server/Makefile.am:
5672 * gst/rtsp-server/rtsp-address-pool.c:
5673 * gst/rtsp-server/rtsp-address-pool.h:
5674 address-pool: add object to manage multicast addresses
5675 Make an object that can manage a rage of multicast addresses and ports.
5677 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5679 * gst/rtsp-server/rtsp-server.c:
5680 server: set default max-threads property
5682 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5684 * gst/rtsp-server/rtsp-media.c:
5685 media: wait for concurrent _prepare
5686 If a prepare is busy, wait for the result.
5688 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5690 * gst/rtsp-server/rtsp-media.c:
5691 media: add lock around message handler
5692 We don't want to dispatch messages while we are still processing the result of
5695 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5697 * gst/rtsp-server/rtsp-media.c:
5698 * gst/rtsp-server/rtsp-media.h:
5699 media: add lock to protect state changes
5701 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5703 * gst/rtsp-server/rtsp-stream.c:
5704 * gst/rtsp-server/rtsp-stream.h:
5707 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5709 * gst/rtsp-server/rtsp-stream-transport.c:
5710 * gst/rtsp-server/rtsp-stream-transport.h:
5711 * gst/rtsp-server/rtsp-stream.c:
5712 stream-transport: add keep-alive method
5714 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5716 * gst/rtsp-server/rtsp-stream-transport.c:
5717 * gst/rtsp-server/rtsp-stream-transport.h:
5718 * gst/rtsp-server/rtsp-stream.c:
5719 stream-transport: add method to handle RTP/RTCP
5720 Call new methods instead of poking into the structures directly.
5722 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5724 * gst/rtsp-server/rtsp-session-media.c:
5725 * gst/rtsp-server/rtsp-session-media.h:
5726 session-media: add locking
5728 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5730 * gst/rtsp-server/rtsp-session.c:
5731 * gst/rtsp-server/rtsp-session.h:
5732 session: add locking
5734 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5736 * gst/rtsp-server/rtsp-server.c:
5737 server: free old socket
5739 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5741 * gst/rtsp-server/rtsp-media-mapping.c:
5742 * gst/rtsp-server/rtsp-media-mapping.h:
5743 mapping: add locking
5745 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5747 * gst/rtsp-server/rtsp-media-factory.c:
5748 media-factory: add locking
5750 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5752 * gst/rtsp-server/rtsp-auth.c:
5753 * gst/rtsp-server/rtsp-auth.h:
5756 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5758 * gst/rtsp-server/rtsp-server.c:
5759 * gst/rtsp-server/rtsp-server.h:
5760 server: add max-thread property
5762 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5764 * gst/rtsp-server/rtsp-server.c:
5765 * gst/rtsp-server/rtsp-server.h:
5766 server: use a threadpool for the mainloops
5768 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5770 * gst/rtsp-server/rtsp-client.c:
5771 * gst/rtsp-server/rtsp-client.h:
5772 client: rename method
5773 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
5774 don't really create the client from the socket, we use the socket for the
5777 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5779 * gst/rtsp-server/rtsp-client.c:
5780 * gst/rtsp-server/rtsp-client.h:
5781 * gst/rtsp-server/rtsp-server.c:
5782 server: rework maincontext handling in clients
5783 Make a separate method to attach a client to a MainContext.
5784 Let the server decide in what GMainContext the client will operate and give this
5785 context to the client in attach. Then the server can later decide to use a
5786 separate thread for each client or just use the mainthread.
5788 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5790 * gst/rtsp-server/rtsp-client.c:
5791 * gst/rtsp-server/rtsp-session.c:
5792 * gst/rtsp-server/rtsp-session.h:
5793 session: move session header code in session object
5795 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
5799 * examples/test-auth.c:
5800 * examples/test-launch.c:
5801 * examples/test-mp4.c:
5802 * examples/test-ogg.c:
5803 * examples/test-readme.c:
5804 * examples/test-sdp.c:
5805 * examples/test-uri.c:
5806 * examples/test-video.c:
5807 * gst/rtsp-server/rtsp-auth.c:
5808 * gst/rtsp-server/rtsp-auth.h:
5809 * gst/rtsp-server/rtsp-client.c:
5810 * gst/rtsp-server/rtsp-client.h:
5811 * gst/rtsp-server/rtsp-media-factory-uri.c:
5812 * gst/rtsp-server/rtsp-media-factory-uri.h:
5813 * gst/rtsp-server/rtsp-media-factory.c:
5814 * gst/rtsp-server/rtsp-media-factory.h:
5815 * gst/rtsp-server/rtsp-media-mapping.c:
5816 * gst/rtsp-server/rtsp-media-mapping.h:
5817 * gst/rtsp-server/rtsp-media.c:
5818 * gst/rtsp-server/rtsp-media.h:
5819 * gst/rtsp-server/rtsp-params.c:
5820 * gst/rtsp-server/rtsp-params.h:
5821 * gst/rtsp-server/rtsp-sdp.c:
5822 * gst/rtsp-server/rtsp-sdp.h:
5823 * gst/rtsp-server/rtsp-server.c:
5824 * gst/rtsp-server/rtsp-server.h:
5825 * gst/rtsp-server/rtsp-session-media.c:
5826 * gst/rtsp-server/rtsp-session-media.h:
5827 * gst/rtsp-server/rtsp-session-pool.c:
5828 * gst/rtsp-server/rtsp-session-pool.h:
5829 * gst/rtsp-server/rtsp-session.c:
5830 * gst/rtsp-server/rtsp-session.h:
5831 * gst/rtsp-server/rtsp-stream-transport.c:
5832 * gst/rtsp-server/rtsp-stream-transport.h:
5833 * gst/rtsp-server/rtsp-stream.c:
5834 * gst/rtsp-server/rtsp-stream.h:
5835 * tests/check/gst/rtspserver.c:
5836 * tests/test-cleanup.c:
5839 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
5841 * gst/rtsp-server/rtsp-media.c:
5842 * gst/rtsp-server/rtsp-session-media.c:
5843 * gst/rtsp-server/rtsp-session.c:
5844 rtsp-server: added annotations to indicate type of ownership transfer of return values
5845 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5847 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
5850 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
5852 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
5855 * bindings/Makefile.am:
5856 * bindings/vala/Makefile.am:
5857 * bindings/vala/gst-rtsp-server-0.10.deps:
5858 * bindings/vala/gst-rtsp-server-0.10.vapi:
5859 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
5860 * bindings/vala/packages/gst-rtsp-server-0.10.files:
5861 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
5862 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
5863 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
5865 bindings: remove vala bindings
5866 They'll be reunited with the other GStreamer bindings
5867 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5869 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5871 * gst/rtsp-server/rtsp-client.c:
5872 * gst/rtsp-server/rtsp-session-media.c:
5873 * gst/rtsp-server/rtsp-session-media.h:
5874 * gst/rtsp-server/rtsp-stream-transport.c:
5875 * gst/rtsp-server/rtsp-stream-transport.h:
5876 rtsp: only create transport when needed
5877 Only create the StreamTransport when configured.
5879 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5881 * gst/rtsp-server/rtsp-client.c:
5882 client: small cleanup
5884 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5886 * gst/rtsp-server/rtsp-client.c:
5887 * gst/rtsp-server/rtsp-client.h:
5888 * gst/rtsp-server/rtsp-stream-transport.c:
5889 * gst/rtsp-server/rtsp-stream-transport.h:
5890 rtsp: refactor configuration of transport
5891 Move the configuration of the transport to a place where it makes
5894 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5896 * gst/rtsp-server/rtsp-client.c:
5897 client: refactor transport parsing
5899 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5901 * gst/rtsp-server/rtsp-client.c:
5902 client: refuse to change the MTU on shared media
5903 If we change the MTU of chared media, it changes for all clients.
5904 We don't want to set the MTU to something large for clients that
5907 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5909 * examples/test-mp4.c:
5910 * gst/rtsp-server/rtsp-media.c:
5911 small fixes to docs and debug
5913 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5915 * gst/rtsp-server/rtsp-stream.c:
5916 stream: transports must already have been removed
5918 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5920 * gst/rtsp-server/rtsp-media.c:
5921 * gst/rtsp-server/rtsp-stream.c:
5922 * gst/rtsp-server/rtsp-stream.h:
5923 stream: improve join and leave of the pipeline
5925 Do the cleanup properly
5928 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5930 * gst/rtsp-server/rtsp-media.c:
5931 media: move unprepare below default implementation
5932 Makes it easier to find the default implementation
5934 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5936 * gst/rtsp-server/rtsp-media.c:
5937 media: signal unprepared when we actually finish
5939 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5941 * gst/rtsp-server/rtsp-media.c:
5942 media: no need to unlock, unprepare does that when needed
5944 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5946 * docs/libs/gst-rtsp-server-sections.txt:
5947 * gst/rtsp-server/rtsp-media-factory.h:
5948 * gst/rtsp-server/rtsp-media-mapping.c:
5949 * gst/rtsp-server/rtsp-media.h:
5950 * gst/rtsp-server/rtsp-params.c:
5951 * gst/rtsp-server/rtsp-server.c:
5952 * gst/rtsp-server/rtsp-session-pool.h:
5953 * gst/rtsp-server/rtsp-session.c:
5954 * gst/rtsp-server/rtsp-session.h:
5955 * gst/rtsp-server/rtsp-stream-transport.h:
5956 * gst/rtsp-server/rtsp-stream.h:
5959 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5961 * gst/rtsp-server/rtsp-client.c:
5962 * gst/rtsp-server/rtsp-media-mapping.h:
5963 * gst/rtsp-server/rtsp-media.c:
5964 * gst/rtsp-server/rtsp-media.h:
5965 * gst/rtsp-server/rtsp-server.h:
5966 * gst/rtsp-server/rtsp-stream.c:
5967 * gst/rtsp-server/rtsp-stream.h:
5968 rtsp: fix MTU setting
5969 Fix setting of the MTU. There is no need for a vmethod.
5971 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5976 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5979 configure: bump version number after refactoring
5981 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5983 * gst/rtsp-server/Makefile.am:
5984 * gst/rtsp-server/rtsp-client.c:
5985 * gst/rtsp-server/rtsp-client.h:
5986 * gst/rtsp-server/rtsp-media-factory-uri.c:
5987 * gst/rtsp-server/rtsp-media-factory.c:
5988 * gst/rtsp-server/rtsp-media-factory.h:
5989 * gst/rtsp-server/rtsp-media.c:
5990 * gst/rtsp-server/rtsp-media.h:
5991 * gst/rtsp-server/rtsp-sdp.c:
5992 * gst/rtsp-server/rtsp-session-media.c:
5993 * gst/rtsp-server/rtsp-session-media.h:
5994 * gst/rtsp-server/rtsp-session.c:
5995 * gst/rtsp-server/rtsp-session.h:
5996 * gst/rtsp-server/rtsp-stream-transport.c:
5997 * gst/rtsp-server/rtsp-stream-transport.h:
5998 * gst/rtsp-server/rtsp-stream.c:
5999 * gst/rtsp-server/rtsp-stream.h:
6000 rtsp: massive refactoring
6001 Make GObjects from the remaining simple structures.
6002 Remove GstRTSPSessionStream, it's not needed.
6003 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
6004 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
6005 a GstRTSPStream should be transported to a client.
6006 Rename GstRTSPMediaFactory::get_element -> create_element because that
6007 more accurately describes what it does.
6008 Make nice methods instead of poking in the structures.
6009 Move some methods inside the relevant object source code.
6010 Use GPtrArray to store objects instead of plain arrays, it is more
6011 natural and allows us to more easily clean up.
6012 Move the allocation of udp ports to the Stream object. The Stream object
6013 contains the elements needed to stream the media to a client.
6014 Improve the prepare and unprepare methods. Unprepare should now undo
6015 everything prepare did. Improve also async unprepare when doing EOS on
6016 shutdown. Make sure we always unprepare correctly.
6018 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
6020 * gst/rtsp-server/rtsp-client.c:
6021 rtsp-client: Unref server address clients connected to
6022 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
6024 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
6026 * gst/rtsp-server/rtsp-server.c:
6027 rtsp-server: don't ref server socket if it is NULL
6028 Fixes test_bind_already_in_use unit test again after commit 6a497440.
6029 https://bugzilla.gnome.org/show_bug.cgi?id=686644
6031 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
6033 * tests/check/Makefile.am:
6034 tests: Add libgio link dependency
6035 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
6037 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6039 * gst/rtsp-server/rtsp-media-mapping.c:
6040 * gst/rtsp-server/rtsp-media-mapping.h:
6041 rtsp-media-mapping: rename find_media vfunc to find_factory
6042 The virtual method and class method should have the same name
6043 so it is correctly represented in GIR file
6044 https://bugzilla.gnome.org/show_bug.cgi?id=680777
6046 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6048 * gst/rtsp-server/rtsp-auth.c:
6049 * gst/rtsp-server/rtsp-client.c:
6050 * gst/rtsp-server/rtsp-media-factory-uri.c:
6051 * gst/rtsp-server/rtsp-media-factory.c:
6052 * gst/rtsp-server/rtsp-media-mapping.c:
6053 * gst/rtsp-server/rtsp-media.c:
6054 * gst/rtsp-server/rtsp-server.c:
6055 * gst/rtsp-server/rtsp-session-pool.c:
6056 * gst/rtsp-server/rtsp-session.c:
6057 rtsp-server: fixed comments and GIR annotations
6058 https://bugzilla.gnome.org/show_bug.cgi?id=680777
6060 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
6062 * gst/rtsp-server/rtsp-media-mapping.c:
6063 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
6065 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
6067 * gst/rtsp-server/rtsp-server.c:
6068 rtsp-server: allow binding on port 0 (binds on a random port)
6070 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
6072 * gst/rtsp-server/rtsp-server.c:
6073 * gst/rtsp-server/rtsp-server.h:
6074 rtsp-server: add bound-port property
6075 bound-port can be used to retrieve the port number when the server is bound on
6076 port 0, which binds on a random port.
6078 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
6080 * gst/rtsp-server/rtsp-media-factory.c:
6081 * gst/rtsp-server/rtsp-media-factory.h:
6082 rtsp-media-factory: make ::get_element overridable by GI bindings
6083 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
6084 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
6085 as the invoker for ::get_element(), making it overridable by GI generated
6088 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
6090 * gst/rtsp-server/rtsp-media-factory-uri.c:
6091 rtsp-media-factory-uri: don't autoplug parsers in a loop
6092 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
6095 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
6097 * gst/rtsp-server/Makefile.am:
6098 Explicitly link against gio. Fix link error on mac.
6100 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
6102 * gst/rtsp-server/rtsp-session.c:
6103 session: add ttl to the transport header in SETUP
6104 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
6106 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
6108 * gst/rtsp-server/rtsp-client.c:
6109 * gst/rtsp-server/rtsp-client.h:
6110 * gst/rtsp-server/rtsp-media.c:
6111 client: Use client transport settings for multicast if allowed.
6112 This patch makes it possible for the client to send transport settings for
6113 multicast (destination && ttl). Client settings must be explicitly allowed or
6114 the server will use its own settings.
6115 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
6117 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
6120 Automatic update of common submodule
6121 From 6c0b52c to 6bb6951
6123 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
6125 * gst/rtsp-server/rtsp-client.c:
6126 rtsp-client: do not destroy the rtsp watch
6127 Don't destroy the client watch while dispatching. The rtsp watch is
6128 automatically destroyed after the rtsp watch function closed() has
6130 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
6132 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
6135 Automatic update of common submodule
6136 From 4f962f7 to 6c0b52c
6138 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
6140 * gst/rtsp-server/rtsp-media.c:
6141 media: fix check for seekability
6143 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6145 * gst/rtsp-server/rtsp-client.c:
6146 client: use more GIO
6147 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
6149 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6151 * gst/rtsp-server/rtsp-server.c:
6152 server: remove obsolete includes
6154 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
6156 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
6157 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
6158 be available in "on_new_ssrc". The transports are added in
6159 gst_rtsp_media_set_state when going to PLAYING state. However,
6160 "on_new_ssrc" might be called before this happens.
6161 https://bugzilla.gnome.org/show_bug.cgi?id=683304
6163 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
6165 * gst/rtsp-server/rtsp-client.c:
6166 * gst/rtsp-server/rtsp-client.h:
6167 rtsp-client: add signals for rtsp requests (fixes #683287)
6169 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
6171 * gst/rtsp-server/rtsp-client.c:
6172 * gst/rtsp-server/rtsp-client.h:
6173 add new-session signal to rtsp-client (fixes #683058)
6175 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
6178 Automatic update of common submodule
6179 From 668acee to 4f962f7
6181 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
6183 * gst/rtsp-server/rtsp-server.c:
6184 * tests/check/gst/rtspserver.c:
6185 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
6186 Do not assume that *error is set in g_socket_address_enumerator_next.
6187 Added test_bind_already_in_use unit-test.
6188 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
6190 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
6193 Automatic update of common submodule
6194 From 94ccf4c to 668acee
6196 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
6198 * gst/rtsp-server/rtsp-client.c:
6199 * gst/rtsp-server/rtsp-client.h:
6200 rtsp-client: make create_sdp virtual method
6201 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
6203 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6206 Automatic update of common submodule
6207 From 98e386f to 94ccf4c
6209 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6211 * gst/rtsp-server/rtsp-client.c:
6214 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
6216 * gst/rtsp-server/rtsp-client.c:
6217 * gst/rtsp-server/rtsp-client.h:
6218 * gst/rtsp-server/rtsp-server.c:
6219 * gst/rtsp-server/rtsp-server.h:
6220 rtsp-server: use an existing socket to establish HTTP tunnel
6221 Make it possible to transfer a socket from an HTTP server to be used as
6222 an RTSP over HTTP tunnel.
6224 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
6226 * gst/rtsp-server/rtsp-client.c:
6227 * gst/rtsp-server/rtsp-media.c:
6228 * gst/rtsp-server/rtsp-media.h:
6229 rtsp: Handle the blocksize parameter
6230 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
6232 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
6234 * tests/check/Makefile.am:
6235 * tests/check/gst/rtspserver.c:
6236 Have unit test get header from source dir, not installed dir
6237 This makes compilation of unit tests work in a build directory other
6238 than the source directory.
6239 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
6241 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
6243 * gst/rtsp-server/rtsp-media.c:
6244 rtsp-media: update for gst_element_make_from_uri() changes
6246 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
6249 * tests/Makefile.am:
6250 * tests/check/Makefile.am:
6251 * tests/check/gst/rtspserver.c:
6253 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
6255 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
6257 * gst/rtsp-server/rtsp-media.c:
6258 rtsp-media: don't collect media stats when going to NULL
6259 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
6261 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6263 * gst/rtsp-server/rtsp-client.c:
6264 client: don't leak transports
6266 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
6268 * gst/rtsp-server/rtsp-client.c:
6269 rtsp-client: free transport on no_stream in SETUP handler
6271 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
6273 * gst/rtsp-server/rtsp-client.c:
6274 rtsp-client: changed session media iteration
6275 In client_unlink_session: now don't iterate in session->medias
6276 list where items are removed by gst_rtsp_session_release_media.
6277 Instead, repeatedly remove the first item.
6279 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
6281 * gst/rtsp-server/rtsp-client.c:
6282 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
6283 GstRTSPSessionMedia is not a GObject type. When the
6284 GstRTSPSession is freed, it will free the media.
6286 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
6288 * gst/rtsp-server/rtsp-media-factory.c:
6289 factory: plug pad leak in collect_streams
6290 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
6291 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
6292 will take one reference, and the other reference will otherwise
6295 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
6298 configure: suppress some warnings when debug is disabled
6299 Warnings about unused variables should be suppressed if core has the
6300 debug system disabled.
6301 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
6303 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6305 * docs/libs/Makefile.am:
6306 docs: fix build in uninstalled setup
6307 Include gst-plugins-base libs properly.
6309 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
6311 * docs/libs/gst-rtsp-server.types:
6312 docs: include headers defining rtsp-server object types
6313 Fixes compiler warnings during docs build.
6314 https://bugzilla.gnome.org/show_bug.cgi?id=676824
6316 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
6319 configure: Add warning flags for compiler when configuring
6320 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
6322 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6325 Automatic update of common submodule
6326 From 03a0e57 to 98e386f
6328 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6331 Automatic update of common submodule
6332 From 1fab359 to 03a0e57
6334 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
6336 * gst/rtsp-server/rtsp-client.c:
6337 client: fix GSocketAddress leak in gst_rtsp_client_accept
6338 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
6340 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6343 Automatic update of common submodule
6344 From f1b5a96 to 1fab359
6346 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6349 Automatic update of common submodule
6350 From 92b7266 to f1b5a96
6352 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6355 Automatic update of common submodule
6356 From ec1c4a8 to 92b7266
6358 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6361 Automatic update of common submodule
6362 From 3429ba6 to ec1c4a8
6364 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
6366 * gst/rtsp-server/rtsp-auth.c:
6367 * gst/rtsp-server/rtsp-client.c:
6368 * gst/rtsp-server/rtsp-media-factory-uri.c:
6369 * gst/rtsp-server/rtsp-server.c:
6370 rtsp: fix compiler warnings
6371 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
6373 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6376 Automatic update of common submodule
6377 From dc70203 to 3429ba6
6379 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6381 * gst/rtsp-server/rtsp-client.c:
6382 * gst/rtsp-server/rtsp-media-factory.c:
6383 * gst/rtsp-server/rtsp-media-factory.h:
6384 * gst/rtsp-server/rtsp-media.c:
6385 * gst/rtsp-server/rtsp-media.h:
6386 * gst/rtsp-server/rtsp-server.c:
6387 * gst/rtsp-server/rtsp-server.h:
6388 * gst/rtsp-server/rtsp-session-pool.c:
6389 * gst/rtsp-server/rtsp-session-pool.h:
6390 rtsp-server: port to new thread API
6392 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6395 Automatic update of common submodule
6396 From 6db25be to dc70203
6398 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6400 * gst/rtsp-server/rtsp-auth.c:
6401 * gst/rtsp-server/rtsp-auth.h:
6402 * gst/rtsp-server/rtsp-client.c:
6403 rtsp-server: Fix compilation and compiler warnings
6405 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6409 * gst/rtsp-server/Makefile.am:
6410 configure: Modernize autotools setup a bit
6411 Also we now only create tar.bz2 and tar.xz tarballs.
6413 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6416 Automatic update of common submodule
6417 From 464fe15 to 6db25be
6419 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6422 Automatic update of common submodule
6423 From 7fda524 to 464fe15
6425 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6428 * docs/libs/Makefile.am:
6429 * docs/version.entities.in:
6431 * gst/rtsp-server/Makefile.am:
6432 * pkgconfig/Makefile.am:
6433 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
6434 * pkgconfig/gstreamer-rtsp-server.pc.in:
6435 * tests/Makefile.am:
6436 rtsp-server: Update versioning
6438 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6440 Merge remote-tracking branch 'origin/0.10'
6442 gst/rtsp-server/rtsp-session-pool.c
6444 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6446 * gst/rtsp-server/rtsp-session-pool.c:
6447 rtsp-server: Don't use deprecated GLib API
6449 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6451 Replace master with 0.11
6453 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6455 Merge branch 'master' into 0.11
6457 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6459 Merge branch 'master' into 0.11
6461 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
6464 A couple minor typo fixes
6466 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6468 * gst/rtsp-server/rtsp-media.c:
6469 media: fix state of the appqueue
6471 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6473 * gst/rtsp-server/rtsp-media-factory-uri.c:
6474 factory: use videoconvert
6476 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6478 * gst/rtsp-server/rtsp-media-factory-uri.c:
6479 factory: change to new style caps
6481 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6483 * gst/rtsp-server/rtsp-client.c:
6484 * gst/rtsp-server/rtsp-client.h:
6485 * gst/rtsp-server/rtsp-media-factory-uri.c:
6486 * gst/rtsp-server/rtsp-media.c:
6487 * gst/rtsp-server/rtsp-server.c:
6488 * gst/rtsp-server/rtsp-server.h:
6489 * gst/rtsp-server/rtsp-session-pool.c:
6490 rtsp-server: port to GIO
6493 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6496 configure: fix build
6498 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6501 docs: fix for gst_rtsp_server_set_port() -> _set_service()
6502 https://bugzilla.gnome.org/show_bug.cgi?id=666548
6504 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6507 * examples/Makefile.am:
6508 First rule of gst-rtsp-server club: don't talk about gst-phonon
6510 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6513 * pkgconfig/Makefile.am:
6514 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
6515 * pkgconfig/gst-rtsp-server.pc.in:
6516 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
6517 * pkgconfig/gstreamer-rtsp-server.pc.in:
6518 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
6519 For consistency with all other modules.
6521 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6523 * gst/rtsp-server/rtsp-client.c:
6524 rtsp-client: update for new map API
6526 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6529 * bindings/Makefile.am:
6530 * bindings/python/Makefile.am:
6531 * bindings/python/arg-types.py:
6532 * bindings/python/codegen/Makefile.am:
6533 * bindings/python/codegen/__init__.py:
6534 * bindings/python/codegen/argtypes.py:
6535 * bindings/python/codegen/code-coverage.py:
6536 * bindings/python/codegen/codegen.py:
6537 * bindings/python/codegen/definitions.py:
6538 * bindings/python/codegen/defsparser.py:
6539 * bindings/python/codegen/docextract.py:
6540 * bindings/python/codegen/docgen.py:
6541 * bindings/python/codegen/fileprefix.override:
6542 * bindings/python/codegen/fileprefixmodule.c:
6543 * bindings/python/codegen/h2def.py:
6544 * bindings/python/codegen/mergedefs.py:
6545 * bindings/python/codegen/mkskel.py:
6546 * bindings/python/codegen/override.py:
6547 * bindings/python/codegen/reversewrapper.py:
6548 * bindings/python/codegen/scmexpr.py:
6549 * bindings/python/rtspserver-types.defs:
6550 * bindings/python/rtspserver.defs:
6551 * bindings/python/rtspserver.override:
6552 * bindings/python/rtspservermodule.c:
6553 * bindings/python/test.py:
6555 python: remove pygst-based python bindings
6556 pygi is the future, apparently.
6558 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
6561 Automatic update of common submodule
6562 From c463bc0 to 7fda524
6564 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6567 Automatic update of common submodule
6568 From 2a59016 to c463bc0
6570 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6573 Automatic update of common submodule
6574 From 0807187 to 2a59016
6576 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6579 Automatic update of common submodule
6580 From 11f0cd5 to 0807187
6582 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6584 * examples/test-auth.c:
6585 example: update for new caps
6587 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6589 * examples/test-video.c:
6590 * gst/rtsp-server/rtsp-client.c:
6591 * gst/rtsp-server/rtsp-media-factory-uri.c:
6592 * gst/rtsp-server/rtsp-media.c:
6593 * gst/rtsp-server/rtsp-media.h:
6594 * gst/rtsp-server/rtsp-session.c:
6595 * gst/rtsp-server/rtsp-session.h:
6596 rtsp-server: port some more to 0.11
6598 Remove bufferlist stuff
6600 Add queue before appsink now that preroll-queue-len is gone.
6601 Update for request pad changes.
6603 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6605 Merge branch 'master' into 0.11
6607 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
6609 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6610 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
6611 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6613 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
6615 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6616 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
6617 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6619 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6621 Merge branch 'master' into 0.11
6623 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6625 * gst/rtsp-server/rtsp-media.c:
6626 * gst/rtsp-server/rtsp-media.h:
6627 media: add a seekable boolean
6628 Maintain the seekable state with a new variable instead of reusing the
6631 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
6633 * gst/rtsp-server/rtsp-media.c:
6634 Disallow seek in live media
6636 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6638 Merge branch 'master' into 0.11
6640 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
6642 * gst/rtsp-server/rtsp-server.c:
6643 #ifdef statements for windows socket creation were missing
6645 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
6648 Automatic update of common submodule
6649 From a39eb83 to 11f0cd5
6651 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
6654 Automatic update of common submodule
6655 From 605cd9a to a39eb83
6657 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6659 Merge branch 'master' into 0.11
6661 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6663 * gst/rtsp-server/rtsp-client.c:
6664 client: use method to access property
6666 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6668 * gst/rtsp-server/rtsp-media-factory.c:
6669 * gst/rtsp-server/rtsp-media-factory.h:
6670 media-factory: add protocols property
6671 Add a property to configure the allowed protocols in the media created from the
6674 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6676 * gst/rtsp-server/rtsp-media-factory.c:
6677 * gst/rtsp-server/rtsp-media-factory.h:
6678 media-factory: add media-configure signal
6679 Add signal to allow the application to configure the media after it was created
6682 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6684 * gst/rtsp-server/rtsp-client.c:
6685 client: use method to access property
6687 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6689 * gst/rtsp-server/rtsp-media-factory.c:
6690 * gst/rtsp-server/rtsp-media-factory.h:
6691 media-factory: add protocols property
6692 Add a property to configure the allowed protocols in the media created from the
6695 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6697 * gst/rtsp-server/rtsp-media-factory.c:
6698 * gst/rtsp-server/rtsp-media-factory.h:
6699 media-factory: add media-configure signal
6700 Add signal to allow the application to configure the media after it was created
6703 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6705 Merge branch 'master' into 0.11
6707 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6709 * gst/rtsp-server/rtsp-client.c:
6710 client: use media multicast group
6712 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6714 * gst/rtsp-server/rtsp-media-factory.h:
6715 * gst/rtsp-server/rtsp-server.h:
6716 * gst/rtsp-server/rtsp-session-pool.h:
6717 * gst/rtsp-server/rtsp-session.h:
6720 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6722 * gst/rtsp-server/rtsp-client.c:
6723 * gst/rtsp-server/rtsp-sdp.h:
6724 sdp: copy and free the server ip address
6725 Copy and free the server ip address to make memory management easier later.
6727 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6729 * gst/rtsp-server/rtsp-media-factory.c:
6730 media-factory: configure multicast in media
6732 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6734 * gst/rtsp-server/rtsp-media.c:
6735 * gst/rtsp-server/rtsp-media.h:
6736 media: add property for multicast group
6737 Add a property to configure the multicast group in the media.
6738 Based on patches from Marc Leeman and Robert Krakora.
6740 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6742 * gst/rtsp-server/rtsp-media-factory.c:
6743 * gst/rtsp-server/rtsp-media-factory.h:
6744 media-factory: add property for multicast group
6745 Add a property to configure the multicast group in the media factory.
6746 Based on patches from Marc Leeman and Robert Krakora.
6748 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6750 * gst/rtsp-server/rtsp-client.c:
6751 client: do configuration of transport in one place
6752 Move the configuration of the transport destination address to where we also
6753 configure the other bits.
6755 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6757 * gst/rtsp-server/rtsp-client.c:
6758 client: use media multicast group
6760 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6762 * gst/rtsp-server/rtsp-media-factory.h:
6763 * gst/rtsp-server/rtsp-server.h:
6764 * gst/rtsp-server/rtsp-session-pool.h:
6765 * gst/rtsp-server/rtsp-session.h:
6768 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6770 * gst/rtsp-server/rtsp-client.c:
6771 * gst/rtsp-server/rtsp-sdp.h:
6772 sdp: copy and free the server ip address
6773 Copy and free the server ip address to make memory management easier later.
6775 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6777 * gst/rtsp-server/rtsp-media-factory.c:
6778 media-factory: configure multicast in media
6780 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6782 * gst/rtsp-server/rtsp-media.c:
6783 * gst/rtsp-server/rtsp-media.h:
6784 media: add property for multicast group
6785 Add a property to configure the multicast group in the media.
6786 Based on patches from Marc Leeman and Robert Krakora.
6788 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6790 * gst/rtsp-server/rtsp-media-factory.c:
6791 * gst/rtsp-server/rtsp-media-factory.h:
6792 media-factory: add property for multicast group
6793 Add a property to configure the multicast group in the media factory.
6794 Based on patches from Marc Leeman and Robert Krakora.
6796 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6798 * gst/rtsp-server/rtsp-client.c:
6799 client: do configuration of transport in one place
6800 Move the configuration of the transport destination address to where we also
6801 configure the other bits.
6803 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6805 Merge branch 'master' into 0.11
6807 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6809 * gst/rtsp-server/rtsp-client.c:
6810 client: destroy pipeline on client disconnect with no prior TEARDOWN.
6811 The problem occurs when the client abruptly closes the connection without
6812 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
6813 server is where the pipeline gets torn down. Since this handler is not called,
6814 the pipeline remains and is up and running. Subsequent clients get their own
6815 pipelines and if the do not issue TEARDOWNs then those pipelines will also
6816 remain up and running. This is a resource leak.
6818 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6820 Merge branch 'master' into 0.11
6822 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
6824 * gst/rtsp-server/rtsp-media-factory.c:
6825 * gst/rtsp-server/rtsp-media-factory.h:
6826 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
6827 For example, it can be used to retrieve source elements like appsrc, in a more
6828 convenient way than subclassing get_element.
6830 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6832 Merge branch 'master' into 0.11
6834 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
6836 * gst/rtsp-server/rtsp-server.c:
6837 rtsp-server: hold on to reference while using object
6839 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6841 * gst/rtsp-server/rtsp-media.c:
6844 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6847 configure: use unstable api
6849 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
6851 * gst/rtsp-server/rtsp-client.c:
6852 client: fix reference counting
6854 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
6856 * gst/rtsp-server/rtsp-client.c:
6857 * gst/rtsp-server/rtsp-media.c:
6858 fix compiler warnings about unused variables
6860 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
6862 * examples/test-launch.c:
6863 * examples/test-readme.c:
6864 * examples/test-uri.c:
6865 * examples/test-video.c:
6866 examples: tell rtsp uri when ready
6868 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
6871 Automatic update of common submodule
6872 From 69b981f to 605cd9a
6874 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6876 * gst/rtsp-server/rtsp-client.c:
6877 client: update for buffer API change
6879 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6881 * gst/rtsp-server/Makefile.am:
6882 Makefile.am: 0.10 => @GST_MAJORMINOR@
6884 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6886 * gst/rtsp-server/rtsp-media-factory-uri.c:
6887 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
6889 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6891 * gst/rtsp-server/.gitignore:
6892 .gitignore: 0.10 => 0.11
6894 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6896 * gst/rtsp-server/Makefile.am:
6897 Makefile.am: 0.10 => @GST_MAJORMINOR@
6899 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6901 Merge branch 'master' into 0.11
6903 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
6906 Automatic update of common submodule
6907 From 9e5bbd5 to 69b981f
6909 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
6912 Automatic update of common submodule
6913 From fd35073 to 9e5bbd5
6915 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
6918 Automatic update of common submodule
6919 From 46dfcea to fd35073
6921 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6923 * gst/rtsp-server/rtsp-media-factory-uri.c:
6924 * gst/rtsp-server/rtsp-media.c:
6925 media: port to new caps API
6927 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6929 Merge branch 'master' into 0.11
6931 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
6933 * bindings/vala/gst-rtsp-server-0.10.vapi:
6934 Updated Vala bindings.
6935 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6937 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
6939 * gst/rtsp-server/rtsp-server.c:
6940 * gst/rtsp-server/rtsp-server.h:
6941 Add a signal for newly connected clients.
6942 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6944 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
6946 * bindings/python/rtspserver.override:
6947 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
6949 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6951 * gst/rtsp-server/Makefile.am:
6952 * gst/rtsp-server/rtsp-client.c:
6953 * gst/rtsp-server/rtsp-funnel.c:
6954 * gst/rtsp-server/rtsp-funnel.h:
6955 * gst/rtsp-server/rtsp-media.c:
6956 rtsp-server: port to 0.11
6958 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6963 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6965 Merge branch 'master' into 0.11
6970 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6973 Automatic update of common submodule
6974 From c3cafe1 to 46dfcea
6976 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
6978 * bindings/python/Makefile.am:
6979 * bindings/python/rtspserver.defs:
6980 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
6982 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
6984 * bindings/python/arg-types.py:
6985 python bindings: add GstRTSPUrlParam
6986 Needed to implement MediaFactory virtual proxies
6988 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
6990 * bindings/python/arg-types.py:
6991 python bindings: fix returning GstRTSPUrl types
6993 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
6995 * bindings/python/arg-types.py:
6996 python bindings: add arg type for GstRTSPUrl
6998 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
7000 * bindings/python/rtspserver.defs:
7001 python bindings: fix the definition of MediaFactory.collect_stream
7003 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
7006 Automatic update of common submodule
7007 From 1ccbe09 to c3cafe1
7009 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7012 Automatic update of common submodule
7013 From 193b717 to 1ccbe09
7015 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
7018 Automatic update of common submodule
7019 From b77e2bf to 193b717
7021 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7024 build: Include lcov.mak to allow test coverage report generation
7026 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7029 Automatic update of common submodule
7030 From d8814b6 to b77e2bf
7032 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7035 Automatic update of common submodule
7036 From 6aaa286 to d8814b6
7038 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
7041 Automatic update of common submodule
7042 From 6aec6b9 to 6aaa286
7044 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
7047 autogen: wingo signed comment
7049 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
7051 * gst/rtsp-server/rtsp-session-pool.c:
7052 session: use full charset for RTSP session ID
7053 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
7054 session ID more difficult.
7055 https://bugzilla.gnome.org/show_bug.cgi?id=643812
7057 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7059 * gst/rtsp-server/Makefile.am:
7060 rtsp-server: Don't install the funnel header
7062 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
7065 Automatic update of common submodule
7066 From 1de7f6a to 6aec6b9
7068 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7071 configure: require core/base 0.10.31
7072 Needed at least for gst_plugin_feature_rank_compare_func().
7074 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
7077 Automatic update of common submodule
7078 From f94d739 to 1de7f6a
7080 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7082 * gst/rtsp-server/rtsp-media.c:
7083 media: remove more unused code
7085 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7087 * gst/rtsp-server/rtsp-media.c:
7088 * gst/rtsp-server/rtsp-media.h:
7089 media: remove duplicate filtering
7090 Remove the duplicate filtering code now that we have a released -good version.
7091 Give a warning instead.
7093 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7095 * gst/rtsp-server/rtsp-media-factory.c:
7096 * gst/rtsp-server/rtsp-media.c:
7097 media: fix default buffer size
7099 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7101 * gst/rtsp-server/rtsp-media-factory.c:
7102 * gst/rtsp-server/rtsp-media-factory.h:
7103 media-factory: add property to configure the buffer-size
7104 Add a property to configure the kernel UDP buffer size.
7106 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7108 * gst/rtsp-server/rtsp-media.c:
7109 * gst/rtsp-server/rtsp-media.h:
7110 media: add property to configure kernel buffer sizes
7111 Add a property to configure the kernel UDP buffer size.
7113 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7116 configure: set PYGOBJECT_REQ before using it
7117 https://bugzilla.gnome.org/show_bug.cgi?id=640641
7119 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7122 docs: recursive into sub-directories on 'make upload'
7124 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7126 * docs/libs/gst-rtsp-server-docs.sgml:
7127 * docs/version.entities.in:
7128 docs: mention full version these docs are for, not just major-minor
7130 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7135 === release 0.10.8 ===
7137 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7142 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7144 * gst/rtsp-server/rtsp-server.c:
7145 rtsp-server: clarify docs a little
7147 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7149 * gst/rtsp-server/rtsp-media.c:
7150 media: init debug category before starting thread
7152 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7154 * gst/rtsp-server/rtsp-auth.c:
7155 auth: add realm to make it more spec compliant
7157 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7159 * gst/rtsp-server/rtsp-server.c:
7160 * gst/rtsp-server/rtsp-server.h:
7163 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7165 * examples/test-video.c:
7166 example: improve example docs a little
7168 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7170 * gst/rtsp-server/rtsp-server.c:
7171 server: ensure the watch has a ref to the server
7173 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7175 * gst/rtsp-server/rtsp-server.c:
7176 server: simpify channel function
7178 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7180 * gst/rtsp-server/rtsp-server.c:
7181 * gst/rtsp-server/rtsp-server.h:
7182 server: simplify management of channel and source
7183 We don't need to keep around the channel and source objects. Let the mainloop
7184 and the source manage the source and channel respectively.
7186 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7192 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7195 * tests/Makefile.am:
7196 * tests/test-cleanup.c:
7197 tests: add tests directory and cleanup test
7199 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7201 * gst/rtsp-server/rtsp-media-factory-uri.c:
7202 * gst/rtsp-server/rtsp-media-factory.c:
7203 * gst/rtsp-server/rtsp-media-mapping.c:
7204 * gst/rtsp-server/rtsp-media.c:
7205 * gst/rtsp-server/rtsp-session-pool.c:
7206 * gst/rtsp-server/rtsp-session.c:
7207 server: improve debugging in various objects
7209 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7211 * gst/rtsp-server/rtsp-server.c:
7212 server: chain up to the parent finalize
7214 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
7216 * bindings/python/rtspserver-types.defs:
7217 * bindings/python/rtspserver.defs:
7218 * bindings/python/rtspserver.override:
7219 * bindings/python/test.py:
7220 gst-rtsp-server: update python bindings
7222 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7224 * gst/rtsp-server/rtsp-client.c:
7225 client: use the response from the clientstate
7226 Create the response object only once and store in the client state.
7227 Make all methods use the state response,
7229 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7231 * gst/rtsp-server/rtsp-server.c:
7232 server: use signal to keep track of clients
7233 Keep track of all the clients that the server creates and remove them when they
7234 fire the 'closed' signal.
7236 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7238 * gst/rtsp-server/rtsp-client.c:
7239 * gst/rtsp-server/rtsp-client.h:
7240 client: emit signal when closing
7242 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7244 * examples/.gitignore:
7245 * examples/Makefile.am:
7246 * examples/test-auth.c:
7247 * examples/test-video.c:
7248 * gst/rtsp-server/rtsp-auth.c:
7249 * gst/rtsp-server/rtsp-auth.h:
7250 * gst/rtsp-server/rtsp-client.c:
7251 * gst/rtsp-server/rtsp-media-factory.c:
7252 * gst/rtsp-server/rtsp-media.c:
7253 * gst/rtsp-server/rtsp-media.h:
7254 * gst/rtsp-server/rtsp-session-pool.h:
7255 * gst/rtsp-server/rtsp-session.h:
7256 media: enable per factory authorisations
7257 Allow for adding a GstRTSPAuth on the factory and media level and check
7258 permissions when accessing the factory.
7259 Add hints to the auth methods for future more fine grained authorisation.
7260 Add example application for per factory authentication.
7262 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7264 * gst/rtsp-server/rtsp-auth.c:
7265 * gst/rtsp-server/rtsp-auth.h:
7266 * gst/rtsp-server/rtsp-client.c:
7267 * gst/rtsp-server/rtsp-client.h:
7268 * gst/rtsp-server/rtsp-params.c:
7269 * gst/rtsp-server/rtsp-params.h:
7270 rtsp-server: Pass ClientState structure arround
7271 Pass the collected information for the ongoing request in a GstRTSPClientState
7272 structure that we can then pass around to simplify the method arguments. This
7273 will also be handy when we implement logging functionality.
7275 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7277 * gst/rtsp-server/rtsp-media-factory.c:
7278 * gst/rtsp-server/rtsp-media-factory.h:
7279 media-factory: add methods to configure authorisation
7281 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7283 * gst/rtsp-server/rtsp-client.c:
7284 client: unref auth in finalize
7286 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7288 * gst/rtsp-server/rtsp-server.c:
7289 server: unref auth in finalize
7291 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7293 * docs/libs/gst-rtsp-server-docs.sgml:
7294 * docs/libs/gst-rtsp-server-sections.txt:
7295 * docs/libs/gst-rtsp-server.types:
7298 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7300 * gst/rtsp-server/rtsp-server.c:
7301 * gst/rtsp-server/rtsp-server.h:
7302 server: separate create and accept
7303 Create separate create and accept methods so that subclasses can create custom
7305 Configure the server in the client object and prepare for keeping track of
7308 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7310 * gst/rtsp-server/rtsp-client.c:
7311 * gst/rtsp-server/rtsp-client.h:
7312 client: add support for setting the server.
7313 Add support for keeping a ref to the server that started this client
7316 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7318 * gst/rtsp-server/rtsp-auth.c:
7319 auth: fix memleak and add some docs
7320 Fix a memleak of the basic auth token.
7321 Add docs for the helper function
7323 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7325 * gst/rtsp-server/rtsp-auth.c:
7326 * gst/rtsp-server/rtsp-auth.h:
7327 * gst/rtsp-server/rtsp-client.c:
7328 client: delegate setup of auth to the manager
7329 Delegate the configuration of the authentication tokens to the manager object
7332 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7334 * examples/test-video.c:
7335 * gst/rtsp-server/Makefile.am:
7336 * gst/rtsp-server/rtsp-auth.c:
7337 * gst/rtsp-server/rtsp-auth.h:
7338 * gst/rtsp-server/rtsp-client.c:
7339 * gst/rtsp-server/rtsp-client.h:
7340 * gst/rtsp-server/rtsp-server.c:
7341 * gst/rtsp-server/rtsp-server.h:
7342 auth: add authentication object
7343 Add an object that can check the authorization of requests.
7344 Implement basic authentication.
7345 Add example authentication to test-video
7347 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7349 * gst/rtsp-server/rtsp-server.c:
7350 * gst/rtsp-server/rtsp-server.h:
7351 server: move includes back
7352 the includes are needed for sockaddr_in.
7354 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7356 * gst/rtsp-server/rtsp-client.c:
7357 * gst/rtsp-server/rtsp-client.h:
7358 * gst/rtsp-server/rtsp-server.c:
7359 * gst/rtsp-server/rtsp-server.h:
7360 rtsp: move network includes where they are needed
7362 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
7364 * gst/rtsp-server/rtsp-media.h:
7365 rtsp-media.h: Minor corrections in comments.
7368 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
7371 Automatic update of common submodule
7372 From e572c87 to f94d739
7374 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7378 * docs/libs/.gitignore:
7379 * examples/.gitignore:
7380 * gst/rtsp-server/.gitignore:
7383 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7385 * docs/libs/Makefile.am:
7386 docs: We don't build ps/pdf for API reference docs
7388 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7391 Automatic update of common submodule
7392 From ccbaa85 to e572c87
7394 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7397 Automatic update of common submodule
7398 From 46445ad to ccbaa85
7400 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7402 * gst/rtsp-server/Makefile.am:
7403 * gst/rtsp-server/fs-funnel.c:
7404 * gst/rtsp-server/fs-funnel.h:
7405 * gst/rtsp-server/rtsp-funnel.c:
7406 * gst/rtsp-server/rtsp-funnel.h:
7407 * gst/rtsp-server/rtsp-media.c:
7408 funnel: rename fsfunnel to rtspfunnel
7409 Rename the funnel to avoid conflicts with the farsight one.
7411 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7413 * gst/rtsp-server/Makefile.am:
7414 * gst/rtsp-server/fs-funnel.c:
7415 * gst/rtsp-server/fs-funnel.h:
7416 * gst/rtsp-server/rtsp-media.c:
7417 rtsp-media: add and use fsfunnel
7418 Add a copy of fsfunnel to the build because input-selector removed the (broken)
7419 select-all property that we need.
7421 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7423 * gst/rtsp-server/Makefile.am:
7424 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
7425 Use PKG_CONFIG_PATH specified at configure time (if any) as well
7426 for the g-ir-compiler, rather than just assuming the env var has
7429 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7436 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
7438 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7441 * gst/rtsp-server/Makefile.am:
7442 gobject-introspection: fix g-i build for uninstalled setup
7443 Requires gst-plugins-base git (> 0.10.31.2).
7445 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7447 * examples/test-uri.c:
7448 examples: add some more options and comments
7450 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7452 * gst/rtsp-server/rtsp-media-factory-uri.c:
7453 factory-uri: use right property type
7455 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7457 * gst/rtsp-server/rtsp-media-factory-uri.c:
7458 factory-uri: attempt to configure buffer-lists
7459 Attempt to configure buffer lists in the payloader for improved performance.
7461 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7463 * gst/rtsp-server/rtsp-media.c:
7464 media: attempt to configure bigger UDP buffers
7465 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
7466 send buffers with high bitrate streams.
7468 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
7470 * gst/rtsp-server/rtsp-client.c:
7471 client: use the socket length from getsockname
7472 Use the length returned by getsockname to perform the getnameinfo call because
7473 the size can depend on the socket type and platform.
7476 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7478 * docs/libs/gst-rtsp-server-docs.sgml:
7479 * docs/libs/gst-rtsp-server-sections.txt:
7480 docs: add uri factory to the docs
7482 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7484 * gst/rtsp-server/rtsp-client.c:
7485 * gst/rtsp-server/rtsp-media.h:
7488 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7490 * gst/rtsp-server/rtsp-client.c:
7491 * gst/rtsp-server/rtsp-media.c:
7492 * gst/rtsp-server/rtsp-media.h:
7493 * gst/rtsp-server/rtsp-session.c:
7494 * gst/rtsp-server/rtsp-session.h:
7495 rtsp-server: add support for buffer lists
7496 Add support for sending bufferlists received from appsink.
7499 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7501 * gst/rtsp-server/rtsp-client.c:
7502 * gst/rtsp-server/rtsp-media.c:
7503 * gst/rtsp-server/rtsp-media.h:
7504 * gst/rtsp-server/rtsp-sdp.c:
7505 media: make method to retrieve the play range
7506 Make a method to retrieve the playback range so that we can conditionally create
7507 a different range for the SDP and the PLAY requests.
7509 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7511 * gst/rtsp-server/rtsp-media.c:
7512 * gst/rtsp-server/rtsp-media.h:
7513 media: add signal to notify of state changes
7515 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7517 * gst/rtsp-server/rtsp-client.h:
7518 client: cleanup headers
7520 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7522 * gst/rtsp-server/rtsp-client.c:
7525 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7527 * gst/rtsp-server/rtsp-media-factory-uri.c:
7528 * gst/rtsp-server/rtsp-media-factory-uri.h:
7529 factory-uri: add support for gstpay
7530 Add an option to prefer gstpay over decoder + raw payloader.
7532 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7534 * gst/rtsp-server/rtsp-media-factory-uri.c:
7535 * gst/rtsp-server/rtsp-media-factory-uri.h:
7536 factory-uri: rework the autoplugger.
7537 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
7540 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7542 * gst/rtsp-server/rtsp-media-factory-uri.c:
7543 factory-uri: use better factory filter
7544 Make better payloader filter based on autoplug rank and RTP use case.
7546 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7549 Automatic update of common submodule
7550 From 169462a to 46445ad
7552 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7554 * gst/rtsp-server/rtsp-server.c:
7555 server: set SO_REUSEADDR before bind
7556 Set the SO_REUSEADDR _before_ bind() to make it actually work.
7558 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7560 * gst/rtsp-server/rtsp-media.c:
7561 * gst/rtsp-server/rtsp-media.h:
7562 media: emit prepared signal when prepared
7563 Make a 'prepared' signal and emit it when we successfully prepared the element.
7564 This signal can be used to configure the media object after it has been prepared
7567 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
7570 Automatic update of common submodule
7571 From 011bcc8 to 169462a
7573 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
7575 python an optional dependency
7576 * configure.ac: Move up valgrind and g-i checks. Make the python
7577 dependency optional, as it was before.
7579 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7581 Merge branch 'master' into 0.11
7586 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7588 * gst/rtsp-server/rtsp-media.c:
7589 media: update range when active clients changed
7590 When we changed the number of active clients, update the current range
7591 information because we want the second client connecting to a shared resource
7592 continue from where the stream currently.
7594 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7596 * gst/rtsp-server/rtsp-media-factory-uri.c:
7597 * gst/rtsp-server/rtsp-media-factory-uri.h:
7598 factory-uri: add colorspace and fix pt
7599 Rework the way we pass data to the autoplugger.
7600 When we have raw caps, plug a converter element to make pluggin to raw
7601 payloaders more successful.
7602 Make sure all dynamically plugged payloaders have a unique payload types.
7604 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7606 * examples/Makefile.am:
7607 * examples/test-uri.c:
7608 example: add example of the uri factory
7610 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7612 * gst/rtsp-server/Makefile.am:
7613 * gst/rtsp-server/rtsp-media-factory-uri.c:
7614 * gst/rtsp-server/rtsp-media-factory-uri.h:
7615 * gst/rtsp-server/rtsp-server.h:
7616 factory-uri: add a factory to stream any URI
7617 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
7620 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7622 * gst/rtsp-server/rtsp-media.c:
7623 * gst/rtsp-server/rtsp-media.h:
7624 media: ignore spurious ASYNC_DONE messages
7625 When we are dynamically adding pads, the addition of the udpsrc elements will
7626 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
7627 the real ASYNC_DONE when everything is prerolled.
7629 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7631 * gst/rtsp-server/rtsp-media-factory.c:
7632 * gst/rtsp-server/rtsp-media-factory.h:
7633 media-factory: make lock macro
7635 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
7637 * gst/rtsp-server/rtsp-client.c:
7638 rtsp-server: Remove unused variable and dead assignment
7640 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
7642 * examples/test-launch.c:
7643 * examples/test-mp4.c:
7644 * examples/test-ogg.c:
7645 * examples/test-readme.c:
7646 * examples/test-sdp.c:
7647 * examples/test-video.c:
7648 examples: Run gst-indent
7650 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
7652 * gst/rtsp-server/rtsp-client.c:
7653 * gst/rtsp-server/rtsp-media-factory.c:
7654 * gst/rtsp-server/rtsp-media-mapping.c:
7655 * gst/rtsp-server/rtsp-media.c:
7656 * gst/rtsp-server/rtsp-params.c:
7657 * gst/rtsp-server/rtsp-sdp.c:
7658 * gst/rtsp-server/rtsp-server.c:
7659 * gst/rtsp-server/rtsp-session-pool.c:
7660 * gst/rtsp-server/rtsp-session.c:
7661 rtsp-server: Run gst-indent
7662 Since it wasn't using the upstream common previously, there was no
7663 indentation check before commiting.
7665 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
7667 * gst/rtsp-server/rtsp-media-mapping.h:
7668 * gst/rtsp-server/rtsp-media.c:
7669 * gst/rtsp-server/rtsp-media.h:
7670 * gst/rtsp-server/rtsp-sdp.c:
7671 * gst/rtsp-server/rtsp-session-pool.h:
7672 * gst/rtsp-server/rtsp-session.c:
7673 * gst/rtsp-server/rtsp-session.h:
7674 rtsp-server: Some more doc fixups
7676 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7679 Makefile: Add cruft-cleaning support
7681 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7686 * docs/libs/Makefile.am:
7687 * docs/libs/gst-rtsp-server-docs.sgml:
7688 * docs/libs/gst-rtsp-server-sections.txt:
7689 * docs/libs/gst-rtsp-server.types:
7690 * docs/version.entities.in:
7691 docs: Add gtk-doc build system
7693 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7695 * gst/rtsp-server/Makefile.am:
7696 Makefile.am: Use standard GIR make behaviour
7698 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7702 autogen/configure: Bring more in sync to standard gst module behaviour
7704 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7706 * gst/rtsp-server/rtsp-media.c:
7707 media: warn and fail when gstrtpbin is not found
7709 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7712 configure: open 0.11 branch
7714 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
7718 Add common submodule
7720 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
7723 * common/Makefile.am:
7724 * common/c-to-xml.py:
7726 * common/coverage/coverage-report-entry.pl:
7727 * common/coverage/coverage-report.pl:
7728 * common/coverage/coverage-report.xsl:
7729 * common/coverage/lcov.mak:
7730 * common/gettext.patch:
7731 * common/glib-gen.mak:
7732 * common/gst-autogen.sh:
7733 * common/gst-xmlinspect.py:
7735 * common/gstdoc-scangobj:
7736 * common/gtk-doc-plugins.mak:
7737 * common/gtk-doc.mak:
7738 * common/m4/.gitignore:
7739 * common/m4/Makefile.am:
7741 * common/m4/as-ac-expand.m4:
7742 * common/m4/as-auto-alt.m4:
7743 * common/m4/as-compiler-flag.m4:
7744 * common/m4/as-compiler.m4:
7745 * common/m4/as-docbook.m4:
7746 * common/m4/as-libtool-tags.m4:
7747 * common/m4/as-libtool.m4:
7748 * common/m4/as-python.m4:
7749 * common/m4/as-scrub-include.m4:
7750 * common/m4/as-version.m4:
7751 * common/m4/ax_create_stdint_h.m4:
7752 * common/m4/check.m4:
7753 * common/m4/glib-gettext.m4:
7754 * common/m4/gst-arch.m4:
7755 * common/m4/gst-args.m4:
7756 * common/m4/gst-check.m4:
7757 * common/m4/gst-debuginfo.m4:
7758 * common/m4/gst-default.m4:
7759 * common/m4/gst-doc.m4:
7760 * common/m4/gst-error.m4:
7761 * common/m4/gst-feature.m4:
7762 * common/m4/gst-function.m4:
7763 * common/m4/gst-gettext.m4:
7764 * common/m4/gst-glib2.m4:
7765 * common/m4/gst-libxml2.m4:
7766 * common/m4/gst-plugindir.m4:
7767 * common/m4/gst-valgrind.m4:
7768 * common/m4/gtk-doc.m4:
7769 * common/m4/introspection.m4:
7771 * common/mangle-tmpl.py:
7772 * common/plugins.xsl:
7774 * common/release.mak:
7775 * common/scangobj-merge.py:
7776 * common/upload.mak:
7777 common: Remove static version
7779 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
7781 * common/m4/introspection.m4:
7782 Update introspection.m4 to match usage
7784 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7788 Remove old stuff from the README
7790 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7795 === release 0.10.7 ===
7797 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7802 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7804 * examples/test-ogg.c:
7805 test-ogg: remove parsers
7806 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
7807 buffers with timestamps. Using the parsers also seems to break things.
7809 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7811 * bindings/vala/gst-rtsp-server-0.10.vapi:
7812 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7813 Updated Vala bindings
7815 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7817 * common/m4/introspection.m4:
7819 * gst/rtsp-server/Makefile.am:
7820 Added initial gobject-introspection support
7822 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7824 * gst/rtsp-server/rtsp-media-factory.c:
7825 media-factory: don't use host for shared hash key
7826 When we generate the key to share made between connections, don't include the
7827 host used to connect so that we can share media even if between clients that
7828 connected with localhost and ones with the ip address.
7830 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7832 * bindings/vala/Makefile.am:
7833 build: fix distcheck
7835 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7837 * bindings/vala/gst-rtsp-server-0.10.vapi:
7838 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7839 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7840 Update Vala bindings
7842 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7844 * bindings/vala/Makefile.am:
7846 Fix configure checks and installation location for Vala bindings
7849 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7854 === release 0.10.6 ===
7856 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7859 configure: release 0.10.6
7861 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7863 * gst/rtsp-server/rtsp-media.c:
7864 media: help the compiler a little
7866 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7868 * gst/rtsp-server/rtsp-media.c:
7869 * gst/rtsp-server/rtsp-media.h:
7870 * gst/rtsp-server/rtsp-session.c:
7871 media: cleanup media transport before freeing
7872 Cleanup the media transport data before freeing. In particular, remove the qdata
7873 from the rtpsource object.
7875 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7877 * gst/rtsp-server/rtsp-media-factory.c:
7878 * gst/rtsp-server/rtsp-media-factory.h:
7879 * gst/rtsp-server/rtsp-media.c:
7880 * gst/rtsp-server/rtsp-media.h:
7881 media-factory: add eos-shutdown property
7882 Add an eos-shutdown property that will send an EOS to the pipeline before
7883 shutting it down. This allows for nice cleanup in case of a muxer.
7886 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7888 * gst/rtsp-server/rtsp-media.c:
7889 * gst/rtsp-server/rtsp-media.h:
7890 media: use multiudpsink send-duplicates when we can
7891 If we have a new enough multiudpsink with the send-duplicates property, use this
7892 instead of doing our own filtering. Our custom filtering code should eventually
7893 be removed when we can depend on a released -good.
7895 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7897 * gst/rtsp-server/rtsp-media.c:
7898 media: don't leak destinations
7899 Refactor and cleanup the destinations array when the stream is destroyed.
7901 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7903 * gst/rtsp-server/rtsp-media.c:
7904 * gst/rtsp-server/rtsp-media.h:
7905 media: don't add udp addresses multiple times
7906 Keep track of the udp addresses we added to udpsink and never add the same udp
7907 destination twice. This avoids duplicate packets when using multicast.
7909 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7911 * gst/rtsp-server/rtsp-server.c:
7912 server: disable use of SO_LINGER
7913 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
7914 server close()s the connection.
7916 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7918 * gst/rtsp-server/rtsp-server.c:
7919 server: use 5 second linger period in SO_LINGER
7920 Wait 5 seconds before clearing the send buffers and reseting the connection with
7921 the client when we do a close. This should be enough time to get the message to
7925 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
7927 * gst/rtsp-server/rtsp-server.c:
7928 server: use SO_LINGER
7929 SO_LINGER on the socket will make sure that any pending data on the socket is
7930 flushed ASAP and that the socket connection is reset. This makes sure that the
7931 socket can be reused immediately.
7934 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7937 README: add blurb about shared media factories
7939 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
7941 * gst/rtsp-server/rtsp-media.c:
7942 Add stdlib.h for atoi()
7944 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7946 * bindings/python/Makefile.am:
7947 * bindings/vala/Makefile.am:
7948 build: distcheck fixes
7949 Fix 'make distcheck', somewhat (it still fails because it tries to
7950 install files into /usr/share/vala/vapi/ irrespective of the
7953 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7956 configure: bump core/base requirements to released version
7957 Makes things less confusing for people.
7959 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7962 configure: fail if GStreamer core/base requirements are not met
7964 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7966 * gst/rtsp-server/rtsp-client.c:
7967 client: improve client cleanups
7968 Make sure the session does not timeout when using TCP. We need to do this
7969 because quicktime player does not send RTCP for some reason in tunneled
7971 Refactor some cleanup code.
7974 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7976 * gst/rtsp-server/rtsp-session.c:
7977 * gst/rtsp-server/rtsp-session.h:
7978 session: add support for prevent session timeouts
7979 Add an atomix counter to prevent session timeouts when we are, for example,
7982 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7984 * gst/rtsp-server/rtsp-client.c:
7985 client: fix unlink on session timeouts
7986 When our session times out, make sure we unlink all streams in this
7988 Remove the tunnelid when closing the connection.
7990 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7992 * gst/rtsp-server/rtsp-session.c:
7993 session: small cleanups
7995 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7997 * gst/rtsp-server/rtsp-client.c:
7998 client: handle lost_tunnel callbacks
7999 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
8000 hashtable so that we can reuse it for when the client reopens the POST
8002 Close the connection after a TEARDOWN.
8003 Make sure or watchid is cleared when the watch is removed.
8006 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8008 * gst/rtsp-server/rtsp-client.c:
8009 * gst/rtsp-server/rtsp-media.c:
8010 * gst/rtsp-server/rtsp-sdp.c:
8011 rtsp-server: add more support for multicast
8013 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8016 * gst/rtsp-server/rtsp-media.c:
8017 * gst/rtsp-server/rtsp-media.h:
8018 media: allow configuration of allowed lower transport
8020 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8022 * gst/rtsp-server/rtsp-client.h:
8023 * gst/rtsp-server/rtsp-media.c:
8024 * gst/rtsp-server/rtsp-media.h:
8025 * gst/rtsp-server/rtsp-sdp.c:
8026 * gst/rtsp-server/rtsp-sdp.h:
8027 * gst/rtsp-server/rtsp-server.c:
8028 rtsp: keep track of server ip and ipv6
8029 Keep track of how the client connected to the server and setup the udp ports
8030 with the same protocol.
8031 Copy the server ip address in the SDP so that clients can send RTCP back to
8034 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8036 * gst/rtsp-server/rtsp-session.c:
8039 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8041 * gst/rtsp-server/rtsp-client.c:
8042 client: use right size for malloc
8044 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8046 * gst/rtsp-server/rtsp-server.c:
8047 server: comment ipv6 server listening address
8049 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8051 * gst/rtsp-server/rtsp-media.c:
8052 media: allow for ipv6 sockets
8054 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8056 * gst/rtsp-server/rtsp-server.c:
8057 * gst/rtsp-server/rtsp-server.h:
8058 server: rework server part
8059 Allow setting a bind address, make sure we can deal with ipv6.
8060 Remove the port property and change with the service property.
8062 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8064 * gst/rtsp-server/rtsp-media.h:
8065 media: update comments a little
8067 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8069 * gst/rtsp-server/rtsp-client.c:
8070 client: make content-base better
8071 Use the URI formatting functions to make a content-base. Also make sure that
8072 there is a trailing / at the end.
8074 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8076 * gst/rtsp-server/rtsp-client.c:
8077 client: guard against invalid paths
8079 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8081 * examples/test-video.c:
8082 test: catch server bind errors
8084 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
8086 * gst/rtsp-server/rtsp-media.c:
8087 rtspmedia: emit "unprepared" if _prepare fails.
8088 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
8089 media object is removed from its factory's cache.
8091 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8093 * gst/rtsp-server/rtsp-media.c:
8094 media: collect media position when seek completes
8096 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
8098 * gst/rtsp-server/rtsp-client.c:
8099 client: call unlink_streams in client finalize
8102 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8104 * gst/rtsp-server/rtsp-media.c:
8105 media: limit the time to wait to something huge
8106 Avoid waiting forever but limit the timeout to 20 seconds.
8108 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8110 * gst/rtsp-server/rtsp-sdp.c:
8111 sdp: reindent and check for prepared status
8113 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8115 * gst/rtsp-server/rtsp-media.c:
8116 * gst/rtsp-server/rtsp-media.h:
8117 * gst/rtsp-server/rtsp-session.c:
8118 media: avoid doing _get_state() for state changes
8119 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
8120 until the media is prerolled or in error. This avoids doing a blocking call of
8121 gst_element_get_state() that can cause lockups when there is an error.
8124 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8126 * gst/rtsp-server/rtsp-media.c:
8129 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8131 * gst/rtsp-server/rtsp-media-factory.c:
8132 media-factory: better error handling
8133 Improve the error handling a bit.
8135 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8137 * gst/rtsp-server/rtsp-client.c:
8138 client: rework transport parsing
8139 Rework the transport parsing code so that we can ignore transports we don't
8140 support instead of just picking the first one we can parse.
8141 Configure a (for now hardcoded) destination for multicast transports.
8143 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8145 * gst/rtsp-server/rtsp-media.c:
8146 media: set multicast sink parameters
8147 Disable loop and automatic multicast join on the udpsink elements.
8148 Add some more debug info.
8149 Reset some state variables in the right place.
8150 Use the right port numbers for multicast.
8152 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8154 * gst/rtsp-server/rtsp-session.c:
8155 session: handle transport setup correctly
8156 Handle UDP, MCAST and TCP transport negotiation more correctly.
8157 Store the server session SSRC in the transport.
8159 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8161 * gst/rtsp-server/rtsp-client.c:
8162 rtsp-client: implement error_full
8163 Implement error_full to avoid some segfaults when the rtspconnection calls it.
8166 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8169 * gst/rtsp-server/rtsp-client.c:
8170 * gst/rtsp-server/rtsp-server.c:
8171 docs: update docs and comments
8173 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
8175 * gst/rtsp-server/rtsp-sdp.c:
8176 sdp: make server work better when behind a proxy
8178 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8180 * gst/rtsp-server/rtsp-client.c:
8181 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
8183 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8185 * gst/rtsp-server/rtsp-client.c:
8186 * gst/rtsp-server/rtsp-media-factory.c:
8187 * gst/rtsp-server/rtsp-media-mapping.c:
8188 * gst/rtsp-server/rtsp-media.c:
8189 * gst/rtsp-server/rtsp-server.c:
8190 * gst/rtsp-server/rtsp-session-pool.c:
8191 * gst/rtsp-server/rtsp-session.c:
8192 Use GStreamer's debugging subsystem
8194 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8196 * gst/rtsp-server/rtsp-media-factory.c:
8197 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
8199 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8204 === release 0.10.5 ===
8206 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8211 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8214 configure: bump required versions
8216 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
8218 * gst/rtsp-server/rtsp-client.c:
8219 client: call weak-unref on client->sessions from finalize
8222 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8224 * gst/rtsp-server/rtsp-media.c:
8225 media: Fixed crasher where caps got unref'ed too often
8227 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8230 * pkgconfig/.gitignore:
8231 * pkgconfig/Makefile.am:
8232 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
8233 Added pkg-config file to use gst-rtsp-server uninstalled
8235 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8237 * gst/rtsp-server/rtsp-media.c:
8238 media: add some docs
8240 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
8242 * gst/rtsp-server/rtsp-client.c:
8243 rtsp: Use gst_rtsp_watch_send_message().
8244 Use gst_rtsp_watch_send_message() since the old API which used
8245 gst_rtsp_watch_queue_message() has been deprecated.
8247 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8252 === release 0.10.4 ===
8254 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8259 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8261 * gst/rtsp-server/rtsp-client.c:
8262 * gst/rtsp-server/rtsp-session.c:
8263 * gst/rtsp-server/rtsp-session.h:
8264 rtsp: allocate channels in TCP mode
8265 When the client does not provide us with channels in TCP mode, allocate channels
8268 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8270 * gst/rtsp-server/rtsp-client.c:
8271 client: don't crash when tunnelid is missing
8272 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
8273 don't crash but return an error response to the client.
8276 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8278 * bindings/vala/gst-rtsp-server-0.10.vapi:
8279 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8280 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8281 bindings: update vala bindings with new method
8283 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8285 * gst/rtsp-server/rtsp-session-pool.c:
8286 * gst/rtsp-server/rtsp-session-pool.h:
8287 sessionpool: add function to filter sessions
8288 Add generic function to retrieve/remove sessions.
8290 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8293 configure: bump core/base requirements to release
8295 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8297 * gst/rtsp-server/rtsp-media.c:
8298 media: fix indentation
8300 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8302 * gst/rtsp-server/rtsp-media.c:
8303 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
8305 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8307 * gst/rtsp-server/rtsp-media.c:
8308 set state and remove elements of media in for loop
8310 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
8312 * bindings/vala/gst-rtsp-server-0.10.vapi:
8313 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8314 Added gst_rtsp_media_remove_elements function to Vala bindings
8316 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
8318 * gst/rtsp-server/rtsp-media.c:
8319 * gst/rtsp-server/rtsp-media.h:
8320 Added gst_rtsp_media_remove_elements function
8322 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
8324 * gst/rtsp-server/rtsp-media.c:
8325 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
8327 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8329 * bindings/vala/gst-rtsp-server-0.10.vapi:
8330 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8331 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8332 Updated Vala bindings
8334 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8336 * gst/rtsp-server/rtsp-media.c:
8337 * gst/rtsp-server/rtsp-media.h:
8338 Added vmethod unprepare to GstRTSPMedia
8339 The default implementation sets the state of the pipeline to GST_STATE_NULL
8341 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8343 * gst/rtsp-server/rtsp-media-factory.c:
8344 * gst/rtsp-server/rtsp-media-factory.h:
8345 Made collect_streams function public
8347 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8349 * gst/rtsp-server/rtsp-media-factory.c:
8350 * gst/rtsp-server/rtsp-media-factory.h:
8351 * gst/rtsp-server/rtsp-media.c:
8352 Added vmethod create_pipeline to GstRTSPMediaFactory
8353 The pipeline is created in this method and the GstRTSPMedia's element is added to it
8355 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8357 * gst/rtsp-server/rtsp-client.c:
8358 client: use g_source_destroy()
8359 We need to use g_source_destroy() because we might have added the source to a
8360 different main context than the default one.
8362 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8364 * gst/rtsp-server/Makefile.am:
8365 * gst/rtsp-server/rtsp-client.c:
8366 * gst/rtsp-server/rtsp-params.c:
8367 * gst/rtsp-server/rtsp-params.h:
8368 rtsp: prepare for handling GET/SET_PARAMETER
8369 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
8371 Fix return codes of handlers.
8373 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8375 * gst/rtsp-server/rtsp-media.c:
8376 media: don't leak session pads
8378 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8380 * gst/rtsp-server/rtsp-media.c:
8381 media: clean up the messages a bit
8383 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8385 * gst/rtsp-server/rtsp-sdp.c:
8386 sdp: warn and skip streams without media
8388 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8390 * bindings/vala/gst-rtsp-server-0.10.vapi:
8391 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8392 vala: Fixed typo in header file of RTSPMediaStream
8394 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8396 * gst/rtsp-server/rtsp-media.c:
8399 Make dumping RTCP stats configurable
8401 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8403 * gst/rtsp-server/rtsp-media.c:
8404 media: be less verbose and leak less
8406 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8408 * gst/rtsp-server/rtsp-media.c:
8409 media: don't leak the destination address
8411 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8413 * gst/rtsp-server/rtsp-client.c:
8414 * gst/rtsp-server/rtsp-media.c:
8415 * gst/rtsp-server/rtsp-media.h:
8416 * gst/rtsp-server/rtsp-session.c:
8417 * gst/rtsp-server/rtsp-session.h:
8418 rtsp: use RTCP to keep the session alive
8419 Use the RTCP rtcp-from stats field to find the associated session and use this
8420 to keep the session alive.
8422 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8424 * gst/rtsp-server/rtsp-session.c:
8425 session: add 5sec to the real session timeout
8426 Allow the session to live 5sec longer before really timing out. This should give
8427 clients some extra time to keep the session active.
8429 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8431 * gst/rtsp-server/rtsp-client.c:
8432 client: replay OK to GET/SET_PARAMETER
8433 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
8434 so that we return OK for those requests.
8436 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8438 * gst/rtsp-server/rtsp-media.c:
8439 * gst/rtsp-server/rtsp-media.h:
8440 media: keep track of active transports
8441 Keep track of which transport is active to avoid closing the connection too
8443 Remove the destination transport also when going to NULL.
8444 Print some stats about the SDES and other RTCP messages we receive from the
8447 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8449 * examples/.gitignore:
8450 * examples/Makefile.am:
8451 * examples/test-sdp.c:
8452 example: add SDP relay example
8454 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8456 * gst/rtsp-server/rtsp-media.c:
8457 media: also count active TCP connections
8459 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8461 * gst/rtsp-server/rtsp-media-factory.c:
8462 * gst/rtsp-server/rtsp-media.c:
8463 * gst/rtsp-server/rtsp-media.h:
8464 rtsp: add support for dynamic elements
8465 Add support for dynamic elements.
8466 Don't set live pipelines back to paused.
8468 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8470 * gst/rtsp-server/rtsp-sdp.c:
8471 sdp: don't add encoding name when absent in caps
8473 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8475 * gst/rtsp-server/rtsp-client.c:
8476 client: warn when we can't do RTP-Info
8478 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8480 * gst/rtsp-server/rtsp-media-factory.c:
8481 factory: factor out the stream construction
8483 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8485 * gst/rtsp-server/rtsp-client.c:
8486 client: only add RTP-Info when we have the info
8487 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
8490 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8495 === release 0.10.3 ===
8497 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8501 - Fixes a bug where it put the wrong verion in pkgconfig
8502 - Link RTP and RTCP sources
8504 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8506 * gst/rtsp-server/rtsp-media.c:
8507 * gst/rtsp-server/rtsp-media.h:
8508 media: link the RTP udpsrc to the session manager
8509 Link the RTP udpsrc and the appsrc to the session manager so that they don't
8510 shut down when the client sends a packet to open firewalls.
8512 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8514 * pkgconfig/gst-rtsp-server.pc.in:
8515 Don't use hard-coded version number in pkg-config file
8517 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8522 === release 0.10.2 ===
8524 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8529 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8532 * common/m4/.gitignore:
8533 * examples/.gitignore:
8534 * pkgconfig/.gitignore:
8535 add some .gitignore files
8537 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8539 * gst/rtsp-server/rtsp-media.c:
8540 media: seek to key frames
8542 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8544 * gst/rtsp-server/rtsp-media.c:
8545 media: emit the unprepared signal by id
8546 Emit the unprepared signal by id instead of name and set the media as
8549 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8551 * gst/rtsp-server/rtsp-media.c:
8552 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
8554 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8556 * gst/rtsp-server/rtsp-server.c:
8557 Added finalize function to GstRTPSPServer to unref session pool and media mapping
8559 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8561 * bindings/vala/gst-rtsp-server-0.10.vapi:
8562 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8563 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8564 Updated vala bindings
8566 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8568 * gst/rtsp-server/Makefile.am:
8569 * gst/rtsp-server/rtsp-client.c:
8570 * gst/rtsp-server/rtsp-media.c:
8571 server: use appsink and appsrc with the API
8572 Use the appsink/appsrc API instead of the signals for higher
8575 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8577 * examples/test-ogg.c:
8578 tests: set the payload type correctly
8580 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8582 * gst/rtsp-server/rtsp-media-factory.c:
8583 factory: connect to the unprepare signal
8584 Connect to the unprepare signal for non-reusable media so that we can remove
8585 them from the cache.
8587 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8589 * gst/rtsp-server/rtsp-media.c:
8590 * gst/rtsp-server/rtsp-media.h:
8591 media: add signal to notify of unprepare
8593 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8595 * gst/rtsp-server/rtsp-media.c:
8596 * gst/rtsp-server/rtsp-media.h:
8597 media: more work on making the media shared
8598 Add a reusable flag to medias, indicating that they can be reused after a state
8602 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8604 * examples/test-readme.c:
8605 examples: mark the example as shared for testing
8607 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8609 * gst/rtsp-server/rtsp-media.c:
8610 * gst/rtsp-server/rtsp-media.h:
8611 client: support shared media
8612 Always perform the state actions even if the target state of the pipeline is
8613 already correct, we still want to add/remove the transports when we are dealing
8615 Keep a counter of the number of active transports for a media so that we can use
8616 this to perform a state change when needed.
8617 Perform a state change of the pipeline only when the first transport was added
8618 or when there are no active transports.
8620 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8622 * gst/rtsp-server/rtsp-client.c:
8623 client: fix refcounting crasher
8624 Don't need to remove the weak refs in the finalize methods, they are already
8625 removed in the dispose.
8626 Don't register the callback with a DestroyNofity.
8628 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8630 * gst/rtsp-server/rtsp-client.c:
8631 Fix rtsp client refcount management in TCP mode.
8632 Don't unref a client ref we never had. Fixes an unref
8633 of an already-free client object after a client
8634 teardown request for me.
8636 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8638 * gst/rtsp-server/rtsp-session.c:
8639 docs: fix typo in API docs
8641 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8643 * gst/rtsp-server/rtsp-media.c:
8645 Keep the udp sources in playing even if we go to paused. unlock the sources when
8647 Add some more debug info.
8648 Only seek when we need to.
8649 Keep track of the position when we go to paused.
8651 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8653 * gst/rtsp-server/rtsp-client.c:
8654 * gst/rtsp-server/rtsp-media.c:
8655 * gst/rtsp-server/rtsp-media.h:
8656 Add beginnings of seeking.
8657 Parse the Range header and perform a seek on the pipeline for the requested
8658 position. It's disabled currently until I figure out what's going wrong.
8660 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8662 * gst/rtsp-server/rtsp-client.c:
8663 allow pause requests for now.
8666 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8668 * gst/rtsp-server/rtsp-client.c:
8669 Remove weak ref on the session in teardown
8670 We need to remove our weakref from the session when we do a teardown because
8671 else we close the TCP connection prematurely.
8673 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8675 * gst/rtsp-server/rtsp-client.c:
8676 * gst/rtsp-server/rtsp-client.h:
8677 * gst/rtsp-server/rtsp-session-pool.c:
8678 Do some more session cleanup
8679 Make session timeout kill the TCP connection that currently watches the
8681 Remove the client timeout property.
8683 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8685 * gst/rtsp-server/rtsp-client.c:
8686 * gst/rtsp-server/rtsp-client.h:
8687 * gst/rtsp-server/rtsp-media.c:
8688 * gst/rtsp-server/rtsp-media.h:
8689 * gst/rtsp-server/rtsp-server.c:
8690 * gst/rtsp-server/rtsp-session.c:
8691 * gst/rtsp-server/rtsp-session.h:
8693 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
8696 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8698 * examples/Makefile.am:
8699 * examples/test-launch.c:
8700 Add example server that takes launch lines
8701 Add an example server that streams any -launch line.
8703 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8705 * examples/test-readme.c:
8706 * gst/rtsp-server/rtsp-client.c:
8707 * gst/rtsp-server/rtsp-media.c:
8708 * gst/rtsp-server/rtsp-media.h:
8709 Add support for live streams
8710 Add support for live streams and ranges
8711 Start on handling TCP data transfer.
8713 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8715 * gst/rtsp-server/rtsp-media.c:
8716 Free the pipeline before other things
8719 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8721 * gst/rtsp-server/rtsp-client.c:
8722 Only free the pending tunnel if there is one
8725 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8727 * gst/rtsp-server/rtsp-client.c:
8728 * gst/rtsp-server/rtsp-client.h:
8729 * gst/rtsp-server/rtsp-media.c:
8730 rtsp-server: Add support for tunneling
8731 Add support for tunneling over HTTP.
8732 Use new connection methods to retrieve the url.
8733 Dispatch messages based on the message type instead of blindly
8734 assuming it's always a request.
8735 Keep track of the watch id so that we can remove it later.
8736 Set the media pipeline to NULL before unreffing the pipeline.
8738 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8740 * gst/rtsp-server/rtsp-client.c:
8741 * gst/rtsp-server/rtsp-client.h:
8742 Fix for channel -> watch rename in gstreamer
8743 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
8745 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8747 * gst/rtsp-server/rtsp-client.c:
8748 * gst/rtsp-server/rtsp-client.h:
8750 Use the async RTSP channels instead of spawning a new thread for each client.
8751 If a sessionid is specified in a request, fail if we don't have the session.
8753 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8755 * gst/rtsp-server/rtsp-media.c:
8756 Add better debug info
8757 Add some better debug info.
8759 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8761 * examples/test-video.c:
8763 Add support for session timeouts in the example.
8765 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8767 * gst/rtsp-server/rtsp-session-pool.c:
8768 * gst/rtsp-server/rtsp-session-pool.h:
8769 Pass GTimeVal around for performance reasons
8770 Get the current time only once and pass it around so that sessions don't have to
8771 get the current time anymore.
8772 Add experimental support for a GSource that dispatches when the session needs to
8775 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8777 * gst/rtsp-server/rtsp-session.c:
8778 * gst/rtsp-server/rtsp-session.h:
8779 Add better support for session timeouts
8780 Add a method to request the number of milliseconds when a session will timeout.
8782 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8784 * gst/rtsp-server/rtsp-media.c:
8785 * gst/rtsp-server/rtsp-media.h:
8786 Add suport for RTP manager monitoring
8787 Add the first stage in monitoring the rtp manager.
8788 Make sure we don't update the state to something we don't want.
8790 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8792 * gst/rtsp-server/rtsp-client.c:
8793 Add support for session keepalive
8794 Get and update the session timeout for all requests. get the session as early as
8797 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8799 * gst/rtsp-server/rtsp-media-factory.h:
8800 * gst/rtsp-server/rtsp-media.c:
8801 * gst/rtsp-server/rtsp-media.h:
8802 Handle media bus messages
8803 Handle media bus messages in a custom mainloop and dispatch them to the
8804 RTSPMedia objects. Let the default implementation handle some common messages.
8806 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8808 * gst/rtsp-server/rtsp-client.c:
8809 * gst/rtsp-server/rtsp-session-pool.c:
8810 * gst/rtsp-server/rtsp-session.c:
8811 Some more session timeout handling
8812 Move the session header setting code to a central place so that we always add
8813 the timeout parameter too.
8814 Handle timeouts by running the session cleanup code.
8815 Stop media before cleaning up.
8817 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8819 * gst/rtsp-server/rtsp-client.c:
8820 * gst/rtsp-server/rtsp-client.h:
8821 Add timeout property
8822 Add a timeout property ot the client and make the other properties into GObject
8825 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8827 * gst/rtsp-server/rtsp-session-pool.c:
8828 Use getters and setters in property code
8829 Use the getters and setters for the timeout property instead of locking
8832 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8834 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
8836 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8838 * gst/rtsp-server/rtsp-session-pool.c:
8839 * gst/rtsp-server/rtsp-session-pool.h:
8840 * gst/rtsp-server/rtsp-session.c:
8841 * gst/rtsp-server/rtsp-session.h:
8842 Add more timeout stuff
8843 Add method to check if a session is expired.
8844 Add method to perform cleanup on a session pool.
8846 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8848 * gst/rtsp-server/rtsp-client.c:
8849 * gst/rtsp-server/rtsp-session-pool.c:
8850 * gst/rtsp-server/rtsp-session-pool.h:
8851 * gst/rtsp-server/rtsp-session.c:
8852 * gst/rtsp-server/rtsp-session.h:
8853 Add beginnings of session timeouts and limits
8854 Add the timeout value to the Session header for unusual timeout values.
8855 Allow us to configure a limit to the amount of active sessions in a pool. Set a
8856 limit on the amount of retry we do after a sessionid collision.
8857 Add properties to the sessionid and the timeout of a session. Keep track of
8858 creation time and last access time for sessions.
8860 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8862 * gst/rtsp-server/rtsp-client.c:
8863 * gst/rtsp-server/rtsp-media.c:
8864 * gst/rtsp-server/rtsp-media.h:
8865 * gst/rtsp-server/rtsp-sdp.c:
8866 * gst/rtsp-server/rtsp-session-pool.c:
8867 * gst/rtsp-server/rtsp-session.c:
8868 * gst/rtsp-server/rtsp-session.h:
8869 Cleanup of sessions and more
8870 Fix the refcounting of media and sessions in the client. Properly clean up the
8871 session data when the client performs a teardown.
8872 Add Server header to responses.
8873 Allow for multiple uri setups in one session.
8874 Add Range header to the PLAY response and add the range attribute to the SDP
8876 Fix the session pool remove method, it used the wrong key in the hashtable. Also
8877 give the ownership of the sessionid to the session object.
8879 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8881 * gst/rtsp-server/rtsp-server.c:
8882 * gst/rtsp-server/rtsp-server.h:
8884 Rename the 'server_port' variable to simply 'port'.
8886 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8889 * gst/rtsp-server/rtsp-client.c:
8890 * gst/rtsp-server/rtsp-media.c:
8891 * gst/rtsp-server/rtsp-media.h:
8892 * gst/rtsp-server/rtsp-session.c:
8893 * gst/rtsp-server/rtsp-session.h:
8894 Rework the way we handle transports for streams
8895 Make the media accept an array of transports for the streams that we have
8896 configured for the play/pause requests.
8897 Implement server states for a client and its media.
8898 Require 0.10.22.1 (git HEAD) of gstreamer.
8900 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8902 * gst/rtsp-server/rtsp-client.c:
8903 * gst/rtsp-server/rtsp-media-factory.c:
8904 Drop const from functions dealing with urls
8905 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
8906 have the right const in them.
8908 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8910 * gst/rtsp-server/rtsp-client.c:
8911 * gst/rtsp-server/rtsp-media.c:
8912 * gst/rtsp-server/rtsp-sdp.c:
8916 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8918 * gst/rtsp-server/rtsp-client.c:
8919 * gst/rtsp-server/rtsp-media-factory.c:
8920 * gst/rtsp-server/rtsp-media.c:
8921 * gst/rtsp-server/rtsp-media.h:
8923 Don't keep a reference to the GstRTSPMedia in the stream.
8924 Free more things when freeing the GstRTSPMedia.
8926 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8929 * gst/rtsp-server/rtsp-media-factory.c:
8930 * gst/rtsp-server/rtsp-media-factory.h:
8931 * gst/rtsp-server/rtsp-media.c:
8932 * gst/rtsp-server/rtsp-media.h:
8933 * gst/rtsp-server/rtsp-server.c:
8934 * gst/rtsp-server/rtsp-server.h:
8935 More docs and small cleanups
8936 Add some more docs and update the README
8937 Cleanup some method names.
8938 Remove an unneeded idx field in the GstRTSPMediaStream
8940 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8943 * examples/Makefile.am:
8944 * examples/test-readme.c:
8945 Add a README and more example code
8946 Add a README file that contains a small introduction on how to use the server
8947 along with the example code explained in the readme.
8949 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8951 * gst/rtsp-server/rtsp-media.c:
8952 * gst/rtsp-server/rtsp-server.c:
8953 Fix some leaks and change default port
8954 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
8955 we finished the initial preroll. If we keep them locked, setting the pipeline to
8956 NULL will not stop and clean up the sources correctly.
8957 Change the default RTSP port to 8554 aka the official alternative RTSP port.
8959 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8961 * gst/rtsp-server/rtsp-session.c:
8962 * gst/rtsp-server/rtsp-session.h:
8963 Cleanups to the session object
8964 Remove some unneeded variables in the session state of a stream such as the
8965 owner media and the server transport.
8966 Get the configuration of a media stream in a session based on the media_stream
8967 in the original object instead of our cached index.
8968 Free more data in the finalize method.
8970 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8972 * gst/rtsp-server/rtsp-client.c:
8973 * gst/rtsp-server/rtsp-client.h:
8974 Cleanups and reuse media from DESCRIBE
8975 Handle thread create errors.
8976 Rename some internal methods to better match what they actually do.
8977 Handle misconfiguration of session_pool and media_mapping gracefully.
8978 Cache the DESCRIBE media and uri in the client connection and reuse them when
8979 we receive a SETUP request in the same connection for the same uri.
8980 Cleanup the client connection object.
8982 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8984 * gst/rtsp-server/rtsp-media-factory.c:
8985 * gst/rtsp-server/rtsp-media-factory.h:
8986 * gst/rtsp-server/rtsp-media.c:
8987 * gst/rtsp-server/rtsp-media.h:
8988 Add shared properties to media and factory
8989 Add the shared property to media.
8990 Implement some simple caching in the factory depending on if the media is shared
8993 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8995 * gst/rtsp-server/rtsp-client.c:
8996 Add a little comment
8997 Add some comment about the content-base header.
8999 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9001 * examples/Makefile.am:
9003 * examples/test-mp4.c:
9004 * examples/test-ogg.c:
9005 * examples/test-video.c:
9006 * gst/rtsp-server/Makefile.am:
9007 * gst/rtsp-server/rtsp-client.c:
9008 * gst/rtsp-server/rtsp-client.h:
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-sdp.c:
9014 * gst/rtsp-server/rtsp-sdp.h:
9015 * gst/rtsp-server/rtsp-server.c:
9016 * gst/rtsp-server/rtsp-server.h:
9017 * gst/rtsp-server/rtsp-session.c:
9018 * gst/rtsp-server/rtsp-session.h:
9019 Reorganize things, prepare for media sharing
9020 Added various other test server examples
9021 Move the SDP message generation to a separate helper.
9022 Refactor common code for finding the session.
9023 Add content-base for realplayer compatibility
9024 Clean up request uris before processing for better vlc compatibility.
9025 Move prerolling and pipeline construction to the RTSPMedia object.
9026 Use multiudpsink for future pipeline reuse.
9028 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9034 === release 0.10.1 ===
9036 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9042 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9044 * bindings/vala/Makefile.am:
9046 Add more directories and files to the dist.
9048 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9050 * bindings/python/Makefile.am:
9051 * bindings/python/rtspserver.override:
9052 Fixed compile error of python bindings
9054 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9056 * bindings/vala/gst-rtsp-server-0.10.vapi:
9057 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9058 Marked values as nullable accordingly
9060 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9062 * bindings/vala/gst-rtsp-server-0.10.vapi:
9063 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
9064 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9065 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9066 Updated Vala bindings
9068 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9070 * gst/rtsp-server/rtsp-client.c:
9071 * gst/rtsp-server/rtsp-media-mapping.c:
9072 * gst/rtsp-server/rtsp-media-mapping.h:
9073 * gst/rtsp-server/rtsp-media.h:
9074 * gst/rtsp-server/rtsp-session-pool.h:
9075 Cleanups and doc updates
9076 Add some more documentation and do some minor cleanups here and there.
9078 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9080 * gst/rtsp-server/rtsp-client.c:
9081 * gst/rtsp-server/rtsp-media-factory.c:
9082 * gst/rtsp-server/rtsp-media-factory.h:
9083 * gst/rtsp-server/rtsp-media.c:
9084 * gst/rtsp-server/rtsp-media.h:
9085 * gst/rtsp-server/rtsp-session.c:
9086 * gst/rtsp-server/rtsp-session.h:
9088 Rename GstRTSPMediaBin to GstRTSPMedia
9089 Parse the request url into a GstRTSPUri object and pass this object to the
9090 various handlers and methods that require the uri.
9092 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9096 Add some more docs and remove some old code from the example.
9098 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9100 * gst/rtsp-server/rtsp-client.c:
9101 Handle state change failures better
9102 Handle state change failures better when changing the state of the pipeline to
9105 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9107 * gst/rtsp-server/rtsp-media-factory.c:
9108 * gst/rtsp-server/rtsp-media-factory.h:
9109 Make element creation more extendible
9110 Add get_element vmethod to the default MediaFactory so that subclasses can just
9111 override that method and still use the default logic for making a MediaBin from
9114 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9117 * gst/rtsp-server/Makefile.am:
9118 * gst/rtsp-server/rtsp-client.c:
9119 * gst/rtsp-server/rtsp-client.h:
9120 * gst/rtsp-server/rtsp-media-factory.c:
9121 * gst/rtsp-server/rtsp-media-factory.h:
9122 * gst/rtsp-server/rtsp-media-mapping.c:
9123 * gst/rtsp-server/rtsp-media-mapping.h:
9124 * gst/rtsp-server/rtsp-media.c:
9125 * gst/rtsp-server/rtsp-media.h:
9126 * gst/rtsp-server/rtsp-server.c:
9127 * gst/rtsp-server/rtsp-server.h:
9128 * gst/rtsp-server/rtsp-session.c:
9129 * gst/rtsp-server/rtsp-session.h:
9130 Make the server handle arbitrary pipelines
9131 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
9132 The GstMediaBin object has a handle to a bin with elements and to a list of
9133 GstMediaStream objects that this bin produces.
9134 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
9135 with methods to register and remove those mappings.
9136 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
9137 used by the server instance.
9138 Modify the example application so that it shows how to create custom pipelines
9139 attached to a specific mount point.
9140 Various misc cleanps.
9142 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9144 * gst/rtsp-server/rtsp-server.c:
9145 * gst/rtsp-server/rtsp-server.h:
9146 Allow setting a custom media factory for a server
9148 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9150 * gst/rtsp-server/rtsp-client.c:
9151 * gst/rtsp-server/rtsp-client.h:
9152 Allow setting a custom media factory for a client.
9154 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9156 * gst/rtsp-server/Makefile.am:
9157 Add Makefile entry for the media factory
9159 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9161 * gst/rtsp-server/rtsp-media-factory.c:
9162 * gst/rtsp-server/rtsp-media-factory.h:
9163 Add media factory to map urls to media pipeline objects.
9165 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9167 * gst/rtsp-server/rtsp-media.c:
9168 * gst/rtsp-server/rtsp-media.h:
9169 Add comments. Remove unused field
9171 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9173 * gst/rtsp-server/rtsp-session-pool.c:
9174 * gst/rtsp-server/rtsp-session-pool.h:
9175 Allow custom session pools to override the session id allocation algorithms Add some comments.
9177 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9179 * gst/rtsp-server/rtsp-session.h:
9182 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9184 * gst/rtsp-server/rtsp-client.c:
9185 * gst/rtsp-server/rtsp-client.h:
9186 Move the connection code in one place Add some comments
9188 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9190 * gst/rtsp-server/rtsp-server.c:
9191 * gst/rtsp-server/rtsp-server.h:
9192 Make vmethod to create and accept new clients. Add some docs.
9194 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9196 * gst/rtsp-server/rtsp-server.c:
9197 * gst/rtsp-server/rtsp-server.h:
9198 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
9200 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9202 * gst/rtsp-server/rtsp-client.c:
9203 * gst/rtsp-server/rtsp-client.h:
9204 Name the parameters more appropriately.
9206 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9208 * gst/rtsp-server/rtsp-session-pool.c:
9209 Do some more cleanup of the session pool.
9211 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9213 * gst/rtsp-server/Makefile.am:
9214 * gst/rtsp-server/rtsp-client.c:
9215 Check if return value of gst_rtsp_session_get_media is not NULL
9217 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9219 * gst/rtsp-server/Makefile.am:
9220 Install rtsp-session and rtsp-session-pool headers
9222 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9227 * bindings/python/Makefile.am:
9228 * bindings/python/arg-types.py:
9229 * bindings/python/codegen/Makefile.am:
9230 * bindings/python/codegen/__init__.py:
9231 * bindings/python/codegen/argtypes.py:
9232 * bindings/python/codegen/code-coverage.py:
9233 * bindings/python/codegen/codegen.py:
9234 * bindings/python/codegen/definitions.py:
9235 * bindings/python/codegen/defsparser.py:
9236 * bindings/python/codegen/docextract.py:
9237 * bindings/python/codegen/docgen.py:
9238 * bindings/python/codegen/fileprefix.override:
9239 * bindings/python/codegen/fileprefixmodule.c:
9240 * bindings/python/codegen/h2def.py:
9241 * bindings/python/codegen/mergedefs.py:
9242 * bindings/python/codegen/mkskel.py:
9243 * bindings/python/codegen/override.py:
9244 * bindings/python/codegen/reversewrapper.py:
9245 * bindings/python/codegen/scmexpr.py:
9246 * bindings/python/rtspserver-types.defs:
9247 * bindings/python/rtspserver.defs:
9248 * bindings/python/rtspserver.override:
9249 * bindings/python/rtspservermodule.c:
9251 Add python bindings.
9253 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9255 * bindings/Makefile.am:
9257 Don't go into python dir when requirements for python bindings are missing
9259 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9261 * bindings/Makefile.am:
9262 * bindings/vala/Makefile.am:
9264 Install Vala bindings if vala is available
9266 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9268 * bindings/vala/gst-rtsp-server-0.10.deps:
9269 * bindings/vala/gst-rtsp-server-0.10.vapi:
9270 * bindings/vala/gst-rtsp-server.vapi:
9271 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
9272 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
9273 * bindings/vala/packages/gst-rtsp-server-0.10.files:
9274 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9275 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9276 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
9277 * bindings/vala/packages/gst-rtsp-server.deps:
9278 * bindings/vala/packages/gst-rtsp-server.excludes:
9279 * bindings/vala/packages/gst-rtsp-server.files:
9280 * bindings/vala/packages/gst-rtsp-server.gi:
9281 * bindings/vala/packages/gst-rtsp-server.metadata:
9282 * bindings/vala/packages/gst-rtsp-server.namespace:
9283 Regenerated Vala bindings
9285 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9287 * bindings/vala/gst-rtsp-server.vapi:
9288 * bindings/vala/packages/gst-rtsp-server.metadata:
9289 Fixed typo in included headers for vala bindings
9291 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9295 * pkgconfig/Makefile.am:
9296 * pkgconfig/gst-rtsp-server.pc.in:
9297 Added pkgconfig file
9299 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
9301 * bindings/vala/gst-rtsp-server.vapi:
9302 * bindings/vala/packages/gst-rtsp-server.excludes:
9303 * bindings/vala/packages/gst-rtsp-server.gi:
9304 * bindings/vala/packages/gst-rtsp-server.metadata:
9305 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
9307 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
9309 * bindings/vala/gst-rtsp-server.vapi:
9310 * bindings/vala/packages/gst-rtsp-server.deps:
9311 * bindings/vala/packages/gst-rtsp-server.files:
9312 * bindings/vala/packages/gst-rtsp-server.gi:
9313 * bindings/vala/packages/gst-rtsp-server.metadata:
9314 * bindings/vala/packages/gst-rtsp-server.namespace:
9317 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
9319 * gst/rtsp-server/rtsp-session.c:
9320 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
9322 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
9324 * examples/Makefile.am:
9325 * gst/rtsp-server/Makefile.am:
9326 Put GStreamer version in library name
9328 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9330 * examples/Makefile.am:
9331 * gst/rtsp-server/Makefile.am:
9332 Fix some issues to pass distcheck
9334 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9336 * gst/rtsp-server/rtsp-server.c:
9337 Added port property to GstRTSPServer class.
9339 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9344 * examples/Makefile.am:
9347 * gst/rtsp-server/Makefile.am:
9348 * gst/rtsp-server/rtsp-client.c:
9349 * gst/rtsp-server/rtsp-client.h:
9350 * gst/rtsp-server/rtsp-media.c:
9351 * gst/rtsp-server/rtsp-media.h:
9352 * gst/rtsp-server/rtsp-server.c:
9353 * gst/rtsp-server/rtsp-server.h:
9354 * gst/rtsp-server/rtsp-session-pool.c:
9355 * gst/rtsp-server/rtsp-session-pool.h:
9356 * gst/rtsp-server/rtsp-session.c:
9357 * gst/rtsp-server/rtsp-session.h:
9360 * src/rtsp-client.c:
9361 * src/rtsp-client.h:
9364 * src/rtsp-server.c:
9365 * src/rtsp-server.h:
9366 * src/rtsp-session-pool.c:
9367 * src/rtsp-session-pool.h:
9368 * src/rtsp-session.c:
9369 * src/rtsp-session.h:
9370 Split in library and example program
9372 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
9374 * src/rtsp-client.h:
9375 Removed obsolete variable
9377 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
9379 * src/rtsp-client.c:
9380 * src/rtsp-client.h:
9381 Removed pipeline variable GstRTSPClient, because it's only used in one function
9383 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9386 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
9388 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
9390 * src/rtsp-session.c:
9391 Initialize some more vars.
9393 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
9395 * src/rtsp-session.c:
9396 Initialize variable to avoid compiler warning.
9398 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
9401 Add a reasonable generic .gitignore