3 2016-02-19 Sebastian Dröge <slomo@coaxion.net>
8 2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
10 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
11 uninstalled.pc: add support for non libtool build systems
12 Currently the .la path is provided which requires to use libtool as
13 mentioned in the GStreamer manual section-helloworld-compilerun.html.
14 It is fine as long as the application is built using libtool.
15 So currently it is not possible to compile a GStreamer application
16 within gst-uninstalled with CMake or other build system different
18 This patch allows to do the following in gst-uninstalled env:
19 gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
20 gstreamer-rtsp-server-1.0)
21 Previously it required to prepend libtool --mode=link
22 https://bugzilla.gnome.org/show_bug.cgi?id=720778
24 2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
26 * gst/rtsp-sink/gstrtspclientsink.c:
27 rtspclientsink: remove check for impossible condition
28 Goto error label checks stream to see if it needs to be unreferenced before
29 returning, but this goto jumps happens before the stream is ever set, so it
30 will always be NULL in this error label.
33 2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
35 * gst/rtsp-sink/gstrtspclientsink.c:
36 rtspclientsink: clean switch statements
37 Coverity demands for fallthrough statements to be clearly commented,
38 to distinguish from accidental fall throughs. And it also needs all
39 cases to finish with a break, even if the break is never going to be
40 executed like in the case of a continue jump.
44 2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
46 * tests/check/Makefile.am:
47 tests: extend the AM_TESTS_ENVIRONMENT from check.mak
48 To get the CK_DEFAULT_TIMEOUT defined for all tests
49 Also removes a 120 seconds timeout that was set as default
50 explicitly in this module
51 https://bugzilla.gnome.org/show_bug.cgi?id=761472
53 2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
57 Automatic update of common submodule
58 From 86e4663 to b64f03f
60 2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
62 * gst/rtsp-server/rtsp-media.c:
63 rtsp-media: fix state_lock not locked again when preroll fails
64 https://bugzilla.gnome.org/show_bug.cgi?id=761399
66 2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
69 configure: Move plugin specific flags below all the others
70 They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
71 -no-undefined. And -no-undefined is required on Windows to build DLLs.
73 2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
75 * gst/rtsp-sink/gstrtspclientsink.c:
76 rtspclientsink: Simplify slightly using new -base API
77 Use the new Mikey and SDP API in the base plugins libs
78 to simplify some code.
79 https://bugzilla.gnome.org/show_bug.cgi?id=758180
81 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
86 * gst/rtsp-sink/Makefile.am:
87 * gst/rtsp-sink/gstrtspclientsink.c:
88 * gst/rtsp-sink/gstrtspclientsink.h:
89 * gst/rtsp-sink/plugin.c:
90 * tests/check/Makefile.am:
91 * tests/check/gst/rtspclientsink.c:
92 rtspsink: Add rtspclientsink element
93 Add an rtspclientsink element that accepts streams for which
94 there is a registered payloader and sends them to
95 an RTSP server using RECORD.
96 Sending is synchronised to the pipeline clock. Payload-types
97 are automatically selected. The 'new-payloader' signal is fired
98 for custom configuration of payloaders when they are created.
99 Can now stream a movie like this:
101 ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
102 decodebin name=depay1 ! audioconvert ! autoaudiosink )"
104 gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
105 queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
106 https://bugzilla.gnome.org/show_bug.cgi?id=758180
108 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
110 * gst/rtsp-server/rtsp-stream.c:
111 * gst/rtsp-server/rtsp-stream.h:
112 rtsp-stream: Add functions for using rtsp-stream from the client
113 Add a boolean to indicate that the rtsp-stream is running on the
114 'client' side of an RTSP connection, for sending streams via
115 RECORD. In that case, the roles of the client/server ports
116 in transport setup are swapped.
117 https://bugzilla.gnome.org/show_bug.cgi?id=758180
119 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
121 * gst/rtsp-server/rtsp-sdp.c:
122 * gst/rtsp-server/rtsp-sdp.h:
123 rtsp-sdp: Add gst_rtsp_sdp_from_stream()
124 A new function that adds info from a GstRTSPStream into an SDP message.
125 https://bugzilla.gnome.org/show_bug.cgi?id=758180
127 2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
129 * gst/rtsp-server/rtsp-media.c:
130 rtsp-media: Fix mutex beeing unlocked while they should be locked
131 https://bugzilla.gnome.org/show_bug.cgi?id=761226
133 2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
135 * gst/rtsp-server/rtsp-media-factory.c:
136 rtsp-media-factory: add missing break in "clock" property setter
139 2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
141 * gst/rtsp-server/rtsp-stream.c:
142 rtsp-stream: fixed assert during update transport
143 When RTSP server trying update transport during multicast, it throws an
144 assert. The assert is thrown because it is trying to get the parent of
145 an non-existing funnel element.
146 https://bugzilla.gnome.org/show_bug.cgi?id=760150
148 2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
150 * gst/rtsp-server/rtsp-permissions.h:
151 * gst/rtsp-server/rtsp-thread-pool.h:
152 * gst/rtsp-server/rtsp-token.h:
153 docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
154 gtk-doc can handle static inline functions just fine these days,
155 there's no need for this stuff any more.
157 2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
159 * gst/rtsp-server/rtsp-media.c:
160 * gst/rtsp-server/rtsp-sdp.c:
161 sdp: replace duplicated codes to call new base sdp apis
162 https://bugzilla.gnome.org/show_bug.cgi?id=745880
164 2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
166 * examples/test-netclock.c:
167 test-netclock: Use the new API to configure a clock directly
169 2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
171 * gst/rtsp-server/rtsp-media-factory.c:
172 * gst/rtsp-server/rtsp-media-factory.h:
173 * gst/rtsp-server/rtsp-media.c:
174 * gst/rtsp-server/rtsp-media.h:
175 rtsp-media: Add API to directly configure a clock on the media pipelines
177 2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
179 * gst/rtsp-server/rtsp-media.c:
180 rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
182 2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
184 * gst/rtsp-server/rtsp-media-factory.c:
185 rtsp-media-factory: Add FIXME for 2.0
187 2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
189 * gst/rtsp-server/rtsp-stream.c:
190 rtsp-stream: Fix indentation
192 2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
194 * gst/rtsp-server/rtsp-media.c:
195 rtsp-media: Do not prepare media after media times out
196 Deferred calls to start_prepare() can be deferred past the point until
197 which wait_preroll() and by proxy gst_rtsp_media_get_status() is
198 prepared to wait. Previously there was no lock and no check for this
199 situation. This meant that a media could be prepared and unprepared
200 simultaneously by two different threads. Now a lock is in place and a
201 suitable check is done.
202 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
204 2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
206 * gst/rtsp-server/rtsp-client.c:
207 * gst/rtsp-server/rtsp-media-factory.c:
208 * gst/rtsp-server/rtsp-media-factory.h:
209 * gst/rtsp-server/rtsp-media.c:
210 * gst/rtsp-server/rtsp-media.h:
211 rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
212 Without TEARDOWN it might be desireable to keep the media running and continue
213 sending data to the client, even if the RTSP connection itself is
215 Only do this for session medias that have only UDP transports. If there's at
216 least on TCP transport, it will stop working and cause problems when the
217 connection is disconnected.
218 https://bugzilla.gnome.org/show_bug.cgi?id=758999
220 2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
225 === release 1.7.1 ===
227 2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
233 * gst-rtsp-server.doap:
236 2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
239 configure: Make -Bsymbolic check work with clang.
240 Update the -Bsymbolic check with the version glib has. This version
242 https://bugzilla.gnome.org/show_bug.cgi?id=759713
244 2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
246 * gst/rtsp-server/rtsp-session-pool.c:
247 rtsp-session-pool: Avoid dollar sign ($) in session ids
248 Live555 in VLC strips off dollar signs and then gets very confused,
249 we don't loose too much entropy by just skipping it.
251 2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
253 * gst/rtsp-server/rtsp-address-pool.h:
254 * gst/rtsp-server/rtsp-auth.h:
255 * gst/rtsp-server/rtsp-client.h:
256 * gst/rtsp-server/rtsp-media-factory-uri.h:
257 * gst/rtsp-server/rtsp-media-factory.h:
258 * gst/rtsp-server/rtsp-media.h:
259 * gst/rtsp-server/rtsp-mount-points.h:
260 * gst/rtsp-server/rtsp-permissions.h:
261 * gst/rtsp-server/rtsp-server.h:
262 * gst/rtsp-server/rtsp-session-media.h:
263 * gst/rtsp-server/rtsp-session-pool.h:
264 * gst/rtsp-server/rtsp-session.h:
265 * gst/rtsp-server/rtsp-stream-transport.h:
266 * gst/rtsp-server/rtsp-stream.h:
267 * gst/rtsp-server/rtsp-thread-pool.h:
268 * gst/rtsp-server/rtsp-token.h:
269 rtsp-server: Add g_autoptr() support to all types
270 https://bugzilla.gnome.org/show_bug.cgi?id=754464
272 2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
274 * gst/rtsp-server/rtsp-stream.c:
275 rtsp-stream: fixed valgrind error
276 Fixed the valgrind error in unit test. The UDP source created during
277 gst_rtsp_stream_join_bin() was not released while destroying the rtp
279 https://bugzilla.gnome.org/show_bug.cgi?id=759010
281 2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
285 Automatic update of common submodule
286 From b319909 to 86e4663
288 2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
290 * gst/rtsp-server/rtsp-client.c:
291 rtsp-client: suspend media during setup request
292 SETUP request from clients needs to suspend the media to clear the
293 prerolled buffers. Otherwise it will not affect the prerolled buffer
294 and the prerolled buffers will be incorrect (for example block-size
295 from setup request will not affect the prerolled buffer unless the
297 https://bugzilla.gnome.org/show_bug.cgi?id=758268
299 2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
301 * gst/rtsp-server/rtsp-stream.c:
302 rtsp-stream: create stream pipeline based on transport
303 Based on the protocol, create the rtsp stream pipeline. If only TCP or
304 only UDP is set as the transport protocol, it will not add the extra tee
305 or queue element to the pipeline. Both these elements will be added, if
306 it supports both TCP and UDP protocols. This improves the pipeline
307 performance when one protocol is present.
308 https://bugzilla.gnome.org/show_bug.cgi?id=758179
310 2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
312 * gst/rtsp-server/rtsp-stream.c:
313 rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
314 Adding them when not needed will start some logic inside rtpbin that might be
315 problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
316 would start up a rtpjitterbuffer and behave in weird ways.
317 We still set up the UDP sources for RTP receiving for a sender media to be
318 able to receive any packets sent by the client for NAT traversal. They will
319 all go to a fakesink though.
320 Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
321 NO_PREROLL, which will cause deadlocks when seeking the media as it will never
322 receive ASYNC_DONE after a seek.
323 https://bugzilla.gnome.org/show_bug.cgi?id=758319
325 2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
327 * gst/rtsp-server/rtsp-stream.c:
328 rtsp-stream: Disable multicast loopback for the multicast udp sources too
329 On POSIX this setting is for sender sockets, on Windows for receiver sockets.
330 Previously we were only setting this for sender sockets, which caused looped
331 back packets to be received on Windows if a multicast transport was used.
333 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
335 * examples/test-record-auth.c:
336 * examples/test-record.c:
337 examples: Actually use the provided port in the record examples
339 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
341 * examples/test-record-auth.c:
342 test-record-auth: Add the option to build in TLS support
344 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
346 * examples/test-auth.c:
347 test-auth: Use an 'anonymous' user for unauthenticated default
348 There's a comment on one of the resources that 'user' and 'admin'
349 shouldn't even be able to see it, but they can if the default
350 token is 'admin2', since that gives them access anyway.
352 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
354 * examples/.gitignore:
355 * examples/Makefile.am:
356 * examples/test-record-auth.c:
357 Add test-record-auth example
359 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
361 * gst/rtsp-server/rtsp-client.c:
362 * tests/check/gst/client.c:
363 rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
365 2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
367 * gst/rtsp-server/rtsp-server.c:
368 rtsp-server: Change the logic so we don't pop a NULL context
369 When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
370 will sometimes fail. This call is made before any context is pushed
371 resulting in an attempt to pop a NULL context.
372 https://bugzilla.gnome.org/show_bug.cgi?id=757949
374 2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
376 * tests/check/gst/rtspserver.c:
377 rtspserver: Add udp-mcast transport SETUP test
378 Refactor utility functions in the test file so they can handle
379 more than UDP and TCP as lower transport.
380 https://bugzilla.gnome.org/show_bug.cgi?id=756969
382 2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
384 * gst/rtsp-server/rtsp-stream.c:
385 rtsp-stream: Always unref return value of gst_object_get_parent()
386 Fixes a leak of a GstBin in the udp-mcast case.
387 https://bugzilla.gnome.org/show_bug.cgi?id=756968
389 2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
392 Automatic update of common submodule
393 From b99800a to b319909
395 2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
398 Use new GST_ENABLE_EXTRA_CHECKS #define
399 https://bugzilla.gnome.org/show_bug.cgi?id=756870
401 2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
404 Automatic update of common submodule
405 From 6babecd to b99800a
407 2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
410 Update GLib dependency to 2.40.0
412 2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
414 * examples/test-mp4.c:
415 * gst/rtsp-server/rtsp-stream.c:
416 stream: listen to sender ssrc signals
417 https://bugzilla.gnome.org/show_bug.cgi?id=746747
419 2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
422 common: update for new suppression
423 Makes check-valgrind pass with glib 2.46
425 2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
427 * gst/rtsp-server/rtsp-media.c:
428 rtsp-media: Take reference to media that will be prepared
429 default_prepare() takes a transfer-none reference GstRTSPMedia object.
430 Later on a g_idle_source_new() is created and a pointer to the media
431 object is passed as user data. If the media is freed before the idle
432 source is dispatched the media object pointer is invalid, but the idle
433 source callback expects it to still be valid. To fix this a reference to
434 the media object is taken when registering the source callback function
435 and a corresponding release of the reference is done when the souce is
437 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
439 2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
441 * examples/test-launch.c:
442 * examples/test-mp4.c:
443 * examples/test-ogg.c:
444 * examples/test-record.c:
445 * examples/test-uri.c:
446 rtsp-server: Fix memory leaks when context parse fails
447 When g_option_context_parse fails, context and error variables are not getting free'd
448 which results in memory leaks. Free'ing the same.
449 And replacing g_error_free with g_clear_error, which checks if the error being passed
450 is not NULL and sets the variable to NULL on free'ing.
451 https://bugzilla.gnome.org/show_bug.cgi?id=753863
453 2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
458 === release 1.6.0 ===
460 2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
466 * gst-rtsp-server.doap:
469 === release 1.5.91 ===
471 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
477 * gst-rtsp-server.doap:
480 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
482 * docs/libs/gst-rtsp-server-sections.txt:
483 * gst/rtsp-server/rtsp-stream.c:
484 stream: fix docs for recently-added get/set_buffer_size API
485 https://bugzilla.gnome.org/show_bug.cgi?id=749095
487 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
489 * gst/rtsp-server/rtsp-media.c:
490 rtsp-media: Don't crash on encrypted RTX SDP
491 In parse_keymgmt(), don't mutate the input string that's been passed
492 as const, especially since we might need the original value again if
493 the same key info applies to multiple streams (RTX, for example).
494 https://bugzilla.gnome.org/show_bug.cgi?id=754753
496 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
498 * examples/test-mp4.c:
499 test-mp4: Support filenames with spaces in them. Error out on too few arguments
501 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
503 * examples/test-record.c:
504 test-record: Check parameter count and print out help
505 If no launch pipeline was supplied, print out some help
507 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
509 * gst/rtsp-server/rtsp-media.c:
510 * gst/rtsp-server/rtsp-stream.c:
511 * gst/rtsp-server/rtsp-stream.h:
512 rtsp-stream: Implement UDP buffer size setting.
513 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
515 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
516 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
518 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
520 * gst/rtsp-server/rtsp-media.h:
521 rtsp-media: Fix small typo causing gtk-doc to complain
523 === release 1.5.90 ===
525 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
531 * gst-rtsp-server.doap:
534 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
536 * gst/rtsp-server/rtsp-media-factory.c:
537 media-factory: get port number through gst_rtsp_url_get_port
538 https://bugzilla.gnome.org/show_bug.cgi?id=753473
540 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
542 * tests/check/gst/media.c:
543 media-test: Removing unnecessary assertion
544 https://bugzilla.gnome.org/show_bug.cgi?id=753385
546 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
548 * gst/rtsp-server/rtsp-server.c:
549 Document that source keeps a ref on server until it's destroyed
550 https://bugzilla.gnome.org/show_bug.cgi?id=749227
552 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
554 * tests/check/gst/media.c:
555 media-test: Test for multiple dynamic payload
556 https://bugzilla.gnome.org/show_bug.cgi?id=753385
558 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
560 * gst/rtsp-server/rtsp-media.c:
561 media: Only add fakesink once per pipeline
562 The intention is to prevent going PLAYING state before pads are created.
563 If there was mutilple dynamic payload, it would leak few fakesink and
564 actually prevent from ever reaching playing state.
565 https://bugzilla.gnome.org/show_bug.cgi?id=753385
567 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
569 * gst/rtsp-server/rtsp-media.c:
570 Revert "rtsp-media: Only add 1 fakesink per pipeline"
571 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
573 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
575 * gst/rtsp-server/rtsp-media.c:
576 rtsp-media: Only add 1 fakesink per pipeline
577 There should be only one fakesink per pipeline, not per dynpay. This
578 would lead to element naming clash.
580 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
582 * gst/rtsp-server/rtsp-media.c:
583 rtsp-media: assertion error due to wrong condition check
584 In media to caps function, reserved_keys array is being used for variable i,
585 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
586 changed it to variable j
587 https://bugzilla.gnome.org/show_bug.cgi?id=753009
589 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
591 * gst/rtsp-server/rtsp-media.c:
592 rtsp-media: Strip keys from the fmtp that we use internally in our caps
593 Skip keys from the fmtp, which we already use ourselves for the
594 caps. Some software is adding random things like clock-rate into
595 the fmtp, and we would otherwise here set a string-typed clock-rate
596 in the caps... and thus fail to create valid RTP caps
597 https://bugzilla.gnome.org/show_bug.cgi?id=753009
599 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
601 * gst/rtsp-server/rtsp-thread-pool.c:
602 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
603 https://bugzilla.gnome.org/show_bug.cgi?id=752640
605 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
608 Automatic update of common submodule
609 From f74b2df to 9aed1d7
611 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
616 === release 1.5.2 ===
618 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
624 * gst-rtsp-server.doap:
627 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
629 * gst/rtsp-server/rtsp-client.c:
630 * gst/rtsp-server/rtsp-client.h:
631 * tests/check/gst/client.c:
632 rtsp-client: allow application to decide what requirements are supported
633 Add "check-requirements" signal and vfunc to allow application
634 (and subclasses) to check the requirements.
635 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
636 https://bugzilla.gnome.org/show_bug.cgi?id=749417
638 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
641 Automatic update of common submodule
642 From 6015d26 to f74b2df
644 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
646 * gst/rtsp-server/rtsp-media.c:
647 rtsp-media: Always use real payloader when creating streams
648 A bin that contains the real payloader might be used as payloader. In this
649 case we have to get the real payloader for the various properties it provides.
650 Example use cases for this are bins that payload some media and then have
651 additional elements that add metadata or RTP extension headers to the stream.
652 https://bugzilla.gnome.org/show_bug.cgi?id=750800
654 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
656 * examples/test-netclock-client.c:
657 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
659 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
661 * examples/test-netclock-client.c:
662 * examples/test-netclock.c:
663 test-netclock: Use new ntp-time-source property on rtpbin
664 Select the clock time to be used as NTP time source. This allows proper
665 synchronization between receivers, independent of sharing base times, and just
666 requires them to use the same clock.
668 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
670 * examples/test-netclock-client.c:
671 * examples/test-netclock.c:
672 test-netclock: Setting the same base time on sender and receiver is not necessary
673 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
675 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
677 * gst/rtsp-server/rtsp-stream.c:
678 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
679 https://bugzilla.gnome.org/show_bug.cgi?id=750764
681 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
683 * docs/libs/gst-rtsp-server.types:
684 docs: add missing types
685 https://bugzilla.gnome.org/show_bug.cgi?id=750764
687 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
689 * docs/libs/gst-rtsp-server-sections.txt:
690 docs: add missing apis
691 https://bugzilla.gnome.org/show_bug.cgi?id=750764
693 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
695 * examples/test-netclock-client.c:
696 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
698 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
700 * docs/libs/gst-rtsp-server-sections.txt:
701 * gst/rtsp-server/rtsp-auth.c:
702 * gst/rtsp-server/rtsp-auth.h:
703 GstRTSPAuth: Add client certificate authentication support
704 https://bugzilla.gnome.org/show_bug.cgi?id=750471
706 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
708 * examples/test-netclock-client.c:
709 test-netclock-client: Use new GstClock API to wait for clock synchronization
711 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
713 * examples/test-netclock-client.c:
714 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
715 A mainloop is needed to get glimagesink to display something on OSX, and
716 the source-setup signal just makes things a little bit easier.
718 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
721 Automatic update of common submodule
722 From d9a3353 to 6015d26
724 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
727 Automatic update of common submodule
728 From d37af32 to d9a3353
730 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
733 Automatic update of common submodule
734 From 21ba2e5 to d37af32
736 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
739 Automatic update of common submodule
740 From c408583 to 21ba2e5
742 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
744 * docs/libs/Makefile.am:
745 docs: remove variables that we define in the snippet from common
746 This is syncing our Makefile.am with upstream gtkdoc.
748 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
751 Automatic update of common submodule
752 From 44a3517 to c408583
754 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
759 === release 1.5.1 ===
761 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
767 * gst-rtsp-server.doap:
770 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
772 * gst/rtsp-server/rtsp-client.c:
773 rtsp-client: No flush during Teardown.
774 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
775 backlog is empty it can happen that just a part of a message will be
776 sent and rest is in backlog queue. If then flush during teardown
777 just a part of message will be sent.This can lead to client miss
778 teardown response since it expect to get the last part of message.
779 The flushing during teardown was introduced to fix a deadlock that now
780 is fixed more generally in handle_request by temporary setting backlog
782 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
784 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
786 * tests/check/Makefile.am:
787 tests: Use AM_TESTS_ENVIRONMENT
788 Needed by the new automake test runner and the
789 current version of the common submodule.
791 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
793 * gst/rtsp-server/rtsp-media.h:
794 * gst/rtsp-server/rtsp-stream.h:
795 rtsp-server: Use single-include rtsp header to make sure we get all definitions
797 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
799 * gst/rtsp-server/rtsp-media.c:
800 rtsp-media: Mark some more functions static
802 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
804 * gst/rtsp-server/rtsp-media.c:
805 rtsp-media: Only unblock the media in suspend() when actually changing the state
806 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
808 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
810 * examples/test-video-rtx.c:
811 examples: Use AVPF profile for the RTX example
813 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
815 * gst/rtsp-server/rtsp-sdp.c:
816 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
818 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
820 * gst/rtsp-server/rtsp-stream.c:
821 rtsp-stream: get valid clock-rate from last-sample
822 clock-rate in last-sample's caps is integer, not unsigned.
823 To get this value properly, variable needs to be type-casted to int.
824 https://bugzilla.gnome.org/show_bug.cgi?id=747614
826 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
830 autogen.sh: only run autopoint if gettext requested in configure.ac
831 Not just because there happens to be a po directory.
832 https://bugzilla.gnome.org/show_bug.cgi?id=748058
834 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
837 Revert "configure.ac: uncomment gettext version setup"
838 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
839 We don't need a gettext setup here and there's no po
840 directory either, so no reason why autopoint would be
841 run in the first place.
842 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
844 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
846 * examples/test-multicast.c:
847 * examples/test-multicast2.c:
848 * examples/test-sdp.c:
849 * examples/test-video-rtx.c:
850 * examples/test-video.c:
851 * tests/test-cleanup.c:
852 * tests/test-reuse.c:
853 Fix timeout function signatures across tests and examples
855 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
857 * tests/check/Makefile.am:
858 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
859 Make sure the test environment is set up.
860 https://bugzilla.gnome.org//show_bug.cgi?id=747624
862 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
865 configure: bump automake requirement to 1.14 and autoconf to 2.69
866 This is only required for builds from git, people can still
867 build tarballs if they only have older autotools.
868 https://bugzilla.gnome.org//show_bug.cgi?id=747624
870 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
873 configure.ac: uncomment gettext version setup
874 Fixes autogen.sh. It would run autopoint, which would complain
875 that it could not find the gettext version in configure.ac.
876 https://bugzilla.gnome.org/show_bug.cgi?id=748058
878 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
880 * examples/test-video-rtx.c:
881 test-video-rtx: set exact payload type to PCMA payloader
882 Setting wrong payload type causes failure to do retransmission through audio stream
883 https://bugzilla.gnome.org/show_bug.cgi?id=747839
885 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
887 * gst/rtsp-server/rtsp-media.c:
888 * gst/rtsp-server/rtsp-stream.c:
889 * gst/rtsp-server/rtsp-stream.h:
890 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
891 Because of duplicated g_signal_connect for request-aux-sender signal,
892 wrong stream pointer is passed to the signal handler.
893 Instead of passing each stream, pass stream array and get the relevant stream.
894 https://bugzilla.gnome.org/show_bug.cgi?id=747839
896 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
900 Update autogen.sh to latest version from common
901 Fixes build after aclocal_check etc. helpers have been removed.
903 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
906 Automatic update of common submodule
907 From bc76a8b to c8fb372
909 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
911 * gst/rtsp-server/rtsp-stream.c:
912 rtsp-stream: Limit the queues to 1 buffer
913 We only need them to be able to pre-roll, queueing up more data here
914 is only going to harm latency and memory usage.
916 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
918 * gst/rtsp-server/rtsp-stream.c:
919 rtsp-stream: Update comment and ASCII art to the latest code
920 We have a queue in front of the udpsink too to prevent the pipeline from
923 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
925 * gst/rtsp-server/rtsp-stream.c:
926 rtsp-media: Properly return first rtptime
927 Instead we where returning first GstBuffer timestamp. This would result
928 in clock skew and unwanted behaviour in RTSP playback.
929 https://bugzilla.gnome.org/show_bug.cgi?id=746479
931 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
933 * gst/rtsp-server/rtsp-stream.c:
934 rtsp-stream: Don't leave buffer mapped
935 If the seq is NULL, the RTP buffer was left mapped. We should always
938 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
943 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
945 * gst/rtsp-server/rtsp-media-factory.c:
946 * tests/check/gst/client.c:
947 Fix double semicolons
949 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
951 * gst/rtsp-server/rtsp-stream.c:
952 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
953 This gives more accurate values than asking the payloader. There might be
954 queueing happening between the payloader and the sink.
955 https://bugzilla.gnome.org/show_bug.cgi?id=745704
957 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
959 * gst/rtsp-server/rtsp-media.c:
960 rtsp-media: Don't seek for PLAY if the position will not change
961 https://bugzilla.gnome.org/show_bug.cgi?id=745704
963 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
965 * gst/rtsp-server/rtsp-media.c:
966 rtsp-media: Don't include payload type in the caps for framesize
967 When the sdp media attribute framesize are converted to caps
968 the <payload> should not be included.
969 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
970 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
972 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
974 * gst/rtsp-server/rtsp-sdp.c:
975 rtsp-sdp: add payload type to the sdp framesize attribute
976 The sdp framesize attribute is desribed in RFC6064. It is specified
977 for payloading of H263 and has the following form
978 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
979 should be added to the caps in a payloader and the <payload type> should
980 be added by the rtsp-server.
981 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
983 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
985 * examples/test-uri.c:
986 examples: test-uri: fix tainted variable
987 Insignificant but this keeps Coverity happy.
990 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
992 * examples/.gitignore:
993 * examples/Makefile.am:
994 * examples/test-netclock-client.c:
995 * examples/test-netclock.c:
996 examples: Add a simple example of network synch for live streams.
997 An example server and client that works for synchronising live streams
998 only - as it can't support pause/play.
1000 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
1002 * gst/rtsp-server/rtsp-media-factory.c:
1003 * gst/rtsp-server/rtsp-media-factory.h:
1004 rtsp-media-factory: Add functions to set/get the media gtype
1005 Allow specifying the GType of a GstRtspMedia subclass to create
1006 as a simpler way to get the factory to create a custom
1007 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
1009 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
1011 * gst/rtsp-server/rtsp-media.c:
1012 rtsp-media: fix double unlock in _get_buffer_size()
1013 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
1014 because of double g_mutex_unlock () usage.
1015 https://bugzilla.gnome.org/show_bug.cgi?id=745434
1017 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
1019 * gst/rtsp-server/rtsp-session-pool.c:
1020 * gst/rtsp-server/rtsp-session.c:
1021 * gst/rtsp-server/rtsp-session.h:
1022 rtsp-session: Use monotonic time for RTSP session timeout
1023 Changed RTSP session timeout handling to monotonic time
1024 and deprecating the API for current system time.
1025 This fixes timeouts when the system time changes.
1026 https://bugzilla.gnome.org/show_bug.cgi?id=743346
1028 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
1030 * gst/rtsp-server/rtsp-client.c:
1031 * gst/rtsp-server/rtsp-media.c:
1032 rtsp-client: Only error out in PLAY if seeking actually failed
1033 If the media was just not seekable, we continue from whatever position we are
1034 and let the client decide if that is what is wanted or not.
1035 Only if the actual seek failed, we can't really recover and should error out.
1037 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
1039 * gst/rtsp-server/rtsp-stream.c:
1040 rtsp-stream: Add necessary queues between tee and multiudpsink
1041 https://bugzilla.gnome.org/show_bug.cgi?id=744379
1043 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
1045 * gst/rtsp-server/rtsp-client.c:
1046 * gst/rtsp-server/rtsp-media.c:
1047 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
1048 Instead error out properly the same way as if the SEEKING query already
1051 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
1053 * gst/rtsp-server/rtsp-stream.h:
1054 rtsp-stream: minor code formatting fix
1056 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
1058 * gst/rtsp-server/rtsp-media.c:
1059 rtsp-media: fix logic for collect_streams
1060 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
1061 all streams it knows if it got any, and can check if the transport mode is OK.
1064 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
1066 * gst/rtsp-server/rtsp-media.c:
1067 rtsp-media: Don't set the transport mode based on what elements we find
1068 Just print a warning if the one that was set before disagrees with what
1069 elements we found. It must already be set to something before as this
1070 function is called after we received the SDP from ANNOUNCE in RECORD mode,
1071 and we would reject ANNOUNCE if the RECORD flag was not set.
1073 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
1075 * tests/check/gst/rtspserver.c:
1076 tests: rtspserver: rename shadowed variable
1077 We have two different 'sink' variables here,
1078 rename one of them for clarity.
1080 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
1082 * gst/rtsp-server/rtsp-client.c:
1083 rtsp-client: fix awkward if clause
1085 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
1087 * examples/test-uri.c:
1088 examples: test-uri: improve uri argument handling and accept file names
1089 Print an error if the argument passed is not a URI and can't
1090 be converted into one, or no arguments have been provided.
1092 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
1094 * examples/test-uri.c:
1095 examples: test-uri: don't remove mount point after 10 seconds
1096 It's very irritating when trying to test stuff repeatedly
1097 and serves no real purpose other than showing that it can
1100 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
1102 * examples/.gitignore:
1103 examples: add new test-record to .gitignore
1105 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
1107 * examples/test-record.c:
1108 * gst/rtsp-server/rtsp-client.c:
1109 * gst/rtsp-server/rtsp-media-factory.c:
1110 * gst/rtsp-server/rtsp-media-factory.h:
1111 * gst/rtsp-server/rtsp-media.c:
1112 * gst/rtsp-server/rtsp-media.h:
1113 * tests/check/gst/rtspserver.c:
1114 rtsp-media: Use flags to distinguish between PLAY and RECORD media
1116 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
1118 * examples/test-record.c:
1119 test-record: Set latency for playback-style example to 2s instead of 200ms
1121 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
1123 * tests/check/gst/rtspserver.c:
1124 tests: add some unit tests for ANNOUNCE and RECORD
1125 https://bugzilla.gnome.org/show_bug.cgi?id=743175
1127 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
1129 * gst/rtsp-server/rtsp-client.c:
1130 rtsp-client: fix a couple of leaks in handle_announce
1132 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
1134 * gst/rtsp-server/rtsp-media-factory.c:
1135 * gst/rtsp-server/rtsp-media-factory.h:
1136 * gst/rtsp-server/rtsp-media.c:
1137 * gst/rtsp-server/rtsp-media.h:
1138 rtsp-media: Expose latency setting for setting the rtpbin latency
1140 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
1142 * examples/test-record.c:
1143 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
1145 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
1147 * gst/rtsp-server/rtsp-stream.c:
1148 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
1150 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
1152 * examples/Makefile.am:
1153 * examples/test-record.c:
1154 * gst/rtsp-server/rtsp-client.c:
1155 * gst/rtsp-server/rtsp-client.h:
1156 * gst/rtsp-server/rtsp-media-factory.c:
1157 * gst/rtsp-server/rtsp-media-factory.h:
1158 * gst/rtsp-server/rtsp-media.c:
1159 * gst/rtsp-server/rtsp-media.h:
1160 * gst/rtsp-server/rtsp-session-media.c:
1161 * gst/rtsp-server/rtsp-stream.c:
1162 * gst/rtsp-server/rtsp-stream.h:
1163 Add initial support for RECORD
1164 We currently only support media that is RECORD or PLAY only, not both at once.
1165 https://bugzilla.gnome.org/show_bug.cgi?id=743175
1167 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
1169 * gst/rtsp-server/rtsp-stream.c:
1170 rtsp-stream: RTCP and RTP transport cache cookies seperated
1171 RTCP packets were not sent because the same tr_cache_cookie was used for
1172 both RTP and RTCP. So only one of the tr_cache lists were populated
1173 depending on which one was sent first. If the tr_cache list is not
1174 populated then no packets can be sent. Most often this happened to be
1175 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
1176 resulted in both the tr_cache_lists to be populated regardless of which
1178 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
1180 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
1182 * gst/rtsp-server/rtsp-stream.c:
1183 rtsp-stream: fix false compiler warning
1184 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
1186 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
1188 * gst/rtsp-server/rtsp-client.c:
1189 rtsp-client: log interleaved data received
1191 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
1193 * gst/rtsp-server/rtsp-client.c:
1194 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
1196 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
1198 * gst/rtsp-server/rtsp-client.c:
1199 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
1201 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
1203 * gst/rtsp-server/rtsp-client.c:
1204 rtsp-client: Use a random session ID in the SDP
1205 RFC4566 Section 5.2 says that it should make the username, session id,
1206 nettype, addrtype and unicast address tuple globally unique. Always using
1207 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
1208 Instead let's create a 64 bit random number, which at least brings us
1209 closer to the goal of global uniqueness.
1210 https://tools.ietf.org/html/rfc4566#section-5.2
1212 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
1214 * examples/test-launch.c:
1215 * examples/test-mp4.c:
1216 * examples/test-ogg.c:
1217 * examples/test-uri.c:
1218 examples: Don't call gst_init() and gst_get_option_group()
1219 The latter calls the former at the appropriate time.
1221 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
1223 * gst/rtsp-server/rtsp-client.c:
1224 rtsp-client: Drop trailing \0 of RTSP DATA messages
1225 We add a trailing \0 in GstRTSPConnection to make parsing of
1226 string message bodies easier (e.g. the SDP from DESCRIBE) but
1227 for actual data this means we have to drop it or otherwise
1228 create invalid data.
1230 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
1232 * gst/rtsp-server/rtsp-stream.c:
1233 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
1234 Fixes crash when two threads access handle_new_sample() at the same
1235 time, one for RTP, one for RTCP.
1236 Otherwise, when iterating over the transports cache, it might be modified by
1237 another thread at the same time if the transports cookie has changed.
1238 https://bugzilla.gnome.org/show_bug.cgi?id=742954
1240 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
1242 * gst/rtsp-server/rtsp-stream.c:
1243 rtsp-stream: Set format=TIME on our app sources for TCP
1245 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
1247 * gst/rtsp-server/rtsp-session-pool.c:
1248 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
1249 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
1250 RFC 2326 states that session IDs may consist of alphanumeric as well as
1251 the safe characters $-_.+ -- N.B. the percent character is not allowed.
1252 Previously the session ID was URI-escaped, this meant that any character
1253 which was not alphanumeric or any of the characters +-._~ would be
1254 percent encoded. While the RFC (surprisingly) mentions that linear white
1255 space in session IDs should be URI-escaped, it does not say anything
1256 about other characters. Moreover no white space is allowed in the
1257 session ID. Finally the percent character which is the result of
1258 URI-escaping is not allowed in a session ID.
1259 So there is no reason to do any URI-escaping, and now it is removed.
1260 https://bugzilla.gnome.org/show_bug.cgi?id=742869
1262 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
1265 Automatic update of common submodule
1266 From f2c6b95 to bc76a8b
1268 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
1271 Fix 'make check' from top-level directory
1273 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
1275 * examples/test-launch.c:
1276 * examples/test-mp4.c:
1277 * examples/test-ogg.c:
1278 * examples/test-uri.c:
1279 examples: Add command-line parsing and take a 'port' argument
1280 This allows users to run multiple servers on different ports for testing.
1281 Only done for examples that actually take arguments and hence are capable of
1282 outputting different streams for each instance on each port.
1283 https://bugzilla.gnome.org/show_bug.cgi?id=742115
1285 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
1287 * gst/rtsp-server/rtsp-client.c:
1288 * gst/rtsp-server/rtsp-client.h:
1289 rtsp-client: Add a send_message default signal handler
1290 This allows subclasses to easily hook into the response sending
1291 mechanism without doing everything from a signal, which seems
1292 awkward from subclasses.
1294 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
1297 Automatic update of common submodule
1298 From ef1ffdc to f2c6b95
1300 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1304 configure: add --disable-examples switch
1305 https://bugzilla.gnome.org/show_bug.cgi?id=741678
1307 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
1309 * examples/.gitignore:
1310 * examples/Makefile.am:
1311 * examples/test-video-rtx.c:
1312 examples: add a retransmisison example implementing RFC4588
1313 Currently only SSRC-multiplexed rtx streams are supported
1315 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
1317 * gst/rtsp-server/rtsp-stream.c:
1318 rtsp-stream: Fix some minor memory leaks
1320 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
1322 * gst/rtsp-server/rtsp-media.c:
1323 rtsp-media: Some minor cleanup
1325 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
1327 * gst/rtsp-server/rtsp-stream.c:
1328 rtsp-stream: Fix compiler warnings
1329 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
1330 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1332 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
1333 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1336 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
1338 * docs/libs/gst-rtsp-server-sections.txt:
1339 * gst/rtsp-server/rtsp-media-factory.c:
1340 * gst/rtsp-server/rtsp-media-factory.h:
1341 * gst/rtsp-server/rtsp-media.c:
1342 * gst/rtsp-server/rtsp-media.h:
1343 * gst/rtsp-server/rtsp-sdp.c:
1344 * gst/rtsp-server/rtsp-stream.c:
1345 * gst/rtsp-server/rtsp-stream.h:
1346 media: implement ssrc-multiplexed retransmission support
1347 based off RFC 4588 and the server-rtpaux example in -good
1349 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
1351 * gst/rtsp-server/rtsp-client.c:
1352 * gst/rtsp-server/rtsp-stream-transport.c:
1353 * gst/rtsp-server/rtsp-stream.c:
1354 rtsp: Ref transports in hash table.
1355 Also ref streams for transports.
1356 This solves a crash when reciving a rtcp after teardown but before
1358 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
1360 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
1363 Automatic update of common submodule
1364 From 7bb2bce to ef1ffdc
1366 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
1368 * gst/rtsp-server/rtsp-client.c:
1369 client: refactor cleanup of cached media
1371 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
1373 * tests/check/gst/client.c:
1375 The session leak is now fixed, lets remove those FIXME comments.
1377 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
1379 * tests/check/gst/rtspserver.c:
1380 tests: Test to setup two sessions on one connection
1381 https://bugzilla.gnome.org/show_bug.cgi?id=739112
1383 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
1385 * tests/check/gst/rtspserver.c:
1386 tests: Test setup with tcp transport
1387 https://bugzilla.gnome.org/show_bug.cgi?id=739112
1389 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
1391 * gst/rtsp-server/rtsp-client.c:
1392 client: Configure transport after creating session media
1393 The default implementation of configure_client_transport() in
1394 rtsp-client uses the session media when it chooses channels for
1395 interleaved traffic.
1396 https://bugzilla.gnome.org/show_bug.cgi?id=739112
1398 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
1400 * gst/rtsp-server/rtsp-client.c:
1401 * gst/rtsp-server/rtsp-session-media.c:
1402 client: Stop caching media in client when doing setup
1403 If the media has been managed by a session media, it should not be
1404 cached in the client any longer. The GstRTSPSessionMedia object is now
1405 responsible for unpreparing the GstRTSPMedia object using
1406 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
1408 https://bugzilla.gnome.org/show_bug.cgi?id=739112
1410 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1412 * gst/rtsp-server/rtsp-stream.c:
1413 rtsp-stream: unref srtp decoder when leaving bin
1414 https://bugzilla.gnome.org/show_bug.cgi?id=739481
1416 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1418 * gst/rtsp-server/rtsp-client.c:
1419 rtsp-client: mikey memory leaks
1420 https://bugzilla.gnome.org/show_bug.cgi?id=739383
1422 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
1425 Automatic update of common submodule
1426 From 84d06cd to 7bb2bce
1428 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
1431 Parallelise 'make check-valgrind'
1433 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
1436 Automatic update of common submodule
1437 From a8c8939 to 84d06cd
1439 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
1442 Automatic update of common submodule
1443 From 36388a1 to a8c8939
1445 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
1447 * gst/rtsp-server/rtsp-media.c:
1448 rtsp-media: deactivate media when shutting down from paused
1449 This was only done when going directly from playing.
1450 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
1452 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1454 * gst/rtsp-server/rtsp-client.c:
1455 * gst/rtsp-server/rtsp-context.h:
1456 rtsp-client: add stream transport to context
1457 We add the stream transport to the context so we can get the configured
1458 client stream transport in the setup request signal.
1459 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
1461 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1463 * gst/rtsp-server/rtsp-stream.c:
1464 stream: release lock even not all transports have been removed
1465 We don't want to keep the lock even we return FALSE because not all the
1466 transports have been removed. This could lead into a deadlock.
1467 https://bugzilla.gnome.org/show_bug.cgi?id=737797
1469 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
1471 * gst/rtsp-server/rtsp-sdp.c:
1472 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
1473 These were renamed in GstRTPBasePayload in 1.0
1475 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1477 * gst/rtsp-server/rtsp-client.c:
1478 client: set session media to NULL without the lock
1479 We need to set session medias to NULL without the client lock otherwise
1480 we can end up in a deadlock if another thread is waiting for the lock
1481 and media unprepare is also waiting for that thread to end.
1482 https://bugzilla.gnome.org/show_bug.cgi?id=737690
1484 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
1486 * gst/rtsp-server/rtsp-media.c:
1487 rtsp-media: Set state to UNPREPARING in all cases
1489 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
1491 * gst/rtsp-server/rtsp-media.c:
1492 media: set state to unpreparing when unprepare is initiated
1493 https://bugzilla.gnome.org/show_bug.cgi?id=737675
1495 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
1497 * gst/rtsp-server/rtsp-client.c:
1498 rtsp-client: Remove backlog limit while processings requests
1499 If the backlog limit is kept two cases of deadlocks may be
1500 encountered when streaming over TCP. Without the backlog
1501 limit this deadlocks can not happen, at the expence of
1503 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
1505 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
1507 * gst/rtsp-server/rtsp-client.c:
1508 rtsp-client: do not free main context before rtsp watch
1509 https://bugzilla.gnome.org/show_bug.cgi?id=737110
1511 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
1513 * tests/check/gst/rtspserver.c:
1514 tests: Extend unit test timeout to accomodate for valgrind
1515 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1517 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
1519 * gst/rtsp-server/rtsp-client.c:
1520 * gst/rtsp-server/rtsp-session.c:
1521 * gst/rtsp-server/rtsp-stream-transport.c:
1522 rtsp-*: Treat sending packets to clients as keepalive
1523 As long as gst-rtsp-server can successfully send RTP/RTCP data to
1524 clients then the client must be reading. This change makes the server
1525 timeout the connection if the client stops reading.
1526 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1528 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
1530 * gst/rtsp-server/rtsp-client.c:
1531 rtsp-client: Allow backlog to grow while expiring session
1532 Allow the send backlog in the RTSP watch to grow to unlimited size while
1533 attempting to bring the media pipeline to NULL due to a session
1534 expiring. Without this change the appsink element cannot change state
1535 because it is blocked while rendering data in the new_sample callback.
1536 This callback will block until it has successfully put the data into the
1537 send backlog. There is a chance that the send backlog is full at this
1538 point which means that the callback may block for a long time, possibly
1539 forever. Therefore the media pipeline may also be prevented from
1540 changing state for a long time.
1541 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1543 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
1545 * gst/rtsp-server/rtsp-client.c:
1546 rtsp-client: Make old compilers happy
1547 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
1548 Just in case that guint8 doesn't fit in a pointer. Just in case ...
1550 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
1552 * gst/rtsp-server/rtsp-client.c:
1553 client: raise the backlog limits before pausing
1554 We need to raise the backlog limits before pausing the pipeline or else
1555 the appsink might be blocking in the render method in wait_backlog() and
1556 we would deadlock waiting for paused.
1557 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
1559 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
1561 * gst/rtsp-server/rtsp-client.c:
1562 client: make define for the WATCH_BACKLOG
1563 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
1565 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
1567 * gst/rtsp-server/rtsp-client.c:
1568 client: simplify session transport handling
1569 link/unlink of the transport in a session was done to keep track of all
1570 TCP transports and to send RTP/RTCP data to the streams. We can simplify
1571 that by putting all the TCP transports in a hashtable indexed with the
1573 We also don't need to link/unlink the transports when we pause/resume
1574 the streams. The same effect is already achieved when we pause/play the
1575 media. Indeed, when we pause the media, the transport is removed from
1576 the media and the callbacks will not be called anymore.
1577 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
1579 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
1581 * gst/rtsp-server/rtsp-stream-transport.c:
1582 * gst/rtsp-server/rtsp-stream-transport.h:
1583 stream-transport: make method to handle received data
1584 Make a method to handle the data received on a channel. It sends the
1585 data to the stream of the transport on the RTP or RTCP pads based on
1588 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
1590 * examples/test-mp4.c:
1591 test: add example of dumping RTCP reports
1593 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
1595 * gst/rtsp-server/rtsp-media.c:
1596 * gst/rtsp-server/rtsp-stream.c:
1597 * gst/rtsp-server/rtsp-stream.h:
1598 rtsp-media: Make sure that sequence numbers are monotonic after pause
1599 The sequence number is not monotonic for RTP packets after pause. The
1600 reason is basepayloader generates a randon sequence number when the
1601 pipeline goes from ready to pause. With this fix generation of sequence
1602 number will be monotonic when going from pause to play request.
1603 https://bugzilla.gnome.org/show_bug.cgi?id=736017
1605 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
1607 * gst/rtsp-server/rtsp-client.c:
1608 rtsp-client: Protect saved clients watch with a mutex
1609 Fixes a crash when close() is called while merging clients
1610 in handle_tunnel(). In that case close() would destroy the
1611 watch while it is still being used in handle_tunnel().
1612 https://bugzilla.gnome.org/show_bug.cgi?id=735570
1614 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
1616 * gst/rtsp-server/rtsp-stream.c:
1617 rtsp-stream: Remove the multicast group udp sources when removing from the bin
1619 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
1621 * gst/rtsp-server/rtsp-media.c:
1622 * gst/rtsp-server/rtsp-stream.c:
1623 * gst/rtsp-server/rtsp-stream.h:
1624 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
1625 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
1626 seeking and will always continue counting the time. This leads to
1627 the NPT after a backwards seek to be something completely different
1628 to the actual seek position.
1629 https://bugzilla.gnome.org/show_bug.cgi?id=732644
1631 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
1633 * examples/test-appsrc.c:
1634 examples: fix another reference leak
1635 gst_rtsp_media_get_element() returns a new ref.
1637 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1639 * examples/test-appsrc.c:
1640 examples: unref element after usage
1641 gst_bin_get_by_name_recurse_up() returns an element
1642 reference that must be unreffed after usage.
1643 https://bugzilla.gnome.org/show_bug.cgi?id=734546
1645 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
1647 * gst/rtsp-server/rtsp-media.c:
1648 signals: Fix copy-pasto in target-state signal offset
1650 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
1654 Makefile: Add usage of build-checks step
1655 Allows building checks without running them
1657 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
1659 * gst/rtsp-server/rtsp-stream.c:
1660 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
1661 When a UDP multicast transport is used it is expected that the server listens
1662 for RTP and RTCP packets on the multicast group with the corresponding port.
1663 Without this we will never get RTCP packets from clients in multicast mode.
1664 https://bugzilla.gnome.org/show_bug.cgi?id=732238
1666 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
1671 === release 1.4.0 ===
1673 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
1679 * gst-rtsp-server.doap:
1682 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
1684 * gst/rtsp-server/rtsp-media.h:
1685 media: correct misspelled words in description
1686 https://bugzilla.gnome.org/show_bug.cgi?id=733244
1688 === release 1.3.91 ===
1690 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
1696 * gst-rtsp-server.doap:
1699 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
1701 * docs/libs/gst-rtsp-server-sections.txt:
1704 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
1706 * gst/rtsp-server/rtsp-server.c:
1707 server: implement client REMOVE filter
1709 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
1711 * gst/rtsp-server/rtsp-client.c:
1712 * gst/rtsp-server/rtsp-client.h:
1713 client: expose _close() method
1714 Expose a previously internal close method to close the client
1717 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
1719 * gst/rtsp-server/rtsp-session-pool.c:
1720 session-pool: signal session-removed outside of the lock
1721 Release the lock before emiting the session-removed signal.
1723 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
1725 * gst/rtsp-server/rtsp-client.c:
1726 * gst/rtsp-server/rtsp-server.c:
1727 * gst/rtsp-server/rtsp-session-pool.c:
1728 * gst/rtsp-server/rtsp-session.c:
1729 * gst/rtsp-server/rtsp-stream.c:
1730 filter: Release lock in filter functions
1731 Release the object lock before calling the filter functions. We need to
1732 keep a cookie to detect when the list changed during the filter
1733 callback. We also keep a hashtable to make sure we only call the filter
1734 function once for each object in case of concurrent modification.
1735 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
1737 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
1739 * gst/rtsp-server/rtsp-client.c:
1740 client: check if watch is set in handle_teardown()
1741 The unit tests run without a watch
1743 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
1745 * tests/check/gst/client.c:
1746 client tests: send teardown to cleanup session
1748 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
1750 * tests/check/gst/rtspserver.c:
1751 server tests: send teardown to cleanup session
1753 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
1755 * gst/rtsp-server/rtsp-client.c:
1756 client: keep ref to client for the session removed handler
1757 This extra ref will be dropped when all client sessions have been
1758 removed. A session is removed when a client sends teardown, closes its
1759 endpoint of the TCP connection or the sessions expires.
1760 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
1762 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
1764 * gst/rtsp-server/rtsp-client.c:
1765 * gst/rtsp-server/rtsp-session.c:
1766 * tests/check/gst/client.c:
1767 client: manage media in session as a last step
1768 Once we manage a media in a session, we can't unmanage it anymore
1769 without destroying it. Therefore, first check everything before we
1770 manage the media, otherwise if something is wrong we have no way to
1772 If we created a new session and something went wrong, remove the session
1773 again. Fixes a leak in the unit test.
1775 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
1777 * examples/test-mp4.c:
1778 * examples/test-ogg.c:
1779 examples: print 'stream ready at url' for mp4 and ogg example
1781 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
1783 * gst/rtsp-server/rtsp-client.c:
1784 * gst/rtsp-server/rtsp-sdp.c:
1785 rtsp: fix for MIKEY api change
1787 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
1789 * gst/rtsp-server/rtsp-client.c:
1790 client: free watch context only once
1791 The watch context is freed when the source is destroyed. Avoids
1792 a CRITICAL when we try to unref the context twice.
1794 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
1796 * gst/rtsp-server/rtsp-client.c:
1799 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
1801 * gst/rtsp-server/rtsp-client.c:
1802 client: protect sessions with lock
1803 Protect the list of sessions with the lock.
1804 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
1806 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
1808 * gst/rtsp-server/rtsp-client.c:
1809 Client: keep a ref to the session
1810 Don't just keep a weak ref to the session objects but use a hard ref. We
1811 will be notified when a session is removed from the pool (expired) with
1812 the new session-removed signal.
1813 Don't automatically close the RTSP connection when all the sessions of
1814 a client are removed, a client can continue to operate and it can create
1815 a new session if it wants. If you want to remove the client from the
1816 server, you have to use gst_rtsp_server_client_filter() now.
1817 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
1818 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
1820 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
1822 * gst/rtsp-server/rtsp-session-pool.c:
1823 * gst/rtsp-server/rtsp-session-pool.h:
1824 session-pool: add session-removed signal
1825 Add a signal to be notified when a session is removed from the pool.
1827 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
1829 * gst/rtsp-server/Makefile.am:
1830 * gst/rtsp-server/rtsp-server.h:
1831 Make rtsp-server.h a single-include header, use it for G-I
1832 https://bugzilla.gnome.org/show_bug.cgi?id=732411
1834 === release 1.3.90 ===
1836 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
1842 * gst-rtsp-server.doap:
1845 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
1847 * gst/rtsp-server/rtsp-stream.c:
1848 stream: crypto can be NULL
1850 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
1852 * gst/rtsp-server/rtsp-client.c:
1853 * gst/rtsp-server/rtsp-media.c:
1854 * gst/rtsp-server/rtsp-mount-points.c:
1855 introspection: add missing allow-none annotations
1856 https://bugzilla.gnome.org/show_bug.cgi?id=730952
1858 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
1860 * gst/rtsp-server/rtsp-address-pool.c:
1861 * gst/rtsp-server/rtsp-media.c:
1862 * gst/rtsp-server/rtsp-session-media.c:
1863 * gst/rtsp-server/rtsp-session-pool.c:
1864 * gst/rtsp-server/rtsp-stream-transport.c:
1865 * gst/rtsp-server/rtsp-stream.c:
1866 * gst/rtsp-server/rtsp-token.c:
1867 introspection: add (nullable) annotations to return values
1868 https://bugzilla.gnome.org/show_bug.cgi?id=730952
1870 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
1872 * gst/rtsp-server/rtsp-client.c:
1873 * gst/rtsp-server/rtsp-stream.c:
1874 gi: improve annotations
1875 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
1877 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
1879 * gst/rtsp-server/rtsp-client.c:
1880 * gst/rtsp-server/rtsp-media-factory.c:
1881 * gst/rtsp-server/rtsp-media.c:
1882 * gst/rtsp-server/rtsp-server.c:
1883 signals: use generic marshal function
1884 Use the generic C marshal function.
1885 Use more explicit type instead of G_TYPE_POINTER
1887 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
1889 * gst/rtsp-server/rtsp-context.h:
1890 context: add type macro
1892 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
1894 * gst/rtsp-server/rtsp-client.c:
1895 * gst/rtsp-server/rtsp-sdp.c:
1896 * gst/rtsp-server/rtsp-sdp.h:
1897 sdp: hide key length defines
1898 They don't have a namespace.
1900 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
1905 === release 1.3.3 ===
1907 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
1913 * gst-rtsp-server.doap:
1916 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1918 * gst/rtsp-server/rtsp-client.c:
1919 * gst/rtsp-server/rtsp-sdp.c:
1920 * gst/rtsp-server/rtsp-sdp.h:
1921 mikey: add different key length parameters
1922 Add encryption and authentication key length parameters to MIKEY. For
1923 the encoders, the key lengths are obtained from the cipher and auth
1924 algorithms set in the caps. For the decoders, they are obtained while
1925 parsing the key management from the client.
1926 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
1928 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
1930 * tests/check/gst/stream.c:
1931 stream tests: Make sure we get right multicast address from stream
1932 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
1934 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
1936 * gst/rtsp-server/rtsp-client.c:
1937 client: ref the context until rtsp watch is alive
1938 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
1940 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
1942 * gst/rtsp-server/rtsp-client.c:
1943 client: Destroy the rtsp watch after connection close
1945 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
1947 * gst/rtsp-server/rtsp-media.c:
1948 media: fix confusing comment
1950 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
1952 * gst/rtsp-server/rtsp-session.c:
1953 rtsp-session: Timeout in header.
1954 Adding the possbilty to always have timout in header.
1955 This is configurabe with setting "timeout-always-visible".
1956 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
1958 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
1963 === release 1.3.2 ===
1965 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
1972 * gst-rtsp-server.doap:
1975 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
1978 Automatic update of common submodule
1979 From 211fa5f to 1f5d3c3
1981 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
1983 * gst/rtsp-server/rtsp-client.c:
1984 client: store TCP ports in transport
1985 Store the TCP ports in the transport when we are doing RTSP over TCP.
1986 This way, we can easily get to the ports from the transport.
1987 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
1989 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1991 * gst/rtsp-server/rtsp-stream.c:
1992 stream: add signals for new RTP/RTCP encoders
1993 New signals to allow the user to configure the dynamically created
1995 https://bugzilla.gnome.org/show_bug.cgi?id=730228
1997 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
1999 * gst/rtsp-server/rtsp-media.c:
2000 * gst/rtsp-server/rtsp-media.h:
2001 media: Make suspend()/unsuspend() virtual
2002 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
2004 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2006 * gst/rtsp-server/rtsp-client.c:
2007 client: fix send-message signal marshaller
2008 Use generic marshalling for the send-message signal. It has
2009 two POINTER arguments, not just one.
2010 https://bugzilla.gnome.org/show_bug.cgi?id=729900
2012 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
2014 * tests/check/gst/media.c:
2015 tests: add and remove pads only once
2016 In this test we simulate a dynamic pad by watching the caps event.
2017 Because of renegotiation in the base payloader now, this caps is sent
2018 multiple times but we can only deal with 1 invocation, use a variable to
2019 only 'add and remove' the pad once.
2021 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
2023 * tests/check/gst/rtspserver.c:
2024 tests: add unit test for correct handling of Require headers
2025 https://bugzilla.gnome.org/show_bug.cgi?id=729426
2027 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
2029 * gst/rtsp-server/rtsp-client.c:
2030 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
2031 Servers must handle Require headers and must report a failure
2032 if they don't handle any of the Required options, see RFC 2326,
2033 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
2034 https://bugzilla.gnome.org/show_bug.cgi?id=729426
2036 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
2041 === release 1.3.1 ===
2043 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
2049 * gst-rtsp-server.doap:
2052 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
2055 Automatic update of common submodule
2056 From bcb1518 to 211fa5f
2058 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
2063 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
2065 * tests/check/gst/sessionmedia.c:
2066 tests: fix memory leak in sessionmedia unit test
2068 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
2070 * gst/rtsp-server/rtsp-client.c:
2071 client: emit a signal before sending a message
2072 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
2074 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
2076 * gst/rtsp-server/rtsp-client.c:
2077 client: pass context to send_message
2078 Pass the current context to send_message, we will need it later.
2080 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
2082 * gst/rtsp-server/rtsp-client.c:
2083 client: fix typo in comment
2085 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
2087 * gst/rtsp-server/rtsp-media.c:
2088 media: Do not stop thread twice if default_prepare() fails
2090 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
2092 * gst/rtsp-server/rtsp-client.c:
2093 client: set the watch to flushing before going to NULL
2094 First set the watch to flushing so that we unblock any current and
2095 future attempt to send data on the watch, Then set the pipeline to
2097 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
2099 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
2101 * gst/rtsp-server/rtsp-session-pool.c:
2102 * tests/check/gst/sessionpool.c:
2103 rtsp-session-pool: Fixes annotation
2104 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
2105 in the sessionpool test.
2106 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
2108 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
2110 * gst/rtsp-server/rtsp-media.c:
2111 * gst/rtsp-server/rtsp-media.h:
2112 media: make media_prepare virtual
2113 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
2115 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
2117 * gst/rtsp-server/rtsp-media.c:
2118 * tests/check/gst/media.c:
2119 media: stop the thread in more error cases
2121 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
2123 * gst/rtsp-server/rtsp-media.c:
2124 * tests/check/gst/media.c:
2125 media: allow NULL as the thread
2126 Use the default context whan passing a NULL thread.
2128 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
2130 * gst/rtsp-server/rtsp-client.c:
2131 rtsp-client: indent cleanup
2132 Coverity was moaning about unreachable code, and I think it was just
2133 confused by { being before the label. We'll see if it pops up again.
2136 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
2138 * gst/rtsp-server/rtsp-client.c:
2139 * gst/rtsp-server/rtsp-media.c:
2140 client: Add drop-backlog property
2141 When we have too many messages queued for a client (currently hardcoded
2142 to 100) we overflow and drop the messages. Add a drop-backlog property
2143 to control this behaviour. Setting this property to FALSE will retry
2144 to send the messages to the client by waiting for more room in the
2146 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
2148 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
2150 * gst/rtsp-server/rtsp-client.c:
2151 client: support for POST before GET when setting up a tunnel
2153 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
2155 * gst/rtsp-server/rtsp-client.c:
2156 client: remove watch of the second client after http tunnel setup
2157 The second client will be freed after the HTTP tunnel has been set up.
2158 Make sure it's RTSP watch is never dispatched again.
2159 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
2161 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
2163 * gst/rtsp-server/rtsp-media.c:
2164 * tests/check/gst/media.c:
2165 media: Make media_prepare() fail if port allocation fails
2166 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
2168 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
2170 * tests/check/gst/media.c:
2171 media test: cleanup the thread pool in tests
2173 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
2175 * gst/rtsp-server/rtsp-media.c:
2176 * tests/check/gst/media.c:
2177 rtsp-media: Unblock blocked streams in unprepare
2178 The streams will be blocked when a live media is prepared.
2179 The streams should be unblocked in gst_rtsp_media_unprepare.
2180 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
2182 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
2184 * gst/rtsp-server/rtsp-media.c:
2185 media: release the state lock when going to NULL
2186 Set our state to UNPREPARING and release the state-lock before
2187 setting the pipeline to the NULL state. This way, any pad-added
2188 callback will be able to take the state-lock and check that we are now
2189 unpreparing instead of deadlocking.
2190 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
2192 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
2194 * gst/rtsp-server/rtsp-media.c:
2195 media: protect status with lock
2196 Make sure we only update the status with the lock.
2198 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
2200 * gst/rtsp-server/rtsp-client.c:
2201 * gst/rtsp-server/rtsp-sdp.c:
2202 rtsp: update for MIKEY API changes
2204 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
2206 * gst/rtsp-server/rtsp-client.c:
2207 client: parse the mikey response from the client
2208 Parse the mikey response from the client and update the policy for
2211 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
2213 * gst/rtsp-server/rtsp-stream.c:
2214 * gst/rtsp-server/rtsp-stream.h:
2215 stream: add method to set crypto info
2216 Make a method to configure the crypto information of a stream.
2217 Set udpsrc in READY instead of PAUSED so that we can configure caps
2220 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
2222 * gst/rtsp-server/rtsp-client.c:
2223 client: cleanup error paths
2225 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
2227 * gst/rtsp-server/rtsp-media.c:
2230 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
2232 * examples/test-video.c:
2233 test: enable SRTP only on RTSPS
2234 We only want to enable SRTP when doing rtsp over TLS so that we can
2235 exchange the keys in a secure way.
2237 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
2239 * examples/test-video.c:
2240 test: print an error on failure
2242 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
2245 * examples/test-video.c:
2246 * gst/rtsp-server/rtsp-sdp.c:
2247 * gst/rtsp-server/rtsp-stream.c:
2248 * tests/check/Makefile.am:
2249 stream: add SRTP support
2250 Install srtp encoder and decoder elements in rtpbin
2253 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2255 * tests/check/Makefile.am:
2256 * tests/check/gst/sessionpool.c:
2257 tests: Add unit tests for sessionpool
2258 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
2260 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2262 * tests/check/gst/threadpool.c:
2263 tests: Improve code coverage of rtsp-threadpool tests
2264 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
2266 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2268 * tests/check/gst/sessionmedia.c:
2269 tests: Improve code coverage for rtsp-session-media
2270 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
2272 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2274 gobject-introspection: Add annotations to support language bindings
2275 In addition a few cosmetic changes:
2276 * Adjust the order of arguments
2277 * Fix typo: occured -> occurred
2278 * Fix indentation after Return:-clauses
2279 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
2281 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2283 * gst/rtsp-server/rtsp-stream.c:
2284 rtsp-stream: Don't mix IPv4 and IPv6 addresses
2285 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
2287 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
2289 * gst/rtsp-server/rtsp-stream.c:
2290 stream: take caps after the session manager
2291 Take the caps for the SDP after they leave the rtpbin so that we can
2292 also get the properties added by rtpbin elements.
2294 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
2296 * gst/rtsp-server/rtsp-stream.c:
2297 stream: release lock while pushing out packets
2298 Keep a cache of the transports and use this to iterate the transport
2299 while pushing packets. This allows us to release the lock early.
2300 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
2302 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
2304 * gst/rtsp-server/rtsp-client.c:
2305 * gst/rtsp-server/rtsp-client.h:
2306 rtsp-client: vmethod for modifying tunnel GET response
2307 Add a vmethod tunnel_http_response where the response to the HTTP GET
2308 for tunneled connections can be modified.
2309 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
2311 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
2313 * gst/rtsp-server/rtsp-sdp.c:
2314 sdp: make 1 media line per profile
2315 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
2316 line in the SDP for each profile. The client is then supposed to pick
2317 one of the profiles in the SETUP request. Because the m= lines have the
2318 same pt, the client also knows that only 1 option is possible.
2320 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
2322 * gst/rtsp-server/rtsp-media-factory.c:
2323 * gst/rtsp-server/rtsp-media-factory.h:
2324 * gst/rtsp-server/rtsp-media.c:
2325 factory: add profile property and pass to media and streams
2327 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
2329 * examples/test-multicast.c:
2330 * gst/rtsp-server/rtsp-sdp.c:
2331 sdp: pass multicast connection for multicast-only stream
2332 Pass the multicast address of the stream in the connection info in the
2333 SDP so that clients try a multicast connection first.
2334 Only allow multicast connections in the test-multicast example. Also
2335 increase the TTL a little.
2337 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2340 .gitignore: Ignore gcov intermediate files
2341 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
2343 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
2345 * gst/rtsp-server/rtsp-stream.c:
2346 stream: release some locks in error cases
2348 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2350 docs: Enable and fix gtk-doc warnings
2351 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
2352 * addresspool/mediafactory: Add missing annotation colon
2353 * stream: Annotate return value
2354 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
2356 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
2359 Automatic update of common submodule
2360 From fe1672e to bcb1518
2362 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
2365 Automatic update of common submodule
2366 From 1a07da9 to fe1672e
2368 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
2370 * examples/Makefile.am:
2371 examples: use LDADD for libs instead of LDFLAGS
2373 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
2376 configure: make sure releases are in .doap file
2378 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
2380 * examples/test-cgroups.c:
2381 examples: test-cgroups: don't put code with side effects into g_assert()
2382 The g_assert() might get compiled out with the right
2383 compiler/preprocessor flags.
2385 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
2387 * examples/.gitignore:
2388 examples: add cgroup test binary to .gitignore
2390 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
2392 * examples/test-cgroups.c:
2393 examples: fix cgroup test build
2394 Fixes build failure caused by compiler warning:
2395 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
2397 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
2400 .gitignore: ignore temp files created in the course of 'make check'
2402 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
2404 * gst/rtsp-server/rtsp-media.c:
2405 rtsp-media: don't loose frames handling new PLAY request
2406 If client supplied a range check if the range specifies the start point.
2407 If not, then do an accurate seek to the current position. If a start
2408 point was specified do do a key unit seek to make sure the streaming
2409 starts with decodeable frames.
2410 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
2412 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
2414 * gst/rtsp-server/rtsp-media.c:
2415 Revert "media: only flush when setting a new start position"
2416 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
2417 We need to do the flush in all cases, demuxer block currently for
2420 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
2422 * gst/rtsp-server/rtsp-media.c:
2423 media: only flush when setting a new start position
2424 Only flush the pipeline when we change the start position with
2426 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
2428 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
2430 * gst/rtsp-server/rtsp-stream.c:
2431 stream: set ttl-mc before adding the socket
2432 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
2433 never be set on socket.
2434 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
2436 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
2438 * gst/rtsp-server/rtsp-media.c:
2439 media: stop thread if media is already prepared
2440 in gst_rtsp_media_prepare() the thread is not used if media is already
2441 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
2443 https://bugzilla.gnome.org/show_bug.cgi?id=724182
2445 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
2448 build: Ship gst-rtsp-server.doap file
2450 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
2452 * tests/check/gst/rtspserver.c:
2453 tests: Fix another compiler warning with gcc
2455 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
2457 * gst/rtsp-server/rtsp-client.c:
2458 * gst/rtsp-server/rtsp-mount-points.c:
2459 * gst/rtsp-server/rtsp-stream.c:
2460 * tests/check/gst/client.c:
2461 rtsp-server: Fix lots of compiler warnings with clang
2463 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
2466 * gst-rtsp-server.doap:
2467 * tests/Makefile.am:
2468 configure: Synchronise with the configure scripts of the other modules
2470 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
2473 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
2475 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
2477 * gst/rtsp-server/rtsp-media.c:
2478 * gst/rtsp-server/rtsp-stream.c:
2479 Revert "rtsp-server: support build against last stable release"
2480 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
2481 Let us require 1.2.3 now, which is going to be released in a few
2484 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
2486 * gst/rtsp-server/rtsp-session-media.c:
2487 * gst/rtsp-server/rtsp-stream-transport.c:
2488 session: improve RTP-Info
2489 Ignore streams that can't generate RTP-Info instead of failing.
2490 Don't return the empty string when all streams are unconfigured but
2491 return NULL so that we don't generate and empty RTP-Info header.
2492 Improve docs a little.
2494 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
2496 * gst/rtsp-server/rtsp-session-media.c:
2497 Don't free rtpinfo GString when it is NULL
2498 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
2500 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
2502 * gst/rtsp-server/rtsp-media.c:
2503 media: only set keyframe flag when modifying start
2504 Only set the keyframe flag when we modify the start position. The
2505 keyframe flag should probably be ignored when no change is requested but
2506 until we can claim this is all documented properly and all demuxer
2507 implement this, avoid setting the flag.
2508 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
2510 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
2512 * gst/rtsp-server/rtsp-thread-pool.c:
2513 thread-pool: Unref source after mainloop has quit to avoid races in GLib
2514 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
2516 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
2518 * gst/rtsp-server/rtsp-stream.c:
2519 stream: handle NULL seqnum and rtptime arguments
2521 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
2523 * gst/rtsp-server/rtsp-thread-pool.c:
2524 * tests/check/gst/threadpool.c:
2525 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
2526 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
2528 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
2530 * gst/rtsp-server/rtsp-stream.c:
2531 stream: add fallback for missing stats property
2532 Use a fallback when the payloader does not have a stats property
2533 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
2535 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
2538 Automatic update of common submodule
2539 From f7bc1c3 to 1a07da9
2541 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
2543 * gst/rtsp-server/rtsp-stream.c:
2544 stream: don't leak stats structure
2545 Don't leak the stats structure and deal with NULL stats.
2547 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
2549 * gst/rtsp-server/rtsp-stream.c:
2550 stream: Get rtpinfo properties atomically from payloader
2551 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
2553 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
2555 * gst/rtsp-server/rtsp-media.c:
2556 media: refactor state change functions and signals
2557 Make functions to set the target state and the pipeline state and emit
2558 the signals from those functions.
2560 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
2562 * gst/rtsp-server/rtsp-media.c:
2563 * gst/rtsp-server/rtsp-media.h:
2564 media: add signal to notify of pending state changes
2566 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
2568 * gst/rtsp-server/rtsp-media.c:
2569 * gst/rtsp-server/rtsp-stream.c:
2570 rtsp-server: support build against last stable release
2571 Until 1.2.3 is out with the new get_type function and we
2574 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
2576 * gst/rtsp-server/rtsp-stream.c:
2577 stream: fix compilation
2579 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
2581 * gst/rtsp-server/rtsp-media.c:
2582 * gst/rtsp-server/rtsp-media.h:
2583 * gst/rtsp-server/rtsp-stream.c:
2584 * gst/rtsp-server/rtsp-stream.h:
2585 stream: add property to configure profiles
2587 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
2589 * gst/rtsp-server/rtsp-client.c:
2590 client: let stream check supported transport
2591 Delegate the check if a transport is allowed to the stream.
2592 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
2594 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
2596 * gst/rtsp-server/rtsp-stream.c:
2597 * gst/rtsp-server/rtsp-stream.h:
2598 stream: add method to check supported transport
2599 Add a method to check if a transport is supported
2601 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
2604 configure.ac: Only check for gstreamer-check, not check
2605 We include check in gstreamer-check since quite some time now.
2607 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
2609 * gst/rtsp-server/rtsp-session-media.c:
2610 * gst/rtsp-server/rtsp-stream-transport.c:
2611 * gst/rtsp-server/rtsp-stream.c:
2612 * gst/rtsp-server/rtsp-stream.h:
2613 stream: return clock-rate from get_rtpinfo
2614 And use it to correct the rtptime to the requested start-time.
2615 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
2617 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
2619 * gst/rtsp-server/rtsp-session-media.c:
2620 * gst/rtsp-server/rtsp-stream-transport.c:
2621 * gst/rtsp-server/rtsp-stream-transport.h:
2622 session-media: calculate start-time
2624 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
2626 * gst/rtsp-server/rtsp-stream-transport.c:
2627 * gst/rtsp-server/rtsp-stream.c:
2628 * gst/rtsp-server/rtsp-stream.h:
2629 stream: also return the running-time
2630 Return the running-time in the rtpinfo as well.
2632 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
2634 * gst/rtsp-server/rtsp-client.c:
2635 * gst/rtsp-server/rtsp-session-media.c:
2636 * gst/rtsp-server/rtsp-session-media.h:
2637 * gst/rtsp-server/rtsp-stream-transport.c:
2638 * gst/rtsp-server/rtsp-stream-transport.h:
2639 session-media: let the session-media make the RTPInfo
2640 Add method to create the RTPInfo for a stream-transport.
2641 Add method to create the RTPInfo for all stream-transports in a
2643 Use the session-media RTPInfo code in client. This allows us to refactor
2644 another method to link the TCP callbacks.
2646 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
2648 mount-points: sort sequence before g_sequence_lookup
2649 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
2650 sort sequence if dirty, otherwise lookup will fail.
2651 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
2653 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
2656 configure: rename package from gst-rtsp to gst-rtsp-server
2657 To match git module name and avoid confusion with the
2658 rtsp lib in gst-plugins-base and rtsp plugin in -good.
2660 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
2663 configure: bump core/base/good requirement to 1.2.0
2664 Bump to released stable version and make implicit
2665 requirements explicit.
2667 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
2672 Fix broken gettext setup which is not used anyway
2674 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
2677 Automatic update of common submodule
2678 From dbedaa0 to d48bed3
2680 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
2682 * gst/rtsp-server/rtsp-client.c:
2683 * gst/rtsp-server/rtsp-media.c:
2684 * gst/rtsp-server/rtsp-media.h:
2685 media: add setup_sdp vmethod
2686 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
2687 gst_rtsp_media_setup_sdp.
2688 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
2690 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
2692 * gst/rtsp-server/rtsp-stream.c:
2693 rtsp-stream: Check return value of sscanf
2694 streamid is only valid if sscanf matched something.
2696 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
2698 * gst/rtsp-server/rtsp-client.c:
2699 rtsp-client: Fix iteration
2700 Wouldn't even enter the code block otherwise (i++ was used as the check
2701 and not the postfix).
2703 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
2705 * gst/rtsp-server/rtsp-client.c:
2706 * gst/rtsp-server/rtsp-client.h:
2707 client: add vmethod to configure media and streams
2708 Implement a vmethod that can be used to configure the media and the
2709 streams based on the current context. Handle the blocksize handling in
2710 the default handler.
2711 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
2713 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
2716 Make git ignore more unit test binaries
2718 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
2720 * gst/rtsp-server/rtsp-address-pool.h:
2721 * gst/rtsp-server/rtsp-auth.h:
2722 * gst/rtsp-server/rtsp-client.h:
2723 * gst/rtsp-server/rtsp-context.h:
2724 * gst/rtsp-server/rtsp-media-factory-uri.h:
2725 * gst/rtsp-server/rtsp-media-factory.h:
2726 * gst/rtsp-server/rtsp-media.h:
2727 * gst/rtsp-server/rtsp-mount-points.h:
2728 * gst/rtsp-server/rtsp-server.h:
2729 * gst/rtsp-server/rtsp-session-media.h:
2730 * gst/rtsp-server/rtsp-session-pool.h:
2731 * gst/rtsp-server/rtsp-session.h:
2732 * gst/rtsp-server/rtsp-stream-transport.h:
2733 * gst/rtsp-server/rtsp-stream.h:
2734 * gst/rtsp-server/rtsp-thread-pool.h:
2735 * gst/rtsp-server/rtsp-token.h:
2736 rtsp-server: add padding to many public structures
2737 Not mini objects though, since they are not subclassable
2738 anyway, nor kept on the stack or inlined in a structure.
2740 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
2742 media: add new create_rtpbin vmethod
2743 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
2744 https://bugzilla.gnome.org/show_bug.cgi?id=719734
2746 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
2748 * tests/check/gst/media.c:
2749 tests: fix memory leak, free test's thread pool
2750 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
2752 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
2754 * gst/rtsp-server/rtsp-stream-transport.c:
2755 stream-transport: free url in finalize
2757 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
2759 * gst/rtsp-server/rtsp-media.c:
2760 media: also do state change in suspended state
2762 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
2764 * gst/rtsp-server/rtsp-client.c:
2765 * gst/rtsp-server/rtsp-media.c:
2766 media: also handle prepare and range in suspended state
2767 When we are suspended, we are already prepared.
2768 We can get the range in the suspended state.
2770 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
2772 * tests/check/Makefile.am:
2773 * tests/check/gst/sessionmedia.c:
2774 check: add test for uri in setup
2775 Added unit tests for the new functionality in GstRTSPStreamTransport.
2776 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
2778 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
2780 * gst/rtsp-server/rtsp-client.c:
2781 client: store setup uri and use in PLAY response
2782 Store the uri used when doing the setup and use that in the PLAY
2784 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
2786 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
2788 * gst/rtsp-server/rtsp-stream-transport.c:
2789 * gst/rtsp-server/rtsp-stream-transport.h:
2790 stream-transport: add method to get/set url
2792 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
2794 * gst/rtsp-server/rtsp-client.c:
2795 client: suspend after SDP and unsuspend before PLAYING
2796 Based on patches by Ognyan Tonchev <ognyan@axis.com>
2797 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
2799 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
2801 * gst/rtsp-server/rtsp-media-factory.c:
2802 * gst/rtsp-server/rtsp-media-factory.h:
2803 * gst/rtsp-server/rtsp-media.c:
2804 * gst/rtsp-server/rtsp-media.h:
2805 * gst/rtsp-server/rtsp-session-media.c:
2806 * gst/rtsp-server/rtsp-session.c:
2807 * tests/check/gst/media.c:
2808 * tests/check/gst/mediafactory.c:
2809 media: add suspend modes
2810 Add support for different suspend modes. The stream is suspended right after
2811 producing the SDP and after PAUSE. Different suspend modes are available that
2812 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
2813 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
2814 state and RESET will bring the pipeline to the NULL state.
2815 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
2816 this means that the pipeline needs to be prerolled again.
2817 Base on patches by Ognyan Tonchev <ognyan@axis.com>
2818 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2820 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
2822 * gst/rtsp-server/rtsp-media.c:
2823 media: start live streams in blocked state
2824 Start live streams in the blocked state and make them preroll using the
2825 messages. This ensure that no data is played by the sink until we explicitly
2826 unblock the stream right before going to PLAYING.
2827 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2829 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
2831 * gst/rtsp-server/rtsp-media.c:
2832 media: refactor starting and waiting for preroll
2833 Based on patches from Ognyan Tonchev <ognyan@axis.com>
2834 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2836 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
2838 * gst/rtsp-server/rtsp-stream.c:
2839 * gst/rtsp-server/rtsp-stream.h:
2840 stream: add API to block streams
2841 Add an API to block on the streams and make it post a message.
2842 Based on patch by Ognyan Tonchev <ognyan@axis.com>
2843 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2845 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
2847 * docs/libs/Makefile.am:
2848 docs: Specify the override file
2849 Even if it's empty (for now) it avoids make distcheck complaining
2851 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
2853 * gst/rtsp-server/rtsp-media.c:
2854 media: move default implementations to where they are used
2856 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
2858 * gst/rtsp-server/rtsp-media.c:
2859 media: take the right lock in gst_rtsp_media_set_pipeline_state()
2860 We need to take the state_lock when calling this method.
2862 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
2864 * gst/rtsp-server/rtsp-media.c:
2865 media: handle add-added on non-bins too
2866 Handle dynamic payloaders that are not bins, as used in the unit-test.
2868 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2870 * gst/rtsp-server/rtsp-media-factory.c:
2871 * gst/rtsp-server/rtsp-media-factory.h:
2872 * gst/rtsp-server/rtsp-media.c:
2873 rtsp-media/-factory: Fix request pad name comments
2874 These must be escaped for gtk-doc to parse the comments without warnings.
2876 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
2878 rtsp-media: remove transports if media is in error status
2879 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
2880 trying to change to GST_STATE_NULL and media is in error status, we
2881 remove all transports.
2882 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
2884 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
2886 * gst/rtsp-server/rtsp-media.c:
2887 rtsp-media: use element metadata to find payloader
2888 Use the element metadata to find the payloader instead of checking
2890 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
2892 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
2894 rtsp-stream: add getter for payload type
2895 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
2896 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
2897 element and create the stream with this one instead of the dynpay%d
2899 https://bugzilla.gnome.org/show_bug.cgi?id=712396
2901 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2903 * gst/rtsp-server/rtsp-client.c:
2904 * gst/rtsp-server/rtsp-context.h:
2905 * gst/rtsp-server/rtsp-media.c:
2906 * gst/rtsp-server/rtsp-mount-points.c:
2907 * gst/rtsp-server/rtsp-server.c:
2908 * gst/rtsp-server/rtsp-token.c:
2909 rtsp-*: Refer to NULL as a constant in comments
2911 https://bugzilla.gnome.org/show_bug.cgi?id=714988
2913 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2915 rtsp-*: Fix type name typos in comments
2916 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
2917 * rtsp-auth: Refer to part of constant name as text
2918 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
2919 * rtsp-session-media: Fix GstRTSPSessionMedia typo
2920 * rtsp-stream: Fix typo when refering to GstBin
2921 https://bugzilla.gnome.org/show_bug.cgi?id=714988
2923 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2926 * docs/libs/gst-rtsp-server-docs.sgml:
2927 * docs/libs/gst-rtsp-server-sections.txt:
2928 docs: Improve documentation
2929 * Include annotation-glossary to quiet gtk-doc
2930 * Rename remaining ClientState -> Context
2931 * Rename object hierarchy file
2932 * Remove stale chapter references
2933 * Add missing function and object references
2934 * Include missing GstRTSPAddressPoolResult
2935 https://bugzilla.gnome.org/show_bug.cgi?id=714988
2937 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
2939 * gst/rtsp-server/rtsp-client.c:
2940 * gst/rtsp-server/rtsp-server.c:
2941 * gst/rtsp-server/rtsp-session-pool.c:
2942 * gst/rtsp-server/rtsp-session.c:
2943 * gst/rtsp-server/rtsp-stream.c:
2944 rtsp-server: sprinkle some allow-none annotations for g-i
2946 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
2948 * gst/rtsp-server/rtsp-stream.c:
2949 * gst/rtsp-server/rtsp-stream.h:
2950 stream: add method to filter transports
2951 Add a method to safely iterate and collect the stream transports
2952 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
2954 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
2956 * gst/rtsp-server/rtsp-client.c:
2957 * gst/rtsp-server/rtsp-server.c:
2958 * gst/rtsp-server/rtsp-session-pool.c:
2959 * gst/rtsp-server/rtsp-session.c:
2960 rtsp: allow NULL func in filters
2961 Passing a null function make the filters return a list of
2964 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
2966 * gst/rtsp-server/rtsp-address-pool.c:
2967 * tests/check/gst/addresspool.c:
2968 address-pool: fix address increment
2969 Use a guint instead of guint8 to increment the address. It's still not
2970 completely correct because a guint might not be able to hold the complete
2971 address range, but that's an enhacement for later.
2972 Add unit test to test improved behaviour.
2973 https://bugzilla.gnome.org/show_bug.cgi?id=708237
2975 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
2977 * gst/rtsp-server/rtsp-client.c:
2978 * tests/check/gst/client.c:
2979 client: allow absolute path in requests
2980 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
2982 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
2984 * gst/rtsp-server/rtsp-client.c:
2985 * gst/rtsp-server/rtsp-client.h:
2986 client: make make_path_from_uri a vmethod
2988 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
2990 * docs/libs/gst-rtsp-server-sections.txt:
2991 * gst/rtsp-server/rtsp-stream.c:
2992 * gst/rtsp-server/rtsp-stream.h:
2993 * tests/check/Makefile.am:
2994 * tests/check/gst/stream.c:
2995 stream: Add functions to get rtp and rtcp sockets
2996 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
2998 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
3000 * gst/rtsp-server/rtsp-context.c:
3001 * gst/rtsp-server/rtsp-context.h:
3002 context: defing a GType for the context
3003 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
3005 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
3007 * gst/rtsp-server/Makefile.am:
3008 * gst/rtsp-server/rtsp-auth.c:
3009 * gst/rtsp-server/rtsp-context.c:
3010 * gst/rtsp-server/rtsp-media.c:
3011 * gst/rtsp-server/rtsp-mount-points.c:
3012 * gst/rtsp-server/rtsp-server.h:
3013 * gst/rtsp-server/rtsp-session-media.c:
3014 * gst/rtsp-server/rtsp-session.c:
3015 * gst/rtsp-server/rtsp-stream.c:
3016 Fixed several GIR warnings
3018 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
3020 * gst/rtsp-server/rtsp-auth.c:
3023 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3025 * tests/check/Makefile.am:
3026 * tests/check/gst/token.c:
3027 tests: Add unit tests for token
3028 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
3030 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3032 * gst/rtsp-server/rtsp-token.c:
3033 token: Validate args for gst_rtsp_token_is_allowed
3034 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
3036 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3038 * gst/rtsp-server/rtsp-token.c:
3039 token: Fix bug when creating empty token
3040 We always want to have a valid GstStructure in the token.
3041 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
3043 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
3045 * gst/rtsp-server/rtsp-thread-pool.c:
3046 thread-pool: avoid race in shutdown
3047 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
3048 don't actually stop the mainloop ever. Solve this race by adding an idle source
3049 to the mainloop that calls the _quit. This way we immediately exit the mainloop
3050 if quit was called before we started it.
3052 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3054 * tests/check/Makefile.am:
3055 * tests/check/gst/permissions.c:
3056 tests: Add unit tests for permissions
3057 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
3059 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3061 * tests/check/gst/mediafactory.c:
3062 tests: Test mediafactory permissions
3063 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
3065 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3067 * gst/rtsp-server/rtsp-permissions.c:
3068 permissions: Fix refcounting when adding/removing roles
3069 Previously a role that was removed was unreffed twice, and when
3070 replacing an existing role the replaced role was freed while still being
3071 referenced. Both bugs are now fixed.
3072 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
3074 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3076 * tests/check/gst/media.c:
3077 * tests/check/gst/mediafactory.c:
3078 * tests/check/gst/rtspserver.c:
3079 tests: Check gst_rtsp_url_parse return value
3080 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
3082 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
3085 Automatic update of common submodule
3086 From 865aa20 to dbedaa0
3088 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
3090 * gst/rtsp-server/rtsp-server.c:
3091 rtsp-server: Fix socket leak
3092 https://bugzilla.gnome.org/show_bug.cgi?id=710088
3094 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
3096 * gst/rtsp-server/rtsp-session-pool.c:
3097 rtsp-session-pool: Make sure session IDs are properly URI-escaped
3098 https://bugzilla.gnome.org/show_bug.cgi?id=643812
3100 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
3102 * examples/.gitignore:
3103 * examples/test-video.c:
3104 examples: fix compilation when WITH_AUTH is defined
3105 https://bugzilla.gnome.org/show_bug.cgi?id=710228
3107 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
3110 gitignore: Add new test binary
3112 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
3114 * tests/check/Makefile.am:
3115 * tests/check/gst/threadpool.c:
3116 thread-pool: Add unit test for the thread pools
3117 https://bugzilla.gnome.org/show_bug.cgi?id=710228
3119 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
3121 * gst/rtsp-server/rtsp-thread-pool.c:
3122 thread-pool: Fix thread leak when reusing threads
3123 https://bugzilla.gnome.org/show_bug.cgi?id=709730
3125 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
3127 * gst/rtsp-server/rtsp-server.c:
3128 * tests/check/gst/rtspserver.c:
3129 tests: fixed racy behavior in rtspserver tests
3130 https://bugzilla.gnome.org/show_bug.cgi?id=710078
3132 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3134 * tests/check/gst/addresspool.c:
3135 tests: Improve address pool unit tests
3136 Add a range with mixed IPV4 and IPV6 addresses to pool.
3137 Get an IPV4 address from an IPV6-only pool.
3138 Get an IPV6 address from an IPV4-only pool.
3139 Reserve a IPV6 address from an IPV4-only pool.
3140 Check for unicast addresses in multicast-only pool.
3141 Check for unicast addresses in uni-/multicast-mixed pool.
3142 https://bugzilla.gnome.org/show_bug.cgi?id=710128
3144 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3146 * gst/rtsp-server/rtsp-client.c:
3147 client: append query string in PAUSE/PLAY/TEARDOWN as well
3149 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
3151 * gst/rtsp-server/rtsp-client.c:
3152 client: Add query to control path
3153 If the SETUP url contains a query it must be appended to the control
3154 path so that it matches any already created stream in the media. The
3155 query will also be appended to the session media path.
3157 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3159 * gst/rtsp-server/rtsp-media.c:
3160 rtsp-media: remove old line
3162 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
3164 * gst/rtsp-server/rtsp-stream.c:
3165 stream: Correct control comparison
3166 https://bugzilla.gnome.org/show_bug.cgi?id=709176
3168 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3170 * gst/rtsp-server/rtsp-media.c:
3171 media: Check dynamically if the pipeline supports seeking
3172 We should not depend on whether or not the pipeline state change
3173 returned NO_PREROLL or not. A media could dynamically change its
3174 element and switch from seekable to non seekable so it's best to test
3175 the seekable nature of the pipeline dynamically when we try to do a seek.
3177 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3179 * gst/rtsp-server/rtsp-media.c:
3180 media: Return FALSE if seeking is not supported
3182 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3184 * gst/rtsp-server/rtsp-media.c:
3185 rtsp-media: don't seek accurate by default
3186 Accurate seeking is perhaps a little overkill in the most common situation and
3187 causes some formats (mp3) over slow media to seek extremely slowly.
3189 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
3191 * tests/check/gst/rtspserver.c:
3192 tests: fix unit test
3193 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
3195 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
3197 * gst/rtsp-server/rtsp-client.c:
3198 client: Reply 400 if media cannot be constructed
3199 Reply 400 Bad Request instead of 503 Service Unavailable if media
3200 cannot be constructed in SETUP.
3201 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
3203 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
3205 * gst/rtsp-server/rtsp-client.c:
3206 client: Send setup reply once only
3207 If find_media() failed in handle_setup_request() two replies was sent.
3208 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
3210 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
3213 Automatic update of common submodule
3214 From 6b03ba7 to 865aa20
3216 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
3218 * gst/rtsp-server/rtsp-server.c:
3219 server: Emit client-connected signal earlier
3220 Emit client-connected before the client ref is given to a GSource,
3221 otherwise client-connected can be emitted after the client object has
3224 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
3226 * gst/rtsp-server/rtsp-address-pool.c:
3227 * gst/rtsp-server/rtsp-address-pool.h:
3228 * gst/rtsp-server/rtsp-stream.c:
3229 * tests/check/gst/addresspool.c:
3230 addresspool: return reason of failure
3231 Let gst_rtsp_address_pool_reserve_address() return the reason why
3232 the address could not be reserved.
3233 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
3235 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
3238 autogen.sh: Sync behaviour with other GStreamer modules
3239 Allows building from outside of tree amongst other things
3241 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
3244 Automatic update of common submodule
3245 From b613661 to 6b03ba7
3247 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
3250 Automatic update of common submodule
3251 From 74a6857 to b613661
3253 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
3256 Automatic update of common submodule
3257 From 01a7a46 to 74a6857
3259 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
3261 * gst/rtsp-server/rtsp-client.c:
3262 client: Do not read beyond end of path string
3263 If the setup was done without a control url, make sure we don't try to read the
3264 non-existing control string and crash.
3266 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3268 * gst/rtsp-server/rtsp-client.c:
3269 client: Fix RTPInfo header
3270 Refactor the method to make the content_base.
3271 Use the content-base and the control url to construct the RTPInfo
3274 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3276 * gst/rtsp-server/rtsp-client.c:
3277 client: map url to path only in describe
3278 Only map the request url to a path in the DESCRIBE method. The SDP then
3279 contains the base and control urls that should be used to SETUP/PAUSE/
3280 PLAY/TEARDOWN the media.
3282 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3284 * gst/rtsp-server/rtsp-client.c:
3285 Revert "client: map URL to path in requests"
3286 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
3287 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
3288 contains the base and control urls which are used in the SETUP, PLAY,
3289 PAUSE and TEARDOWN requests.
3291 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3293 * gst/rtsp-server/rtsp-client.c:
3294 client: map URL to path in requests
3296 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3298 * gst/rtsp-server/rtsp-client.c:
3299 * gst/rtsp-server/rtsp-mount-points.c:
3300 * gst/rtsp-server/rtsp-mount-points.h:
3301 mount-points: make vmethod to make path from uri
3302 Make a vmethod to transform an url into a path. The path is then used to lookup
3303 the factory. This makes it possible to also use other bits of the url, such as
3304 the query parameters, to locate the factory.
3306 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
3308 * gst/rtsp-server/rtsp-thread-pool.c:
3309 * gst/rtsp-server/rtsp-thread-pool.h:
3310 thread-pool: Add cleanup to wait for the threadpool to finish
3311 Also fix race condition if two threads are asking for the first
3312 thread from the thread pool at once. This would case two internal
3313 GThreadPools to be created.
3314 https://bugzilla.gnome.org/show_bug.cgi?id=707753
3316 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
3318 * gst/rtsp-server/rtsp-client.c:
3319 * tests/check/gst/client.c:
3320 client: free threadpool
3321 https://bugzilla.gnome.org/show_bug.cgi?id=707638
3323 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
3325 * tests/check/gst/mountpoints.c:
3326 mountpoints tests: unref matched factories
3327 https://bugzilla.gnome.org/show_bug.cgi?id=707638
3329 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
3331 * tests/check/gst/media.c:
3332 media tests: unref thread pool and caps
3333 https://bugzilla.gnome.org/show_bug.cgi?id=707638
3335 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
3337 * gst/rtsp-server/rtsp-auth.c:
3338 * gst/rtsp-server/rtsp-media-factory.c:
3339 * gst/rtsp-server/rtsp-media.c:
3340 auth, media, media-factory: unref permissions
3341 https://bugzilla.gnome.org/show_bug.cgi?id=707638
3343 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3345 * examples/Makefile.am:
3346 Makefile: add rule for appsrc example
3348 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3350 * examples/test-appsrc.c:
3351 tests: add appsrc example
3352 Add an example on how to use appsrc to feed the server pipeline with data.
3354 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
3356 * gst/rtsp-server/rtsp-client.c:
3357 rtsp-client: remove query part from content-base string
3358 Make sure that after the control url has been resolved, it's
3359 not a part of the query-string.
3360 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
3362 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3364 * gst/rtsp-server/rtsp-client.c:
3365 client: don't check url in response
3366 There is no url or method in the response to check
3368 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3370 * gst/rtsp-server/rtsp-client.c:
3371 * gst/rtsp-server/rtsp-client.h:
3372 Add handle-response signal for when we receive a GET_PARAMETER response
3374 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3376 * gst/rtsp-server/rtsp-server.c:
3377 Fix gst_rtsp_server_client_filter, using wrong variable type
3379 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
3381 * gst/rtsp-server/rtsp-media-factory-uri.c:
3382 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
3383 For AAC we need to check for framed=true instead of parsed=true.
3384 https://bugzilla.gnome.org/show_bug.cgi?id=701384
3386 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3388 * gst/rtsp-server/rtsp-stream.c:
3389 stream: optimize pipeline for protocols
3390 When TCP is not an allowed protocol for the stream, avoid creating the
3391 appsrc/appsink/queue and tee elements.
3393 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3395 * gst/rtsp-server/rtsp-media.c:
3396 media: set protocols on streams
3398 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3400 * gst/rtsp-server/rtsp-client.c:
3401 client: use protocols supported by stream
3403 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3405 * gst/rtsp-server/rtsp-media-factory.c:
3406 * gst/rtsp-server/rtsp-media.c:
3407 * gst/rtsp-server/rtsp-stream.c:
3408 media-factory: allow all protocols
3410 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3412 * gst/rtsp-server/rtsp-media.c:
3413 media: configure protocols in new streams
3415 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3417 * gst/rtsp-server/rtsp-stream.c:
3418 * gst/rtsp-server/rtsp-stream.h:
3419 stream: add protocols property
3421 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3423 * gst/rtsp-server/rtsp-media.c:
3424 rtsp-media: send state in "new-state" signal
3425 https://bugzilla.gnome.org/show_bug.cgi?id=705110
3427 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
3430 build: add subdir-objects to AM_INIT_AUTOMAKE
3431 Fixes warnings with automake 1.14
3432 https://bugzilla.gnome.org/show_bug.cgi?id=705350
3434 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3436 * docs/libs/gst-rtsp-server-sections.txt:
3437 * gst/rtsp-server/rtsp-client.c:
3438 * gst/rtsp-server/rtsp-server.c:
3439 * gst/rtsp-server/rtsp-server.h:
3440 server: add method to iterate clients of server
3442 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3444 * gst/rtsp-server/rtsp-media.c:
3445 * gst/rtsp-server/rtsp-media.h:
3446 Add vmethod for rtsp-media subclass to access rtpbin
3448 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3450 * gst/rtsp-server/rtsp-client.h:
3451 small documentation fix
3453 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3455 * gst/rtsp-server/rtsp-client.c:
3456 Do not take range header if range is invalid
3458 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3460 * docs/libs/gst-rtsp-server-sections.txt:
3461 * gst/rtsp-server/rtsp-media.c:
3462 media: add docs for new method
3464 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3466 * gst/rtsp-server/rtsp-media.c:
3467 * gst/rtsp-server/rtsp-media.h:
3468 Add API to rtsp-media set the pipeline's state
3470 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3472 * gst/rtsp-server/rtsp-media.c:
3473 Update current position/duration when gst_rtsp_media_get_range_string is called
3475 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3477 * examples/test-cgroups.c:
3478 tests: add some more docs
3480 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3482 * examples/test-cgroups.c:
3483 * gst/rtsp-server/Makefile.am:
3484 * gst/rtsp-server/rtsp-auth.c:
3485 * gst/rtsp-server/rtsp-auth.h:
3486 * gst/rtsp-server/rtsp-client.c:
3487 * gst/rtsp-server/rtsp-client.h:
3488 * gst/rtsp-server/rtsp-context.c:
3489 * gst/rtsp-server/rtsp-context.h:
3490 * gst/rtsp-server/rtsp-params.c:
3491 * gst/rtsp-server/rtsp-params.h:
3492 * gst/rtsp-server/rtsp-server.c:
3493 * gst/rtsp-server/rtsp-thread-pool.c:
3494 * gst/rtsp-server/rtsp-thread-pool.h:
3495 * tests/check/gst/client.c:
3496 ClientState -> Context
3497 Rename the clientstate to context and put the code in a separate file.
3499 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3501 * examples/test-auth.c:
3502 * gst/rtsp-server/rtsp-auth.c:
3503 * gst/rtsp-server/rtsp-auth.h:
3504 auth: add support for default token
3505 The default token is used when the user is not authenticated and can be used to
3506 give minimal permissions.
3508 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3510 * examples/test-auth.c:
3511 * gst/rtsp-server/rtsp-auth.c:
3512 auth: use defines when possible
3514 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3516 * gst/rtsp-server/rtsp-address-pool.c:
3517 address-pool: improve docs
3519 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3521 * gst/rtsp-server/rtsp-permissions.c:
3522 permissions: add the role to the copy
3524 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
3526 * gst/rtsp-server/rtsp-permissions.c:
3527 permissions: Also copy the roles
3529 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
3531 * gst/rtsp-server/rtsp-permissions.c:
3532 permissions: Make it build
3534 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3536 * gst/rtsp-server/rtsp-address-pool.h:
3539 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3541 * docs/libs/gst-rtsp-server-sections.txt:
3542 * gst/rtsp-server/rtsp-auth.c:
3543 * gst/rtsp-server/rtsp-auth.h:
3544 * gst/rtsp-server/rtsp-media.c:
3545 * gst/rtsp-server/rtsp-session-media.c:
3546 * gst/rtsp-server/rtsp-stream-transport.c:
3547 * gst/rtsp-server/rtsp-stream-transport.h:
3548 * gst/rtsp-server/rtsp-stream.c:
3549 * tests/check/gst/client.c:
3552 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3554 * docs/libs/gst-rtsp-server-sections.txt:
3555 * gst/rtsp-server/rtsp-address-pool.c:
3556 * gst/rtsp-server/rtsp-address-pool.h:
3557 * tests/check/gst/addresspool.c:
3558 * tests/check/gst/rtspserver.c:
3559 address-pool: cleanups
3560 Remove redundant method, improve docs.
3562 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3564 * docs/libs/gst-rtsp-server-sections.txt:
3565 * gst/rtsp-server/rtsp-auth.h:
3566 * gst/rtsp-server/rtsp-permissions.c:
3567 * gst/rtsp-server/rtsp-permissions.h:
3568 * gst/rtsp-server/rtsp-token.c:
3571 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3573 * gst/rtsp-server/rtsp-permissions.c:
3574 permissions: implement _remove_role
3576 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3578 * gst/rtsp-server/rtsp-permissions.c:
3579 permissions: update docs
3581 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3583 * tests/check/gst/client.c:
3584 tests: simplify tests
3585 Client settings are now disabled by default so we don't need an auth
3586 module to disable them.
3588 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3590 * gst/rtsp-server/rtsp-auth.c:
3591 auth: add default authorizations
3592 When no auth module is specified, use our table of defaults to look up the
3593 default value of the check instead of always allowing everything. This was
3594 we can disallow client settings by default.
3596 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3599 README: update readme
3601 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3603 * gst/rtsp-server/rtsp-thread-pool.c:
3604 * gst/rtsp-server/rtsp-thread-pool.h:
3605 thread-pool: add more docs
3607 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3609 * gst/rtsp-server/rtsp-thread-pool.c:
3610 * gst/rtsp-server/rtsp-thread-pool.h:
3611 thread-pool: fix race in thread reuse
3612 If we try to reuse a thread right after we made it stop, we end up using a
3613 stopped thread. Catch this case and only reuse threads that are not stopping.
3615 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3617 * gst/rtsp-server/rtsp-server.c:
3618 server: add small debug
3620 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3622 * tests/check/gst/client.c:
3624 Add some permissions to media so we can use the auth and enable
3627 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3629 * gst/rtsp-server/rtsp-client.c:
3630 client: support pushed context in handle_request
3631 If we already have a pushed state, reuse it and add our own things. This makes
3632 it easier to write tests.
3634 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3636 * gst/rtsp-server/rtsp-auth.c:
3637 auth: don't auth on methods
3638 Don't authorize on methods anymore but on the resources that we
3639 try to access, this is more flexible.
3640 Move the authorization checks to where they are needed and let the
3641 check return the response on error.
3643 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3645 * gst/rtsp-server/rtsp-mount-points.c:
3646 mount-points: add some debug
3648 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3650 * tests/check/gst/client.c:
3651 tests: almost fix test
3653 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3655 * gst/rtsp-server/rtsp-auth.c:
3656 * gst/rtsp-server/rtsp-auth.h:
3657 * gst/rtsp-server/rtsp-client.c:
3658 * gst/rtsp-server/rtsp-client.h:
3659 * gst/rtsp-server/rtsp-server.c:
3660 * gst/rtsp-server/rtsp-server.h:
3661 auth: let the auth module check client_settings
3662 Let the auth module decide if client settings are allowed for the
3665 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3667 * gst/rtsp-server/rtsp-token.c:
3668 * gst/rtsp-server/rtsp-token.h:
3669 token: add method to check boolean permission
3671 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3673 * examples/test-auth.c:
3674 * examples/test-cgroups.c:
3675 * gst/rtsp-server/rtsp-token.c:
3676 * gst/rtsp-server/rtsp-token.h:
3677 token: simplify token constructor
3678 Use variable arguments to make easier API.
3680 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3682 * examples/test-auth.c:
3683 * examples/test-cgroups.c:
3684 * gst/rtsp-server/rtsp-media-factory.c:
3685 * gst/rtsp-server/rtsp-media-factory.h:
3686 media-factory: add convenience API for factory
3688 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3690 * examples/test-auth.c:
3691 * examples/test-cgroups.c:
3692 * gst/rtsp-server/rtsp-permissions.c:
3693 * gst/rtsp-server/rtsp-permissions.h:
3694 permissions: simplify API a little
3695 Avoid passing GstStructure in the add_role method, use varargs instead
3696 to construct the structure behind the scenes. We can then also use the
3697 structure name as the role and simplify some more logic.
3699 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3701 * gst/rtsp-server/rtsp-auth.c:
3704 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3706 * gst/rtsp-server/rtsp-auth.c:
3707 * gst/rtsp-server/rtsp-auth.h:
3708 * gst/rtsp-server/rtsp-client.c:
3709 auth: handle unauthorized response
3710 Move handling of the unauthorized response to the auth module, it can add
3711 the appropriate headers to request authorization for the required method
3712 much better than the client.
3714 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3716 * gst/rtsp-server/rtsp-client.c:
3717 * gst/rtsp-server/rtsp-client.h:
3718 client: allow for sending any message, not only requests
3719 Change the _send_request() method to _send_message() so that we
3720 can both send requests and replies.
3722 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3724 * docs/libs/gst-rtsp-server-sections.txt:
3725 * gst/rtsp-server/rtsp-server.h:
3728 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3730 * examples/test-video.c:
3731 * gst/rtsp-server/rtsp-auth.c:
3732 * gst/rtsp-server/rtsp-auth.h:
3733 * gst/rtsp-server/rtsp-server.c:
3734 * gst/rtsp-server/rtsp-server.h:
3735 auth: move TLS handling to auth module
3736 Remove the TLS settings on the server and move it to the auth module because
3737 that is where security related bits go.
3739 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3741 * gst/rtsp-server/rtsp-client.c:
3742 * gst/rtsp-server/rtsp-client.h:
3743 client: add state push/pop
3745 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3747 * gst/rtsp-server/rtsp-client.c:
3748 * gst/rtsp-server/rtsp-client.h:
3749 client: add connection to state
3751 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3753 * gst/rtsp-server/rtsp-mount-points.c:
3754 mount-points: fix debug
3756 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3758 * tests/check/gst/media.c:
3759 tests: fix media test
3761 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3763 * gst/rtsp-server/rtsp-thread-pool.c:
3764 thread-pool: we don't require a state
3766 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3768 * gst/rtsp-server/rtsp-server.c:
3769 server: let context ref the server
3770 So that we don't risk losing the server object early anc crash.
3772 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3774 * tests/check/gst/client.c:
3775 tests: fix client test
3777 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3780 * docs/libs/gst-rtsp-server-docs.sgml:
3781 * docs/libs/gst-rtsp-server-sections.txt:
3782 * gst/rtsp-server/rtsp-address-pool.c:
3783 * gst/rtsp-server/rtsp-auth.c:
3784 * gst/rtsp-server/rtsp-client.c:
3785 * gst/rtsp-server/rtsp-client.h:
3786 * gst/rtsp-server/rtsp-media-factory-uri.c:
3787 * gst/rtsp-server/rtsp-media-factory.c:
3788 * gst/rtsp-server/rtsp-media-factory.h:
3789 * gst/rtsp-server/rtsp-media.c:
3790 * gst/rtsp-server/rtsp-mount-points.c:
3791 * gst/rtsp-server/rtsp-params.c:
3792 * gst/rtsp-server/rtsp-permissions.c:
3793 * gst/rtsp-server/rtsp-sdp.c:
3794 * gst/rtsp-server/rtsp-server.c:
3795 * gst/rtsp-server/rtsp-server.h:
3796 * gst/rtsp-server/rtsp-session-media.c:
3797 * gst/rtsp-server/rtsp-session-pool.c:
3798 * gst/rtsp-server/rtsp-session.c:
3799 * gst/rtsp-server/rtsp-stream-transport.c:
3800 * gst/rtsp-server/rtsp-stream.c:
3801 * gst/rtsp-server/rtsp-thread-pool.c:
3802 * gst/rtsp-server/rtsp-token.c:
3805 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3807 * gst/rtsp-server/rtsp-session-pool.c:
3808 * gst/rtsp-server/rtsp-session-pool.h:
3809 session-pool: make vmethod to create a session
3810 Make a vmethod to create a sessions so that subclasses can create
3811 custom session objects
3813 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3815 * gst/rtsp-server/rtsp-auth.c:
3816 * gst/rtsp-server/rtsp-media-factory.h:
3817 * gst/rtsp-server/rtsp-media.h:
3818 * gst/rtsp-server/rtsp-mount-points.h:
3819 * gst/rtsp-server/rtsp-session-pool.h:
3820 * gst/rtsp-server/rtsp-stream.h:
3823 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3825 * docs/libs/gst-rtsp-server-docs.sgml:
3826 * docs/libs/gst-rtsp-server-sections.txt:
3827 * gst/rtsp-server/rtsp-address-pool.c:
3828 * gst/rtsp-server/rtsp-address-pool.h:
3829 * gst/rtsp-server/rtsp-auth.c:
3830 * gst/rtsp-server/rtsp-client.h:
3831 * gst/rtsp-server/rtsp-media-factory.h:
3832 * gst/rtsp-server/rtsp-media.c:
3833 * gst/rtsp-server/rtsp-media.h:
3834 * gst/rtsp-server/rtsp-permissions.c:
3835 * gst/rtsp-server/rtsp-permissions.h:
3836 * gst/rtsp-server/rtsp-server.h:
3837 * gst/rtsp-server/rtsp-session-media.c:
3838 * gst/rtsp-server/rtsp-session-media.h:
3839 * gst/rtsp-server/rtsp-session-pool.h:
3840 * gst/rtsp-server/rtsp-session.h:
3841 * gst/rtsp-server/rtsp-stream-transport.h:
3842 * gst/rtsp-server/rtsp-stream.c:
3843 * gst/rtsp-server/rtsp-thread-pool.h:
3846 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3849 * examples/Makefile.am:
3850 configure: compile cgroup example conditionally
3851 Only compile the cgroup example when we have libcgroup
3853 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3856 * examples/Makefile.am:
3857 * examples/test-cgroups.c:
3858 examples: add cgroups example
3860 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3862 * tests/check/gst/rtspserver.c:
3863 tests: fix compilation
3865 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3867 * gst/rtsp-server/rtsp-thread-pool.c:
3868 thread-pool: fix vmethod invocation
3870 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3872 * gst/rtsp-server/rtsp-thread-pool.c:
3873 * gst/rtsp-server/rtsp-thread-pool.h:
3874 thread-pool: store thread type in thread
3876 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3878 * gst/rtsp-server/rtsp-client.c:
3879 client: pass thread from pool to media _prepare
3880 Get a thread from the configured threadpool and pass it to the prepare method of
3883 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3885 * gst/rtsp-server/rtsp-media.c:
3886 * gst/rtsp-server/rtsp-media.h:
3887 media: Accept a thread in _prepare
3888 Remove out own threadpool handling and use the provided thread and
3889 maincontext for the bus messages and the state changes.
3891 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3893 * gst/rtsp-server/rtsp-server.c:
3894 server: configure client thread pool
3896 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3898 * gst/rtsp-server/rtsp-client.c:
3899 * gst/rtsp-server/rtsp-client.h:
3900 client: add method to configure thread pool
3902 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3904 * gst/rtsp-server/rtsp-client.h:
3905 * gst/rtsp-server/rtsp-server.c:
3906 * gst/rtsp-server/rtsp-server.h:
3907 server: use thread pool
3908 Use the thread pool instead of doing our own thing.
3910 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3912 * gst/rtsp-server/Makefile.am:
3913 * gst/rtsp-server/rtsp-thread-pool.c:
3914 * gst/rtsp-server/rtsp-thread-pool.h:
3915 thread-pool: add object to manage threads
3916 Add an object to manage the client and media threads.
3918 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3920 * gst/rtsp-server/rtsp-auth.c:
3921 auth: debug authorization check
3923 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3925 * gst/rtsp-server/rtsp-media.c:
3926 media: start media pipeline in context
3927 Start the media pipeline in the provided context (or our default one
3928 when NULL). This makes sure that we run the bus thread in this context and that
3929 all media threads are children of this context.
3931 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3933 * gst/rtsp-server/rtsp-media-factory.c:
3934 factory: pass permissions to media by default
3936 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3938 * examples/test-auth.c:
3939 test: add permissions to auth test
3940 Ass some permissions to the media factory in the test.
3942 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3944 * gst/rtsp-server/rtsp-auth.c:
3945 * gst/rtsp-server/rtsp-auth.h:
3946 * gst/rtsp-server/rtsp-client.c:
3947 auth: simplify auth checks
3948 Remove client from methods, it's now in the state
3949 Perform the check specified by the string, use the information from the
3950 thread local context.
3952 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3954 * gst/rtsp-server/rtsp-client.c:
3955 * gst/rtsp-server/rtsp-client.h:
3956 client: add state to current thread
3957 Add the client to the ClientState object.
3958 Place the ClientState on the current thread.
3960 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3962 * gst/rtsp-server/rtsp-media-factory.c:
3963 * gst/rtsp-server/rtsp-media-factory.h:
3964 * gst/rtsp-server/rtsp-media.c:
3965 * gst/rtsp-server/rtsp-media.h:
3966 media: make it possible to set permissions
3967 Make it possible to set permissions on media and media factory objects
3969 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3971 * gst/rtsp-server/Makefile.am:
3972 * gst/rtsp-server/rtsp-permissions.c:
3973 * gst/rtsp-server/rtsp-permissions.h:
3974 permissions: add permissions object
3975 Add a mini object to store permissions based on a role.
3977 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3979 * examples/test-auth.c:
3980 * gst/rtsp-server/rtsp-auth.c:
3981 * gst/rtsp-server/rtsp-auth.h:
3982 * gst/rtsp-server/rtsp-client.c:
3983 auth: add auth checks
3984 Add an enum with auth checks and implement the checks in the auth object.
3985 Perform the checks from the client.
3987 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3989 * examples/test-auth.c:
3990 * gst/rtsp-server/rtsp-auth.c:
3991 * gst/rtsp-server/rtsp-auth.h:
3992 * gst/rtsp-server/rtsp-client.h:
3993 auth: use the token after authentication
3994 After we authenticated a user, keep the Token around in the state.
3996 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3998 * gst/rtsp-server/rtsp-client.c:
3999 * gst/rtsp-server/rtsp-media.c:
4000 * gst/rtsp-server/rtsp-media.h:
4001 * tests/check/gst/media.c:
4002 media: add optional context for bus messages
4003 Add an optional mainloop to _prepare that will handle the bus messages instead
4004 of always using the shared mainloop.
4006 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4008 * gst/rtsp-server/Makefile.am:
4009 * gst/rtsp-server/rtsp-token.c:
4010 * gst/rtsp-server/rtsp-token.h:
4011 token: add authorization token
4012 Add a simply miniobject that contains the authorizations. The object contains a
4013 GstStructure that hold all authorization fields. When a user is authenticated,
4014 the auth module will create a Token for the user. The token is then used to
4015 check what operations the user is allowed to do and various other configuration
4018 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4020 * examples/test-auth.c:
4021 * gst/rtsp-server/rtsp-auth.c:
4022 * gst/rtsp-server/rtsp-auth.h:
4023 * gst/rtsp-server/rtsp-client.c:
4024 * gst/rtsp-server/rtsp-client.h:
4025 * gst/rtsp-server/rtsp-media-factory.c:
4026 * gst/rtsp-server/rtsp-media-factory.h:
4027 * gst/rtsp-server/rtsp-media.c:
4028 * gst/rtsp-server/rtsp-media.h:
4029 auth: remove auth from media and factory
4030 Remove the auth object from media and factory. We want to have the RTSPClient
4031 authenticate and authorize resources, there is no need to place another auth
4032 manager on the media/factory.
4034 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4036 * examples/test-auth.c:
4037 * gst/rtsp-server/rtsp-auth.c:
4038 * gst/rtsp-server/rtsp-auth.h:
4039 * gst/rtsp-server/rtsp-client.h:
4040 auth: add support for multiple basic auth tokens
4041 Make it possible to add multiple basic authorisation tokens to one authorization
4042 object. Associate with each token an authorization group that will define what
4043 capabilities are allowed.
4045 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4047 * gst/rtsp-server/rtsp-client.c:
4048 client: error out on non-aggregate control
4049 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
4051 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4053 * gst/rtsp-server/rtsp-client.c:
4054 client: rework setup request a little
4055 Cache the media in DESCRIBE based on the longest matching path with the uri
4056 that we can find in the mount points.
4057 Rework the setup request a little to get the media from the session or from
4058 the longest matching path, this way we can derive the control string as
4059 everything after the path instead of hardcoding it.
4060 Find the stream based on the control string and only open a session when all
4063 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4065 * gst/rtsp-server/rtsp-media.c:
4066 * gst/rtsp-server/rtsp-media.h:
4067 media: add method to find a stream by control url
4069 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4071 * gst/rtsp-server/rtsp-stream.c:
4072 * gst/rtsp-server/rtsp-stream.h:
4073 stream: add method to check control url of stream
4075 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4077 * gst/rtsp-server/rtsp-client.c:
4078 * gst/rtsp-server/rtsp-session-media.c:
4079 * gst/rtsp-server/rtsp-session-media.h:
4080 * gst/rtsp-server/rtsp-session.c:
4081 * gst/rtsp-server/rtsp-session.h:
4082 session: use path matching for session media
4083 Use a path string instead of a uri to lookup session media in the sessions. Also
4084 use path matching to find the largest possible path that matches.
4086 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4088 * gst/rtsp-server/rtsp-client.c:
4089 * gst/rtsp-server/rtsp-mount-points.c:
4090 * gst/rtsp-server/rtsp-mount-points.h:
4091 * tests/check/gst/mountpoints.c:
4092 mount-points: remove useless vmethod
4093 Making lookups in the mount points should not be done with a URL, if there is a
4094 mapping to be done from URL to mount points, we'll need to do it somewhere
4097 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4099 * gst/rtsp-server/rtsp-mount-points.c:
4100 * gst/rtsp-server/rtsp-mount-points.h:
4101 * tests/check/gst/mountpoints.c:
4102 mount-points: improve mount point searching
4103 Use a GSequence to keep track of the mount points.
4104 Match a URL to the longest matching registered mount point. This should be the
4105 URL to perform aggreagate control and the remainder is the stream specific
4107 Add some unit tests for this.
4109 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
4111 * gst/rtsp-server/Makefile.am:
4112 rtsp-server: Allow building of static library
4114 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4116 * tests/check/gst/mediafactory.c:
4117 tests: fix compilation
4119 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4121 * gst/rtsp-server/rtsp-sdp.c:
4122 sdp: get control string from stream
4123 Use the control string as configured in the stream.
4125 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4127 * gst/rtsp-server/rtsp-stream.c:
4128 * gst/rtsp-server/rtsp-stream.h:
4129 stream: add methods and property to set control string
4131 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4133 * gst/rtsp-server/rtsp-client.c:
4135 Rename variables for clarity
4136 Keep media in state when we can
4138 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4140 * gst/rtsp-server/rtsp-client.c:
4141 * gst/rtsp-server/rtsp-stream.c:
4142 * gst/rtsp-server/rtsp-stream.h:
4143 stream: add more support for IPv6
4144 Rename _get_address to _get_multicast_address in GstRTSPStream to
4145 make it clear that this function only deals with multicast.
4146 Make it possible to have both an IPv4 and IPv6 multicast address on
4147 a stream. Give the client an IPv4 or IPv6 address depending on the
4148 address it used to connect to the server.
4149 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
4151 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4153 * gst/rtsp-server/rtsp-client.c:
4156 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4158 * gst/rtsp-server/rtsp-stream.c:
4159 stream: handle failed port allocation
4160 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
4161 can't allocate any family at all. Also keep track of what port families we
4163 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
4165 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4167 * gst/rtsp-server/rtsp-stream.c:
4168 stream: improve docs
4170 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4172 * gst/rtsp-server/rtsp-stream-transport.c:
4173 stream-transport: remove old if 0 block
4175 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
4177 * tests/check/gst/client.c:
4179 gst_rtsp_client_get_uri() has been removed
4180 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
4182 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4184 * gst/rtsp-server/rtsp-client.c:
4185 * gst/rtsp-server/rtsp-client.h:
4186 client: add method to filter managed sessions
4187 Add a method to filter the sessions managed by this client connection.
4188 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
4190 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4192 * gst/rtsp-server/rtsp-client.c:
4193 * gst/rtsp-server/rtsp-client.h:
4194 client: remove _get_uri() method
4195 Remove the get_uri() method on the client. A client has no uri, the uri
4196 property is an internal property to manage the last cached media for
4199 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4201 * gst/rtsp-server/rtsp-media-factory.h:
4202 media-factory: fix typo
4204 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
4206 * gst/rtsp-server/rtsp-media.c:
4207 rtsp-media: Do not leak the query in default_query_stop
4208 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
4210 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4212 * gst/rtsp-server/rtsp-media.c:
4213 media: don't unlock when conversion fails
4214 Don't unlock the state lock when conversion fails because it was not locked.
4216 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4218 * gst/rtsp-server/rtsp-media.c:
4219 * gst/rtsp-server/rtsp-media.h:
4220 Add query_position and query_stop vmethods to rtsp-media
4222 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4224 * gst/rtsp-server/rtsp-media.c:
4225 Fix typo in property install for rtsp-media's time-provider
4227 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4229 * gst/rtsp-server/rtsp-client.c:
4230 * gst/rtsp-server/rtsp-client.h:
4231 client: clean some variables
4232 Clean some variables and add some guards to _send_request()
4234 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4236 * gst/rtsp-server/rtsp-client.c:
4237 * gst/rtsp-server/rtsp-client.h:
4238 Add gst_rtsp_client_send_request API
4239 This makes it possible to send arbitrary messages to a client, such as
4240 SET_PARAMETER or GET_PARAMETER
4242 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4244 * gst/rtsp-server/rtsp-media.c:
4245 * gst/rtsp-server/rtsp-media.h:
4246 media: add _get_element() method
4247 Add method to get the element used when creating the media.
4248 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
4250 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4252 * gst/rtsp-server/rtsp-media.c:
4255 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4257 * gst/rtsp-server/rtsp-stream.c:
4258 * gst/rtsp-server/rtsp-stream.h:
4259 stream: allow access to the rtp session
4260 https://bugzilla.gnome.org/show_bug.cgi?id=703004
4262 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
4264 * gst/rtsp-server/rtsp-stream.c:
4265 * gst/rtsp-server/rtsp-stream.h:
4266 dscp qos support in gst-rtsp-stream
4267 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
4269 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4271 * tests/check/gst/rtspserver.c:
4273 Actually do what the comment says. Also keep the old code around, not sure what
4274 should happen when you get a 454 from a TEARDOWN, does it close the connection?
4275 it currently doesn't.
4277 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4279 * gst/rtsp-server/rtsp-client.c:
4280 client: also watch newly created session
4281 When we newly created a session, start watching it immediately instead of
4282 on the next request.
4284 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
4286 * tests/check/gst/client.c:
4287 tests: add unit test for new-session
4288 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
4290 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4292 * gst/rtsp-server/rtsp-client.c:
4293 client: emit new-session when new session is created
4294 Only emit new-session when we created a new session for a client, not when a
4295 client picked up a previous session.
4296 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
4298 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
4300 * gst/rtsp-server/rtsp-client.c:
4301 client: handle asterisk as path in requests
4302 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
4304 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4306 * gst/rtsp-server/rtsp-media.c:
4307 media: handle segment query format mismatch
4308 It's possible that the segment query returns with a different format than what
4309 we asked for, handle this case also.
4311 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
4313 * gst/rtsp-server/rtsp-media.c:
4314 media: use segment stop in collect_media_stats
4315 Use segment stop instead of duration as range end point.
4316 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
4318 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
4320 * gst/rtsp-server/rtsp-media.c:
4321 * tests/check/gst/media.c:
4322 rtsp-media: Do not leak the element in take_pipeline
4323 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
4325 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
4327 * gst/rtsp-server/rtsp-client.c:
4328 * gst/rtsp-server/rtsp-client.h:
4329 rtsp-client: Make configure_client_transport virtual
4330 This patch makes configure_client_transport virtual. The functionality is
4331 needed to handle some weird clients sending multicast transport settings as url
4333 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
4335 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
4337 * gst/rtsp-server/rtsp-client.c:
4338 * gst/rtsp-server/rtsp-client.h:
4339 rtsp-client: Make param_set and param_get virtual
4340 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
4342 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
4344 * gst/rtsp-server/rtsp-client.c:
4345 * gst/rtsp-server/rtsp-media.c:
4346 * gst/rtsp-server/rtsp-media.h:
4347 media: convert_range replaces get_range_times
4348 get_range_times worked for handling UTC ranges for seeks, but we also
4349 need to convert back from NPT to the requested unit in
4350 get_range_string. convert_range is now used for both.
4351 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
4353 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4355 * gst/rtsp-server/rtsp-client.c:
4356 * gst/rtsp-server/rtsp-sdp.c:
4357 * gst/rtsp-server/rtsp-sdp.h:
4358 sdp: cleanup sdp info
4359 We don't need to pass the proto, we can more easily check a boolean.
4360 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
4362 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
4364 * gst/rtsp-server/rtsp-sdp.c:
4365 use 0.0.0.0 or :: for c= line instead of server address
4367 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
4369 * gst/rtsp-server/rtsp-client.c:
4370 use local address, not remote, in SDP
4371 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
4373 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4376 Automatic update of common submodule
4377 From 098c0d7 to 01a7a46
4379 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
4381 * gst/rtsp-server/rtsp-media.c:
4382 * gst/rtsp-server/rtsp-media.h:
4383 media: possibility to override range time conversion
4384 Make it possible to override the conversion from GstRTSPTimeRange to
4385 GstClockTimes, that is done before seeking on the media
4386 pipeline. Overriding can be useful for UTC ranges, where the default
4387 conversion gives nanoseconds since 1900.
4388 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
4390 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
4392 * gst/rtsp-server/rtsp-server.c:
4393 * gst/rtsp-server/rtsp-server.h:
4394 rtsp-server: Expose the use_client_settings API
4395 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
4397 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
4399 * gst/rtsp-server/rtsp-client.c:
4400 * gst/rtsp-server/rtsp-stream.c:
4401 * gst/rtsp-server/rtsp-stream.h:
4402 rtspstream: handle both ipv4 and ipv6 clients
4403 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
4405 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4407 * gst/rtsp-server/rtsp-sdp.c:
4408 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
4409 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
4410 We already have a way to place extra attributes in the SDP by using a string
4411 property with prefix x- or a- in the caps.
4413 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4415 * gst/rtsp-server/rtsp-sdp.c:
4416 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
4417 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
4418 We already have a way to place extra attributes in the SDP, just make a string
4419 property in the payloader with a- or x- prefix.
4421 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4423 * gst/rtsp-server/rtsp-sdp.c:
4424 rtsp: place a- and x- properties as attributes
4425 application/x-rtp has properties with a- and x- prefixes that should be
4426 placed as attributes in the SDP for the media instead of being added to the
4429 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4431 * examples/Makefile.am:
4432 * examples/test-video.c:
4433 example: add TLS example
4435 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4437 * gst/rtsp-server/rtsp-server.c:
4438 * gst/rtsp-server/rtsp-server.h:
4439 server: add support for TLS
4440 Add methods to set and get a TLS certificate.
4441 Add vmethod to configure a new connection. By default, configure the TLS
4442 certificate in a new connection if needed.
4444 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4446 * gst/rtsp-server/rtsp-server.c:
4447 * gst/rtsp-server/rtsp-server.h:
4448 server: remove accept_client vmethod
4449 This vmethod is not very useful so remove it.
4451 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4453 * gst/rtsp-server/rtsp-server.c:
4454 server: don't crash on NULL GError
4456 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
4458 * gst/rtsp-server/rtsp-session-pool.c:
4459 rtsp-session-pool: corrected session timeout detection
4460 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
4462 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4464 * gst/rtsp-server/rtsp-client.c:
4465 client: improve debug
4467 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4469 * gst/rtsp-server/rtsp-client.c:
4470 * gst/rtsp-server/rtsp-client.h:
4471 * gst/rtsp-server/rtsp-server.c:
4472 server: refactor connection setup
4473 Let the server accept the socket connection and construct a GstRTSPConnection
4474 from it. Remove the code from the client and let the client only deal with
4475 a fully configure GstRTSPConnection object.
4476 We will need this later when the server will configure the connection for
4479 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4481 * gst/rtsp-server/rtsp-stream.c:
4482 stream: keep the transport object alive
4483 Keep the transport object alive while we have it as qdata on the
4486 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
4488 * gst/rtsp-server/rtsp-client.c:
4489 * gst/rtsp-server/rtsp-server.c:
4490 rtsp-server: Do not crash on nmapping of server
4491 * generate error when gst_rtsp_connection_accept fails
4492 * do not stop accepting incoming connections because
4493 accepting a client fails
4494 https://bugzilla.gnome.org/show_bug.cgi?id=701072
4496 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
4498 * gst/rtsp-server/rtsp-client.c:
4499 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
4500 https://bugzilla.gnome.org/show_bug.cgi?id=700953
4502 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
4504 * gst/rtsp-server/rtsp-sdp.c:
4505 rtsp-sdp: Parse framerate caps field and set SDP attribute
4506 The SDP attribute and its format is described in RFC4566.
4507 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
4509 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
4511 * gst/rtsp-server/rtsp-sdp.c:
4512 rtsp-sdp: Parse width/height from caps and set SDP attribute
4513 The SDP attribute and its format is described in RFC6064.
4514 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
4516 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
4518 * gst/rtsp-server/rtsp-sdp.c:
4519 * tests/check/gst/client.c:
4520 rtsp-sdp: add bandwidth line
4521 https://bugzilla.gnome.org/show_bug.cgi?id=699220
4523 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4526 Automatic update of common submodule
4527 From 5edcd85 to 098c0d7
4529 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
4531 * tests/check/gst/media.c:
4532 tests: add dynamic payloader prepare/unprepare check
4534 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4536 * gst/rtsp-server/rtsp-media.c:
4537 media: release lock when removing fakesink
4539 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4541 * gst/rtsp-server/rtsp-stream.c:
4542 stream: set elements to NULL before removing
4543 When removing a stream, set the elements to NULL first. This avoids
4544 element-is-not-in-NULL-state errors when we dispose the elements.
4546 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
4549 Automatic update of common submodule
4550 From 3cb3d3c to 5edcd85
4552 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4554 * gst/rtsp-server/rtsp-media.c:
4555 * gst/rtsp-server/rtsp-media.h:
4556 media: listen to pad-removed signals
4557 Listen to the pad-removed signal and remove the stream associated with the
4559 Add signal to be notified of the removed pad.
4560 Remove the fakesink in unprepare()
4561 Fix signatures of the signal methods
4563 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4565 * examples/test-sdp.c:
4566 tests: add example of reusable pipelines
4568 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
4570 * gst/rtsp-server/rtsp-stream.c:
4571 * gst/rtsp-server/rtsp-stream.h:
4572 stream: add method to get the srcpad
4574 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
4576 * tests/check/gst/media.c:
4577 check: add media prepare/unprepare test
4578 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4580 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
4582 * gst/rtsp-server/rtsp-media.c:
4583 media: disconnect from signal handlers in unprepare()
4584 We connected to the pad-added and no-more-pads signals in prepare() so
4585 we need to disconnect from them in unprepare().
4586 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4588 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
4590 * gst/rtsp-server/rtsp-media.c:
4591 media: don't free streams array
4592 Don't free the streams array in the unprepare() method, they were not
4594 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4596 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
4598 * gst/rtsp-server/rtsp-media.c:
4599 media: don't unref the pipeline in unprepare
4600 Unprepare() should undo what prepare() does. Because the pipeline is
4601 not created in prepare(), we should not unref it in unprepare()
4603 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
4605 * gst/rtsp-server/rtsp-stream.c:
4606 stream: clear session and caps for reuse
4607 Set the session and caps to NULL after unref otherwise we might unref
4609 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4611 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
4613 * gst/rtsp-server/rtsp-client.c:
4614 client: send out teardown signal before tearing down
4615 The advantage is that in the signal handler you get direct access to
4616 information about what streams are about to get torn down (in the
4617 GstRTSPClientState).
4618 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
4620 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
4622 * gst/rtsp-server/rtsp-client.c:
4623 * gst/rtsp-server/rtsp-client.h:
4624 client: expose connection
4625 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
4627 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
4630 Automatic update of common submodule
4631 From aed87ae to 3cb3d3c
4633 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4635 * gst/rtsp-server/rtsp-media.c:
4636 * gst/rtsp-server/rtsp-media.h:
4637 * gst/rtsp-server/rtsp-session-media.c:
4638 * gst/rtsp-server/rtsp-session-media.h:
4639 media: add method to get the base_time of the pipeline
4640 Together with a shared clock, this base-time could eventually be sent to
4641 the client so that it can reconstruct the exact running-time of the clock
4644 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4646 * gst/rtsp-server/Makefile.am:
4647 * gst/rtsp-server/rtsp-media.c:
4648 * gst/rtsp-server/rtsp-media.h:
4649 * gst/rtsp-server/rtsp-sdp.c:
4650 media: add GstNetTimeProvider support
4651 Add a property to let the media provide a GstNetTimeProvider for its clock.
4652 Make methods to get the clock and nettimeprovider
4653 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
4654 provider and also the current time of the clock. This should make it possible
4655 for (GStreamer) clients to slave their clock to the server clock.
4657 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
4660 Automatic update of common submodule
4661 From 04c7a1e to aed87ae
4663 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4665 * gst/rtsp-server/rtsp-media.c:
4666 media: wait for buffering to complete
4667 Wait for buffering to complete before changing the state to the target state.
4669 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4671 * gst/rtsp-server/rtsp-media.c:
4672 media: small cleanup
4674 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
4676 * tests/check/gst/rtspserver.c:
4677 tests: remove extra unref in test_setup_non_existing_stream
4678 The unref is not needed anymore, teardown runs without it.
4679 https://bugzilla.gnome.org/show_bug.cgi?id=696542
4681 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
4683 * tests/check/gst/rtspserver.c:
4684 tests: GSocketService cleanup in test_bind_already_in_use
4685 Use g_socket_service_stop so the rtspserver test stops listening for
4686 incoming connections in test_bind_already_in_use.
4687 https://bugzilla.gnome.org/show_bug.cgi?id=696541
4689 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
4691 * gst/rtsp-server/rtsp-media-factory.c:
4692 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
4693 Instead use a GWeakRef which is safe to use
4694 This is a known GLib bug, see:
4695 https://bugzilla.gnome.org/show_bug.cgi?id=667145
4697 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
4699 * gst/rtsp-server/rtsp-client.c:
4700 * gst/rtsp-server/rtsp-media.c:
4701 * gst/rtsp-server/rtsp-media.h:
4702 * gst/rtsp-server/rtsp-sdp.c:
4703 * tests/check/gst/media.c:
4704 * tests/check/gst/rtspserver.c:
4705 rtsp-media/client: Reply to PLAY request with same type of Range
4706 Remember the type of Range from the PLAY request and use the same type for
4709 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
4711 * gst/rtsp-server/rtsp-client.c:
4712 * gst/rtsp-server/rtsp-client.h:
4713 * tests/check/gst/client.c:
4714 rtsp-client: expose uri
4716 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
4718 * tests/check/gst/mediafactory.c:
4719 tests: Hold ref while creating second media
4720 To test if the media aren't shared, make sure we keep the first one while creating a second
4721 otherwise the same memory address may be reused.
4723 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
4726 configure: remove out-of-date comment
4728 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
4731 .gitignore: ignore more build files
4733 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
4735 * tests/check/Makefile.am:
4736 tests: use right _LIBS variable for gst-plugins-base libs
4738 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4740 * tests/check/Makefile.am:
4741 check: add librtp to libs
4743 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
4745 * tests/check/gst/rtspserver.c:
4746 tests: Add test to check selecting a port the server will send from
4748 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
4750 * tests/check/gst/rtspserver.c:
4751 tests: Make sure packets are actually received
4753 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4755 * gst/rtsp-server/rtsp-stream.c:
4756 stream: Select unicast address from pool if appropriate
4758 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
4760 * gst/rtsp-server/rtsp-stream.c:
4761 stream: Properties are always there in Gst 1.0
4763 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4765 * tests/check/gst/addresspool.c:
4766 tests: Add tests for unicast addresses in pool
4768 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
4770 * gst/rtsp-server/rtsp-address-pool.c:
4771 * tests/check/gst/addresspool.c:
4772 address-pool: Verify that multicast addresses are used for multicast and vice-versa
4774 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
4776 * docs/libs/gst-rtsp-server-sections.txt:
4777 * gst/rtsp-server/rtsp-address-pool.c:
4778 * gst/rtsp-server/rtsp-address-pool.h:
4779 * gst/rtsp-server/rtsp-stream.c:
4780 * tests/check/gst/addresspool.c:
4781 address-pool: Add unicast addresses
4783 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
4786 * gst/rtsp-server/rtsp-server.c:
4787 * tests/check/gst/rtspserver.c:
4788 rtsp-server: Limit the number of threads per server instance
4789 If we exceed the maximum, just round robin the clients over the existing
4792 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
4794 * gst/rtsp-server/rtsp-server.c:
4795 rtsp-server: No need to store the GMainContext in the client context
4797 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
4799 * tests/check/gst/rtspserver.c:
4800 tests: Add test for client disconnection
4802 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
4804 * tests/check/gst/rtspserver.c:
4805 tests: Test client and session timeouts with multiple threads
4807 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
4809 * gst/rtsp-server/rtsp-address-pool.c:
4810 * gst/rtsp-server/rtsp-auth.c:
4811 * gst/rtsp-server/rtsp-client.c:
4812 * gst/rtsp-server/rtsp-media-factory-uri.c:
4813 * gst/rtsp-server/rtsp-media-factory.c:
4814 * gst/rtsp-server/rtsp-media.c:
4815 * gst/rtsp-server/rtsp-mount-points.c:
4816 * gst/rtsp-server/rtsp-server.c:
4817 * gst/rtsp-server/rtsp-session-media.c:
4818 * gst/rtsp-server/rtsp-session-pool.c:
4819 * gst/rtsp-server/rtsp-session.c:
4820 Document locking and its order
4822 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
4824 * tests/check/gst/rtspserver.c:
4825 tests: Test that slow DESCRIBE don't block other clients
4827 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
4829 * tests/check/gst/client.c:
4830 tests: Add tests for client-requested multicast address
4832 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
4834 * docs/libs/gst-rtsp-server-sections.txt:
4835 docs: Put the various functions in the right sections
4837 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
4839 * docs/libs/gst-rtsp-server-docs.sgml:
4840 * docs/libs/gst-rtsp-server-sections.txt:
4841 * gst/rtsp-server/rtsp-address-pool.c:
4842 * gst/rtsp-server/rtsp-address-pool.h:
4843 docs: Generate docs for GstRTSPAddressPool
4845 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4847 * gst/rtsp-server/rtsp-client.c:
4848 * gst/rtsp-server/rtsp-stream.c:
4849 * gst/rtsp-server/rtsp-stream.h:
4850 client: Check client provided addresses against the address pool
4852 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
4854 * gst/rtsp-server/rtsp-address-pool.c:
4855 * gst/rtsp-server/rtsp-address-pool.h:
4856 * tests/check/gst/addresspool.c:
4857 address-pool: Add API to request a specific address from the pool
4858 Also add relevant unit tests.
4860 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
4862 * tests/check/gst/mediafactory.c:
4863 tests: Check the passing around of a RTSPAddressPool
4864 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
4865 way down to the stream.
4867 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
4869 * tests/check/gst/addresspool.c:
4870 tests: Add more tests for the address pool
4872 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
4874 * gst/rtsp-server/rtsp-address-pool.c:
4875 address-pool: Fix off by one error
4876 When splitting a port range, the port after a skip is not part of range.
4878 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
4881 Automatic update of common submodule
4882 From 2de221c to 04c7a1e
4884 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
4887 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
4888 AM_CONFIG_HEADER was removed in automake 1.13
4889 https://bugzilla.gnome.org/show_bug.cgi?id=693368
4891 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
4894 Automatic update of common submodule
4895 From a942293 to 2de221c
4897 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4899 * gst/rtsp-server/rtsp-client.c:
4900 client: make sure the watch exists while sending data
4901 Protect the send_func with a lock. This allows us to wait for sending
4902 to complete before changing the send_func and user_data. We add an
4903 extra ref to the watch to make sure that it remains valid during
4905 When closing the connection, set the send_func to NULL
4906 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
4908 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4910 * tests/check/Makefile.am:
4911 tests: use GST_*_1_0 environment variables everywhere
4912 The _1_0 suffixed environment variables override the
4913 non-suffixed ones, so if we're in an environment that
4914 sets the _1_0 suffixed ones, such as jhbuild, we need
4915 to set those to make sure ours actually always get
4918 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4921 Automatic update of common submodule
4922 From acb04d9 to a942293
4924 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4926 * gst/rtsp-server/rtsp-client.c:
4927 rtsp-client: set the client backlog
4928 Set the client backlog to a reasonable default
4930 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
4932 * gst/rtsp-server/rtsp-media.c:
4933 rtsp-media: Make the element a constructor parameter
4934 https://bugzilla.gnome.org/show_bug.cgi?id=689594
4936 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4938 * docs/libs/Makefile.am:
4939 docs: Link with gcov library when gcov is enabled
4940 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
4942 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4944 * gst/rtsp-server/rtsp-media.c:
4945 media: match prepare with unprepare
4946 Really unprepare when there were an equal amount of prepare calls.
4948 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4950 * gst/rtsp-server/rtsp-media.c:
4951 media: media has to be unprepared in finalize
4952 Because unprepare takes away the last ref on the media.
4954 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4956 * gst/rtsp-server/rtsp-client.c:
4957 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
4958 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
4959 We can't use the refcount to trigger unprepare because it is the unprepare call
4960 that removes the last refcount after all messages are consumed. What we should
4961 probably do is make a prepared refcount and only unprepare when the refcount
4964 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4966 * gst/rtsp-server/rtsp-media.c:
4967 media: let the source unref the last media ref
4968 the last ref to the media is held by the source so we don't need to add more ref
4969 and unrefs, we simply destroy the media when the source is gone.
4971 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4973 * gst/rtsp-server/rtsp-media.c:
4974 media: improve debug
4976 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4978 * gst/rtsp-server/rtsp-media.c:
4980 Make sure we are in the right state when collecting the position and duration.
4981 Only make ourselves PREPARED when we were previously PREPARING.
4983 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4985 * gst/rtsp-server/rtsp-media.c:
4986 media: use g_object_ref/unref for GObjects
4988 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
4990 * gst/rtsp-server/rtsp-client.c:
4991 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
4992 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
4993 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
4994 isn't being used anymore.
4996 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
4998 * gst/rtsp-server/rtsp-media.c:
4999 Fix compiler warning
5001 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
5003 * gst/rtsp-server/rtsp-media-factory-uri.c:
5004 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
5006 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5008 * gst/rtsp-server/rtsp-session-media.h:
5011 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5013 * gst/rtsp-server/rtsp-media.c:
5014 * tests/check/gst/media.c:
5015 media: avoid element leak
5017 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5019 * gst/rtsp-server/rtsp-media.c:
5020 media: require an element in media constructor
5022 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5024 * gst/rtsp-server/rtsp-client.c:
5025 Revert "client: TEARDOWN brings that state to Init again"
5026 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
5027 The object is already disposed, there is no point in setting the state.
5029 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5031 * gst/rtsp-server/rtsp-client.c:
5032 client: TEARDOWN brings that state to Init again
5034 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5036 * docs/libs/gst-rtsp-server-sections.txt:
5037 * examples/test-auth.c:
5038 * gst/rtsp-server/rtsp-auth.c:
5039 * gst/rtsp-server/rtsp-auth.h:
5040 * gst/rtsp-server/rtsp-client.c:
5041 * gst/rtsp-server/rtsp-client.h:
5042 * gst/rtsp-server/rtsp-media-factory-uri.c:
5043 * gst/rtsp-server/rtsp-media-factory-uri.h:
5044 * gst/rtsp-server/rtsp-media-factory.c:
5045 * gst/rtsp-server/rtsp-media-factory.h:
5046 * gst/rtsp-server/rtsp-media.c:
5047 * gst/rtsp-server/rtsp-media.h:
5048 * gst/rtsp-server/rtsp-mount-points.c:
5049 * gst/rtsp-server/rtsp-mount-points.h:
5050 * gst/rtsp-server/rtsp-sdp.c:
5051 * gst/rtsp-server/rtsp-server.c:
5052 * gst/rtsp-server/rtsp-server.h:
5053 * gst/rtsp-server/rtsp-session-media.c:
5054 * gst/rtsp-server/rtsp-session-media.h:
5055 * gst/rtsp-server/rtsp-session-pool.c:
5056 * gst/rtsp-server/rtsp-session-pool.h:
5057 * gst/rtsp-server/rtsp-session.c:
5058 * gst/rtsp-server/rtsp-session.h:
5059 * gst/rtsp-server/rtsp-stream-transport.c:
5060 * gst/rtsp-server/rtsp-stream-transport.h:
5061 * gst/rtsp-server/rtsp-stream.c:
5062 * gst/rtsp-server/rtsp-stream.h:
5063 * tests/check/gst/media.c:
5064 rtsp: make object details private
5065 Make all object details private
5066 Add methods to access private bits
5068 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5070 * tests/check/Makefile.am:
5071 * tests/check/gst/media.c:
5072 tests: add media tests
5074 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5076 * gst/rtsp-server/rtsp-media.c:
5077 media: check if prepared for some methods
5078 Check that the media object is prepared before doing seek and getting the
5079 current position etc.
5080 Add some g_return checks.
5082 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5084 * tests/check/Makefile.am:
5085 * tests/check/gst/mediafactory.c:
5086 tests: add mediafactory test
5088 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5090 * gst/rtsp-server/rtsp-stream.c:
5091 stream: improve debug
5093 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5095 * gst/rtsp-server/rtsp-media.c:
5096 * gst/rtsp-server/rtsp-media.h:
5097 media: unref pipeline in finalize to avoid leaking it
5099 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5101 * gst/rtsp-server/rtsp-media-factory-uri.c:
5102 * gst/rtsp-server/rtsp-media.c:
5103 rtsp: use gst_object_unref on GstObjects
5105 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5107 * gst/rtsp-server/rtsp-media-factory.c:
5108 media-factory: require an url
5110 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5112 * examples/test-uri.c:
5113 examples: fix include
5115 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5117 * gst/rtsp-server/rtsp-server.h:
5118 server: remove unused include
5120 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5122 * tests/check/Makefile.am:
5123 * tests/check/gst/mountpoints.c:
5124 tests: add test for mountpoints
5126 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5128 * gst/rtsp-server/rtsp-client.c:
5129 client: fix factory leak
5130 Keep the factory in the state object only for authorization checks and make
5131 sure we unref it on failure. Also don't keep invalid objects in the state
5134 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5136 * gst/rtsp-server/rtsp-mount-points.c:
5137 mounts: add g_return_if guards
5139 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5141 * tests/check/gst/client.c:
5142 tests: add more tests
5144 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5146 * gst/rtsp-server/rtsp-client.c:
5147 client: improve debug
5149 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5151 * gst/rtsp-server/rtsp-client.c:
5152 client: improve debug and fix leaks
5153 Cleanup the uri and session when there is a bad request.
5155 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5160 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5162 * tests/check/gst/client.c:
5163 test: add test for session in options request
5165 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5167 * gst/rtsp-server/rtsp-client.c:
5168 client: use 454 when session can't be found
5169 We should use 454 when a session can't be found because there was no session
5170 pool configured in the server. This is not a server configuration problem
5171 because the server on which the request is done might not be the same one that
5172 will keep the sessions for us and so it does not need to support sessions.
5174 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5176 * gst/rtsp-server/rtsp-client.c:
5177 client: only free connection when there is one
5178 It's possible that the client doesn't have a connection when we try to free it.
5180 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5182 * tests/check/Makefile.am:
5183 * tests/check/gst/client.c:
5184 tests: add unit test for the client object
5186 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5188 * gst/rtsp-server/rtsp-client.c:
5189 client: small cleanup
5191 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5193 * gst/rtsp-server/rtsp-client.h:
5194 client: remove unused include
5196 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5198 * gst/rtsp-server/rtsp-client.c:
5199 client: fix compilation
5201 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5203 * gst/rtsp-server/rtsp-client.c:
5204 client: call destroy without the lock
5206 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5208 * gst/rtsp-server/rtsp-client.c:
5209 * gst/rtsp-server/rtsp-client.h:
5210 client: make the client usable without a socket
5211 Make a method to let the client handle a message and a callback when the client
5212 wants us to send a response message back. This makes it possible to also use the
5213 client object without the sockets, which should make it easier to test.
5215 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5217 * gst/rtsp-server/rtsp-client.c:
5218 * gst/rtsp-server/rtsp-client.h:
5219 client: small cleanup
5221 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5223 * docs/libs/gst-rtsp-server-sections.txt:
5224 * gst/rtsp-server/rtsp-client.c:
5225 * gst/rtsp-server/rtsp-client.h:
5226 * gst/rtsp-server/rtsp-server.c:
5227 client: remove reference to server
5228 We don't need to keep a ref to the server
5230 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5232 * gst/rtsp-server/rtsp-client.c:
5233 * gst/rtsp-server/rtsp-client.h:
5235 Also add some g_return_if()
5237 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5239 * gst/rtsp-server/rtsp-client.c:
5240 client: log more errors
5242 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5244 * gst/rtsp-server/rtsp-client.c:
5245 client: fix compilation
5247 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5249 * gst/rtsp-server/rtsp-client.c:
5250 * gst/rtsp-server/rtsp-client.h:
5251 client: add generic close-after-send support
5252 Add a property to send_response() to close the connection after the response has
5253 been sent to the client.
5255 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5258 * docs/libs/gst-rtsp-server-docs.sgml:
5259 * docs/libs/gst-rtsp-server-sections.txt:
5260 * docs/libs/gst-rtsp-server.types:
5261 * examples/test-auth.c:
5262 * examples/test-launch.c:
5263 * examples/test-mp4.c:
5264 * examples/test-multicast.c:
5265 * examples/test-multicast2.c:
5266 * examples/test-ogg.c:
5267 * examples/test-readme.c:
5268 * examples/test-sdp.c:
5269 * examples/test-uri.c:
5270 * examples/test-video.c:
5271 * gst/rtsp-server/Makefile.am:
5272 * gst/rtsp-server/rtsp-auth.h:
5273 * gst/rtsp-server/rtsp-client.c:
5274 * gst/rtsp-server/rtsp-client.h:
5275 * gst/rtsp-server/rtsp-media-mapping.c:
5276 * gst/rtsp-server/rtsp-media-mapping.h:
5277 * gst/rtsp-server/rtsp-mount-points.c:
5278 * gst/rtsp-server/rtsp-mount-points.h:
5279 * gst/rtsp-server/rtsp-server.c:
5280 * gst/rtsp-server/rtsp-server.h:
5281 * gst/rtsp-server/rtsp-session-media.c:
5282 * gst/rtsp-server/rtsp-session-pool.c:
5283 * gst/rtsp-server/rtsp-session-pool.h:
5284 * tests/check/gst/rtspserver.c:
5285 MediaMapping -> MountPoints
5286 Describes better what the object manages.
5288 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5291 configure: bump required version of -base
5293 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5295 * gst/rtsp-server/rtsp-media.c:
5298 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5300 * gst/rtsp-server/rtsp-media.c:
5301 * gst/rtsp-server/rtsp-media.h:
5302 media: support more Range formats
5303 Use the new -base methods to convert the Range string into a seek start and stop
5306 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5308 * examples/test-launch.c:
5309 examples: fix whitespace
5311 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5313 * examples/test-auth.c:
5314 test-auth: add example of how to remove sessions
5315 Add an example of the session filter api.
5317 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5319 * examples/test-uri.c:
5320 test-uri: remove mapping example
5322 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5324 * examples/test-uri.c:
5325 test-uri: fix callback signature
5327 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5329 * gst/rtsp-server/rtsp-media-factory.c:
5330 factory: keep ref to factory while media active
5331 While the media from a factory is alive, keep a ref to the factory.
5332 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
5334 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5336 * gst/rtsp-server/rtsp-media-factory-uri.c:
5337 factory-uri: add some debug
5339 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5341 * gst/rtsp-server/rtsp-stream.c:
5342 stream: set udp sources to PLAYING
5343 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
5344 so that it doesn't cause our pipeline to produce ASYNC-DONE.
5346 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5348 * gst/rtsp-server/rtsp-media-factory-uri.c:
5349 factory-uri: take ref to factory
5350 Take a ref to the factory that we place in our list.
5352 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5354 * tests/Makefile.am:
5355 * tests/test-reuse.c:
5356 test: add test for server reuse
5357 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
5359 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
5361 * gst/rtsp-server/rtsp-server.c:
5362 server: start and stop multiple times
5363 Stop listening on the RTSP port when the GSource is removed, so clients
5364 can't connect and the server can be started again.
5365 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
5367 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5369 * gst/rtsp-server/rtsp-server.c:
5370 server: fix small leak
5372 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5374 * gst/rtsp-server/rtsp-media.c:
5375 media: unref source in finish_unprepare
5376 The source is created in prepare, unref it in finish_unprepare.
5377 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
5379 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
5381 * gst/rtsp-server/rtsp-client.c:
5382 * gst/rtsp-server/rtsp-media.c:
5383 rtsp-media: remove bus watch before finalizing
5384 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
5385 * An extra media ref is added for the bus watch. This extra ref is unreffed by
5386 the GDestroyNotify function.
5387 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
5388 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
5389 gst_rtsp_media_unprepare before unreffing the media.
5390 This way, the bus watch will be removed before the media is finalized.
5391 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
5393 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
5395 * gst/rtsp-server/rtsp-client.c:
5396 * gst/rtsp-server/rtsp-client.h:
5397 client: wait until the TEARDOWN response is sent to close the connection
5398 Responses can be sent async so we need to wait until the TEARDOWN response has
5399 been written before we close the connection to the client. This avoids the risk
5400 of writing/polling closed sockets.
5401 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
5403 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
5405 * gst/rtsp-server/rtsp-stream.c:
5406 rtsp-stream: plug socket leak
5407 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
5409 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
5412 Automatic update of common submodule
5413 From 6bb6951 to a72faea
5415 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
5417 * gst/rtsp-server/rtsp-media-factory-uri.c:
5418 rtsp-server: don't use deprecated API
5420 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
5422 * gst/rtsp-server/rtsp-client.c:
5423 rtsp-client: fix unused-but-set-variable compiler warning
5424 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
5426 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5429 * docs/libs/gst-rtsp-server-sections.txt:
5430 * gst/rtsp-server/rtsp-client.c:
5433 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5435 * examples/Makefile.am:
5436 * examples/test-multicast2.c:
5437 examples: add another multicast example
5438 Add an example for how to configure separate multicast ranges for each media
5441 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5443 * examples/test-multicast.c:
5446 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5448 * gst/rtsp-server/rtsp-client.c:
5449 * gst/rtsp-server/rtsp-media.c:
5450 * gst/rtsp-server/rtsp-session-media.c:
5451 * gst/rtsp-server/rtsp-session-media.h:
5452 * gst/rtsp-server/rtsp-stream-transport.c:
5453 * gst/rtsp-server/rtsp-stream-transport.h:
5454 stream: use the address managed by the stream
5455 Use the address managed by the stream for multicast. This allows us to have 1
5456 multicast address for each stream.
5457 Because the address is now managed by the stream we don't have to pass it around
5459 Set the address pool on the streams.
5461 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5463 * gst/rtsp-server/rtsp-client.c:
5464 * gst/rtsp-server/rtsp-media.c:
5465 * gst/rtsp-server/rtsp-stream.c:
5468 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5470 * gst/rtsp-server/rtsp-media.c:
5471 * gst/rtsp-server/rtsp-media.h:
5472 media: add signal for new streams
5473 This allows applications to listen for new streams and configure properties on
5474 them, like the address pool.
5476 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5478 * gst/rtsp-server/rtsp-media.c:
5479 media: configure address pool in new streams
5481 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5483 * gst/rtsp-server/rtsp-stream.c:
5484 * gst/rtsp-server/rtsp-stream.h:
5485 stream: add methods to deal with address pool
5486 Add methods to get and set the address pool for the stream
5487 Add method to allocate and get the multicast addresses for this stream.
5489 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5491 * docs/libs/gst-rtsp-server-sections.txt:
5492 * gst/rtsp-server/rtsp-media.c:
5493 * gst/rtsp-server/rtsp-media.h:
5494 media: remove MTU property
5495 It is a stream property
5497 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5499 * gst/rtsp-server/rtsp-client.c:
5500 client: set blocksize only on stream
5501 Set the blocksize only on the current stream.
5503 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5505 * gst/rtsp-server/rtsp-stream.c:
5506 stream: share src and sink sockets
5507 the allocated socket is in the used-socket property, not socket.
5509 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5511 * gst/rtsp-server/rtsp-address-pool.c:
5512 * gst/rtsp-server/rtsp-address-pool.h:
5513 * gst/rtsp-server/rtsp-client.c:
5514 * gst/rtsp-server/rtsp-session-media.c:
5515 * gst/rtsp-server/rtsp-session-media.h:
5516 * gst/rtsp-server/rtsp-stream-transport.c:
5517 * gst/rtsp-server/rtsp-stream-transport.h:
5518 * tests/check/gst/addresspool.c:
5519 rtsp: make address-pool return an address object
5520 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
5521 store more info in the structure and allows us to more easily return the address
5522 to the right pool when no longer needed.
5523 Pass the address to the StreamTransport so that we can return it to the pool
5524 when the stream transport is freed or changed.
5526 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5528 * examples/Makefile.am:
5529 * examples/test-multicast.c:
5530 examples: add multicast example
5531 Show how to set up the multicast address pool so that media can be
5532 server with multicast.
5534 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5536 * gst/rtsp-server/rtsp-client.c:
5537 * gst/rtsp-server/rtsp-media-factory.c:
5538 * gst/rtsp-server/rtsp-media-factory.h:
5539 * gst/rtsp-server/rtsp-media.c:
5540 * gst/rtsp-server/rtsp-media.h:
5541 rtsp: use AddressPool
5542 Remove the multicast_group property.
5543 Use the configured addresspool to allocate multicast addresses.
5545 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5547 * gst/rtsp-server/rtsp-address-pool.c:
5548 * gst/rtsp-server/rtsp-address-pool.h:
5549 address-pool: add clear method
5551 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5553 * gst/rtsp-server/rtsp-address-pool.c:
5554 address-pool: small cleanups
5556 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5558 * tests/check/Makefile.am:
5559 * tests/check/gst/addresspool.c:
5560 tests: add addresspool unit test
5562 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5564 * gst/rtsp-server/Makefile.am:
5565 * gst/rtsp-server/rtsp-address-pool.c:
5566 * gst/rtsp-server/rtsp-address-pool.h:
5567 address-pool: add object to manage multicast addresses
5568 Make an object that can manage a rage of multicast addresses and ports.
5570 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5572 * gst/rtsp-server/rtsp-server.c:
5573 server: set default max-threads property
5575 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5577 * gst/rtsp-server/rtsp-media.c:
5578 media: wait for concurrent _prepare
5579 If a prepare is busy, wait for the result.
5581 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5583 * gst/rtsp-server/rtsp-media.c:
5584 media: add lock around message handler
5585 We don't want to dispatch messages while we are still processing the result of
5588 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5590 * gst/rtsp-server/rtsp-media.c:
5591 * gst/rtsp-server/rtsp-media.h:
5592 media: add lock to protect state changes
5594 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5596 * gst/rtsp-server/rtsp-stream.c:
5597 * gst/rtsp-server/rtsp-stream.h:
5600 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5602 * gst/rtsp-server/rtsp-stream-transport.c:
5603 * gst/rtsp-server/rtsp-stream-transport.h:
5604 * gst/rtsp-server/rtsp-stream.c:
5605 stream-transport: add keep-alive method
5607 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5609 * gst/rtsp-server/rtsp-stream-transport.c:
5610 * gst/rtsp-server/rtsp-stream-transport.h:
5611 * gst/rtsp-server/rtsp-stream.c:
5612 stream-transport: add method to handle RTP/RTCP
5613 Call new methods instead of poking into the structures directly.
5615 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5617 * gst/rtsp-server/rtsp-session-media.c:
5618 * gst/rtsp-server/rtsp-session-media.h:
5619 session-media: add locking
5621 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5623 * gst/rtsp-server/rtsp-session.c:
5624 * gst/rtsp-server/rtsp-session.h:
5625 session: add locking
5627 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5629 * gst/rtsp-server/rtsp-server.c:
5630 server: free old socket
5632 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5634 * gst/rtsp-server/rtsp-media-mapping.c:
5635 * gst/rtsp-server/rtsp-media-mapping.h:
5636 mapping: add locking
5638 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5640 * gst/rtsp-server/rtsp-media-factory.c:
5641 media-factory: add locking
5643 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5645 * gst/rtsp-server/rtsp-auth.c:
5646 * gst/rtsp-server/rtsp-auth.h:
5649 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5651 * gst/rtsp-server/rtsp-server.c:
5652 * gst/rtsp-server/rtsp-server.h:
5653 server: add max-thread property
5655 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5657 * gst/rtsp-server/rtsp-server.c:
5658 * gst/rtsp-server/rtsp-server.h:
5659 server: use a threadpool for the mainloops
5661 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5663 * gst/rtsp-server/rtsp-client.c:
5664 * gst/rtsp-server/rtsp-client.h:
5665 client: rename method
5666 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
5667 don't really create the client from the socket, we use the socket for the
5670 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5672 * gst/rtsp-server/rtsp-client.c:
5673 * gst/rtsp-server/rtsp-client.h:
5674 * gst/rtsp-server/rtsp-server.c:
5675 server: rework maincontext handling in clients
5676 Make a separate method to attach a client to a MainContext.
5677 Let the server decide in what GMainContext the client will operate and give this
5678 context to the client in attach. Then the server can later decide to use a
5679 separate thread for each client or just use the mainthread.
5681 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5683 * gst/rtsp-server/rtsp-client.c:
5684 * gst/rtsp-server/rtsp-session.c:
5685 * gst/rtsp-server/rtsp-session.h:
5686 session: move session header code in session object
5688 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
5692 * examples/test-auth.c:
5693 * examples/test-launch.c:
5694 * examples/test-mp4.c:
5695 * examples/test-ogg.c:
5696 * examples/test-readme.c:
5697 * examples/test-sdp.c:
5698 * examples/test-uri.c:
5699 * examples/test-video.c:
5700 * gst/rtsp-server/rtsp-auth.c:
5701 * gst/rtsp-server/rtsp-auth.h:
5702 * gst/rtsp-server/rtsp-client.c:
5703 * gst/rtsp-server/rtsp-client.h:
5704 * gst/rtsp-server/rtsp-media-factory-uri.c:
5705 * gst/rtsp-server/rtsp-media-factory-uri.h:
5706 * gst/rtsp-server/rtsp-media-factory.c:
5707 * gst/rtsp-server/rtsp-media-factory.h:
5708 * gst/rtsp-server/rtsp-media-mapping.c:
5709 * gst/rtsp-server/rtsp-media-mapping.h:
5710 * gst/rtsp-server/rtsp-media.c:
5711 * gst/rtsp-server/rtsp-media.h:
5712 * gst/rtsp-server/rtsp-params.c:
5713 * gst/rtsp-server/rtsp-params.h:
5714 * gst/rtsp-server/rtsp-sdp.c:
5715 * gst/rtsp-server/rtsp-sdp.h:
5716 * gst/rtsp-server/rtsp-server.c:
5717 * gst/rtsp-server/rtsp-server.h:
5718 * gst/rtsp-server/rtsp-session-media.c:
5719 * gst/rtsp-server/rtsp-session-media.h:
5720 * gst/rtsp-server/rtsp-session-pool.c:
5721 * gst/rtsp-server/rtsp-session-pool.h:
5722 * gst/rtsp-server/rtsp-session.c:
5723 * gst/rtsp-server/rtsp-session.h:
5724 * gst/rtsp-server/rtsp-stream-transport.c:
5725 * gst/rtsp-server/rtsp-stream-transport.h:
5726 * gst/rtsp-server/rtsp-stream.c:
5727 * gst/rtsp-server/rtsp-stream.h:
5728 * tests/check/gst/rtspserver.c:
5729 * tests/test-cleanup.c:
5732 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
5734 * gst/rtsp-server/rtsp-media.c:
5735 * gst/rtsp-server/rtsp-session-media.c:
5736 * gst/rtsp-server/rtsp-session.c:
5737 rtsp-server: added annotations to indicate type of ownership transfer of return values
5738 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5740 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
5743 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
5745 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
5748 * bindings/Makefile.am:
5749 * bindings/vala/Makefile.am:
5750 * bindings/vala/gst-rtsp-server-0.10.deps:
5751 * bindings/vala/gst-rtsp-server-0.10.vapi:
5752 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
5753 * bindings/vala/packages/gst-rtsp-server-0.10.files:
5754 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
5755 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
5756 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
5758 bindings: remove vala bindings
5759 They'll be reunited with the other GStreamer bindings
5760 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5762 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5764 * gst/rtsp-server/rtsp-client.c:
5765 * gst/rtsp-server/rtsp-session-media.c:
5766 * gst/rtsp-server/rtsp-session-media.h:
5767 * gst/rtsp-server/rtsp-stream-transport.c:
5768 * gst/rtsp-server/rtsp-stream-transport.h:
5769 rtsp: only create transport when needed
5770 Only create the StreamTransport when configured.
5772 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5774 * gst/rtsp-server/rtsp-client.c:
5775 client: small cleanup
5777 2012-10-27 23:49:24 +0200 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-stream-transport.c:
5782 * gst/rtsp-server/rtsp-stream-transport.h:
5783 rtsp: refactor configuration of transport
5784 Move the configuration of the transport to a place where it makes
5787 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5789 * gst/rtsp-server/rtsp-client.c:
5790 client: refactor transport parsing
5792 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5794 * gst/rtsp-server/rtsp-client.c:
5795 client: refuse to change the MTU on shared media
5796 If we change the MTU of chared media, it changes for all clients.
5797 We don't want to set the MTU to something large for clients that
5800 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5802 * examples/test-mp4.c:
5803 * gst/rtsp-server/rtsp-media.c:
5804 small fixes to docs and debug
5806 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5808 * gst/rtsp-server/rtsp-stream.c:
5809 stream: transports must already have been removed
5811 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5813 * gst/rtsp-server/rtsp-media.c:
5814 * gst/rtsp-server/rtsp-stream.c:
5815 * gst/rtsp-server/rtsp-stream.h:
5816 stream: improve join and leave of the pipeline
5818 Do the cleanup properly
5821 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5823 * gst/rtsp-server/rtsp-media.c:
5824 media: move unprepare below default implementation
5825 Makes it easier to find the default implementation
5827 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5829 * gst/rtsp-server/rtsp-media.c:
5830 media: signal unprepared when we actually finish
5832 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5834 * gst/rtsp-server/rtsp-media.c:
5835 media: no need to unlock, unprepare does that when needed
5837 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5839 * docs/libs/gst-rtsp-server-sections.txt:
5840 * gst/rtsp-server/rtsp-media-factory.h:
5841 * gst/rtsp-server/rtsp-media-mapping.c:
5842 * gst/rtsp-server/rtsp-media.h:
5843 * gst/rtsp-server/rtsp-params.c:
5844 * gst/rtsp-server/rtsp-server.c:
5845 * gst/rtsp-server/rtsp-session-pool.h:
5846 * gst/rtsp-server/rtsp-session.c:
5847 * gst/rtsp-server/rtsp-session.h:
5848 * gst/rtsp-server/rtsp-stream-transport.h:
5849 * gst/rtsp-server/rtsp-stream.h:
5852 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5854 * gst/rtsp-server/rtsp-client.c:
5855 * gst/rtsp-server/rtsp-media-mapping.h:
5856 * gst/rtsp-server/rtsp-media.c:
5857 * gst/rtsp-server/rtsp-media.h:
5858 * gst/rtsp-server/rtsp-server.h:
5859 * gst/rtsp-server/rtsp-stream.c:
5860 * gst/rtsp-server/rtsp-stream.h:
5861 rtsp: fix MTU setting
5862 Fix setting of the MTU. There is no need for a vmethod.
5864 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5869 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5872 configure: bump version number after refactoring
5874 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5876 * gst/rtsp-server/Makefile.am:
5877 * gst/rtsp-server/rtsp-client.c:
5878 * gst/rtsp-server/rtsp-client.h:
5879 * gst/rtsp-server/rtsp-media-factory-uri.c:
5880 * gst/rtsp-server/rtsp-media-factory.c:
5881 * gst/rtsp-server/rtsp-media-factory.h:
5882 * gst/rtsp-server/rtsp-media.c:
5883 * gst/rtsp-server/rtsp-media.h:
5884 * gst/rtsp-server/rtsp-sdp.c:
5885 * gst/rtsp-server/rtsp-session-media.c:
5886 * gst/rtsp-server/rtsp-session-media.h:
5887 * gst/rtsp-server/rtsp-session.c:
5888 * gst/rtsp-server/rtsp-session.h:
5889 * gst/rtsp-server/rtsp-stream-transport.c:
5890 * gst/rtsp-server/rtsp-stream-transport.h:
5891 * gst/rtsp-server/rtsp-stream.c:
5892 * gst/rtsp-server/rtsp-stream.h:
5893 rtsp: massive refactoring
5894 Make GObjects from the remaining simple structures.
5895 Remove GstRTSPSessionStream, it's not needed.
5896 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
5897 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
5898 a GstRTSPStream should be transported to a client.
5899 Rename GstRTSPMediaFactory::get_element -> create_element because that
5900 more accurately describes what it does.
5901 Make nice methods instead of poking in the structures.
5902 Move some methods inside the relevant object source code.
5903 Use GPtrArray to store objects instead of plain arrays, it is more
5904 natural and allows us to more easily clean up.
5905 Move the allocation of udp ports to the Stream object. The Stream object
5906 contains the elements needed to stream the media to a client.
5907 Improve the prepare and unprepare methods. Unprepare should now undo
5908 everything prepare did. Improve also async unprepare when doing EOS on
5909 shutdown. Make sure we always unprepare correctly.
5911 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
5913 * gst/rtsp-server/rtsp-client.c:
5914 rtsp-client: Unref server address clients connected to
5915 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
5917 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
5919 * gst/rtsp-server/rtsp-server.c:
5920 rtsp-server: don't ref server socket if it is NULL
5921 Fixes test_bind_already_in_use unit test again after commit 6a497440.
5922 https://bugzilla.gnome.org/show_bug.cgi?id=686644
5924 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
5926 * tests/check/Makefile.am:
5927 tests: Add libgio link dependency
5928 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
5930 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
5932 * gst/rtsp-server/rtsp-media-mapping.c:
5933 * gst/rtsp-server/rtsp-media-mapping.h:
5934 rtsp-media-mapping: rename find_media vfunc to find_factory
5935 The virtual method and class method should have the same name
5936 so it is correctly represented in GIR file
5937 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5939 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
5941 * gst/rtsp-server/rtsp-auth.c:
5942 * gst/rtsp-server/rtsp-client.c:
5943 * gst/rtsp-server/rtsp-media-factory-uri.c:
5944 * gst/rtsp-server/rtsp-media-factory.c:
5945 * gst/rtsp-server/rtsp-media-mapping.c:
5946 * gst/rtsp-server/rtsp-media.c:
5947 * gst/rtsp-server/rtsp-server.c:
5948 * gst/rtsp-server/rtsp-session-pool.c:
5949 * gst/rtsp-server/rtsp-session.c:
5950 rtsp-server: fixed comments and GIR annotations
5951 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5953 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
5955 * gst/rtsp-server/rtsp-media-mapping.c:
5956 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
5958 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
5960 * gst/rtsp-server/rtsp-server.c:
5961 rtsp-server: allow binding on port 0 (binds on a random port)
5963 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
5965 * gst/rtsp-server/rtsp-server.c:
5966 * gst/rtsp-server/rtsp-server.h:
5967 rtsp-server: add bound-port property
5968 bound-port can be used to retrieve the port number when the server is bound on
5969 port 0, which binds on a random port.
5971 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
5973 * gst/rtsp-server/rtsp-media-factory.c:
5974 * gst/rtsp-server/rtsp-media-factory.h:
5975 rtsp-media-factory: make ::get_element overridable by GI bindings
5976 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
5977 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
5978 as the invoker for ::get_element(), making it overridable by GI generated
5981 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
5983 * gst/rtsp-server/rtsp-media-factory-uri.c:
5984 rtsp-media-factory-uri: don't autoplug parsers in a loop
5985 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
5988 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
5990 * gst/rtsp-server/Makefile.am:
5991 Explicitly link against gio. Fix link error on mac.
5993 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
5995 * gst/rtsp-server/rtsp-session.c:
5996 session: add ttl to the transport header in SETUP
5997 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
5999 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
6001 * gst/rtsp-server/rtsp-client.c:
6002 * gst/rtsp-server/rtsp-client.h:
6003 * gst/rtsp-server/rtsp-media.c:
6004 client: Use client transport settings for multicast if allowed.
6005 This patch makes it possible for the client to send transport settings for
6006 multicast (destination && ttl). Client settings must be explicitly allowed or
6007 the server will use its own settings.
6008 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
6010 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
6013 Automatic update of common submodule
6014 From 6c0b52c to 6bb6951
6016 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
6018 * gst/rtsp-server/rtsp-client.c:
6019 rtsp-client: do not destroy the rtsp watch
6020 Don't destroy the client watch while dispatching. The rtsp watch is
6021 automatically destroyed after the rtsp watch function closed() has
6023 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
6025 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
6028 Automatic update of common submodule
6029 From 4f962f7 to 6c0b52c
6031 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
6033 * gst/rtsp-server/rtsp-media.c:
6034 media: fix check for seekability
6036 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6038 * gst/rtsp-server/rtsp-client.c:
6039 client: use more GIO
6040 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
6042 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6044 * gst/rtsp-server/rtsp-server.c:
6045 server: remove obsolete includes
6047 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
6049 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
6050 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
6051 be available in "on_new_ssrc". The transports are added in
6052 gst_rtsp_media_set_state when going to PLAYING state. However,
6053 "on_new_ssrc" might be called before this happens.
6054 https://bugzilla.gnome.org/show_bug.cgi?id=683304
6056 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
6058 * gst/rtsp-server/rtsp-client.c:
6059 * gst/rtsp-server/rtsp-client.h:
6060 rtsp-client: add signals for rtsp requests (fixes #683287)
6062 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
6064 * gst/rtsp-server/rtsp-client.c:
6065 * gst/rtsp-server/rtsp-client.h:
6066 add new-session signal to rtsp-client (fixes #683058)
6068 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
6071 Automatic update of common submodule
6072 From 668acee to 4f962f7
6074 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
6076 * gst/rtsp-server/rtsp-server.c:
6077 * tests/check/gst/rtspserver.c:
6078 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
6079 Do not assume that *error is set in g_socket_address_enumerator_next.
6080 Added test_bind_already_in_use unit-test.
6081 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
6083 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
6086 Automatic update of common submodule
6087 From 94ccf4c to 668acee
6089 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
6091 * gst/rtsp-server/rtsp-client.c:
6092 * gst/rtsp-server/rtsp-client.h:
6093 rtsp-client: make create_sdp virtual method
6094 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
6096 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6099 Automatic update of common submodule
6100 From 98e386f to 94ccf4c
6102 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6104 * gst/rtsp-server/rtsp-client.c:
6107 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
6109 * gst/rtsp-server/rtsp-client.c:
6110 * gst/rtsp-server/rtsp-client.h:
6111 * gst/rtsp-server/rtsp-server.c:
6112 * gst/rtsp-server/rtsp-server.h:
6113 rtsp-server: use an existing socket to establish HTTP tunnel
6114 Make it possible to transfer a socket from an HTTP server to be used as
6115 an RTSP over HTTP tunnel.
6117 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
6119 * gst/rtsp-server/rtsp-client.c:
6120 * gst/rtsp-server/rtsp-media.c:
6121 * gst/rtsp-server/rtsp-media.h:
6122 rtsp: Handle the blocksize parameter
6123 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
6125 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
6127 * tests/check/Makefile.am:
6128 * tests/check/gst/rtspserver.c:
6129 Have unit test get header from source dir, not installed dir
6130 This makes compilation of unit tests work in a build directory other
6131 than the source directory.
6132 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
6134 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
6136 * gst/rtsp-server/rtsp-media.c:
6137 rtsp-media: update for gst_element_make_from_uri() changes
6139 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
6142 * tests/Makefile.am:
6143 * tests/check/Makefile.am:
6144 * tests/check/gst/rtspserver.c:
6146 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
6148 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
6150 * gst/rtsp-server/rtsp-media.c:
6151 rtsp-media: don't collect media stats when going to NULL
6152 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
6154 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6156 * gst/rtsp-server/rtsp-client.c:
6157 client: don't leak transports
6159 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
6161 * gst/rtsp-server/rtsp-client.c:
6162 rtsp-client: free transport on no_stream in SETUP handler
6164 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
6166 * gst/rtsp-server/rtsp-client.c:
6167 rtsp-client: changed session media iteration
6168 In client_unlink_session: now don't iterate in session->medias
6169 list where items are removed by gst_rtsp_session_release_media.
6170 Instead, repeatedly remove the first item.
6172 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
6174 * gst/rtsp-server/rtsp-client.c:
6175 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
6176 GstRTSPSessionMedia is not a GObject type. When the
6177 GstRTSPSession is freed, it will free the media.
6179 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
6181 * gst/rtsp-server/rtsp-media-factory.c:
6182 factory: plug pad leak in collect_streams
6183 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
6184 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
6185 will take one reference, and the other reference will otherwise
6188 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
6191 configure: suppress some warnings when debug is disabled
6192 Warnings about unused variables should be suppressed if core has the
6193 debug system disabled.
6194 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
6196 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6198 * docs/libs/Makefile.am:
6199 docs: fix build in uninstalled setup
6200 Include gst-plugins-base libs properly.
6202 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
6204 * docs/libs/gst-rtsp-server.types:
6205 docs: include headers defining rtsp-server object types
6206 Fixes compiler warnings during docs build.
6207 https://bugzilla.gnome.org/show_bug.cgi?id=676824
6209 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
6212 configure: Add warning flags for compiler when configuring
6213 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
6215 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6218 Automatic update of common submodule
6219 From 03a0e57 to 98e386f
6221 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6224 Automatic update of common submodule
6225 From 1fab359 to 03a0e57
6227 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
6229 * gst/rtsp-server/rtsp-client.c:
6230 client: fix GSocketAddress leak in gst_rtsp_client_accept
6231 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
6233 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6236 Automatic update of common submodule
6237 From f1b5a96 to 1fab359
6239 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6242 Automatic update of common submodule
6243 From 92b7266 to f1b5a96
6245 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6248 Automatic update of common submodule
6249 From ec1c4a8 to 92b7266
6251 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6254 Automatic update of common submodule
6255 From 3429ba6 to ec1c4a8
6257 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
6259 * gst/rtsp-server/rtsp-auth.c:
6260 * gst/rtsp-server/rtsp-client.c:
6261 * gst/rtsp-server/rtsp-media-factory-uri.c:
6262 * gst/rtsp-server/rtsp-server.c:
6263 rtsp: fix compiler warnings
6264 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
6266 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6269 Automatic update of common submodule
6270 From dc70203 to 3429ba6
6272 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6274 * gst/rtsp-server/rtsp-client.c:
6275 * gst/rtsp-server/rtsp-media-factory.c:
6276 * gst/rtsp-server/rtsp-media-factory.h:
6277 * gst/rtsp-server/rtsp-media.c:
6278 * gst/rtsp-server/rtsp-media.h:
6279 * gst/rtsp-server/rtsp-server.c:
6280 * gst/rtsp-server/rtsp-server.h:
6281 * gst/rtsp-server/rtsp-session-pool.c:
6282 * gst/rtsp-server/rtsp-session-pool.h:
6283 rtsp-server: port to new thread API
6285 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6288 Automatic update of common submodule
6289 From 6db25be to dc70203
6291 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6293 * gst/rtsp-server/rtsp-auth.c:
6294 * gst/rtsp-server/rtsp-auth.h:
6295 * gst/rtsp-server/rtsp-client.c:
6296 rtsp-server: Fix compilation and compiler warnings
6298 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6302 * gst/rtsp-server/Makefile.am:
6303 configure: Modernize autotools setup a bit
6304 Also we now only create tar.bz2 and tar.xz tarballs.
6306 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6309 Automatic update of common submodule
6310 From 464fe15 to 6db25be
6312 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6315 Automatic update of common submodule
6316 From 7fda524 to 464fe15
6318 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6321 * docs/libs/Makefile.am:
6322 * docs/version.entities.in:
6324 * gst/rtsp-server/Makefile.am:
6325 * pkgconfig/Makefile.am:
6326 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
6327 * pkgconfig/gstreamer-rtsp-server.pc.in:
6328 * tests/Makefile.am:
6329 rtsp-server: Update versioning
6331 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6333 Merge remote-tracking branch 'origin/0.10'
6335 gst/rtsp-server/rtsp-session-pool.c
6337 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6339 * gst/rtsp-server/rtsp-session-pool.c:
6340 rtsp-server: Don't use deprecated GLib API
6342 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6344 Replace master with 0.11
6346 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6348 Merge branch 'master' into 0.11
6350 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6352 Merge branch 'master' into 0.11
6354 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
6357 A couple minor typo fixes
6359 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6361 * gst/rtsp-server/rtsp-media.c:
6362 media: fix state of the appqueue
6364 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6366 * gst/rtsp-server/rtsp-media-factory-uri.c:
6367 factory: use videoconvert
6369 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6371 * gst/rtsp-server/rtsp-media-factory-uri.c:
6372 factory: change to new style caps
6374 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6376 * gst/rtsp-server/rtsp-client.c:
6377 * gst/rtsp-server/rtsp-client.h:
6378 * gst/rtsp-server/rtsp-media-factory-uri.c:
6379 * gst/rtsp-server/rtsp-media.c:
6380 * gst/rtsp-server/rtsp-server.c:
6381 * gst/rtsp-server/rtsp-server.h:
6382 * gst/rtsp-server/rtsp-session-pool.c:
6383 rtsp-server: port to GIO
6386 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6389 configure: fix build
6391 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6394 docs: fix for gst_rtsp_server_set_port() -> _set_service()
6395 https://bugzilla.gnome.org/show_bug.cgi?id=666548
6397 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6400 * examples/Makefile.am:
6401 First rule of gst-rtsp-server club: don't talk about gst-phonon
6403 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6406 * pkgconfig/Makefile.am:
6407 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
6408 * pkgconfig/gst-rtsp-server.pc.in:
6409 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
6410 * pkgconfig/gstreamer-rtsp-server.pc.in:
6411 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
6412 For consistency with all other modules.
6414 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6416 * gst/rtsp-server/rtsp-client.c:
6417 rtsp-client: update for new map API
6419 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6422 * bindings/Makefile.am:
6423 * bindings/python/Makefile.am:
6424 * bindings/python/arg-types.py:
6425 * bindings/python/codegen/Makefile.am:
6426 * bindings/python/codegen/__init__.py:
6427 * bindings/python/codegen/argtypes.py:
6428 * bindings/python/codegen/code-coverage.py:
6429 * bindings/python/codegen/codegen.py:
6430 * bindings/python/codegen/definitions.py:
6431 * bindings/python/codegen/defsparser.py:
6432 * bindings/python/codegen/docextract.py:
6433 * bindings/python/codegen/docgen.py:
6434 * bindings/python/codegen/fileprefix.override:
6435 * bindings/python/codegen/fileprefixmodule.c:
6436 * bindings/python/codegen/h2def.py:
6437 * bindings/python/codegen/mergedefs.py:
6438 * bindings/python/codegen/mkskel.py:
6439 * bindings/python/codegen/override.py:
6440 * bindings/python/codegen/reversewrapper.py:
6441 * bindings/python/codegen/scmexpr.py:
6442 * bindings/python/rtspserver-types.defs:
6443 * bindings/python/rtspserver.defs:
6444 * bindings/python/rtspserver.override:
6445 * bindings/python/rtspservermodule.c:
6446 * bindings/python/test.py:
6448 python: remove pygst-based python bindings
6449 pygi is the future, apparently.
6451 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
6454 Automatic update of common submodule
6455 From c463bc0 to 7fda524
6457 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6460 Automatic update of common submodule
6461 From 2a59016 to c463bc0
6463 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6466 Automatic update of common submodule
6467 From 0807187 to 2a59016
6469 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6472 Automatic update of common submodule
6473 From 11f0cd5 to 0807187
6475 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6477 * examples/test-auth.c:
6478 example: update for new caps
6480 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6482 * examples/test-video.c:
6483 * gst/rtsp-server/rtsp-client.c:
6484 * gst/rtsp-server/rtsp-media-factory-uri.c:
6485 * gst/rtsp-server/rtsp-media.c:
6486 * gst/rtsp-server/rtsp-media.h:
6487 * gst/rtsp-server/rtsp-session.c:
6488 * gst/rtsp-server/rtsp-session.h:
6489 rtsp-server: port some more to 0.11
6491 Remove bufferlist stuff
6493 Add queue before appsink now that preroll-queue-len is gone.
6494 Update for request pad changes.
6496 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6498 Merge branch 'master' into 0.11
6500 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
6502 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6503 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
6504 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6506 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
6508 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6509 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
6510 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6512 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6514 Merge branch 'master' into 0.11
6516 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6518 * gst/rtsp-server/rtsp-media.c:
6519 * gst/rtsp-server/rtsp-media.h:
6520 media: add a seekable boolean
6521 Maintain the seekable state with a new variable instead of reusing the
6524 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
6526 * gst/rtsp-server/rtsp-media.c:
6527 Disallow seek in live media
6529 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6531 Merge branch 'master' into 0.11
6533 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
6535 * gst/rtsp-server/rtsp-server.c:
6536 #ifdef statements for windows socket creation were missing
6538 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
6541 Automatic update of common submodule
6542 From a39eb83 to 11f0cd5
6544 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
6547 Automatic update of common submodule
6548 From 605cd9a to a39eb83
6550 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6552 Merge branch 'master' into 0.11
6554 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6556 * gst/rtsp-server/rtsp-client.c:
6557 client: use method to access property
6559 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6561 * gst/rtsp-server/rtsp-media-factory.c:
6562 * gst/rtsp-server/rtsp-media-factory.h:
6563 media-factory: add protocols property
6564 Add a property to configure the allowed protocols in the media created from the
6567 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6569 * gst/rtsp-server/rtsp-media-factory.c:
6570 * gst/rtsp-server/rtsp-media-factory.h:
6571 media-factory: add media-configure signal
6572 Add signal to allow the application to configure the media after it was created
6575 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6577 * gst/rtsp-server/rtsp-client.c:
6578 client: use method to access property
6580 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6582 * gst/rtsp-server/rtsp-media-factory.c:
6583 * gst/rtsp-server/rtsp-media-factory.h:
6584 media-factory: add protocols property
6585 Add a property to configure the allowed protocols in the media created from the
6588 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6590 * gst/rtsp-server/rtsp-media-factory.c:
6591 * gst/rtsp-server/rtsp-media-factory.h:
6592 media-factory: add media-configure signal
6593 Add signal to allow the application to configure the media after it was created
6596 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6598 Merge branch 'master' into 0.11
6600 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6602 * gst/rtsp-server/rtsp-client.c:
6603 client: use media multicast group
6605 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6607 * gst/rtsp-server/rtsp-media-factory.h:
6608 * gst/rtsp-server/rtsp-server.h:
6609 * gst/rtsp-server/rtsp-session-pool.h:
6610 * gst/rtsp-server/rtsp-session.h:
6613 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6615 * gst/rtsp-server/rtsp-client.c:
6616 * gst/rtsp-server/rtsp-sdp.h:
6617 sdp: copy and free the server ip address
6618 Copy and free the server ip address to make memory management easier later.
6620 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6622 * gst/rtsp-server/rtsp-media-factory.c:
6623 media-factory: configure multicast in media
6625 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6627 * gst/rtsp-server/rtsp-media.c:
6628 * gst/rtsp-server/rtsp-media.h:
6629 media: add property for multicast group
6630 Add a property to configure the multicast group in the media.
6631 Based on patches from Marc Leeman and Robert Krakora.
6633 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6635 * gst/rtsp-server/rtsp-media-factory.c:
6636 * gst/rtsp-server/rtsp-media-factory.h:
6637 media-factory: add property for multicast group
6638 Add a property to configure the multicast group in the media factory.
6639 Based on patches from Marc Leeman and Robert Krakora.
6641 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6643 * gst/rtsp-server/rtsp-client.c:
6644 client: do configuration of transport in one place
6645 Move the configuration of the transport destination address to where we also
6646 configure the other bits.
6648 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6650 * gst/rtsp-server/rtsp-client.c:
6651 client: use media multicast group
6653 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6655 * gst/rtsp-server/rtsp-media-factory.h:
6656 * gst/rtsp-server/rtsp-server.h:
6657 * gst/rtsp-server/rtsp-session-pool.h:
6658 * gst/rtsp-server/rtsp-session.h:
6661 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6663 * gst/rtsp-server/rtsp-client.c:
6664 * gst/rtsp-server/rtsp-sdp.h:
6665 sdp: copy and free the server ip address
6666 Copy and free the server ip address to make memory management easier later.
6668 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6670 * gst/rtsp-server/rtsp-media-factory.c:
6671 media-factory: configure multicast in media
6673 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6675 * gst/rtsp-server/rtsp-media.c:
6676 * gst/rtsp-server/rtsp-media.h:
6677 media: add property for multicast group
6678 Add a property to configure the multicast group in the media.
6679 Based on patches from Marc Leeman and Robert Krakora.
6681 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6683 * gst/rtsp-server/rtsp-media-factory.c:
6684 * gst/rtsp-server/rtsp-media-factory.h:
6685 media-factory: add property for multicast group
6686 Add a property to configure the multicast group in the media factory.
6687 Based on patches from Marc Leeman and Robert Krakora.
6689 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6691 * gst/rtsp-server/rtsp-client.c:
6692 client: do configuration of transport in one place
6693 Move the configuration of the transport destination address to where we also
6694 configure the other bits.
6696 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6698 Merge branch 'master' into 0.11
6700 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6702 * gst/rtsp-server/rtsp-client.c:
6703 client: destroy pipeline on client disconnect with no prior TEARDOWN.
6704 The problem occurs when the client abruptly closes the connection without
6705 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
6706 server is where the pipeline gets torn down. Since this handler is not called,
6707 the pipeline remains and is up and running. Subsequent clients get their own
6708 pipelines and if the do not issue TEARDOWNs then those pipelines will also
6709 remain up and running. This is a resource leak.
6711 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6713 Merge branch 'master' into 0.11
6715 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
6717 * gst/rtsp-server/rtsp-media-factory.c:
6718 * gst/rtsp-server/rtsp-media-factory.h:
6719 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
6720 For example, it can be used to retrieve source elements like appsrc, in a more
6721 convenient way than subclassing get_element.
6723 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6725 Merge branch 'master' into 0.11
6727 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
6729 * gst/rtsp-server/rtsp-server.c:
6730 rtsp-server: hold on to reference while using object
6732 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6734 * gst/rtsp-server/rtsp-media.c:
6737 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6740 configure: use unstable api
6742 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
6744 * gst/rtsp-server/rtsp-client.c:
6745 client: fix reference counting
6747 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
6749 * gst/rtsp-server/rtsp-client.c:
6750 * gst/rtsp-server/rtsp-media.c:
6751 fix compiler warnings about unused variables
6753 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
6755 * examples/test-launch.c:
6756 * examples/test-readme.c:
6757 * examples/test-uri.c:
6758 * examples/test-video.c:
6759 examples: tell rtsp uri when ready
6761 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
6764 Automatic update of common submodule
6765 From 69b981f to 605cd9a
6767 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6769 * gst/rtsp-server/rtsp-client.c:
6770 client: update for buffer API change
6772 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6774 * gst/rtsp-server/Makefile.am:
6775 Makefile.am: 0.10 => @GST_MAJORMINOR@
6777 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6779 * gst/rtsp-server/rtsp-media-factory-uri.c:
6780 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
6782 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6784 * gst/rtsp-server/.gitignore:
6785 .gitignore: 0.10 => 0.11
6787 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6789 * gst/rtsp-server/Makefile.am:
6790 Makefile.am: 0.10 => @GST_MAJORMINOR@
6792 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6794 Merge branch 'master' into 0.11
6796 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
6799 Automatic update of common submodule
6800 From 9e5bbd5 to 69b981f
6802 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
6805 Automatic update of common submodule
6806 From fd35073 to 9e5bbd5
6808 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
6811 Automatic update of common submodule
6812 From 46dfcea to fd35073
6814 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6816 * gst/rtsp-server/rtsp-media-factory-uri.c:
6817 * gst/rtsp-server/rtsp-media.c:
6818 media: port to new caps API
6820 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6822 Merge branch 'master' into 0.11
6824 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
6826 * bindings/vala/gst-rtsp-server-0.10.vapi:
6827 Updated Vala bindings.
6828 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6830 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
6832 * gst/rtsp-server/rtsp-server.c:
6833 * gst/rtsp-server/rtsp-server.h:
6834 Add a signal for newly connected clients.
6835 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6837 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
6839 * bindings/python/rtspserver.override:
6840 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
6842 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6844 * gst/rtsp-server/Makefile.am:
6845 * gst/rtsp-server/rtsp-client.c:
6846 * gst/rtsp-server/rtsp-funnel.c:
6847 * gst/rtsp-server/rtsp-funnel.h:
6848 * gst/rtsp-server/rtsp-media.c:
6849 rtsp-server: port to 0.11
6851 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6856 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6858 Merge branch 'master' into 0.11
6863 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6866 Automatic update of common submodule
6867 From c3cafe1 to 46dfcea
6869 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
6871 * bindings/python/Makefile.am:
6872 * bindings/python/rtspserver.defs:
6873 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
6875 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
6877 * bindings/python/arg-types.py:
6878 python bindings: add GstRTSPUrlParam
6879 Needed to implement MediaFactory virtual proxies
6881 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
6883 * bindings/python/arg-types.py:
6884 python bindings: fix returning GstRTSPUrl types
6886 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
6888 * bindings/python/arg-types.py:
6889 python bindings: add arg type for GstRTSPUrl
6891 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
6893 * bindings/python/rtspserver.defs:
6894 python bindings: fix the definition of MediaFactory.collect_stream
6896 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
6899 Automatic update of common submodule
6900 From 1ccbe09 to c3cafe1
6902 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6905 Automatic update of common submodule
6906 From 193b717 to 1ccbe09
6908 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
6911 Automatic update of common submodule
6912 From b77e2bf to 193b717
6914 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6917 build: Include lcov.mak to allow test coverage report generation
6919 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6922 Automatic update of common submodule
6923 From d8814b6 to b77e2bf
6925 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6928 Automatic update of common submodule
6929 From 6aaa286 to d8814b6
6931 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
6934 Automatic update of common submodule
6935 From 6aec6b9 to 6aaa286
6937 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
6940 autogen: wingo signed comment
6942 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
6944 * gst/rtsp-server/rtsp-session-pool.c:
6945 session: use full charset for RTSP session ID
6946 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
6947 session ID more difficult.
6948 https://bugzilla.gnome.org/show_bug.cgi?id=643812
6950 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6952 * gst/rtsp-server/Makefile.am:
6953 rtsp-server: Don't install the funnel header
6955 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
6958 Automatic update of common submodule
6959 From 1de7f6a to 6aec6b9
6961 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6964 configure: require core/base 0.10.31
6965 Needed at least for gst_plugin_feature_rank_compare_func().
6967 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
6970 Automatic update of common submodule
6971 From f94d739 to 1de7f6a
6973 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6975 * gst/rtsp-server/rtsp-media.c:
6976 media: remove more unused code
6978 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6980 * gst/rtsp-server/rtsp-media.c:
6981 * gst/rtsp-server/rtsp-media.h:
6982 media: remove duplicate filtering
6983 Remove the duplicate filtering code now that we have a released -good version.
6984 Give a warning instead.
6986 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6988 * gst/rtsp-server/rtsp-media-factory.c:
6989 * gst/rtsp-server/rtsp-media.c:
6990 media: fix default buffer size
6992 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6994 * gst/rtsp-server/rtsp-media-factory.c:
6995 * gst/rtsp-server/rtsp-media-factory.h:
6996 media-factory: add property to configure the buffer-size
6997 Add a property to configure the kernel UDP buffer size.
6999 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7001 * gst/rtsp-server/rtsp-media.c:
7002 * gst/rtsp-server/rtsp-media.h:
7003 media: add property to configure kernel buffer sizes
7004 Add a property to configure the kernel UDP buffer size.
7006 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7009 configure: set PYGOBJECT_REQ before using it
7010 https://bugzilla.gnome.org/show_bug.cgi?id=640641
7012 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7015 docs: recursive into sub-directories on 'make upload'
7017 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7019 * docs/libs/gst-rtsp-server-docs.sgml:
7020 * docs/version.entities.in:
7021 docs: mention full version these docs are for, not just major-minor
7023 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7028 === release 0.10.8 ===
7030 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7035 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7037 * gst/rtsp-server/rtsp-server.c:
7038 rtsp-server: clarify docs a little
7040 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7042 * gst/rtsp-server/rtsp-media.c:
7043 media: init debug category before starting thread
7045 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7047 * gst/rtsp-server/rtsp-auth.c:
7048 auth: add realm to make it more spec compliant
7050 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7052 * gst/rtsp-server/rtsp-server.c:
7053 * gst/rtsp-server/rtsp-server.h:
7056 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7058 * examples/test-video.c:
7059 example: improve example docs a little
7061 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7063 * gst/rtsp-server/rtsp-server.c:
7064 server: ensure the watch has a ref to the server
7066 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7068 * gst/rtsp-server/rtsp-server.c:
7069 server: simpify channel function
7071 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7073 * gst/rtsp-server/rtsp-server.c:
7074 * gst/rtsp-server/rtsp-server.h:
7075 server: simplify management of channel and source
7076 We don't need to keep around the channel and source objects. Let the mainloop
7077 and the source manage the source and channel respectively.
7079 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7085 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7088 * tests/Makefile.am:
7089 * tests/test-cleanup.c:
7090 tests: add tests directory and cleanup test
7092 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7094 * gst/rtsp-server/rtsp-media-factory-uri.c:
7095 * gst/rtsp-server/rtsp-media-factory.c:
7096 * gst/rtsp-server/rtsp-media-mapping.c:
7097 * gst/rtsp-server/rtsp-media.c:
7098 * gst/rtsp-server/rtsp-session-pool.c:
7099 * gst/rtsp-server/rtsp-session.c:
7100 server: improve debugging in various objects
7102 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7104 * gst/rtsp-server/rtsp-server.c:
7105 server: chain up to the parent finalize
7107 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
7109 * bindings/python/rtspserver-types.defs:
7110 * bindings/python/rtspserver.defs:
7111 * bindings/python/rtspserver.override:
7112 * bindings/python/test.py:
7113 gst-rtsp-server: update python bindings
7115 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7117 * gst/rtsp-server/rtsp-client.c:
7118 client: use the response from the clientstate
7119 Create the response object only once and store in the client state.
7120 Make all methods use the state response,
7122 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7124 * gst/rtsp-server/rtsp-server.c:
7125 server: use signal to keep track of clients
7126 Keep track of all the clients that the server creates and remove them when they
7127 fire the 'closed' signal.
7129 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7131 * gst/rtsp-server/rtsp-client.c:
7132 * gst/rtsp-server/rtsp-client.h:
7133 client: emit signal when closing
7135 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7137 * examples/.gitignore:
7138 * examples/Makefile.am:
7139 * examples/test-auth.c:
7140 * examples/test-video.c:
7141 * gst/rtsp-server/rtsp-auth.c:
7142 * gst/rtsp-server/rtsp-auth.h:
7143 * gst/rtsp-server/rtsp-client.c:
7144 * gst/rtsp-server/rtsp-media-factory.c:
7145 * gst/rtsp-server/rtsp-media.c:
7146 * gst/rtsp-server/rtsp-media.h:
7147 * gst/rtsp-server/rtsp-session-pool.h:
7148 * gst/rtsp-server/rtsp-session.h:
7149 media: enable per factory authorisations
7150 Allow for adding a GstRTSPAuth on the factory and media level and check
7151 permissions when accessing the factory.
7152 Add hints to the auth methods for future more fine grained authorisation.
7153 Add example application for per factory authentication.
7155 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7157 * gst/rtsp-server/rtsp-auth.c:
7158 * gst/rtsp-server/rtsp-auth.h:
7159 * gst/rtsp-server/rtsp-client.c:
7160 * gst/rtsp-server/rtsp-client.h:
7161 * gst/rtsp-server/rtsp-params.c:
7162 * gst/rtsp-server/rtsp-params.h:
7163 rtsp-server: Pass ClientState structure arround
7164 Pass the collected information for the ongoing request in a GstRTSPClientState
7165 structure that we can then pass around to simplify the method arguments. This
7166 will also be handy when we implement logging functionality.
7168 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7170 * gst/rtsp-server/rtsp-media-factory.c:
7171 * gst/rtsp-server/rtsp-media-factory.h:
7172 media-factory: add methods to configure authorisation
7174 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7176 * gst/rtsp-server/rtsp-client.c:
7177 client: unref auth in finalize
7179 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7181 * gst/rtsp-server/rtsp-server.c:
7182 server: unref auth in finalize
7184 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7186 * docs/libs/gst-rtsp-server-docs.sgml:
7187 * docs/libs/gst-rtsp-server-sections.txt:
7188 * docs/libs/gst-rtsp-server.types:
7191 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7193 * gst/rtsp-server/rtsp-server.c:
7194 * gst/rtsp-server/rtsp-server.h:
7195 server: separate create and accept
7196 Create separate create and accept methods so that subclasses can create custom
7198 Configure the server in the client object and prepare for keeping track of
7201 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7203 * gst/rtsp-server/rtsp-client.c:
7204 * gst/rtsp-server/rtsp-client.h:
7205 client: add support for setting the server.
7206 Add support for keeping a ref to the server that started this client
7209 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7211 * gst/rtsp-server/rtsp-auth.c:
7212 auth: fix memleak and add some docs
7213 Fix a memleak of the basic auth token.
7214 Add docs for the helper function
7216 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7218 * gst/rtsp-server/rtsp-auth.c:
7219 * gst/rtsp-server/rtsp-auth.h:
7220 * gst/rtsp-server/rtsp-client.c:
7221 client: delegate setup of auth to the manager
7222 Delegate the configuration of the authentication tokens to the manager object
7225 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7227 * examples/test-video.c:
7228 * gst/rtsp-server/Makefile.am:
7229 * gst/rtsp-server/rtsp-auth.c:
7230 * gst/rtsp-server/rtsp-auth.h:
7231 * gst/rtsp-server/rtsp-client.c:
7232 * gst/rtsp-server/rtsp-client.h:
7233 * gst/rtsp-server/rtsp-server.c:
7234 * gst/rtsp-server/rtsp-server.h:
7235 auth: add authentication object
7236 Add an object that can check the authorization of requests.
7237 Implement basic authentication.
7238 Add example authentication to test-video
7240 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7242 * gst/rtsp-server/rtsp-server.c:
7243 * gst/rtsp-server/rtsp-server.h:
7244 server: move includes back
7245 the includes are needed for sockaddr_in.
7247 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7249 * gst/rtsp-server/rtsp-client.c:
7250 * gst/rtsp-server/rtsp-client.h:
7251 * gst/rtsp-server/rtsp-server.c:
7252 * gst/rtsp-server/rtsp-server.h:
7253 rtsp: move network includes where they are needed
7255 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
7257 * gst/rtsp-server/rtsp-media.h:
7258 rtsp-media.h: Minor corrections in comments.
7261 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
7264 Automatic update of common submodule
7265 From e572c87 to f94d739
7267 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7271 * docs/libs/.gitignore:
7272 * examples/.gitignore:
7273 * gst/rtsp-server/.gitignore:
7276 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7278 * docs/libs/Makefile.am:
7279 docs: We don't build ps/pdf for API reference docs
7281 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7284 Automatic update of common submodule
7285 From ccbaa85 to e572c87
7287 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7290 Automatic update of common submodule
7291 From 46445ad to ccbaa85
7293 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7295 * gst/rtsp-server/Makefile.am:
7296 * gst/rtsp-server/fs-funnel.c:
7297 * gst/rtsp-server/fs-funnel.h:
7298 * gst/rtsp-server/rtsp-funnel.c:
7299 * gst/rtsp-server/rtsp-funnel.h:
7300 * gst/rtsp-server/rtsp-media.c:
7301 funnel: rename fsfunnel to rtspfunnel
7302 Rename the funnel to avoid conflicts with the farsight one.
7304 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7306 * gst/rtsp-server/Makefile.am:
7307 * gst/rtsp-server/fs-funnel.c:
7308 * gst/rtsp-server/fs-funnel.h:
7309 * gst/rtsp-server/rtsp-media.c:
7310 rtsp-media: add and use fsfunnel
7311 Add a copy of fsfunnel to the build because input-selector removed the (broken)
7312 select-all property that we need.
7314 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7316 * gst/rtsp-server/Makefile.am:
7317 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
7318 Use PKG_CONFIG_PATH specified at configure time (if any) as well
7319 for the g-ir-compiler, rather than just assuming the env var has
7322 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7329 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
7331 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7334 * gst/rtsp-server/Makefile.am:
7335 gobject-introspection: fix g-i build for uninstalled setup
7336 Requires gst-plugins-base git (> 0.10.31.2).
7338 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7340 * examples/test-uri.c:
7341 examples: add some more options and comments
7343 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7345 * gst/rtsp-server/rtsp-media-factory-uri.c:
7346 factory-uri: use right property type
7348 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7350 * gst/rtsp-server/rtsp-media-factory-uri.c:
7351 factory-uri: attempt to configure buffer-lists
7352 Attempt to configure buffer lists in the payloader for improved performance.
7354 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7356 * gst/rtsp-server/rtsp-media.c:
7357 media: attempt to configure bigger UDP buffers
7358 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
7359 send buffers with high bitrate streams.
7361 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
7363 * gst/rtsp-server/rtsp-client.c:
7364 client: use the socket length from getsockname
7365 Use the length returned by getsockname to perform the getnameinfo call because
7366 the size can depend on the socket type and platform.
7369 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7371 * docs/libs/gst-rtsp-server-docs.sgml:
7372 * docs/libs/gst-rtsp-server-sections.txt:
7373 docs: add uri factory to the docs
7375 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7377 * gst/rtsp-server/rtsp-client.c:
7378 * gst/rtsp-server/rtsp-media.h:
7381 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7383 * gst/rtsp-server/rtsp-client.c:
7384 * gst/rtsp-server/rtsp-media.c:
7385 * gst/rtsp-server/rtsp-media.h:
7386 * gst/rtsp-server/rtsp-session.c:
7387 * gst/rtsp-server/rtsp-session.h:
7388 rtsp-server: add support for buffer lists
7389 Add support for sending bufferlists received from appsink.
7392 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7394 * gst/rtsp-server/rtsp-client.c:
7395 * gst/rtsp-server/rtsp-media.c:
7396 * gst/rtsp-server/rtsp-media.h:
7397 * gst/rtsp-server/rtsp-sdp.c:
7398 media: make method to retrieve the play range
7399 Make a method to retrieve the playback range so that we can conditionally create
7400 a different range for the SDP and the PLAY requests.
7402 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7404 * gst/rtsp-server/rtsp-media.c:
7405 * gst/rtsp-server/rtsp-media.h:
7406 media: add signal to notify of state changes
7408 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7410 * gst/rtsp-server/rtsp-client.h:
7411 client: cleanup headers
7413 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7415 * gst/rtsp-server/rtsp-client.c:
7418 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7420 * gst/rtsp-server/rtsp-media-factory-uri.c:
7421 * gst/rtsp-server/rtsp-media-factory-uri.h:
7422 factory-uri: add support for gstpay
7423 Add an option to prefer gstpay over decoder + raw payloader.
7425 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7427 * gst/rtsp-server/rtsp-media-factory-uri.c:
7428 * gst/rtsp-server/rtsp-media-factory-uri.h:
7429 factory-uri: rework the autoplugger.
7430 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
7433 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7435 * gst/rtsp-server/rtsp-media-factory-uri.c:
7436 factory-uri: use better factory filter
7437 Make better payloader filter based on autoplug rank and RTP use case.
7439 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7442 Automatic update of common submodule
7443 From 169462a to 46445ad
7445 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7447 * gst/rtsp-server/rtsp-server.c:
7448 server: set SO_REUSEADDR before bind
7449 Set the SO_REUSEADDR _before_ bind() to make it actually work.
7451 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7453 * gst/rtsp-server/rtsp-media.c:
7454 * gst/rtsp-server/rtsp-media.h:
7455 media: emit prepared signal when prepared
7456 Make a 'prepared' signal and emit it when we successfully prepared the element.
7457 This signal can be used to configure the media object after it has been prepared
7460 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
7463 Automatic update of common submodule
7464 From 011bcc8 to 169462a
7466 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
7468 python an optional dependency
7469 * configure.ac: Move up valgrind and g-i checks. Make the python
7470 dependency optional, as it was before.
7472 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7474 Merge branch 'master' into 0.11
7479 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7481 * gst/rtsp-server/rtsp-media.c:
7482 media: update range when active clients changed
7483 When we changed the number of active clients, update the current range
7484 information because we want the second client connecting to a shared resource
7485 continue from where the stream currently.
7487 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7489 * gst/rtsp-server/rtsp-media-factory-uri.c:
7490 * gst/rtsp-server/rtsp-media-factory-uri.h:
7491 factory-uri: add colorspace and fix pt
7492 Rework the way we pass data to the autoplugger.
7493 When we have raw caps, plug a converter element to make pluggin to raw
7494 payloaders more successful.
7495 Make sure all dynamically plugged payloaders have a unique payload types.
7497 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7499 * examples/Makefile.am:
7500 * examples/test-uri.c:
7501 example: add example of the uri factory
7503 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7505 * gst/rtsp-server/Makefile.am:
7506 * gst/rtsp-server/rtsp-media-factory-uri.c:
7507 * gst/rtsp-server/rtsp-media-factory-uri.h:
7508 * gst/rtsp-server/rtsp-server.h:
7509 factory-uri: add a factory to stream any URI
7510 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
7513 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7515 * gst/rtsp-server/rtsp-media.c:
7516 * gst/rtsp-server/rtsp-media.h:
7517 media: ignore spurious ASYNC_DONE messages
7518 When we are dynamically adding pads, the addition of the udpsrc elements will
7519 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
7520 the real ASYNC_DONE when everything is prerolled.
7522 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7524 * gst/rtsp-server/rtsp-media-factory.c:
7525 * gst/rtsp-server/rtsp-media-factory.h:
7526 media-factory: make lock macro
7528 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
7530 * gst/rtsp-server/rtsp-client.c:
7531 rtsp-server: Remove unused variable and dead assignment
7533 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
7535 * examples/test-launch.c:
7536 * examples/test-mp4.c:
7537 * examples/test-ogg.c:
7538 * examples/test-readme.c:
7539 * examples/test-sdp.c:
7540 * examples/test-video.c:
7541 examples: Run gst-indent
7543 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
7545 * gst/rtsp-server/rtsp-client.c:
7546 * gst/rtsp-server/rtsp-media-factory.c:
7547 * gst/rtsp-server/rtsp-media-mapping.c:
7548 * gst/rtsp-server/rtsp-media.c:
7549 * gst/rtsp-server/rtsp-params.c:
7550 * gst/rtsp-server/rtsp-sdp.c:
7551 * gst/rtsp-server/rtsp-server.c:
7552 * gst/rtsp-server/rtsp-session-pool.c:
7553 * gst/rtsp-server/rtsp-session.c:
7554 rtsp-server: Run gst-indent
7555 Since it wasn't using the upstream common previously, there was no
7556 indentation check before commiting.
7558 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
7560 * gst/rtsp-server/rtsp-media-mapping.h:
7561 * gst/rtsp-server/rtsp-media.c:
7562 * gst/rtsp-server/rtsp-media.h:
7563 * gst/rtsp-server/rtsp-sdp.c:
7564 * gst/rtsp-server/rtsp-session-pool.h:
7565 * gst/rtsp-server/rtsp-session.c:
7566 * gst/rtsp-server/rtsp-session.h:
7567 rtsp-server: Some more doc fixups
7569 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7572 Makefile: Add cruft-cleaning support
7574 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7579 * docs/libs/Makefile.am:
7580 * docs/libs/gst-rtsp-server-docs.sgml:
7581 * docs/libs/gst-rtsp-server-sections.txt:
7582 * docs/libs/gst-rtsp-server.types:
7583 * docs/version.entities.in:
7584 docs: Add gtk-doc build system
7586 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7588 * gst/rtsp-server/Makefile.am:
7589 Makefile.am: Use standard GIR make behaviour
7591 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7595 autogen/configure: Bring more in sync to standard gst module behaviour
7597 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7599 * gst/rtsp-server/rtsp-media.c:
7600 media: warn and fail when gstrtpbin is not found
7602 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7605 configure: open 0.11 branch
7607 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
7611 Add common submodule
7613 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
7616 * common/Makefile.am:
7617 * common/c-to-xml.py:
7619 * common/coverage/coverage-report-entry.pl:
7620 * common/coverage/coverage-report.pl:
7621 * common/coverage/coverage-report.xsl:
7622 * common/coverage/lcov.mak:
7623 * common/gettext.patch:
7624 * common/glib-gen.mak:
7625 * common/gst-autogen.sh:
7626 * common/gst-xmlinspect.py:
7628 * common/gstdoc-scangobj:
7629 * common/gtk-doc-plugins.mak:
7630 * common/gtk-doc.mak:
7631 * common/m4/.gitignore:
7632 * common/m4/Makefile.am:
7634 * common/m4/as-ac-expand.m4:
7635 * common/m4/as-auto-alt.m4:
7636 * common/m4/as-compiler-flag.m4:
7637 * common/m4/as-compiler.m4:
7638 * common/m4/as-docbook.m4:
7639 * common/m4/as-libtool-tags.m4:
7640 * common/m4/as-libtool.m4:
7641 * common/m4/as-python.m4:
7642 * common/m4/as-scrub-include.m4:
7643 * common/m4/as-version.m4:
7644 * common/m4/ax_create_stdint_h.m4:
7645 * common/m4/check.m4:
7646 * common/m4/glib-gettext.m4:
7647 * common/m4/gst-arch.m4:
7648 * common/m4/gst-args.m4:
7649 * common/m4/gst-check.m4:
7650 * common/m4/gst-debuginfo.m4:
7651 * common/m4/gst-default.m4:
7652 * common/m4/gst-doc.m4:
7653 * common/m4/gst-error.m4:
7654 * common/m4/gst-feature.m4:
7655 * common/m4/gst-function.m4:
7656 * common/m4/gst-gettext.m4:
7657 * common/m4/gst-glib2.m4:
7658 * common/m4/gst-libxml2.m4:
7659 * common/m4/gst-plugindir.m4:
7660 * common/m4/gst-valgrind.m4:
7661 * common/m4/gtk-doc.m4:
7662 * common/m4/introspection.m4:
7664 * common/mangle-tmpl.py:
7665 * common/plugins.xsl:
7667 * common/release.mak:
7668 * common/scangobj-merge.py:
7669 * common/upload.mak:
7670 common: Remove static version
7672 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
7674 * common/m4/introspection.m4:
7675 Update introspection.m4 to match usage
7677 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7681 Remove old stuff from the README
7683 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7688 === release 0.10.7 ===
7690 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7695 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7697 * examples/test-ogg.c:
7698 test-ogg: remove parsers
7699 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
7700 buffers with timestamps. Using the parsers also seems to break things.
7702 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7704 * bindings/vala/gst-rtsp-server-0.10.vapi:
7705 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7706 Updated Vala bindings
7708 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7710 * common/m4/introspection.m4:
7712 * gst/rtsp-server/Makefile.am:
7713 Added initial gobject-introspection support
7715 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7717 * gst/rtsp-server/rtsp-media-factory.c:
7718 media-factory: don't use host for shared hash key
7719 When we generate the key to share made between connections, don't include the
7720 host used to connect so that we can share media even if between clients that
7721 connected with localhost and ones with the ip address.
7723 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7725 * bindings/vala/Makefile.am:
7726 build: fix distcheck
7728 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7730 * bindings/vala/gst-rtsp-server-0.10.vapi:
7731 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7732 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7733 Update Vala bindings
7735 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7737 * bindings/vala/Makefile.am:
7739 Fix configure checks and installation location for Vala bindings
7742 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7747 === release 0.10.6 ===
7749 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7752 configure: release 0.10.6
7754 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7756 * gst/rtsp-server/rtsp-media.c:
7757 media: help the compiler a little
7759 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7761 * gst/rtsp-server/rtsp-media.c:
7762 * gst/rtsp-server/rtsp-media.h:
7763 * gst/rtsp-server/rtsp-session.c:
7764 media: cleanup media transport before freeing
7765 Cleanup the media transport data before freeing. In particular, remove the qdata
7766 from the rtpsource object.
7768 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7770 * gst/rtsp-server/rtsp-media-factory.c:
7771 * gst/rtsp-server/rtsp-media-factory.h:
7772 * gst/rtsp-server/rtsp-media.c:
7773 * gst/rtsp-server/rtsp-media.h:
7774 media-factory: add eos-shutdown property
7775 Add an eos-shutdown property that will send an EOS to the pipeline before
7776 shutting it down. This allows for nice cleanup in case of a muxer.
7779 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7781 * gst/rtsp-server/rtsp-media.c:
7782 * gst/rtsp-server/rtsp-media.h:
7783 media: use multiudpsink send-duplicates when we can
7784 If we have a new enough multiudpsink with the send-duplicates property, use this
7785 instead of doing our own filtering. Our custom filtering code should eventually
7786 be removed when we can depend on a released -good.
7788 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7790 * gst/rtsp-server/rtsp-media.c:
7791 media: don't leak destinations
7792 Refactor and cleanup the destinations array when the stream is destroyed.
7794 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7796 * gst/rtsp-server/rtsp-media.c:
7797 * gst/rtsp-server/rtsp-media.h:
7798 media: don't add udp addresses multiple times
7799 Keep track of the udp addresses we added to udpsink and never add the same udp
7800 destination twice. This avoids duplicate packets when using multicast.
7802 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7804 * gst/rtsp-server/rtsp-server.c:
7805 server: disable use of SO_LINGER
7806 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
7807 server close()s the connection.
7809 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7811 * gst/rtsp-server/rtsp-server.c:
7812 server: use 5 second linger period in SO_LINGER
7813 Wait 5 seconds before clearing the send buffers and reseting the connection with
7814 the client when we do a close. This should be enough time to get the message to
7818 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
7820 * gst/rtsp-server/rtsp-server.c:
7821 server: use SO_LINGER
7822 SO_LINGER on the socket will make sure that any pending data on the socket is
7823 flushed ASAP and that the socket connection is reset. This makes sure that the
7824 socket can be reused immediately.
7827 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7830 README: add blurb about shared media factories
7832 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
7834 * gst/rtsp-server/rtsp-media.c:
7835 Add stdlib.h for atoi()
7837 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7839 * bindings/python/Makefile.am:
7840 * bindings/vala/Makefile.am:
7841 build: distcheck fixes
7842 Fix 'make distcheck', somewhat (it still fails because it tries to
7843 install files into /usr/share/vala/vapi/ irrespective of the
7846 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7849 configure: bump core/base requirements to released version
7850 Makes things less confusing for people.
7852 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7855 configure: fail if GStreamer core/base requirements are not met
7857 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7859 * gst/rtsp-server/rtsp-client.c:
7860 client: improve client cleanups
7861 Make sure the session does not timeout when using TCP. We need to do this
7862 because quicktime player does not send RTCP for some reason in tunneled
7864 Refactor some cleanup code.
7867 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7869 * gst/rtsp-server/rtsp-session.c:
7870 * gst/rtsp-server/rtsp-session.h:
7871 session: add support for prevent session timeouts
7872 Add an atomix counter to prevent session timeouts when we are, for example,
7875 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7877 * gst/rtsp-server/rtsp-client.c:
7878 client: fix unlink on session timeouts
7879 When our session times out, make sure we unlink all streams in this
7881 Remove the tunnelid when closing the connection.
7883 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7885 * gst/rtsp-server/rtsp-session.c:
7886 session: small cleanups
7888 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7890 * gst/rtsp-server/rtsp-client.c:
7891 client: handle lost_tunnel callbacks
7892 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
7893 hashtable so that we can reuse it for when the client reopens the POST
7895 Close the connection after a TEARDOWN.
7896 Make sure or watchid is cleared when the watch is removed.
7899 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7901 * gst/rtsp-server/rtsp-client.c:
7902 * gst/rtsp-server/rtsp-media.c:
7903 * gst/rtsp-server/rtsp-sdp.c:
7904 rtsp-server: add more support for multicast
7906 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7909 * gst/rtsp-server/rtsp-media.c:
7910 * gst/rtsp-server/rtsp-media.h:
7911 media: allow configuration of allowed lower transport
7913 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7915 * gst/rtsp-server/rtsp-client.h:
7916 * gst/rtsp-server/rtsp-media.c:
7917 * gst/rtsp-server/rtsp-media.h:
7918 * gst/rtsp-server/rtsp-sdp.c:
7919 * gst/rtsp-server/rtsp-sdp.h:
7920 * gst/rtsp-server/rtsp-server.c:
7921 rtsp: keep track of server ip and ipv6
7922 Keep track of how the client connected to the server and setup the udp ports
7923 with the same protocol.
7924 Copy the server ip address in the SDP so that clients can send RTCP back to
7927 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7929 * gst/rtsp-server/rtsp-session.c:
7932 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7934 * gst/rtsp-server/rtsp-client.c:
7935 client: use right size for malloc
7937 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7939 * gst/rtsp-server/rtsp-server.c:
7940 server: comment ipv6 server listening address
7942 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7944 * gst/rtsp-server/rtsp-media.c:
7945 media: allow for ipv6 sockets
7947 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7949 * gst/rtsp-server/rtsp-server.c:
7950 * gst/rtsp-server/rtsp-server.h:
7951 server: rework server part
7952 Allow setting a bind address, make sure we can deal with ipv6.
7953 Remove the port property and change with the service property.
7955 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7957 * gst/rtsp-server/rtsp-media.h:
7958 media: update comments a little
7960 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7962 * gst/rtsp-server/rtsp-client.c:
7963 client: make content-base better
7964 Use the URI formatting functions to make a content-base. Also make sure that
7965 there is a trailing / at the end.
7967 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7969 * gst/rtsp-server/rtsp-client.c:
7970 client: guard against invalid paths
7972 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7974 * examples/test-video.c:
7975 test: catch server bind errors
7977 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
7979 * gst/rtsp-server/rtsp-media.c:
7980 rtspmedia: emit "unprepared" if _prepare fails.
7981 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
7982 media object is removed from its factory's cache.
7984 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7986 * gst/rtsp-server/rtsp-media.c:
7987 media: collect media position when seek completes
7989 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
7991 * gst/rtsp-server/rtsp-client.c:
7992 client: call unlink_streams in client finalize
7995 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7997 * gst/rtsp-server/rtsp-media.c:
7998 media: limit the time to wait to something huge
7999 Avoid waiting forever but limit the timeout to 20 seconds.
8001 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8003 * gst/rtsp-server/rtsp-sdp.c:
8004 sdp: reindent and check for prepared status
8006 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8008 * gst/rtsp-server/rtsp-media.c:
8009 * gst/rtsp-server/rtsp-media.h:
8010 * gst/rtsp-server/rtsp-session.c:
8011 media: avoid doing _get_state() for state changes
8012 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
8013 until the media is prerolled or in error. This avoids doing a blocking call of
8014 gst_element_get_state() that can cause lockups when there is an error.
8017 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8019 * gst/rtsp-server/rtsp-media.c:
8022 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8024 * gst/rtsp-server/rtsp-media-factory.c:
8025 media-factory: better error handling
8026 Improve the error handling a bit.
8028 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8030 * gst/rtsp-server/rtsp-client.c:
8031 client: rework transport parsing
8032 Rework the transport parsing code so that we can ignore transports we don't
8033 support instead of just picking the first one we can parse.
8034 Configure a (for now hardcoded) destination for multicast transports.
8036 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8038 * gst/rtsp-server/rtsp-media.c:
8039 media: set multicast sink parameters
8040 Disable loop and automatic multicast join on the udpsink elements.
8041 Add some more debug info.
8042 Reset some state variables in the right place.
8043 Use the right port numbers for multicast.
8045 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8047 * gst/rtsp-server/rtsp-session.c:
8048 session: handle transport setup correctly
8049 Handle UDP, MCAST and TCP transport negotiation more correctly.
8050 Store the server session SSRC in the transport.
8052 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8054 * gst/rtsp-server/rtsp-client.c:
8055 rtsp-client: implement error_full
8056 Implement error_full to avoid some segfaults when the rtspconnection calls it.
8059 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8062 * gst/rtsp-server/rtsp-client.c:
8063 * gst/rtsp-server/rtsp-server.c:
8064 docs: update docs and comments
8066 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
8068 * gst/rtsp-server/rtsp-sdp.c:
8069 sdp: make server work better when behind a proxy
8071 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8073 * gst/rtsp-server/rtsp-client.c:
8074 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
8076 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8078 * gst/rtsp-server/rtsp-client.c:
8079 * gst/rtsp-server/rtsp-media-factory.c:
8080 * gst/rtsp-server/rtsp-media-mapping.c:
8081 * gst/rtsp-server/rtsp-media.c:
8082 * gst/rtsp-server/rtsp-server.c:
8083 * gst/rtsp-server/rtsp-session-pool.c:
8084 * gst/rtsp-server/rtsp-session.c:
8085 Use GStreamer's debugging subsystem
8087 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8089 * gst/rtsp-server/rtsp-media-factory.c:
8090 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
8092 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8097 === release 0.10.5 ===
8099 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8104 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8107 configure: bump required versions
8109 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
8111 * gst/rtsp-server/rtsp-client.c:
8112 client: call weak-unref on client->sessions from finalize
8115 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8117 * gst/rtsp-server/rtsp-media.c:
8118 media: Fixed crasher where caps got unref'ed too often
8120 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8123 * pkgconfig/.gitignore:
8124 * pkgconfig/Makefile.am:
8125 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
8126 Added pkg-config file to use gst-rtsp-server uninstalled
8128 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8130 * gst/rtsp-server/rtsp-media.c:
8131 media: add some docs
8133 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
8135 * gst/rtsp-server/rtsp-client.c:
8136 rtsp: Use gst_rtsp_watch_send_message().
8137 Use gst_rtsp_watch_send_message() since the old API which used
8138 gst_rtsp_watch_queue_message() has been deprecated.
8140 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8145 === release 0.10.4 ===
8147 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8152 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8154 * gst/rtsp-server/rtsp-client.c:
8155 * gst/rtsp-server/rtsp-session.c:
8156 * gst/rtsp-server/rtsp-session.h:
8157 rtsp: allocate channels in TCP mode
8158 When the client does not provide us with channels in TCP mode, allocate channels
8161 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8163 * gst/rtsp-server/rtsp-client.c:
8164 client: don't crash when tunnelid is missing
8165 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
8166 don't crash but return an error response to the client.
8169 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8171 * bindings/vala/gst-rtsp-server-0.10.vapi:
8172 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8173 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8174 bindings: update vala bindings with new method
8176 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8178 * gst/rtsp-server/rtsp-session-pool.c:
8179 * gst/rtsp-server/rtsp-session-pool.h:
8180 sessionpool: add function to filter sessions
8181 Add generic function to retrieve/remove sessions.
8183 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8186 configure: bump core/base requirements to release
8188 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8190 * gst/rtsp-server/rtsp-media.c:
8191 media: fix indentation
8193 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8195 * gst/rtsp-server/rtsp-media.c:
8196 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
8198 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8200 * gst/rtsp-server/rtsp-media.c:
8201 set state and remove elements of media in for loop
8203 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
8205 * bindings/vala/gst-rtsp-server-0.10.vapi:
8206 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8207 Added gst_rtsp_media_remove_elements function to Vala bindings
8209 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
8211 * gst/rtsp-server/rtsp-media.c:
8212 * gst/rtsp-server/rtsp-media.h:
8213 Added gst_rtsp_media_remove_elements function
8215 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
8217 * gst/rtsp-server/rtsp-media.c:
8218 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
8220 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8222 * bindings/vala/gst-rtsp-server-0.10.vapi:
8223 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8224 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8225 Updated Vala bindings
8227 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8229 * gst/rtsp-server/rtsp-media.c:
8230 * gst/rtsp-server/rtsp-media.h:
8231 Added vmethod unprepare to GstRTSPMedia
8232 The default implementation sets the state of the pipeline to GST_STATE_NULL
8234 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8236 * gst/rtsp-server/rtsp-media-factory.c:
8237 * gst/rtsp-server/rtsp-media-factory.h:
8238 Made collect_streams function public
8240 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8242 * gst/rtsp-server/rtsp-media-factory.c:
8243 * gst/rtsp-server/rtsp-media-factory.h:
8244 * gst/rtsp-server/rtsp-media.c:
8245 Added vmethod create_pipeline to GstRTSPMediaFactory
8246 The pipeline is created in this method and the GstRTSPMedia's element is added to it
8248 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8250 * gst/rtsp-server/rtsp-client.c:
8251 client: use g_source_destroy()
8252 We need to use g_source_destroy() because we might have added the source to a
8253 different main context than the default one.
8255 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8257 * gst/rtsp-server/Makefile.am:
8258 * gst/rtsp-server/rtsp-client.c:
8259 * gst/rtsp-server/rtsp-params.c:
8260 * gst/rtsp-server/rtsp-params.h:
8261 rtsp: prepare for handling GET/SET_PARAMETER
8262 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
8264 Fix return codes of handlers.
8266 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8268 * gst/rtsp-server/rtsp-media.c:
8269 media: don't leak session pads
8271 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8273 * gst/rtsp-server/rtsp-media.c:
8274 media: clean up the messages a bit
8276 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8278 * gst/rtsp-server/rtsp-sdp.c:
8279 sdp: warn and skip streams without media
8281 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8283 * bindings/vala/gst-rtsp-server-0.10.vapi:
8284 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8285 vala: Fixed typo in header file of RTSPMediaStream
8287 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8289 * gst/rtsp-server/rtsp-media.c:
8292 Make dumping RTCP stats configurable
8294 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8296 * gst/rtsp-server/rtsp-media.c:
8297 media: be less verbose and leak less
8299 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8301 * gst/rtsp-server/rtsp-media.c:
8302 media: don't leak the destination address
8304 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8306 * gst/rtsp-server/rtsp-client.c:
8307 * gst/rtsp-server/rtsp-media.c:
8308 * gst/rtsp-server/rtsp-media.h:
8309 * gst/rtsp-server/rtsp-session.c:
8310 * gst/rtsp-server/rtsp-session.h:
8311 rtsp: use RTCP to keep the session alive
8312 Use the RTCP rtcp-from stats field to find the associated session and use this
8313 to keep the session alive.
8315 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8317 * gst/rtsp-server/rtsp-session.c:
8318 session: add 5sec to the real session timeout
8319 Allow the session to live 5sec longer before really timing out. This should give
8320 clients some extra time to keep the session active.
8322 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8324 * gst/rtsp-server/rtsp-client.c:
8325 client: replay OK to GET/SET_PARAMETER
8326 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
8327 so that we return OK for those requests.
8329 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8331 * gst/rtsp-server/rtsp-media.c:
8332 * gst/rtsp-server/rtsp-media.h:
8333 media: keep track of active transports
8334 Keep track of which transport is active to avoid closing the connection too
8336 Remove the destination transport also when going to NULL.
8337 Print some stats about the SDES and other RTCP messages we receive from the
8340 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8342 * examples/.gitignore:
8343 * examples/Makefile.am:
8344 * examples/test-sdp.c:
8345 example: add SDP relay example
8347 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8349 * gst/rtsp-server/rtsp-media.c:
8350 media: also count active TCP connections
8352 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8354 * gst/rtsp-server/rtsp-media-factory.c:
8355 * gst/rtsp-server/rtsp-media.c:
8356 * gst/rtsp-server/rtsp-media.h:
8357 rtsp: add support for dynamic elements
8358 Add support for dynamic elements.
8359 Don't set live pipelines back to paused.
8361 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8363 * gst/rtsp-server/rtsp-sdp.c:
8364 sdp: don't add encoding name when absent in caps
8366 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8368 * gst/rtsp-server/rtsp-client.c:
8369 client: warn when we can't do RTP-Info
8371 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8373 * gst/rtsp-server/rtsp-media-factory.c:
8374 factory: factor out the stream construction
8376 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8378 * gst/rtsp-server/rtsp-client.c:
8379 client: only add RTP-Info when we have the info
8380 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
8383 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8388 === release 0.10.3 ===
8390 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8394 - Fixes a bug where it put the wrong verion in pkgconfig
8395 - Link RTP and RTCP sources
8397 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8399 * gst/rtsp-server/rtsp-media.c:
8400 * gst/rtsp-server/rtsp-media.h:
8401 media: link the RTP udpsrc to the session manager
8402 Link the RTP udpsrc and the appsrc to the session manager so that they don't
8403 shut down when the client sends a packet to open firewalls.
8405 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8407 * pkgconfig/gst-rtsp-server.pc.in:
8408 Don't use hard-coded version number in pkg-config file
8410 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8415 === release 0.10.2 ===
8417 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8422 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8425 * common/m4/.gitignore:
8426 * examples/.gitignore:
8427 * pkgconfig/.gitignore:
8428 add some .gitignore files
8430 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8432 * gst/rtsp-server/rtsp-media.c:
8433 media: seek to key frames
8435 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8437 * gst/rtsp-server/rtsp-media.c:
8438 media: emit the unprepared signal by id
8439 Emit the unprepared signal by id instead of name and set the media as
8442 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8444 * gst/rtsp-server/rtsp-media.c:
8445 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
8447 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8449 * gst/rtsp-server/rtsp-server.c:
8450 Added finalize function to GstRTPSPServer to unref session pool and media mapping
8452 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8454 * bindings/vala/gst-rtsp-server-0.10.vapi:
8455 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8456 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8457 Updated vala bindings
8459 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8461 * gst/rtsp-server/Makefile.am:
8462 * gst/rtsp-server/rtsp-client.c:
8463 * gst/rtsp-server/rtsp-media.c:
8464 server: use appsink and appsrc with the API
8465 Use the appsink/appsrc API instead of the signals for higher
8468 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8470 * examples/test-ogg.c:
8471 tests: set the payload type correctly
8473 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8475 * gst/rtsp-server/rtsp-media-factory.c:
8476 factory: connect to the unprepare signal
8477 Connect to the unprepare signal for non-reusable media so that we can remove
8478 them from the cache.
8480 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8482 * gst/rtsp-server/rtsp-media.c:
8483 * gst/rtsp-server/rtsp-media.h:
8484 media: add signal to notify of unprepare
8486 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8488 * gst/rtsp-server/rtsp-media.c:
8489 * gst/rtsp-server/rtsp-media.h:
8490 media: more work on making the media shared
8491 Add a reusable flag to medias, indicating that they can be reused after a state
8495 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8497 * examples/test-readme.c:
8498 examples: mark the example as shared for testing
8500 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8502 * gst/rtsp-server/rtsp-media.c:
8503 * gst/rtsp-server/rtsp-media.h:
8504 client: support shared media
8505 Always perform the state actions even if the target state of the pipeline is
8506 already correct, we still want to add/remove the transports when we are dealing
8508 Keep a counter of the number of active transports for a media so that we can use
8509 this to perform a state change when needed.
8510 Perform a state change of the pipeline only when the first transport was added
8511 or when there are no active transports.
8513 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8515 * gst/rtsp-server/rtsp-client.c:
8516 client: fix refcounting crasher
8517 Don't need to remove the weak refs in the finalize methods, they are already
8518 removed in the dispose.
8519 Don't register the callback with a DestroyNofity.
8521 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8523 * gst/rtsp-server/rtsp-client.c:
8524 Fix rtsp client refcount management in TCP mode.
8525 Don't unref a client ref we never had. Fixes an unref
8526 of an already-free client object after a client
8527 teardown request for me.
8529 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8531 * gst/rtsp-server/rtsp-session.c:
8532 docs: fix typo in API docs
8534 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8536 * gst/rtsp-server/rtsp-media.c:
8538 Keep the udp sources in playing even if we go to paused. unlock the sources when
8540 Add some more debug info.
8541 Only seek when we need to.
8542 Keep track of the position when we go to paused.
8544 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8546 * gst/rtsp-server/rtsp-client.c:
8547 * gst/rtsp-server/rtsp-media.c:
8548 * gst/rtsp-server/rtsp-media.h:
8549 Add beginnings of seeking.
8550 Parse the Range header and perform a seek on the pipeline for the requested
8551 position. It's disabled currently until I figure out what's going wrong.
8553 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8555 * gst/rtsp-server/rtsp-client.c:
8556 allow pause requests for now.
8559 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8561 * gst/rtsp-server/rtsp-client.c:
8562 Remove weak ref on the session in teardown
8563 We need to remove our weakref from the session when we do a teardown because
8564 else we close the TCP connection prematurely.
8566 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8568 * gst/rtsp-server/rtsp-client.c:
8569 * gst/rtsp-server/rtsp-client.h:
8570 * gst/rtsp-server/rtsp-session-pool.c:
8571 Do some more session cleanup
8572 Make session timeout kill the TCP connection that currently watches the
8574 Remove the client timeout property.
8576 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8578 * gst/rtsp-server/rtsp-client.c:
8579 * gst/rtsp-server/rtsp-client.h:
8580 * gst/rtsp-server/rtsp-media.c:
8581 * gst/rtsp-server/rtsp-media.h:
8582 * gst/rtsp-server/rtsp-server.c:
8583 * gst/rtsp-server/rtsp-session.c:
8584 * gst/rtsp-server/rtsp-session.h:
8586 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
8589 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8591 * examples/Makefile.am:
8592 * examples/test-launch.c:
8593 Add example server that takes launch lines
8594 Add an example server that streams any -launch line.
8596 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8598 * examples/test-readme.c:
8599 * gst/rtsp-server/rtsp-client.c:
8600 * gst/rtsp-server/rtsp-media.c:
8601 * gst/rtsp-server/rtsp-media.h:
8602 Add support for live streams
8603 Add support for live streams and ranges
8604 Start on handling TCP data transfer.
8606 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8608 * gst/rtsp-server/rtsp-media.c:
8609 Free the pipeline before other things
8612 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8614 * gst/rtsp-server/rtsp-client.c:
8615 Only free the pending tunnel if there is one
8618 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8620 * gst/rtsp-server/rtsp-client.c:
8621 * gst/rtsp-server/rtsp-client.h:
8622 * gst/rtsp-server/rtsp-media.c:
8623 rtsp-server: Add support for tunneling
8624 Add support for tunneling over HTTP.
8625 Use new connection methods to retrieve the url.
8626 Dispatch messages based on the message type instead of blindly
8627 assuming it's always a request.
8628 Keep track of the watch id so that we can remove it later.
8629 Set the media pipeline to NULL before unreffing the pipeline.
8631 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8633 * gst/rtsp-server/rtsp-client.c:
8634 * gst/rtsp-server/rtsp-client.h:
8635 Fix for channel -> watch rename in gstreamer
8636 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
8638 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8640 * gst/rtsp-server/rtsp-client.c:
8641 * gst/rtsp-server/rtsp-client.h:
8643 Use the async RTSP channels instead of spawning a new thread for each client.
8644 If a sessionid is specified in a request, fail if we don't have the session.
8646 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8648 * gst/rtsp-server/rtsp-media.c:
8649 Add better debug info
8650 Add some better debug info.
8652 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8654 * examples/test-video.c:
8656 Add support for session timeouts in the example.
8658 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8660 * gst/rtsp-server/rtsp-session-pool.c:
8661 * gst/rtsp-server/rtsp-session-pool.h:
8662 Pass GTimeVal around for performance reasons
8663 Get the current time only once and pass it around so that sessions don't have to
8664 get the current time anymore.
8665 Add experimental support for a GSource that dispatches when the session needs to
8668 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8670 * gst/rtsp-server/rtsp-session.c:
8671 * gst/rtsp-server/rtsp-session.h:
8672 Add better support for session timeouts
8673 Add a method to request the number of milliseconds when a session will timeout.
8675 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8677 * gst/rtsp-server/rtsp-media.c:
8678 * gst/rtsp-server/rtsp-media.h:
8679 Add suport for RTP manager monitoring
8680 Add the first stage in monitoring the rtp manager.
8681 Make sure we don't update the state to something we don't want.
8683 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8685 * gst/rtsp-server/rtsp-client.c:
8686 Add support for session keepalive
8687 Get and update the session timeout for all requests. get the session as early as
8690 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8692 * gst/rtsp-server/rtsp-media-factory.h:
8693 * gst/rtsp-server/rtsp-media.c:
8694 * gst/rtsp-server/rtsp-media.h:
8695 Handle media bus messages
8696 Handle media bus messages in a custom mainloop and dispatch them to the
8697 RTSPMedia objects. Let the default implementation handle some common messages.
8699 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8701 * gst/rtsp-server/rtsp-client.c:
8702 * gst/rtsp-server/rtsp-session-pool.c:
8703 * gst/rtsp-server/rtsp-session.c:
8704 Some more session timeout handling
8705 Move the session header setting code to a central place so that we always add
8706 the timeout parameter too.
8707 Handle timeouts by running the session cleanup code.
8708 Stop media before cleaning up.
8710 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8712 * gst/rtsp-server/rtsp-client.c:
8713 * gst/rtsp-server/rtsp-client.h:
8714 Add timeout property
8715 Add a timeout property ot the client and make the other properties into GObject
8718 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8720 * gst/rtsp-server/rtsp-session-pool.c:
8721 Use getters and setters in property code
8722 Use the getters and setters for the timeout property instead of locking
8725 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8727 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
8729 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8731 * gst/rtsp-server/rtsp-session-pool.c:
8732 * gst/rtsp-server/rtsp-session-pool.h:
8733 * gst/rtsp-server/rtsp-session.c:
8734 * gst/rtsp-server/rtsp-session.h:
8735 Add more timeout stuff
8736 Add method to check if a session is expired.
8737 Add method to perform cleanup on a session pool.
8739 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8741 * gst/rtsp-server/rtsp-client.c:
8742 * gst/rtsp-server/rtsp-session-pool.c:
8743 * gst/rtsp-server/rtsp-session-pool.h:
8744 * gst/rtsp-server/rtsp-session.c:
8745 * gst/rtsp-server/rtsp-session.h:
8746 Add beginnings of session timeouts and limits
8747 Add the timeout value to the Session header for unusual timeout values.
8748 Allow us to configure a limit to the amount of active sessions in a pool. Set a
8749 limit on the amount of retry we do after a sessionid collision.
8750 Add properties to the sessionid and the timeout of a session. Keep track of
8751 creation time and last access time for sessions.
8753 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8755 * gst/rtsp-server/rtsp-client.c:
8756 * gst/rtsp-server/rtsp-media.c:
8757 * gst/rtsp-server/rtsp-media.h:
8758 * gst/rtsp-server/rtsp-sdp.c:
8759 * gst/rtsp-server/rtsp-session-pool.c:
8760 * gst/rtsp-server/rtsp-session.c:
8761 * gst/rtsp-server/rtsp-session.h:
8762 Cleanup of sessions and more
8763 Fix the refcounting of media and sessions in the client. Properly clean up the
8764 session data when the client performs a teardown.
8765 Add Server header to responses.
8766 Allow for multiple uri setups in one session.
8767 Add Range header to the PLAY response and add the range attribute to the SDP
8769 Fix the session pool remove method, it used the wrong key in the hashtable. Also
8770 give the ownership of the sessionid to the session object.
8772 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8774 * gst/rtsp-server/rtsp-server.c:
8775 * gst/rtsp-server/rtsp-server.h:
8777 Rename the 'server_port' variable to simply 'port'.
8779 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8782 * gst/rtsp-server/rtsp-client.c:
8783 * gst/rtsp-server/rtsp-media.c:
8784 * gst/rtsp-server/rtsp-media.h:
8785 * gst/rtsp-server/rtsp-session.c:
8786 * gst/rtsp-server/rtsp-session.h:
8787 Rework the way we handle transports for streams
8788 Make the media accept an array of transports for the streams that we have
8789 configured for the play/pause requests.
8790 Implement server states for a client and its media.
8791 Require 0.10.22.1 (git HEAD) of gstreamer.
8793 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8795 * gst/rtsp-server/rtsp-client.c:
8796 * gst/rtsp-server/rtsp-media-factory.c:
8797 Drop const from functions dealing with urls
8798 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
8799 have the right const in them.
8801 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8803 * gst/rtsp-server/rtsp-client.c:
8804 * gst/rtsp-server/rtsp-media.c:
8805 * gst/rtsp-server/rtsp-sdp.c:
8809 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8811 * gst/rtsp-server/rtsp-client.c:
8812 * gst/rtsp-server/rtsp-media-factory.c:
8813 * gst/rtsp-server/rtsp-media.c:
8814 * gst/rtsp-server/rtsp-media.h:
8816 Don't keep a reference to the GstRTSPMedia in the stream.
8817 Free more things when freeing the GstRTSPMedia.
8819 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8822 * gst/rtsp-server/rtsp-media-factory.c:
8823 * gst/rtsp-server/rtsp-media-factory.h:
8824 * gst/rtsp-server/rtsp-media.c:
8825 * gst/rtsp-server/rtsp-media.h:
8826 * gst/rtsp-server/rtsp-server.c:
8827 * gst/rtsp-server/rtsp-server.h:
8828 More docs and small cleanups
8829 Add some more docs and update the README
8830 Cleanup some method names.
8831 Remove an unneeded idx field in the GstRTSPMediaStream
8833 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8836 * examples/Makefile.am:
8837 * examples/test-readme.c:
8838 Add a README and more example code
8839 Add a README file that contains a small introduction on how to use the server
8840 along with the example code explained in the readme.
8842 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8844 * gst/rtsp-server/rtsp-media.c:
8845 * gst/rtsp-server/rtsp-server.c:
8846 Fix some leaks and change default port
8847 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
8848 we finished the initial preroll. If we keep them locked, setting the pipeline to
8849 NULL will not stop and clean up the sources correctly.
8850 Change the default RTSP port to 8554 aka the official alternative RTSP port.
8852 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8854 * gst/rtsp-server/rtsp-session.c:
8855 * gst/rtsp-server/rtsp-session.h:
8856 Cleanups to the session object
8857 Remove some unneeded variables in the session state of a stream such as the
8858 owner media and the server transport.
8859 Get the configuration of a media stream in a session based on the media_stream
8860 in the original object instead of our cached index.
8861 Free more data in the finalize method.
8863 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8865 * gst/rtsp-server/rtsp-client.c:
8866 * gst/rtsp-server/rtsp-client.h:
8867 Cleanups and reuse media from DESCRIBE
8868 Handle thread create errors.
8869 Rename some internal methods to better match what they actually do.
8870 Handle misconfiguration of session_pool and media_mapping gracefully.
8871 Cache the DESCRIBE media and uri in the client connection and reuse them when
8872 we receive a SETUP request in the same connection for the same uri.
8873 Cleanup the client connection object.
8875 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8877 * gst/rtsp-server/rtsp-media-factory.c:
8878 * gst/rtsp-server/rtsp-media-factory.h:
8879 * gst/rtsp-server/rtsp-media.c:
8880 * gst/rtsp-server/rtsp-media.h:
8881 Add shared properties to media and factory
8882 Add the shared property to media.
8883 Implement some simple caching in the factory depending on if the media is shared
8886 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8888 * gst/rtsp-server/rtsp-client.c:
8889 Add a little comment
8890 Add some comment about the content-base header.
8892 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8894 * examples/Makefile.am:
8896 * examples/test-mp4.c:
8897 * examples/test-ogg.c:
8898 * examples/test-video.c:
8899 * gst/rtsp-server/Makefile.am:
8900 * gst/rtsp-server/rtsp-client.c:
8901 * gst/rtsp-server/rtsp-client.h:
8902 * gst/rtsp-server/rtsp-media-factory.c:
8903 * gst/rtsp-server/rtsp-media-factory.h:
8904 * gst/rtsp-server/rtsp-media.c:
8905 * gst/rtsp-server/rtsp-media.h:
8906 * gst/rtsp-server/rtsp-sdp.c:
8907 * gst/rtsp-server/rtsp-sdp.h:
8908 * gst/rtsp-server/rtsp-server.c:
8909 * gst/rtsp-server/rtsp-server.h:
8910 * gst/rtsp-server/rtsp-session.c:
8911 * gst/rtsp-server/rtsp-session.h:
8912 Reorganize things, prepare for media sharing
8913 Added various other test server examples
8914 Move the SDP message generation to a separate helper.
8915 Refactor common code for finding the session.
8916 Add content-base for realplayer compatibility
8917 Clean up request uris before processing for better vlc compatibility.
8918 Move prerolling and pipeline construction to the RTSPMedia object.
8919 Use multiudpsink for future pipeline reuse.
8921 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8927 === release 0.10.1 ===
8929 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8935 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8937 * bindings/vala/Makefile.am:
8939 Add more directories and files to the dist.
8941 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8943 * bindings/python/Makefile.am:
8944 * bindings/python/rtspserver.override:
8945 Fixed compile error of python bindings
8947 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8949 * bindings/vala/gst-rtsp-server-0.10.vapi:
8950 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8951 Marked values as nullable accordingly
8953 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8955 * bindings/vala/gst-rtsp-server-0.10.vapi:
8956 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
8957 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8958 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8959 Updated Vala bindings
8961 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8963 * gst/rtsp-server/rtsp-client.c:
8964 * gst/rtsp-server/rtsp-media-mapping.c:
8965 * gst/rtsp-server/rtsp-media-mapping.h:
8966 * gst/rtsp-server/rtsp-media.h:
8967 * gst/rtsp-server/rtsp-session-pool.h:
8968 Cleanups and doc updates
8969 Add some more documentation and do some minor cleanups here and there.
8971 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8973 * gst/rtsp-server/rtsp-client.c:
8974 * gst/rtsp-server/rtsp-media-factory.c:
8975 * gst/rtsp-server/rtsp-media-factory.h:
8976 * gst/rtsp-server/rtsp-media.c:
8977 * gst/rtsp-server/rtsp-media.h:
8978 * gst/rtsp-server/rtsp-session.c:
8979 * gst/rtsp-server/rtsp-session.h:
8981 Rename GstRTSPMediaBin to GstRTSPMedia
8982 Parse the request url into a GstRTSPUri object and pass this object to the
8983 various handlers and methods that require the uri.
8985 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8989 Add some more docs and remove some old code from the example.
8991 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8993 * gst/rtsp-server/rtsp-client.c:
8994 Handle state change failures better
8995 Handle state change failures better when changing the state of the pipeline to
8998 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9000 * gst/rtsp-server/rtsp-media-factory.c:
9001 * gst/rtsp-server/rtsp-media-factory.h:
9002 Make element creation more extendible
9003 Add get_element vmethod to the default MediaFactory so that subclasses can just
9004 override that method and still use the default logic for making a MediaBin from
9007 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9010 * gst/rtsp-server/Makefile.am:
9011 * gst/rtsp-server/rtsp-client.c:
9012 * gst/rtsp-server/rtsp-client.h:
9013 * gst/rtsp-server/rtsp-media-factory.c:
9014 * gst/rtsp-server/rtsp-media-factory.h:
9015 * gst/rtsp-server/rtsp-media-mapping.c:
9016 * gst/rtsp-server/rtsp-media-mapping.h:
9017 * gst/rtsp-server/rtsp-media.c:
9018 * gst/rtsp-server/rtsp-media.h:
9019 * gst/rtsp-server/rtsp-server.c:
9020 * gst/rtsp-server/rtsp-server.h:
9021 * gst/rtsp-server/rtsp-session.c:
9022 * gst/rtsp-server/rtsp-session.h:
9023 Make the server handle arbitrary pipelines
9024 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
9025 The GstMediaBin object has a handle to a bin with elements and to a list of
9026 GstMediaStream objects that this bin produces.
9027 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
9028 with methods to register and remove those mappings.
9029 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
9030 used by the server instance.
9031 Modify the example application so that it shows how to create custom pipelines
9032 attached to a specific mount point.
9033 Various misc cleanps.
9035 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9037 * gst/rtsp-server/rtsp-server.c:
9038 * gst/rtsp-server/rtsp-server.h:
9039 Allow setting a custom media factory for a server
9041 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9043 * gst/rtsp-server/rtsp-client.c:
9044 * gst/rtsp-server/rtsp-client.h:
9045 Allow setting a custom media factory for a client.
9047 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9049 * gst/rtsp-server/Makefile.am:
9050 Add Makefile entry for the media factory
9052 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9054 * gst/rtsp-server/rtsp-media-factory.c:
9055 * gst/rtsp-server/rtsp-media-factory.h:
9056 Add media factory to map urls to media pipeline objects.
9058 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9060 * gst/rtsp-server/rtsp-media.c:
9061 * gst/rtsp-server/rtsp-media.h:
9062 Add comments. Remove unused field
9064 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9066 * gst/rtsp-server/rtsp-session-pool.c:
9067 * gst/rtsp-server/rtsp-session-pool.h:
9068 Allow custom session pools to override the session id allocation algorithms Add some comments.
9070 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9072 * gst/rtsp-server/rtsp-session.h:
9075 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9077 * gst/rtsp-server/rtsp-client.c:
9078 * gst/rtsp-server/rtsp-client.h:
9079 Move the connection code in one place Add some comments
9081 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9083 * gst/rtsp-server/rtsp-server.c:
9084 * gst/rtsp-server/rtsp-server.h:
9085 Make vmethod to create and accept new clients. Add some docs.
9087 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9089 * gst/rtsp-server/rtsp-server.c:
9090 * gst/rtsp-server/rtsp-server.h:
9091 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
9093 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9095 * gst/rtsp-server/rtsp-client.c:
9096 * gst/rtsp-server/rtsp-client.h:
9097 Name the parameters more appropriately.
9099 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9101 * gst/rtsp-server/rtsp-session-pool.c:
9102 Do some more cleanup of the session pool.
9104 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9106 * gst/rtsp-server/Makefile.am:
9107 * gst/rtsp-server/rtsp-client.c:
9108 Check if return value of gst_rtsp_session_get_media is not NULL
9110 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9112 * gst/rtsp-server/Makefile.am:
9113 Install rtsp-session and rtsp-session-pool headers
9115 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9120 * bindings/python/Makefile.am:
9121 * bindings/python/arg-types.py:
9122 * bindings/python/codegen/Makefile.am:
9123 * bindings/python/codegen/__init__.py:
9124 * bindings/python/codegen/argtypes.py:
9125 * bindings/python/codegen/code-coverage.py:
9126 * bindings/python/codegen/codegen.py:
9127 * bindings/python/codegen/definitions.py:
9128 * bindings/python/codegen/defsparser.py:
9129 * bindings/python/codegen/docextract.py:
9130 * bindings/python/codegen/docgen.py:
9131 * bindings/python/codegen/fileprefix.override:
9132 * bindings/python/codegen/fileprefixmodule.c:
9133 * bindings/python/codegen/h2def.py:
9134 * bindings/python/codegen/mergedefs.py:
9135 * bindings/python/codegen/mkskel.py:
9136 * bindings/python/codegen/override.py:
9137 * bindings/python/codegen/reversewrapper.py:
9138 * bindings/python/codegen/scmexpr.py:
9139 * bindings/python/rtspserver-types.defs:
9140 * bindings/python/rtspserver.defs:
9141 * bindings/python/rtspserver.override:
9142 * bindings/python/rtspservermodule.c:
9144 Add python bindings.
9146 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9148 * bindings/Makefile.am:
9150 Don't go into python dir when requirements for python bindings are missing
9152 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9154 * bindings/Makefile.am:
9155 * bindings/vala/Makefile.am:
9157 Install Vala bindings if vala is available
9159 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9161 * bindings/vala/gst-rtsp-server-0.10.deps:
9162 * bindings/vala/gst-rtsp-server-0.10.vapi:
9163 * bindings/vala/gst-rtsp-server.vapi:
9164 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
9165 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
9166 * bindings/vala/packages/gst-rtsp-server-0.10.files:
9167 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9168 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9169 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
9170 * bindings/vala/packages/gst-rtsp-server.deps:
9171 * bindings/vala/packages/gst-rtsp-server.excludes:
9172 * bindings/vala/packages/gst-rtsp-server.files:
9173 * bindings/vala/packages/gst-rtsp-server.gi:
9174 * bindings/vala/packages/gst-rtsp-server.metadata:
9175 * bindings/vala/packages/gst-rtsp-server.namespace:
9176 Regenerated Vala bindings
9178 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9180 * bindings/vala/gst-rtsp-server.vapi:
9181 * bindings/vala/packages/gst-rtsp-server.metadata:
9182 Fixed typo in included headers for vala bindings
9184 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9188 * pkgconfig/Makefile.am:
9189 * pkgconfig/gst-rtsp-server.pc.in:
9190 Added pkgconfig file
9192 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
9194 * bindings/vala/gst-rtsp-server.vapi:
9195 * bindings/vala/packages/gst-rtsp-server.excludes:
9196 * bindings/vala/packages/gst-rtsp-server.gi:
9197 * bindings/vala/packages/gst-rtsp-server.metadata:
9198 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
9200 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
9202 * bindings/vala/gst-rtsp-server.vapi:
9203 * bindings/vala/packages/gst-rtsp-server.deps:
9204 * bindings/vala/packages/gst-rtsp-server.files:
9205 * bindings/vala/packages/gst-rtsp-server.gi:
9206 * bindings/vala/packages/gst-rtsp-server.metadata:
9207 * bindings/vala/packages/gst-rtsp-server.namespace:
9210 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
9212 * gst/rtsp-server/rtsp-session.c:
9213 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
9215 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
9217 * examples/Makefile.am:
9218 * gst/rtsp-server/Makefile.am:
9219 Put GStreamer version in library name
9221 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9223 * examples/Makefile.am:
9224 * gst/rtsp-server/Makefile.am:
9225 Fix some issues to pass distcheck
9227 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9229 * gst/rtsp-server/rtsp-server.c:
9230 Added port property to GstRTSPServer class.
9232 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9237 * examples/Makefile.am:
9240 * gst/rtsp-server/Makefile.am:
9241 * gst/rtsp-server/rtsp-client.c:
9242 * gst/rtsp-server/rtsp-client.h:
9243 * gst/rtsp-server/rtsp-media.c:
9244 * gst/rtsp-server/rtsp-media.h:
9245 * gst/rtsp-server/rtsp-server.c:
9246 * gst/rtsp-server/rtsp-server.h:
9247 * gst/rtsp-server/rtsp-session-pool.c:
9248 * gst/rtsp-server/rtsp-session-pool.h:
9249 * gst/rtsp-server/rtsp-session.c:
9250 * gst/rtsp-server/rtsp-session.h:
9253 * src/rtsp-client.c:
9254 * src/rtsp-client.h:
9257 * src/rtsp-server.c:
9258 * src/rtsp-server.h:
9259 * src/rtsp-session-pool.c:
9260 * src/rtsp-session-pool.h:
9261 * src/rtsp-session.c:
9262 * src/rtsp-session.h:
9263 Split in library and example program
9265 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
9267 * src/rtsp-client.h:
9268 Removed obsolete variable
9270 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
9272 * src/rtsp-client.c:
9273 * src/rtsp-client.h:
9274 Removed pipeline variable GstRTSPClient, because it's only used in one function
9276 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9279 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
9281 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
9283 * src/rtsp-session.c:
9284 Initialize some more vars.
9286 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
9288 * src/rtsp-session.c:
9289 Initialize variable to avoid compiler warning.
9291 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
9294 Add a reasonable generic .gitignore
projects / platform / upstream / gstreamer.git / blob