3 2017-05-04 Sebastian Dröge <slomo@coaxion.net>
8 === release 1.11.91 ===
10 2017-04-27 17:42:02 +0300 Sebastian Dröge <sebastian@centricular.com>
16 * gst-rtsp-server.doap:
20 2017-04-24 20:30:37 +0100 Tim-Philipp Müller <tim@centricular.com>
23 Automatic update of common submodule
24 From 60aeef6 to 48a5d85
26 2017-04-13 14:20:10 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
28 * gst/rtsp-server/rtsp-media-factory.c:
29 * gst/rtsp-server/rtsp-media.c:
30 * gst/rtsp-server/rtsp-session.c:
31 * gst/rtsp-server/rtsp-stream.c:
32 gi: Fix some annotations and docstrings
34 2017-04-13 13:52:26 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
36 * gst/rtsp-server/meson.build:
41 2017-04-10 23:51:12 +0100 Tim-Philipp Müller <tim@centricular.com>
45 Automatic update of common submodule
46 From 39ac2f5 to 60aeef6
48 === release 1.11.90 ===
50 2017-04-07 16:35:03 +0300 Sebastian Dröge <sebastian@centricular.com>
56 * gst-rtsp-server.doap:
60 2017-03-27 18:19:33 +0100 Tim-Philipp Müller <tim@centricular.com>
62 * examples/test-launch.c:
63 examples: make test-launch pipeline shared by default as well
65 2017-02-27 19:10:44 +0200 Sebastian Dröge <sebastian@centricular.com>
67 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
68 gstreamer-rtsp-server: Add both srcdir and builddir to the include path
69 Just the build dir is not going to work for srcdir!=builddir.
71 2017-02-24 15:59:54 +0200 Sebastian Dröge <sebastian@centricular.com>
76 2017-02-24 15:37:49 +0200 Sebastian Dröge <sebastian@centricular.com>
81 === release 1.11.2 ===
83 2017-02-24 15:10:07 +0200 Sebastian Dröge <sebastian@centricular.com>
89 * gst-rtsp-server.doap:
92 2017-02-14 20:40:26 +0000 Tim-Philipp Müller <tim@centricular.com>
95 meson: dist meson build files
96 Ship meson build files in tarballs, so people who use tarballs
97 in their builds can start playing with meson already.
99 2017-02-07 23:39:37 +1100 Jan Schmidt <jan@centricular.com>
101 * examples/test-record.c:
102 examples/test-record: Add extra line to initial printout
103 Add an example line of how to deliver a stream to the
106 2017-01-19 14:57:19 +0200 Sebastian Dröge <sebastian@centricular.com>
108 * gst/rtsp-server/rtsp-client.c:
109 rtsp-client: Also handle the (S|G)ET_PARAMETER case of size==0 || !data as keep-alive
110 If there is no Content-Length header, no body would be allocated and the
111 '\0' would also not be appended to the body.
113 2017-01-19 14:24:07 +0200 Sebastian Dröge <sebastian@centricular.com>
115 * gst/rtsp-server/rtsp-client.c:
116 rtsp-client: Fix handling of keep-alive GET_PARAMETER/SET_PARAMETER
117 While they logically have 0 bytes length, GstRTSPConnection is appending
118 a '\0' to everything making the size be 1 instead.
120 2017-01-13 12:39:36 +0000 Tim-Philipp Müller <tim@centricular.com>
125 2017-01-12 19:04:23 +0200 Sebastian Dröge <sebastian@centricular.com>
127 * gst/rtsp-server/rtsp-session.c:
128 rtsp-session: Only remove deprecated API if requested to do so, not just when disabling
129 gst_rtsp_session_is_expired() and gst_rtsp_session_next_timeout() were
132 2017-01-12 16:32:59 +0200 Sebastian Dröge <sebastian@centricular.com>
137 === release 1.11.1 ===
139 2017-01-12 16:14:46 +0200 Sebastian Dröge <sebastian@centricular.com>
145 * gst-rtsp-server.doap:
146 * win32/common/libgstrtspserver.def:
149 2017-01-10 08:34:50 +0100 Patricia Muscalu <patricia@axis.com>
151 * gst/rtsp-server/rtsp-stream.c:
152 rtsp-stream: corrected if-statement in _get_server_port()
153 This bug was accidentally introduced while fixing a segfault
154 in _get_server_port() function.
155 https://bugzilla.gnome.org/show_bug.cgi?id=776345
157 2017-01-09 14:12:05 +0100 Patricia Muscalu <patricia@axis.com>
159 * gst/rtsp-server/rtsp-stream.c:
160 * tests/check/gst/stream.c:
161 rtsp-stream: fixed segmenation fault in _get_server_port()
162 Calling function gst_rtsp_stream_get_server_port() results in
163 segmenation fault in the RTP/RTSP/TCP case.
164 Port that the server will use to receive RTCP makes only
165 sense in the UDP case, however the function should handle
166 the TCP case in a nicer way.
167 https://bugzilla.gnome.org/show_bug.cgi?id=776345
169 2017-01-09 12:22:40 +0300 Aleksandr Slobodeniuk <alenuke@yandex.ru>
171 * gst/rtsp-server/rtsp-media-factory.c:
172 dosc: Fix a little typo
173 https://bugzilla.gnome.org/show_bug.cgi?id=777037
175 2017-01-04 16:20:54 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
177 * pkgconfig/Makefile.am:
178 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
179 * pkgconfig/meson.build:
180 meson: generate pkg-config -uninstalled pc files
181 Generating those files is useful for users building the GStreamer stack
182 using meson and having to link it to another project which is still
184 https://bugzilla.gnome.org/show_bug.cgi?id=776810
186 2017-01-04 16:11:08 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
188 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
189 pkgconfig: fix -uninstalled pc file
190 pcfiledir was never defined so the paths were wrong.
191 https://bugzilla.gnome.org/show_bug.cgi?id=776867
193 2016-12-21 13:41:50 +0100 Patricia Muscalu <patricia@axis.com>
195 * gst/rtsp-server/rtsp-stream.c:
196 * tests/check/gst/rtspserver.c:
197 rtsp-stream: Fixed TCP transport case
198 Make sure that the appsink element is actually added to
199 the bin before trying to link it with the elements in it.
200 https://bugzilla.gnome.org/show_bug.cgi?id=776343
202 2016-12-16 17:26:04 +0000 Tim-Philipp Müller <tim@centricular.com>
208 Remove generated .spec file
209 Likely extremely bitrotten, and we should not ship this anyway.
211 2016-12-03 08:21:02 +0100 Edward Hervey <bilboed@bilboed.com>
214 Automatic update of common submodule
215 From f980fd9 to 39ac2f5
217 2016-12-02 15:40:09 +0100 Edward Hervey <edward@centricular.com>
219 * gst/rtsp-server/rtsp-media.c:
220 media: Fix pt map caps
221 Since decryption is handled within rtpbin, all outcoming stream
222 caps will be application/x-rtp (i.e. regular rtp)
223 Fixes RECORD with SRTP streams
225 2016-12-02 15:38:04 +0100 Edward Hervey <edward@centricular.com>
227 * gst/rtsp-server/rtsp-media-factory.c:
228 media-factory: Create media objects with the proper transport mode
229 The function called immediately afterwards (collect_streams()) will
230 need it to work properly
232 2016-12-02 14:36:50 +0200 Sebastian Dröge <sebastian@centricular.com>
234 * gst/rtsp-server/rtsp-auth.c:
235 rtsp-auth: Don't remove digest-auth nonces that already/still have a client connected
237 2016-12-01 18:04:34 +0200 Sebastian Dröge <sebastian@centricular.com>
239 * gst/rtsp-server/rtsp-media-factory.c:
240 rtsp-media-factory: Don't create a pipeline for the media pipeline string
241 We're going to put a pipeline into a pipeline otherwise, which is not
244 2016-10-25 15:41:28 +0300 Kseniia Vasilchuk <vasilchukkseniia@gmail.com>
246 * gst/rtsp-server/rtsp-media.c:
247 media: Fix race condition around finish_unprepare() if called multiple time
248 https://bugzilla.gnome.org/show_bug.cgi?id=755329
250 2016-11-30 14:06:36 +1100 Jan Schmidt <jan@centricular.com>
252 * gst/rtsp-sink/gstrtspclientsink.c:
253 rtspclientsink: Don't leave stale pointer after unref
254 Fix a warning on shutdown - don't keep a pointer to an
255 alread-unreffed object.
257 2016-11-26 11:24:50 +0000 Tim-Philipp Müller <tim@centricular.com>
260 common: use https protocol for common submodule
261 https://bugzilla.gnome.org/show_bug.cgi?id=775110
263 2016-11-21 23:29:56 +1100 Matthew Waters <matthew@centricular.com>
265 * gst/rtsp-server/rtsp-stream.c:
266 stream: block the output of rtpbin instead of the source pipeline
267 85c52e194bcb81928b96614be0ae47d59eccb1ce introduced a more correct
268 detection of the srtp rollover counter to add to the SDP.
269 Unfortunately, it was incomplete for live pipelines where the logic
270 blocks the source bin before creating the SDP and thus would never have
271 the necessary informaiton to create a correct SDP with srtp encryption.
272 Move the pad blocks to rtpbin's output pads instead so that the
273 necessary information can be created before we need the information for
275 https://bugzilla.gnome.org/show_bug.cgi?id=770239
277 2016-11-21 16:02:39 +0100 Dag Gullberg <dagg@axis.com>
279 * gst/rtsp-server/rtsp-client.c:
280 rtsp-client: add IDLE timeout, before session exists
281 The RTSP server will not timeout an idle RTSP connection
282 (note this is different from doing timeout on a RTSP
284 At least for Apache this is a problem when running RTSP over
285 HTTPS since it uses one of the threads (there is a rather
286 limited number) that are available for handling requests.
287 https://bugzilla.gnome.org/show_bug.cgi?id=771830
289 2016-11-23 09:45:08 +0000 Tim-Philipp Müller <tim@centricular.com>
294 2016-11-21 13:05:50 +0100 Göran Jönsson <goranjn@axis.com>
296 * gst/rtsp-server/rtsp-stream.c:
297 rtsp-stream: Set close-socket FALSE on UDP src:es
298 With this RTSP server can use the sockets independent on the udpsrc
300 When the udp src is finalized it will unref socket and when g_socket
301 is finalized the socket will be closed.
302 https://bugzilla.gnome.org/show_bug.cgi?id=765673
304 2016-11-18 17:47:13 +0200 Sebastian Dröge <sebastian@centricular.com>
306 * gst/rtsp-sink/gstrtspclientsink.c:
307 rtspclientsink: Move to new helper function to parse authentication responses
308 https://bugzilla.gnome.org/show_bug.cgi?id=774416
310 2016-11-16 08:42:24 +0200 Sebastian Dröge <sebastian@centricular.com>
312 * examples/Makefile.am:
313 * examples/test-auth-digest.c:
314 * gst/rtsp-server/rtsp-auth.c:
315 * gst/rtsp-server/rtsp-auth.h:
316 * win32/common/libgstrtspserver.def:
317 rtsp-auth: Add support for Digest authentication
318 https://bugzilla.gnome.org/show_bug.cgi?id=774416
320 2016-11-17 09:41:53 -0800 Scott D Phillips <scott.d.phillips@intel.com>
323 * gst/rtsp-server/meson.build:
325 * tests/check/meson.build:
327 * win32/common/libgstrtspserver.def:
328 Enable building with MSVC
329 https://bugzilla.gnome.org/show_bug.cgi?id=774640
331 2016-11-18 20:23:14 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
334 meson: gstreamer gst_check_dep does not exist on windows
336 2016-11-17 09:43:37 -0800 Scott D Phillips <scott.d.phillips@intel.com>
338 * gst/rtsp-server/rtsp-client.c:
339 client: update do_send_message to match type GstRTSPClientSendFunc
340 This type mismatch fails building with MSVC
341 https://bugzilla.gnome.org/show_bug.cgi?id=774640
343 2016-11-11 14:42:08 +0200 Sebastian Dröge <sebastian@centricular.com>
345 * gst/rtsp-server/rtsp-sdp.c:
346 rtsp-sdp: Fix indentation
348 2016-11-10 05:16:00 +0000 Neha Arora <arora.neha@samsung.com>
350 * gst/rtsp-server/rtsp-media.c:
351 rtsp-media: Only signal "new-state" if the state has actually changed
352 https://bugzilla.gnome.org/show_bug.cgi?id=774173
354 2016-08-24 11:39:13 +0200 Branko Subasic <branko@axis.com>
356 * gst/rtsp-server/rtsp-client.c:
357 * gst/rtsp-server/rtsp-client.h:
358 client: emit signal in the beginning of each rtsp request
359 These signals let the application validate the requests, configure the
360 media/stream in a certain way and also generate error status code in
361 case of error or bad request.
362 https://bugzilla.gnome.org/show_bug.cgi?id=758062
364 2016-11-01 18:10:35 +0000 Tim-Philipp Müller <tim@centricular.com>
367 meson: update version
369 === release 1.11.0 ===
371 2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com>
376 === release 1.10.0 ===
378 2016-11-01 18:06:46 +0200 Sebastian Dröge <sebastian@centricular.com>
384 * gst-rtsp-server.doap:
387 2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com>
389 * tests/check/gst/rtspserver.c:
390 * tests/check/gst/stream.c:
391 tests: try to avoid using the same ports in different tests
392 Causes problems with client multicast tests otherwise if
393 tests are run in parallel.
394 https://bugzilla.gnome.org/show_bug.cgi?id=773640
396 2016-10-28 17:50:59 +0100 Tim-Philipp Müller <tim@centricular.com>
398 * tests/check/gst/client.c:
399 tests: client: use fail_unless_equals_foo() for better failure reporting
401 2016-09-26 11:16:04 +0200 Göran Jönsson <goranjn@axis.com>
403 * gst/rtsp-server/rtsp-client.c:
404 rtsp-client: Session filter in unwatch session
405 Call session filter with filter_session_media as paramer in
406 client_unwatch_session if using drop_backlog = FALSE.
407 In client_unwatch_session its allowed to grow the watchs backlog.
408 If using drop_backlog = FALSE and the backlog is full it will cause
409 a deadlock when setting session media state to NULL
410 if the backlog is not allowed to grow.
411 https://bugzilla.gnome.org/show_bug.cgi?id=771983
413 2016-10-20 21:40:18 +0100 Tim-Philipp Müller <tim@centricular.com>
416 meson: add fallbacks for gst modules
419 2016-09-14 17:48:39 +0300 Nikita Bobkov <NikitaDBobkov@gmail.com>
421 * gst/rtsp-server/rtsp-client.c:
422 rtsp-client: Fix factory leaking in find_media() in error cases
423 https://bugzilla.gnome.org/show_bug.cgi?id=771488
425 2016-10-06 11:47:50 -0400 Xavier Claessens <xavier.claessens@collabora.com>
427 * gst/rtsp-server/rtsp-stream.c:
428 stream: Fix randomly missing streams from SDP with dynamic elements
429 When using dynamic elements, gst_rtsp_stream_join_bin() is called from
430 "pad-added" signal. In that case priv->srcpad could already have its caps,
431 and they'll be sent to priv->send_src[0] pad. That means that when it
432 connects "notify::caps" signal, that pad could already have received its
433 caps and the signal won't be emitted anymore.
434 In that case priv->caps stay to NULL and when building the SDP that stream
435 gets ignored. Leading to missing video or audio when playing in client side.
436 https://bugzilla.gnome.org/show_bug.cgi?id=772478
438 2016-09-30 11:42:08 +0100 Tim-Philipp Müller <tim@centricular.com>
441 meson: update version
443 === release 1.9.90 ===
445 2016-09-30 13:04:12 +0300 Sebastian Dröge <sebastian@centricular.com>
451 * gst-rtsp-server.doap:
454 2016-09-17 13:17:19 +0100 Ian Jamison <ian.dev@arkver.com>
456 * gst/rtsp-server/rtsp-media-factory.c:
457 * gst/rtsp-server/rtsp-media.c:
458 * gst/rtsp-server/rtsp-stream.c:
459 rtsp-server: Hint that set_multicast_iface expects the name of the interface
460 To prevent any possibly confusion with IPs or anything else.
461 https://bugzilla.gnome.org/show_bug.cgi?id=771530
463 2016-09-18 09:58:55 -0400 Sebastian Dröge <sebastian@centricular.com>
465 * gst/rtsp-server/rtsp-media-factory.c:
466 * gst/rtsp-server/rtsp-media.c:
467 rtsp-media: Call g_free() instead of g_object_unref() on multicast-iface strings
468 https://bugzilla.gnome.org/show_bug.cgi?id=763000#c5
470 2016-09-14 11:31:15 +0200 Sebastian Dröge <sebastian@centricular.com>
473 configure: Depend on gstreamer 1.9.2.1
475 2016-09-10 20:52:31 +1000 Jan Schmidt <jan@centricular.com>
479 Automatic update of common submodule
480 From b18d820 to f980fd9
482 2016-09-10 09:58:31 +1000 Jan Schmidt <jan@centricular.com>
486 Automatic update of common submodule
487 From 6f2d209 to b18d820
489 2016-09-07 18:44:34 +0300 Sebastian Dröge <sebastian@centricular.com>
491 * gst/rtsp-server/rtsp-stream.c:
492 rtsp-stream: Remove unused _locked() variant of a function
493 It was added during refactoring.
495 2016-09-07 10:21:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
497 * gst/rtsp-server/rtsp-stream.c:
498 stream: cosmetic cleanup
499 https://bugzilla.gnome.org/show_bug.cgi?id=766612
501 2016-09-07 10:16:19 -0400 Xavier Claessens <xavier.claessens@collabora.com>
503 * gst/rtsp-server/rtsp-stream.c:
504 stream: Compare IP addresses case insensitive in more places
505 https://bugzilla.gnome.org/show_bug.cgi?id=766612
507 2016-09-07 10:12:18 -0400 Xavier Claessens <xavier.claessens@collabora.com>
510 * gst/rtsp-server/rtsp-stream.c:
511 stream: Fix leaked joined_bin
512 There is no need to keep a strong ref on it, and _leave_bin() was
513 setting it to NULL before calling g_clear_object() so it was leaked.
514 https://bugzilla.gnome.org/show_bug.cgi?id=766612
516 2016-09-06 19:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
518 * gst/rtsp-server/rtsp-stream.c:
519 rtsp-stream: Compare IP address strings case insensitive
520 Otherwise IPv6 addresses might fail this comparision.
522 2016-09-06 19:10:21 +0300 Sebastian Dröge <sebastian@centricular.com>
524 * gst/rtsp-server/rtsp-stream.c:
525 rtsp-stream: Bind multicast sockets to ANY as before
526 https://bugzilla.gnome.org/show_bug.cgi?id=766612#c48
528 2016-09-05 18:31:36 +0300 Kseniia <vasilchukkseniia@gmail.com>
530 * gst/rtsp-server/rtsp-session.c:
531 rtsp-session: Fix segfault when doing keep-alive after removing the session
532 If keep-alive happens after removing the session but before finalizing the
533 stream transport, we would segfault.
534 https://bugzilla.gnome.org/show_bug.cgi?id=750544
536 2016-09-05 18:04:50 +0300 Sebastian Dröge <sebastian@centricular.com>
538 * gst/rtsp-server/rtsp-stream.c:
539 rtsp-stream: Always create multicast UDP elements if the protocol flag is set
540 Adding them later will cause deadlocks due to
541 1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
542 2) adding the multicast sink
543 3) waiting for it to get data to preroll again
544 3) never happens because the queues after the tee are full.
546 2016-09-05 16:32:57 +0300 Sebastian Dröge <sebastian@centricular.com>
548 * gst/rtsp-server/rtsp-stream.c:
549 rtsp-stream: Fix up various multicast related issues
551 2016-09-05 13:40:59 +0300 Sebastian Dröge <sebastian@centricular.com>
553 * tests/check/gst/stream.c:
554 tests: Fix compilation
556 2016-07-28 15:33:05 -0400 Xavier Claessens <xavier.claessens@collabora.com>
558 * gst/rtsp-server/rtsp-client.c:
559 * gst/rtsp-server/rtsp-stream.c:
560 * tests/check/gst/stream.c:
561 stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast
562 This is basically reverting changes introduced in commit f62a9a7,
563 because it was introducing various regressions:
564 - It introduces a leak of udpsrc elements that got wrongly fixed by adding
565 an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
566 ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
567 - If a mcast client connects, it creates a new socket in SETUP to try to respect
568 the destination/port given by the client in the transport, and overrides the
569 socket already set on the udpsink element. That means that if we already had a
570 client connected, the source address on the udp packets it receives suddenly
572 - If a 2nd mcast client connects, the destination/port in its transport is
573 ignored but its transport wasn't updated.
574 What this patch does:
575 - Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
576 - Always have a tee+queue when udp is enabled. This could be optimized
577 again in a later patch, but is more complicated. If no unicast clients
578 connects then those elements are useless, this could be also optimized
580 - When mcast transport is added, it creates a new set of udpsrc/udpsink,
581 seperated from those for unicast clients. Since we already support only
582 one mcast address, we also create only one set of elements.
583 https://bugzilla.gnome.org/show_bug.cgi?id=766612
585 2016-07-28 15:20:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
587 * gst/rtsp-server/rtsp-stream.c:
588 stream: factor our plug_src function
589 https://bugzilla.gnome.org/show_bug.cgi?id=766612
591 2016-07-21 21:46:16 -0400 Xavier Claessens <xavier.claessens@collabora.com>
593 * gst/rtsp-server/rtsp-stream.c:
594 stream: factor out plug_sink function
595 https://bugzilla.gnome.org/show_bug.cgi?id=766612
597 2016-07-20 23:05:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
599 * gst/rtsp-server/rtsp-stream.c:
600 stream: small documentation clarification
601 https://bugzilla.gnome.org/show_bug.cgi?id=766612
603 2016-07-20 15:35:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
605 * gst/rtsp-server/rtsp-stream.c:
606 stream: rename addr_v4/6 to mcast_addr_v4/6 for clarity
607 https://bugzilla.gnome.org/show_bug.cgi?id=766612
609 2016-07-14 11:10:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
611 * gst/rtsp-server/rtsp-stream.c:
612 stream: Keep a ref on joined bin
613 https://bugzilla.gnome.org/show_bug.cgi?id=766612
615 2016-07-20 15:11:32 -0400 Xavier Claessens <xavier.claessens@collabora.com>
617 * gst/rtsp-server/rtsp-stream.c:
619 https://bugzilla.gnome.org/show_bug.cgi?id=766612
621 2016-07-20 23:18:23 -0400 Xavier Claessens <xavier.claessens@collabora.com>
623 * gst/rtsp-server/rtsp-stream.c:
624 stream: small fix in error code path
625 https://bugzilla.gnome.org/show_bug.cgi?id=766612
627 2016-07-20 20:09:57 -0400 Xavier Claessens <xavier.claessens@collabora.com>
629 * gst/rtsp-server/rtsp-stream.c:
630 Revert "rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc"
631 This partly reverts commit cba045e1b19fad6e689e10206f57903e15f1229a,
632 but keeps unit tests.
633 https://bugzilla.gnome.org/show_bug.cgi?id=766612
635 2016-09-01 12:33:00 +0300 Sebastian Dröge <sebastian@centricular.com>
640 === release 1.9.2 ===
642 2016-09-01 12:32:51 +0300 Sebastian Dröge <sebastian@centricular.com>
648 * gst-rtsp-server.doap:
651 2016-01-27 01:03:52 +0000 Tim-Philipp Müller <tim@centricular.com>
654 * examples/meson.build:
656 * gst/rtsp-server/meson.build:
657 * gst/rtsp-sink/meson.build:
659 * pkgconfig/meson.build:
660 * tests/check/meson.build:
662 Add support for Meson as alternative/parallel build system
663 https://github.com/mesonbuild/meson
665 2016-08-26 21:56:13 +0200 Josep Torra <n770galaxy@gmail.com>
668 * tests/check/Makefile.am:
669 build: silence error about pthread for 'make check' in osx
670 Fixes "clang: error: argument unused during compilation: '-pthread'"
672 2015-09-25 15:04:00 +0000 Nikita Bobkov <NikitaDBobkov@gmail.com>
674 * gst/rtsp-server/rtsp-client.c:
675 rtsp-client: Fix leaking of media in error cases
676 With additional fixes by Kseniya Vasilchuk <vasilchukkseniia@gmail.com>
677 and myself to make the media refcounting a bit easier to follow.
678 https://bugzilla.gnome.org/show_bug.cgi?id=755632
680 2016-08-02 15:08:22 +0300 Sebastian Dröge <sebastian@centricular.com>
682 * gst/rtsp-server/rtsp-client.c:
683 rtsp-client: Fix leaking of session in error cases
684 https://bugzilla.gnome.org/show_bug.cgi?id=755632
686 2016-07-11 21:16:04 +0200 Stefan Sauer <ensonic@users.sf.net>
689 Automatic update of common submodule
690 From f363b32 to f49c55e
692 2016-07-06 13:51:15 +0300 Sebastian Dröge <sebastian@centricular.com>
697 === release 1.9.1 ===
699 2016-07-06 13:28:12 +0300 Sebastian Dröge <sebastian@centricular.com>
705 * gst-rtsp-server.doap:
708 2016-06-24 02:02:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
711 configure: Need to add -DGST_STATIC_COMPILATION when building only statically
712 https://bugzilla.gnome.org/show_bug.cgi?id=767463
714 2016-06-21 11:49:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
717 Automatic update of common submodule
718 From ac2f647 to f363b32
720 2016-04-14 22:56:11 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
722 * gst/rtsp-server/rtsp-sdp.c:
723 * gst/rtsp-server/rtsp-sdp.h:
724 * gst/rtsp-server/rtsp-stream.c:
725 * gst/rtsp-server/rtsp-stream.h:
726 sdp: add rollover counters for all sender SSRC
727 We add different crypto sessions in MIKEY, one for each sender
728 SSRC. Currently, all of them will have the same security policy, 0.
729 The rollover counters are obtained from the srtpenc element using the
731 https://bugzilla.gnome.org/show_bug.cgi?id=730539
733 2016-06-07 20:44:42 +0100 Tim-Philipp Müller <tim@centricular.com>
735 * gst/rtsp-server/rtsp-media-factory.h:
736 * gst/rtsp-server/rtsp-server.h:
739 2016-05-25 10:28:43 +0100 Tim-Philipp Müller <tim@centricular.com>
741 * gst/rtsp-server/Makefile.am:
742 g-i: pass compiler env to g-ir-scanner
743 It's what introspection.mak does as well. Should
744 fix spurious build failures on gnome-continuous
745 (caused by g-ir-scanner getting compiler details
746 via python which is broken in some environments
747 so passing the compiler details bypasses that).
749 2016-05-18 16:48:44 +0100 Ian <ian.arkver.dev@gmail.com>
751 * gst/rtsp-server/rtsp-session.c:
752 rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header
753 This works with rtspsrc and live555, but fails with e.g. ffmpeg.
754 https://bugzilla.gnome.org/show_bug.cgi?id=766619
756 2016-03-07 14:48:38 +0100 Edward Hervey <bilboed@bilboed.com>
758 * gst/rtsp-sink/gstrtspclientsink.c:
759 rtspclientsink: Check return value of sscanf
760 And just make sure we always have 0/0 if we have an error
763 2016-04-25 08:55:25 -0400 Jake Foytik <jake.foytik@ipconfigure.com>
765 * gst/rtsp-server/rtsp-stream.c:
766 * tests/check/gst/rtspserver.c:
767 * tests/check/gst/stream.c:
768 rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
769 - Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
770 - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
771 - Create unit test for shared media.
772 https://bugzilla.gnome.org/show_bug.cgi?id=764744
774 2016-04-11 10:55:23 +0300 Sebastian Dröge <sebastian@centricular.com>
776 * gst/rtsp-server/rtsp-stream.c:
777 rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
778 For IPv6 addresses, binding to a multicast group does not work on Linux
779 either. Always bind to ANY and then later join the multicast group.
780 https://bugzilla.gnome.org/show_bug.cgi?id=764679
782 2016-04-14 10:05:02 +0100 Julien Isorce <j.isorce@samsung.com>
785 Automatic update of common submodule
786 From 6f2d209 to ac2f647
788 2016-04-06 10:09:46 +0200 Patricia Muscalu <patricia@axis.com>
790 * gst/rtsp-server/rtsp-thread-pool.c:
791 rtsp-thread-pool: explained why GSource is a part of ThreadImpl
792 Clarified why it is necessary to add source information to
793 GstRTSPThreadImpl. See the reported bug in GLib:
794 https://bugzilla.gnome.org/show_bug.cgi?id=720186
795 for more information.
796 https://bugzilla.gnome.org/show_bug.cgi?id=761702
798 2016-04-04 12:58:38 +0300 Sebastian Dröge <sebastian@centricular.com>
800 * examples/Makefile.am:
801 examples: Clean up CFLAGS/LDADD even more
802 The internal .la should come first and is part of LDADD, as is
805 2016-04-04 12:39:39 +0300 Sebastian Dröge <sebastian@centricular.com>
807 * examples/Makefile.am:
808 examples: Clean up CFLAGS/LDADD to link with the correct versions of all libraries
810 2016-04-03 12:06:29 +0300 Sebastian Dröge <sebastian@centricular.com>
812 * gst/rtsp-server/Makefile.am:
813 rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS)
815 2015-12-30 18:39:05 +0200 Sebastian Dröge <sebastian@centricular.com>
817 * gst/rtsp-server/rtsp-client.c:
818 * gst/rtsp-server/rtsp-media-factory.c:
819 * gst/rtsp-server/rtsp-media-factory.h:
820 * gst/rtsp-server/rtsp-media.c:
821 * gst/rtsp-server/rtsp-media.h:
822 * gst/rtsp-server/rtsp-sdp.c:
823 * gst/rtsp-server/rtsp-stream.c:
824 * gst/rtsp-server/rtsp-stream.h:
825 rtsp-server: Implement clock signalling according to RFC7273
826 For NTP and PTP clocks we signal the actual clock that is used and signal
827 the direct media clock offset.
828 For all other clocks we at least signal that it's the local sender clock.
829 This allows receivers to know which clock was used to generate the media and
830 its RTP timestamps. Receivers can then implement network synchronization,
831 either absolute or at least relative by getting the sender clock rate directly
832 via NTP/PTP instead of estimating it from RTP timestamps and packet receive
834 https://bugzilla.gnome.org/show_bug.cgi?id=760005
836 2016-03-02 19:42:58 +0200 Sebastian Dröge <sebastian@centricular.com>
838 * gst/rtsp-sink/gstrtspclientsink.c:
839 rtspclientsink: Add support for setting the multicast interface
840 https://bugzilla.gnome.org/show_bug.cgi?id=763000
842 2016-03-02 19:42:13 +0200 Sebastian Dröge <sebastian@centricular.com>
844 * gst/rtsp-server/rtsp-media-factory.c:
845 * gst/rtsp-server/rtsp-media-factory.h:
846 * gst/rtsp-server/rtsp-media.c:
847 * gst/rtsp-server/rtsp-media.h:
848 * gst/rtsp-server/rtsp-stream.c:
849 * gst/rtsp-server/rtsp-stream.h:
850 rtsp-media: Add support for setting the multicast interface
851 https://bugzilla.gnome.org/show_bug.cgi?id=763000
853 2016-03-07 08:50:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
855 * gst/rtsp-sink/gstrtspclientsink.c:
856 rtspclientsink: use new gst_element_class_add_static_pad_template()
857 https://bugzilla.gnome.org/show_bug.cgi?id=763196
859 2016-03-24 13:33:43 +0200 Sebastian Dröge <sebastian@centricular.com>
864 === release 1.8.0 ===
866 2016-03-24 13:00:35 +0200 Sebastian Dröge <sebastian@centricular.com>
872 * gst-rtsp-server.doap:
875 2016-03-16 23:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
877 * gst/rtsp-server/rtsp-stream.c:
878 rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
879 This would get us NO_PREROLL in the bin again and break seeking.
880 Thanks to Carlos Rafael Giani for helping to debug this!
881 https://bugzilla.gnome.org/show_bug.cgi?id=740509
883 === release 1.7.91 ===
885 2016-03-15 12:26:13 +0200 Sebastian Dröge <sebastian@centricular.com>
891 * gst-rtsp-server.doap:
894 2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
896 * gst/rtsp-server/rtsp-stream.c:
897 rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
898 Without this, RECORD pipelines are broken because
899 a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
900 added later. Previously it was there earlier and due to NO_PREROLL caused the
901 pipeline to preroll immediately
902 b) the udpsrc for the pipeline is added later and never set to PLAYING state,
903 as the corresponding code previously was only for PLAY pipelines.
904 https://bugzilla.gnome.org/show_bug.cgi?id=763281
906 2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
908 * gst/rtsp-server/rtsp-stream.c:
909 rtsp-stream: Fix typo in the docstring
910 gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
912 2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
914 * gst/rtsp-server/rtsp-stream.c:
915 rtsp-stream: Disable multicast loopback for all our sockets
916 On Windows this is a receiver-side setting, on Linux a sender-side setting. As
917 we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
918 loopback setting on the socket... while udpsink does which unfortunately has
919 no effect here on Windows but on Linux.
920 https://bugzilla.gnome.org/show_bug.cgi?id=757488
922 2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
924 * tests/check/gst/stream.c:
925 stream tests: added new tests
926 Test a case when the address pool only contains multicast addresses
927 and the client is requesting unicast udp.
928 Added tests for multicast ports allocation.
929 https://bugzilla.gnome.org/show_bug.cgi?id=757488
931 2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
933 * gst/rtsp-server/rtsp-stream.c:
934 rtsp-stream: Only bind multicast sockets to ANY on Windows
935 On Linux it is still needed to bind to the multicast address
936 to filter out random other packets, while on Windows binding
937 to multicast addresses just fails.
939 2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
941 * gst/rtsp-server/rtsp-stream.c:
942 rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
943 Otherwise we fail to allocate UDP ports if the pool only contains multicast
944 addresses, which is something that used to work before. For unicast addresses
945 if the pool contains none, we just allocate them as if there is no pool at
947 https://bugzilla.gnome.org/show_bug.cgi?id=757488
949 2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
951 * gst/rtsp-server/rtsp-client.c:
952 * gst/rtsp-server/rtsp-stream.c:
953 rtsp-server: Fix indentation
955 2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
957 * gst/rtsp-server/rtsp-stream.c:
958 rtsp-stream: Don't bind the sockets to multicast addresses
959 This works on Linux but fails completely on Windows. You're supposed
960 to bind to ANY and then join the multicast group.
961 https://bugzilla.gnome.org/show_bug.cgi?id=757488
963 === release 1.7.90 ===
965 2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
971 * gst-rtsp-server.doap:
974 2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
977 Automatic update of common submodule
978 From b64f03f to 6f2d209
980 2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
982 * gst/rtsp-sink/gstrtspclientsink.c:
983 * tests/check/gst/rtspclientsink.c:
984 rtspsink: Fix some leaks in rtspclientsink and the unit test.
985 https://bugzilla.gnome.org/show_bug.cgi?id=762525
987 2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
989 * tests/check/gst/media.c:
990 * tests/check/gst/rtspclientsink.c:
991 * tests/check/gst/rtspserver.c:
992 * tests/check/gst/stream.c:
993 tests: unit test fixes
994 Removed port allocation test from the media suite.
995 The port allocation failure is now in the stream suite.
997 Make sure that the media is suspended after the DESCRIBE request
998 before reconfiguring the UDP sinks.
1000 In the RECORD case we have to set async property to false
1001 for the appsink element in the test in order to make sure
1002 that the media pipeline doesn't hang in start_preroll().
1003 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1005 2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
1007 * gst/rtsp-server/rtsp-client.c:
1008 * gst/rtsp-server/rtsp-stream.c:
1009 * gst/rtsp-server/rtsp-stream.h:
1010 rtsp-stream: postpone UDP socket allocation until SETUP
1011 Postpone the allocation of the UDP sockets until we know
1012 what transport has been chosen by the client.
1013 Both unicast and multicast UDP sources are created in one
1015 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1017 2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
1019 * gst/rtsp-server/rtsp-stream.c:
1020 rtsp-stream: postpone the creation of the UDP sources
1021 Code refactoring: allocate the UDP ports after the sender and
1022 the reciver parts have been created.
1023 We postpone the creation of the UDP sources until the UDP
1024 ports have been allocated.
1025 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1027 2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
1029 * gst/rtsp-server/rtsp-stream.c:
1030 rtsp-stream: added function for setting UDP sources to PLAYING state
1031 Code refactoring: Introduced a function for setting UDP sources
1033 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1035 2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
1037 * gst/rtsp-server/rtsp-stream.c:
1038 rtsp-stream: added function for creating and configuring UDP sources
1039 Code refactoring: create and configure UDP sources in a separate function.
1040 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1042 2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
1044 * gst/rtsp-server/rtsp-stream.c:
1045 rtsp-stream: added function for RTP/RTCP socket configuration
1046 Code refactoring: configure RTP and RTCP sockets for UDP sinks
1047 in a separate function.
1048 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1050 2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
1052 * gst/rtsp-server/rtsp-stream.c:
1053 rtsp-stream: added function for creating and configuring UDP sinks
1054 Code refactoring: create and configure UDP sinks in a separate function.
1055 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1057 2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
1059 * gst/rtsp-server/rtsp-stream.c:
1060 rtsp-stream: added helper function for creating the sender/receiver parts
1061 Code refactoring: introduced helper function for creating
1062 the receiver and the sender parts of the streaming pipeline.
1063 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1065 2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
1070 === release 1.7.2 ===
1072 2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
1078 * gst-rtsp-server.doap:
1081 2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
1083 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
1084 uninstalled.pc: add support for non libtool build systems
1085 Currently the .la path is provided which requires to use libtool as
1086 mentioned in the GStreamer manual section-helloworld-compilerun.html.
1087 It is fine as long as the application is built using libtool.
1088 So currently it is not possible to compile a GStreamer application
1089 within gst-uninstalled with CMake or other build system different
1091 This patch allows to do the following in gst-uninstalled env:
1092 gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
1093 gstreamer-rtsp-server-1.0)
1094 Previously it required to prepend libtool --mode=link
1095 https://bugzilla.gnome.org/show_bug.cgi?id=720778
1097 2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
1099 * gst/rtsp-sink/gstrtspclientsink.c:
1100 rtspclientsink: remove check for impossible condition
1101 Goto error label checks stream to see if it needs to be unreferenced before
1102 returning, but this goto jumps happens before the stream is ever set, so it
1103 will always be NULL in this error label.
1106 2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
1108 * gst/rtsp-sink/gstrtspclientsink.c:
1109 rtspclientsink: clean switch statements
1110 Coverity demands for fallthrough statements to be clearly commented,
1111 to distinguish from accidental fall throughs. And it also needs all
1112 cases to finish with a break, even if the break is never going to be
1113 executed like in the case of a continue jump.
1117 2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
1119 * tests/check/Makefile.am:
1120 tests: extend the AM_TESTS_ENVIRONMENT from check.mak
1121 To get the CK_DEFAULT_TIMEOUT defined for all tests
1122 Also removes a 120 seconds timeout that was set as default
1123 explicitly in this module
1124 https://bugzilla.gnome.org/show_bug.cgi?id=761472
1126 2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
1130 Automatic update of common submodule
1131 From 86e4663 to b64f03f
1133 2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
1135 * gst/rtsp-server/rtsp-media.c:
1136 rtsp-media: fix state_lock not locked again when preroll fails
1137 https://bugzilla.gnome.org/show_bug.cgi?id=761399
1139 2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
1142 configure: Move plugin specific flags below all the others
1143 They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
1144 -no-undefined. And -no-undefined is required on Windows to build DLLs.
1146 2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
1148 * gst/rtsp-sink/gstrtspclientsink.c:
1149 rtspclientsink: Simplify slightly using new -base API
1150 Use the new Mikey and SDP API in the base plugins libs
1151 to simplify some code.
1152 https://bugzilla.gnome.org/show_bug.cgi?id=758180
1154 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1159 * gst/rtsp-sink/Makefile.am:
1160 * gst/rtsp-sink/gstrtspclientsink.c:
1161 * gst/rtsp-sink/gstrtspclientsink.h:
1162 * gst/rtsp-sink/plugin.c:
1163 * tests/check/Makefile.am:
1164 * tests/check/gst/rtspclientsink.c:
1165 rtspsink: Add rtspclientsink element
1166 Add an rtspclientsink element that accepts streams for which
1167 there is a registered payloader and sends them to
1168 an RTSP server using RECORD.
1169 Sending is synchronised to the pipeline clock. Payload-types
1170 are automatically selected. The 'new-payloader' signal is fired
1171 for custom configuration of payloaders when they are created.
1172 Can now stream a movie like this:
1174 ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
1175 decodebin name=depay1 ! audioconvert ! autoaudiosink )"
1177 gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
1178 queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
1179 https://bugzilla.gnome.org/show_bug.cgi?id=758180
1181 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1183 * gst/rtsp-server/rtsp-stream.c:
1184 * gst/rtsp-server/rtsp-stream.h:
1185 rtsp-stream: Add functions for using rtsp-stream from the client
1186 Add a boolean to indicate that the rtsp-stream is running on the
1187 'client' side of an RTSP connection, for sending streams via
1188 RECORD. In that case, the roles of the client/server ports
1189 in transport setup are swapped.
1190 https://bugzilla.gnome.org/show_bug.cgi?id=758180
1192 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1194 * gst/rtsp-server/rtsp-sdp.c:
1195 * gst/rtsp-server/rtsp-sdp.h:
1196 rtsp-sdp: Add gst_rtsp_sdp_from_stream()
1197 A new function that adds info from a GstRTSPStream into an SDP message.
1198 https://bugzilla.gnome.org/show_bug.cgi?id=758180
1200 2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
1202 * gst/rtsp-server/rtsp-media.c:
1203 rtsp-media: Fix mutex beeing unlocked while they should be locked
1204 https://bugzilla.gnome.org/show_bug.cgi?id=761226
1206 2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
1208 * gst/rtsp-server/rtsp-media-factory.c:
1209 rtsp-media-factory: add missing break in "clock" property setter
1212 2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
1214 * gst/rtsp-server/rtsp-stream.c:
1215 rtsp-stream: fixed assert during update transport
1216 When RTSP server trying update transport during multicast, it throws an
1217 assert. The assert is thrown because it is trying to get the parent of
1218 an non-existing funnel element.
1219 https://bugzilla.gnome.org/show_bug.cgi?id=760150
1221 2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
1223 * gst/rtsp-server/rtsp-permissions.h:
1224 * gst/rtsp-server/rtsp-thread-pool.h:
1225 * gst/rtsp-server/rtsp-token.h:
1226 docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
1227 gtk-doc can handle static inline functions just fine these days,
1228 there's no need for this stuff any more.
1230 2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1232 * gst/rtsp-server/rtsp-media.c:
1233 * gst/rtsp-server/rtsp-sdp.c:
1234 sdp: replace duplicated codes to call new base sdp apis
1235 https://bugzilla.gnome.org/show_bug.cgi?id=745880
1237 2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
1239 * examples/test-netclock.c:
1240 test-netclock: Use the new API to configure a clock directly
1242 2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
1244 * gst/rtsp-server/rtsp-media-factory.c:
1245 * gst/rtsp-server/rtsp-media-factory.h:
1246 * gst/rtsp-server/rtsp-media.c:
1247 * gst/rtsp-server/rtsp-media.h:
1248 rtsp-media: Add API to directly configure a clock on the media pipelines
1250 2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
1252 * gst/rtsp-server/rtsp-media.c:
1253 rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
1255 2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
1257 * gst/rtsp-server/rtsp-media-factory.c:
1258 rtsp-media-factory: Add FIXME for 2.0
1260 2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
1262 * gst/rtsp-server/rtsp-stream.c:
1263 rtsp-stream: Fix indentation
1265 2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1267 * gst/rtsp-server/rtsp-media.c:
1268 rtsp-media: Do not prepare media after media times out
1269 Deferred calls to start_prepare() can be deferred past the point until
1270 which wait_preroll() and by proxy gst_rtsp_media_get_status() is
1271 prepared to wait. Previously there was no lock and no check for this
1272 situation. This meant that a media could be prepared and unprepared
1273 simultaneously by two different threads. Now a lock is in place and a
1274 suitable check is done.
1275 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
1277 2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
1279 * gst/rtsp-server/rtsp-client.c:
1280 * gst/rtsp-server/rtsp-media-factory.c:
1281 * gst/rtsp-server/rtsp-media-factory.h:
1282 * gst/rtsp-server/rtsp-media.c:
1283 * gst/rtsp-server/rtsp-media.h:
1284 rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
1285 Without TEARDOWN it might be desireable to keep the media running and continue
1286 sending data to the client, even if the RTSP connection itself is
1288 Only do this for session medias that have only UDP transports. If there's at
1289 least on TCP transport, it will stop working and cause problems when the
1290 connection is disconnected.
1291 https://bugzilla.gnome.org/show_bug.cgi?id=758999
1293 2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
1298 === release 1.7.1 ===
1300 2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
1306 * gst-rtsp-server.doap:
1309 2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
1312 configure: Make -Bsymbolic check work with clang.
1313 Update the -Bsymbolic check with the version glib has. This version
1315 https://bugzilla.gnome.org/show_bug.cgi?id=759713
1317 2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
1319 * gst/rtsp-server/rtsp-session-pool.c:
1320 rtsp-session-pool: Avoid dollar sign ($) in session ids
1321 Live555 in VLC strips off dollar signs and then gets very confused,
1322 we don't loose too much entropy by just skipping it.
1324 2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
1326 * gst/rtsp-server/rtsp-address-pool.h:
1327 * gst/rtsp-server/rtsp-auth.h:
1328 * gst/rtsp-server/rtsp-client.h:
1329 * gst/rtsp-server/rtsp-media-factory-uri.h:
1330 * gst/rtsp-server/rtsp-media-factory.h:
1331 * gst/rtsp-server/rtsp-media.h:
1332 * gst/rtsp-server/rtsp-mount-points.h:
1333 * gst/rtsp-server/rtsp-permissions.h:
1334 * gst/rtsp-server/rtsp-server.h:
1335 * gst/rtsp-server/rtsp-session-media.h:
1336 * gst/rtsp-server/rtsp-session-pool.h:
1337 * gst/rtsp-server/rtsp-session.h:
1338 * gst/rtsp-server/rtsp-stream-transport.h:
1339 * gst/rtsp-server/rtsp-stream.h:
1340 * gst/rtsp-server/rtsp-thread-pool.h:
1341 * gst/rtsp-server/rtsp-token.h:
1342 rtsp-server: Add g_autoptr() support to all types
1343 https://bugzilla.gnome.org/show_bug.cgi?id=754464
1345 2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
1347 * gst/rtsp-server/rtsp-stream.c:
1348 rtsp-stream: fixed valgrind error
1349 Fixed the valgrind error in unit test. The UDP source created during
1350 gst_rtsp_stream_join_bin() was not released while destroying the rtp
1352 https://bugzilla.gnome.org/show_bug.cgi?id=759010
1354 2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1358 Automatic update of common submodule
1359 From b319909 to 86e4663
1361 2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
1363 * gst/rtsp-server/rtsp-client.c:
1364 rtsp-client: suspend media during setup request
1365 SETUP request from clients needs to suspend the media to clear the
1366 prerolled buffers. Otherwise it will not affect the prerolled buffer
1367 and the prerolled buffers will be incorrect (for example block-size
1368 from setup request will not affect the prerolled buffer unless the
1369 media is suspended).
1370 https://bugzilla.gnome.org/show_bug.cgi?id=758268
1372 2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
1374 * gst/rtsp-server/rtsp-stream.c:
1375 rtsp-stream: create stream pipeline based on transport
1376 Based on the protocol, create the rtsp stream pipeline. If only TCP or
1377 only UDP is set as the transport protocol, it will not add the extra tee
1378 or queue element to the pipeline. Both these elements will be added, if
1379 it supports both TCP and UDP protocols. This improves the pipeline
1380 performance when one protocol is present.
1381 https://bugzilla.gnome.org/show_bug.cgi?id=758179
1383 2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
1385 * gst/rtsp-server/rtsp-stream.c:
1386 rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
1387 Adding them when not needed will start some logic inside rtpbin that might be
1388 problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
1389 would start up a rtpjitterbuffer and behave in weird ways.
1390 We still set up the UDP sources for RTP receiving for a sender media to be
1391 able to receive any packets sent by the client for NAT traversal. They will
1392 all go to a fakesink though.
1393 Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
1394 NO_PREROLL, which will cause deadlocks when seeking the media as it will never
1395 receive ASYNC_DONE after a seek.
1396 https://bugzilla.gnome.org/show_bug.cgi?id=758319
1398 2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
1400 * gst/rtsp-server/rtsp-stream.c:
1401 rtsp-stream: Disable multicast loopback for the multicast udp sources too
1402 On POSIX this setting is for sender sockets, on Windows for receiver sockets.
1403 Previously we were only setting this for sender sockets, which caused looped
1404 back packets to be received on Windows if a multicast transport was used.
1406 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1408 * examples/test-record-auth.c:
1409 * examples/test-record.c:
1410 examples: Actually use the provided port in the record examples
1412 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1414 * examples/test-record-auth.c:
1415 test-record-auth: Add the option to build in TLS support
1417 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1419 * examples/test-auth.c:
1420 test-auth: Use an 'anonymous' user for unauthenticated default
1421 There's a comment on one of the resources that 'user' and 'admin'
1422 shouldn't even be able to see it, but they can if the default
1423 token is 'admin2', since that gives them access anyway.
1425 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1427 * examples/.gitignore:
1428 * examples/Makefile.am:
1429 * examples/test-record-auth.c:
1430 Add test-record-auth example
1432 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1434 * gst/rtsp-server/rtsp-client.c:
1435 * tests/check/gst/client.c:
1436 rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
1438 2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
1440 * gst/rtsp-server/rtsp-server.c:
1441 rtsp-server: Change the logic so we don't pop a NULL context
1442 When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
1443 will sometimes fail. This call is made before any context is pushed
1444 resulting in an attempt to pop a NULL context.
1445 https://bugzilla.gnome.org/show_bug.cgi?id=757949
1447 2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
1449 * tests/check/gst/rtspserver.c:
1450 rtspserver: Add udp-mcast transport SETUP test
1451 Refactor utility functions in the test file so they can handle
1452 more than UDP and TCP as lower transport.
1453 https://bugzilla.gnome.org/show_bug.cgi?id=756969
1455 2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
1457 * gst/rtsp-server/rtsp-stream.c:
1458 rtsp-stream: Always unref return value of gst_object_get_parent()
1459 Fixes a leak of a GstBin in the udp-mcast case.
1460 https://bugzilla.gnome.org/show_bug.cgi?id=756968
1462 2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
1465 Automatic update of common submodule
1466 From b99800a to b319909
1468 2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
1471 Use new GST_ENABLE_EXTRA_CHECKS #define
1472 https://bugzilla.gnome.org/show_bug.cgi?id=756870
1474 2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
1477 Automatic update of common submodule
1478 From 6babecd to b99800a
1480 2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
1483 Update GLib dependency to 2.40.0
1485 2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1487 * examples/test-mp4.c:
1488 * gst/rtsp-server/rtsp-stream.c:
1489 stream: listen to sender ssrc signals
1490 https://bugzilla.gnome.org/show_bug.cgi?id=746747
1492 2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
1495 common: update for new suppression
1496 Makes check-valgrind pass with glib 2.46
1498 2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1500 * gst/rtsp-server/rtsp-media.c:
1501 rtsp-media: Take reference to media that will be prepared
1502 default_prepare() takes a transfer-none reference GstRTSPMedia object.
1503 Later on a g_idle_source_new() is created and a pointer to the media
1504 object is passed as user data. If the media is freed before the idle
1505 source is dispatched the media object pointer is invalid, but the idle
1506 source callback expects it to still be valid. To fix this a reference to
1507 the media object is taken when registering the source callback function
1508 and a corresponding release of the reference is done when the souce is
1510 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
1512 2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
1514 * examples/test-launch.c:
1515 * examples/test-mp4.c:
1516 * examples/test-ogg.c:
1517 * examples/test-record.c:
1518 * examples/test-uri.c:
1519 rtsp-server: Fix memory leaks when context parse fails
1520 When g_option_context_parse fails, context and error variables are not getting free'd
1521 which results in memory leaks. Free'ing the same.
1522 And replacing g_error_free with g_clear_error, which checks if the error being passed
1523 is not NULL and sets the variable to NULL on free'ing.
1524 https://bugzilla.gnome.org/show_bug.cgi?id=753863
1526 2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
1531 === release 1.6.0 ===
1533 2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
1539 * gst-rtsp-server.doap:
1542 === release 1.5.91 ===
1544 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
1550 * gst-rtsp-server.doap:
1553 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
1555 * docs/libs/gst-rtsp-server-sections.txt:
1556 * gst/rtsp-server/rtsp-stream.c:
1557 stream: fix docs for recently-added get/set_buffer_size API
1558 https://bugzilla.gnome.org/show_bug.cgi?id=749095
1560 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
1562 * gst/rtsp-server/rtsp-media.c:
1563 rtsp-media: Don't crash on encrypted RTX SDP
1564 In parse_keymgmt(), don't mutate the input string that's been passed
1565 as const, especially since we might need the original value again if
1566 the same key info applies to multiple streams (RTX, for example).
1567 https://bugzilla.gnome.org/show_bug.cgi?id=754753
1569 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
1571 * examples/test-mp4.c:
1572 test-mp4: Support filenames with spaces in them. Error out on too few arguments
1574 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
1576 * examples/test-record.c:
1577 test-record: Check parameter count and print out help
1578 If no launch pipeline was supplied, print out some help
1580 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
1582 * gst/rtsp-server/rtsp-media.c:
1583 * gst/rtsp-server/rtsp-stream.c:
1584 * gst/rtsp-server/rtsp-stream.h:
1585 rtsp-stream: Implement UDP buffer size setting.
1586 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
1588 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
1589 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
1591 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
1593 * gst/rtsp-server/rtsp-media.h:
1594 rtsp-media: Fix small typo causing gtk-doc to complain
1596 === release 1.5.90 ===
1598 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
1604 * gst-rtsp-server.doap:
1607 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1609 * gst/rtsp-server/rtsp-media-factory.c:
1610 media-factory: get port number through gst_rtsp_url_get_port
1611 https://bugzilla.gnome.org/show_bug.cgi?id=753473
1613 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
1615 * tests/check/gst/media.c:
1616 media-test: Removing unnecessary assertion
1617 https://bugzilla.gnome.org/show_bug.cgi?id=753385
1619 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1621 * gst/rtsp-server/rtsp-server.c:
1622 Document that source keeps a ref on server until it's destroyed
1623 https://bugzilla.gnome.org/show_bug.cgi?id=749227
1625 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1627 * tests/check/gst/media.c:
1628 media-test: Test for multiple dynamic payload
1629 https://bugzilla.gnome.org/show_bug.cgi?id=753385
1631 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1633 * gst/rtsp-server/rtsp-media.c:
1634 media: Only add fakesink once per pipeline
1635 The intention is to prevent going PLAYING state before pads are created.
1636 If there was mutilple dynamic payload, it would leak few fakesink and
1637 actually prevent from ever reaching playing state.
1638 https://bugzilla.gnome.org/show_bug.cgi?id=753385
1640 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1642 * gst/rtsp-server/rtsp-media.c:
1643 Revert "rtsp-media: Only add 1 fakesink per pipeline"
1644 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
1646 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1648 * gst/rtsp-server/rtsp-media.c:
1649 rtsp-media: Only add 1 fakesink per pipeline
1650 There should be only one fakesink per pipeline, not per dynpay. This
1651 would lead to element naming clash.
1653 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
1655 * gst/rtsp-server/rtsp-media.c:
1656 rtsp-media: assertion error due to wrong condition check
1657 In media to caps function, reserved_keys array is being used for variable i,
1658 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
1659 changed it to variable j
1660 https://bugzilla.gnome.org/show_bug.cgi?id=753009
1662 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
1664 * gst/rtsp-server/rtsp-media.c:
1665 rtsp-media: Strip keys from the fmtp that we use internally in our caps
1666 Skip keys from the fmtp, which we already use ourselves for the
1667 caps. Some software is adding random things like clock-rate into
1668 the fmtp, and we would otherwise here set a string-typed clock-rate
1669 in the caps... and thus fail to create valid RTP caps
1670 https://bugzilla.gnome.org/show_bug.cgi?id=753009
1672 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1674 * gst/rtsp-server/rtsp-thread-pool.c:
1675 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
1676 https://bugzilla.gnome.org/show_bug.cgi?id=752640
1678 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
1681 Automatic update of common submodule
1682 From f74b2df to 9aed1d7
1684 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
1689 === release 1.5.2 ===
1691 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
1697 * gst-rtsp-server.doap:
1700 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
1702 * gst/rtsp-server/rtsp-client.c:
1703 * gst/rtsp-server/rtsp-client.h:
1704 * tests/check/gst/client.c:
1705 rtsp-client: allow application to decide what requirements are supported
1706 Add "check-requirements" signal and vfunc to allow application
1707 (and subclasses) to check the requirements.
1708 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
1709 https://bugzilla.gnome.org/show_bug.cgi?id=749417
1711 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1714 Automatic update of common submodule
1715 From 6015d26 to f74b2df
1717 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
1719 * gst/rtsp-server/rtsp-media.c:
1720 rtsp-media: Always use real payloader when creating streams
1721 A bin that contains the real payloader might be used as payloader. In this
1722 case we have to get the real payloader for the various properties it provides.
1723 Example use cases for this are bins that payload some media and then have
1724 additional elements that add metadata or RTP extension headers to the stream.
1725 https://bugzilla.gnome.org/show_bug.cgi?id=750800
1727 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
1729 * examples/test-netclock-client.c:
1730 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
1732 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
1734 * examples/test-netclock-client.c:
1735 * examples/test-netclock.c:
1736 test-netclock: Use new ntp-time-source property on rtpbin
1737 Select the clock time to be used as NTP time source. This allows proper
1738 synchronization between receivers, independent of sharing base times, and just
1739 requires them to use the same clock.
1741 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
1743 * examples/test-netclock-client.c:
1744 * examples/test-netclock.c:
1745 test-netclock: Setting the same base time on sender and receiver is not necessary
1746 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
1748 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1750 * gst/rtsp-server/rtsp-stream.c:
1751 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
1752 https://bugzilla.gnome.org/show_bug.cgi?id=750764
1754 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1756 * docs/libs/gst-rtsp-server.types:
1757 docs: add missing types
1758 https://bugzilla.gnome.org/show_bug.cgi?id=750764
1760 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1762 * docs/libs/gst-rtsp-server-sections.txt:
1763 docs: add missing apis
1764 https://bugzilla.gnome.org/show_bug.cgi?id=750764
1766 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
1768 * examples/test-netclock-client.c:
1769 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
1771 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1773 * docs/libs/gst-rtsp-server-sections.txt:
1774 * gst/rtsp-server/rtsp-auth.c:
1775 * gst/rtsp-server/rtsp-auth.h:
1776 GstRTSPAuth: Add client certificate authentication support
1777 https://bugzilla.gnome.org/show_bug.cgi?id=750471
1779 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
1781 * examples/test-netclock-client.c:
1782 test-netclock-client: Use new GstClock API to wait for clock synchronization
1784 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
1786 * examples/test-netclock-client.c:
1787 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
1788 A mainloop is needed to get glimagesink to display something on OSX, and
1789 the source-setup signal just makes things a little bit easier.
1791 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
1794 Automatic update of common submodule
1795 From d9a3353 to 6015d26
1797 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
1800 Automatic update of common submodule
1801 From d37af32 to d9a3353
1803 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
1806 Automatic update of common submodule
1807 From 21ba2e5 to d37af32
1809 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
1812 Automatic update of common submodule
1813 From c408583 to 21ba2e5
1815 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
1817 * docs/libs/Makefile.am:
1818 docs: remove variables that we define in the snippet from common
1819 This is syncing our Makefile.am with upstream gtkdoc.
1821 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
1824 Automatic update of common submodule
1825 From 44a3517 to c408583
1827 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
1832 === release 1.5.1 ===
1834 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
1840 * gst-rtsp-server.doap:
1843 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
1845 * gst/rtsp-server/rtsp-client.c:
1846 rtsp-client: No flush during Teardown.
1847 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
1848 backlog is empty it can happen that just a part of a message will be
1849 sent and rest is in backlog queue. If then flush during teardown
1850 just a part of message will be sent.This can lead to client miss
1851 teardown response since it expect to get the last part of message.
1852 The flushing during teardown was introduced to fix a deadlock that now
1853 is fixed more generally in handle_request by temporary setting backlog
1855 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
1857 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
1859 * tests/check/Makefile.am:
1860 tests: Use AM_TESTS_ENVIRONMENT
1861 Needed by the new automake test runner and the
1862 current version of the common submodule.
1864 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
1866 * gst/rtsp-server/rtsp-media.h:
1867 * gst/rtsp-server/rtsp-stream.h:
1868 rtsp-server: Use single-include rtsp header to make sure we get all definitions
1870 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
1872 * gst/rtsp-server/rtsp-media.c:
1873 rtsp-media: Mark some more functions static
1875 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
1877 * gst/rtsp-server/rtsp-media.c:
1878 rtsp-media: Only unblock the media in suspend() when actually changing the state
1879 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
1881 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
1883 * examples/test-video-rtx.c:
1884 examples: Use AVPF profile for the RTX example
1886 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
1888 * gst/rtsp-server/rtsp-sdp.c:
1889 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
1891 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1893 * gst/rtsp-server/rtsp-stream.c:
1894 rtsp-stream: get valid clock-rate from last-sample
1895 clock-rate in last-sample's caps is integer, not unsigned.
1896 To get this value properly, variable needs to be type-casted to int.
1897 https://bugzilla.gnome.org/show_bug.cgi?id=747614
1899 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
1903 autogen.sh: only run autopoint if gettext requested in configure.ac
1904 Not just because there happens to be a po directory.
1905 https://bugzilla.gnome.org/show_bug.cgi?id=748058
1907 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
1910 Revert "configure.ac: uncomment gettext version setup"
1911 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
1912 We don't need a gettext setup here and there's no po
1913 directory either, so no reason why autopoint would be
1914 run in the first place.
1915 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
1917 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
1919 * examples/test-multicast.c:
1920 * examples/test-multicast2.c:
1921 * examples/test-sdp.c:
1922 * examples/test-video-rtx.c:
1923 * examples/test-video.c:
1924 * tests/test-cleanup.c:
1925 * tests/test-reuse.c:
1926 Fix timeout function signatures across tests and examples
1928 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
1930 * tests/check/Makefile.am:
1931 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
1932 Make sure the test environment is set up.
1933 https://bugzilla.gnome.org//show_bug.cgi?id=747624
1935 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
1938 configure: bump automake requirement to 1.14 and autoconf to 2.69
1939 This is only required for builds from git, people can still
1940 build tarballs if they only have older autotools.
1941 https://bugzilla.gnome.org//show_bug.cgi?id=747624
1943 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
1946 configure.ac: uncomment gettext version setup
1947 Fixes autogen.sh. It would run autopoint, which would complain
1948 that it could not find the gettext version in configure.ac.
1949 https://bugzilla.gnome.org/show_bug.cgi?id=748058
1951 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1953 * examples/test-video-rtx.c:
1954 test-video-rtx: set exact payload type to PCMA payloader
1955 Setting wrong payload type causes failure to do retransmission through audio stream
1956 https://bugzilla.gnome.org/show_bug.cgi?id=747839
1958 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1960 * gst/rtsp-server/rtsp-media.c:
1961 * gst/rtsp-server/rtsp-stream.c:
1962 * gst/rtsp-server/rtsp-stream.h:
1963 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
1964 Because of duplicated g_signal_connect for request-aux-sender signal,
1965 wrong stream pointer is passed to the signal handler.
1966 Instead of passing each stream, pass stream array and get the relevant stream.
1967 https://bugzilla.gnome.org/show_bug.cgi?id=747839
1969 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
1973 Update autogen.sh to latest version from common
1974 Fixes build after aclocal_check etc. helpers have been removed.
1976 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
1979 Automatic update of common submodule
1980 From bc76a8b to c8fb372
1982 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
1984 * gst/rtsp-server/rtsp-stream.c:
1985 rtsp-stream: Limit the queues to 1 buffer
1986 We only need them to be able to pre-roll, queueing up more data here
1987 is only going to harm latency and memory usage.
1989 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
1991 * gst/rtsp-server/rtsp-stream.c:
1992 rtsp-stream: Update comment and ASCII art to the latest code
1993 We have a queue in front of the udpsink too to prevent the pipeline from
1996 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
1998 * gst/rtsp-server/rtsp-stream.c:
1999 rtsp-media: Properly return first rtptime
2000 Instead we where returning first GstBuffer timestamp. This would result
2001 in clock skew and unwanted behaviour in RTSP playback.
2002 https://bugzilla.gnome.org/show_bug.cgi?id=746479
2004 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
2006 * gst/rtsp-server/rtsp-stream.c:
2007 rtsp-stream: Don't leave buffer mapped
2008 If the seq is NULL, the RTP buffer was left mapped. We should always
2011 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
2016 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
2018 * gst/rtsp-server/rtsp-media-factory.c:
2019 * tests/check/gst/client.c:
2020 Fix double semicolons
2022 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
2024 * gst/rtsp-server/rtsp-stream.c:
2025 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
2026 This gives more accurate values than asking the payloader. There might be
2027 queueing happening between the payloader and the sink.
2028 https://bugzilla.gnome.org/show_bug.cgi?id=745704
2030 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
2032 * gst/rtsp-server/rtsp-media.c:
2033 rtsp-media: Don't seek for PLAY if the position will not change
2034 https://bugzilla.gnome.org/show_bug.cgi?id=745704
2036 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
2038 * gst/rtsp-server/rtsp-media.c:
2039 rtsp-media: Don't include payload type in the caps for framesize
2040 When the sdp media attribute framesize are converted to caps
2041 the <payload> should not be included.
2042 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
2043 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
2045 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
2047 * gst/rtsp-server/rtsp-sdp.c:
2048 rtsp-sdp: add payload type to the sdp framesize attribute
2049 The sdp framesize attribute is desribed in RFC6064. It is specified
2050 for payloading of H263 and has the following form
2051 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
2052 should be added to the caps in a payloader and the <payload type> should
2053 be added by the rtsp-server.
2054 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
2056 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
2058 * examples/test-uri.c:
2059 examples: test-uri: fix tainted variable
2060 Insignificant but this keeps Coverity happy.
2063 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
2065 * examples/.gitignore:
2066 * examples/Makefile.am:
2067 * examples/test-netclock-client.c:
2068 * examples/test-netclock.c:
2069 examples: Add a simple example of network synch for live streams.
2070 An example server and client that works for synchronising live streams
2071 only - as it can't support pause/play.
2073 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
2075 * gst/rtsp-server/rtsp-media-factory.c:
2076 * gst/rtsp-server/rtsp-media-factory.h:
2077 rtsp-media-factory: Add functions to set/get the media gtype
2078 Allow specifying the GType of a GstRtspMedia subclass to create
2079 as a simpler way to get the factory to create a custom
2080 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
2082 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
2084 * gst/rtsp-server/rtsp-media.c:
2085 rtsp-media: fix double unlock in _get_buffer_size()
2086 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
2087 because of double g_mutex_unlock () usage.
2088 https://bugzilla.gnome.org/show_bug.cgi?id=745434
2090 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
2092 * gst/rtsp-server/rtsp-session-pool.c:
2093 * gst/rtsp-server/rtsp-session.c:
2094 * gst/rtsp-server/rtsp-session.h:
2095 rtsp-session: Use monotonic time for RTSP session timeout
2096 Changed RTSP session timeout handling to monotonic time
2097 and deprecating the API for current system time.
2098 This fixes timeouts when the system time changes.
2099 https://bugzilla.gnome.org/show_bug.cgi?id=743346
2101 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
2103 * gst/rtsp-server/rtsp-client.c:
2104 * gst/rtsp-server/rtsp-media.c:
2105 rtsp-client: Only error out in PLAY if seeking actually failed
2106 If the media was just not seekable, we continue from whatever position we are
2107 and let the client decide if that is what is wanted or not.
2108 Only if the actual seek failed, we can't really recover and should error out.
2110 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
2112 * gst/rtsp-server/rtsp-stream.c:
2113 rtsp-stream: Add necessary queues between tee and multiudpsink
2114 https://bugzilla.gnome.org/show_bug.cgi?id=744379
2116 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
2118 * gst/rtsp-server/rtsp-client.c:
2119 * gst/rtsp-server/rtsp-media.c:
2120 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
2121 Instead error out properly the same way as if the SEEKING query already
2124 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
2126 * gst/rtsp-server/rtsp-stream.h:
2127 rtsp-stream: minor code formatting fix
2129 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
2131 * gst/rtsp-server/rtsp-media.c:
2132 rtsp-media: fix logic for collect_streams
2133 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
2134 all streams it knows if it got any, and can check if the transport mode is OK.
2137 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
2139 * gst/rtsp-server/rtsp-media.c:
2140 rtsp-media: Don't set the transport mode based on what elements we find
2141 Just print a warning if the one that was set before disagrees with what
2142 elements we found. It must already be set to something before as this
2143 function is called after we received the SDP from ANNOUNCE in RECORD mode,
2144 and we would reject ANNOUNCE if the RECORD flag was not set.
2146 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
2148 * tests/check/gst/rtspserver.c:
2149 tests: rtspserver: rename shadowed variable
2150 We have two different 'sink' variables here,
2151 rename one of them for clarity.
2153 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
2155 * gst/rtsp-server/rtsp-client.c:
2156 rtsp-client: fix awkward if clause
2158 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
2160 * examples/test-uri.c:
2161 examples: test-uri: improve uri argument handling and accept file names
2162 Print an error if the argument passed is not a URI and can't
2163 be converted into one, or no arguments have been provided.
2165 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
2167 * examples/test-uri.c:
2168 examples: test-uri: don't remove mount point after 10 seconds
2169 It's very irritating when trying to test stuff repeatedly
2170 and serves no real purpose other than showing that it can
2173 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
2175 * examples/.gitignore:
2176 examples: add new test-record to .gitignore
2178 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
2180 * examples/test-record.c:
2181 * gst/rtsp-server/rtsp-client.c:
2182 * gst/rtsp-server/rtsp-media-factory.c:
2183 * gst/rtsp-server/rtsp-media-factory.h:
2184 * gst/rtsp-server/rtsp-media.c:
2185 * gst/rtsp-server/rtsp-media.h:
2186 * tests/check/gst/rtspserver.c:
2187 rtsp-media: Use flags to distinguish between PLAY and RECORD media
2189 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
2191 * examples/test-record.c:
2192 test-record: Set latency for playback-style example to 2s instead of 200ms
2194 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
2196 * tests/check/gst/rtspserver.c:
2197 tests: add some unit tests for ANNOUNCE and RECORD
2198 https://bugzilla.gnome.org/show_bug.cgi?id=743175
2200 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
2202 * gst/rtsp-server/rtsp-client.c:
2203 rtsp-client: fix a couple of leaks in handle_announce
2205 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
2207 * gst/rtsp-server/rtsp-media-factory.c:
2208 * gst/rtsp-server/rtsp-media-factory.h:
2209 * gst/rtsp-server/rtsp-media.c:
2210 * gst/rtsp-server/rtsp-media.h:
2211 rtsp-media: Expose latency setting for setting the rtpbin latency
2213 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
2215 * examples/test-record.c:
2216 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
2218 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
2220 * gst/rtsp-server/rtsp-stream.c:
2221 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
2223 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
2225 * examples/Makefile.am:
2226 * examples/test-record.c:
2227 * gst/rtsp-server/rtsp-client.c:
2228 * gst/rtsp-server/rtsp-client.h:
2229 * gst/rtsp-server/rtsp-media-factory.c:
2230 * gst/rtsp-server/rtsp-media-factory.h:
2231 * gst/rtsp-server/rtsp-media.c:
2232 * gst/rtsp-server/rtsp-media.h:
2233 * gst/rtsp-server/rtsp-session-media.c:
2234 * gst/rtsp-server/rtsp-stream.c:
2235 * gst/rtsp-server/rtsp-stream.h:
2236 Add initial support for RECORD
2237 We currently only support media that is RECORD or PLAY only, not both at once.
2238 https://bugzilla.gnome.org/show_bug.cgi?id=743175
2240 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
2242 * gst/rtsp-server/rtsp-stream.c:
2243 rtsp-stream: RTCP and RTP transport cache cookies seperated
2244 RTCP packets were not sent because the same tr_cache_cookie was used for
2245 both RTP and RTCP. So only one of the tr_cache lists were populated
2246 depending on which one was sent first. If the tr_cache list is not
2247 populated then no packets can be sent. Most often this happened to be
2248 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
2249 resulted in both the tr_cache_lists to be populated regardless of which
2251 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
2253 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
2255 * gst/rtsp-server/rtsp-stream.c:
2256 rtsp-stream: fix false compiler warning
2257 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
2259 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
2261 * gst/rtsp-server/rtsp-client.c:
2262 rtsp-client: log interleaved data received
2264 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
2266 * gst/rtsp-server/rtsp-client.c:
2267 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
2269 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
2271 * gst/rtsp-server/rtsp-client.c:
2272 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
2274 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
2276 * gst/rtsp-server/rtsp-client.c:
2277 rtsp-client: Use a random session ID in the SDP
2278 RFC4566 Section 5.2 says that it should make the username, session id,
2279 nettype, addrtype and unicast address tuple globally unique. Always using
2280 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
2281 Instead let's create a 64 bit random number, which at least brings us
2282 closer to the goal of global uniqueness.
2283 https://tools.ietf.org/html/rfc4566#section-5.2
2285 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
2287 * examples/test-launch.c:
2288 * examples/test-mp4.c:
2289 * examples/test-ogg.c:
2290 * examples/test-uri.c:
2291 examples: Don't call gst_init() and gst_get_option_group()
2292 The latter calls the former at the appropriate time.
2294 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
2296 * gst/rtsp-server/rtsp-client.c:
2297 rtsp-client: Drop trailing \0 of RTSP DATA messages
2298 We add a trailing \0 in GstRTSPConnection to make parsing of
2299 string message bodies easier (e.g. the SDP from DESCRIBE) but
2300 for actual data this means we have to drop it or otherwise
2301 create invalid data.
2303 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
2305 * gst/rtsp-server/rtsp-stream.c:
2306 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
2307 Fixes crash when two threads access handle_new_sample() at the same
2308 time, one for RTP, one for RTCP.
2309 Otherwise, when iterating over the transports cache, it might be modified by
2310 another thread at the same time if the transports cookie has changed.
2311 https://bugzilla.gnome.org/show_bug.cgi?id=742954
2313 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
2315 * gst/rtsp-server/rtsp-stream.c:
2316 rtsp-stream: Set format=TIME on our app sources for TCP
2318 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
2320 * gst/rtsp-server/rtsp-session-pool.c:
2321 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
2322 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
2323 RFC 2326 states that session IDs may consist of alphanumeric as well as
2324 the safe characters $-_.+ -- N.B. the percent character is not allowed.
2325 Previously the session ID was URI-escaped, this meant that any character
2326 which was not alphanumeric or any of the characters +-._~ would be
2327 percent encoded. While the RFC (surprisingly) mentions that linear white
2328 space in session IDs should be URI-escaped, it does not say anything
2329 about other characters. Moreover no white space is allowed in the
2330 session ID. Finally the percent character which is the result of
2331 URI-escaping is not allowed in a session ID.
2332 So there is no reason to do any URI-escaping, and now it is removed.
2333 https://bugzilla.gnome.org/show_bug.cgi?id=742869
2335 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
2338 Automatic update of common submodule
2339 From f2c6b95 to bc76a8b
2341 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
2344 Fix 'make check' from top-level directory
2346 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2348 * examples/test-launch.c:
2349 * examples/test-mp4.c:
2350 * examples/test-ogg.c:
2351 * examples/test-uri.c:
2352 examples: Add command-line parsing and take a 'port' argument
2353 This allows users to run multiple servers on different ports for testing.
2354 Only done for examples that actually take arguments and hence are capable of
2355 outputting different streams for each instance on each port.
2356 https://bugzilla.gnome.org/show_bug.cgi?id=742115
2358 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
2360 * gst/rtsp-server/rtsp-client.c:
2361 * gst/rtsp-server/rtsp-client.h:
2362 rtsp-client: Add a send_message default signal handler
2363 This allows subclasses to easily hook into the response sending
2364 mechanism without doing everything from a signal, which seems
2365 awkward from subclasses.
2367 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
2370 Automatic update of common submodule
2371 From ef1ffdc to f2c6b95
2373 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2377 configure: add --disable-examples switch
2378 https://bugzilla.gnome.org/show_bug.cgi?id=741678
2380 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
2382 * examples/.gitignore:
2383 * examples/Makefile.am:
2384 * examples/test-video-rtx.c:
2385 examples: add a retransmisison example implementing RFC4588
2386 Currently only SSRC-multiplexed rtx streams are supported
2388 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
2390 * gst/rtsp-server/rtsp-stream.c:
2391 rtsp-stream: Fix some minor memory leaks
2393 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
2395 * gst/rtsp-server/rtsp-media.c:
2396 rtsp-media: Some minor cleanup
2398 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
2400 * gst/rtsp-server/rtsp-stream.c:
2401 rtsp-stream: Fix compiler warnings
2402 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
2403 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
2405 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
2406 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
2409 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
2411 * docs/libs/gst-rtsp-server-sections.txt:
2412 * gst/rtsp-server/rtsp-media-factory.c:
2413 * gst/rtsp-server/rtsp-media-factory.h:
2414 * gst/rtsp-server/rtsp-media.c:
2415 * gst/rtsp-server/rtsp-media.h:
2416 * gst/rtsp-server/rtsp-sdp.c:
2417 * gst/rtsp-server/rtsp-stream.c:
2418 * gst/rtsp-server/rtsp-stream.h:
2419 media: implement ssrc-multiplexed retransmission support
2420 based off RFC 4588 and the server-rtpaux example in -good
2422 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
2424 * gst/rtsp-server/rtsp-client.c:
2425 * gst/rtsp-server/rtsp-stream-transport.c:
2426 * gst/rtsp-server/rtsp-stream.c:
2427 rtsp: Ref transports in hash table.
2428 Also ref streams for transports.
2429 This solves a crash when reciving a rtcp after teardown but before
2431 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
2433 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
2436 Automatic update of common submodule
2437 From 7bb2bce to ef1ffdc
2439 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
2441 * gst/rtsp-server/rtsp-client.c:
2442 client: refactor cleanup of cached media
2444 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
2446 * tests/check/gst/client.c:
2448 The session leak is now fixed, lets remove those FIXME comments.
2450 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
2452 * tests/check/gst/rtspserver.c:
2453 tests: Test to setup two sessions on one connection
2454 https://bugzilla.gnome.org/show_bug.cgi?id=739112
2456 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
2458 * tests/check/gst/rtspserver.c:
2459 tests: Test setup with tcp transport
2460 https://bugzilla.gnome.org/show_bug.cgi?id=739112
2462 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
2464 * gst/rtsp-server/rtsp-client.c:
2465 client: Configure transport after creating session media
2466 The default implementation of configure_client_transport() in
2467 rtsp-client uses the session media when it chooses channels for
2468 interleaved traffic.
2469 https://bugzilla.gnome.org/show_bug.cgi?id=739112
2471 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
2473 * gst/rtsp-server/rtsp-client.c:
2474 * gst/rtsp-server/rtsp-session-media.c:
2475 client: Stop caching media in client when doing setup
2476 If the media has been managed by a session media, it should not be
2477 cached in the client any longer. The GstRTSPSessionMedia object is now
2478 responsible for unpreparing the GstRTSPMedia object using
2479 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
2481 https://bugzilla.gnome.org/show_bug.cgi?id=739112
2483 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2485 * gst/rtsp-server/rtsp-stream.c:
2486 rtsp-stream: unref srtp decoder when leaving bin
2487 https://bugzilla.gnome.org/show_bug.cgi?id=739481
2489 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2491 * gst/rtsp-server/rtsp-client.c:
2492 rtsp-client: mikey memory leaks
2493 https://bugzilla.gnome.org/show_bug.cgi?id=739383
2495 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
2498 Automatic update of common submodule
2499 From 84d06cd to 7bb2bce
2501 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
2504 Parallelise 'make check-valgrind'
2506 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
2509 Automatic update of common submodule
2510 From a8c8939 to 84d06cd
2512 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
2515 Automatic update of common submodule
2516 From 36388a1 to a8c8939
2518 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
2520 * gst/rtsp-server/rtsp-media.c:
2521 rtsp-media: deactivate media when shutting down from paused
2522 This was only done when going directly from playing.
2523 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
2525 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2527 * gst/rtsp-server/rtsp-client.c:
2528 * gst/rtsp-server/rtsp-context.h:
2529 rtsp-client: add stream transport to context
2530 We add the stream transport to the context so we can get the configured
2531 client stream transport in the setup request signal.
2532 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
2534 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2536 * gst/rtsp-server/rtsp-stream.c:
2537 stream: release lock even not all transports have been removed
2538 We don't want to keep the lock even we return FALSE because not all the
2539 transports have been removed. This could lead into a deadlock.
2540 https://bugzilla.gnome.org/show_bug.cgi?id=737797
2542 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
2544 * gst/rtsp-server/rtsp-sdp.c:
2545 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
2546 These were renamed in GstRTPBasePayload in 1.0
2548 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2550 * gst/rtsp-server/rtsp-client.c:
2551 client: set session media to NULL without the lock
2552 We need to set session medias to NULL without the client lock otherwise
2553 we can end up in a deadlock if another thread is waiting for the lock
2554 and media unprepare is also waiting for that thread to end.
2555 https://bugzilla.gnome.org/show_bug.cgi?id=737690
2557 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
2559 * gst/rtsp-server/rtsp-media.c:
2560 rtsp-media: Set state to UNPREPARING in all cases
2562 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
2564 * gst/rtsp-server/rtsp-media.c:
2565 media: set state to unpreparing when unprepare is initiated
2566 https://bugzilla.gnome.org/show_bug.cgi?id=737675
2568 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
2570 * gst/rtsp-server/rtsp-client.c:
2571 rtsp-client: Remove backlog limit while processings requests
2572 If the backlog limit is kept two cases of deadlocks may be
2573 encountered when streaming over TCP. Without the backlog
2574 limit this deadlocks can not happen, at the expence of
2576 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
2578 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
2580 * gst/rtsp-server/rtsp-client.c:
2581 rtsp-client: do not free main context before rtsp watch
2582 https://bugzilla.gnome.org/show_bug.cgi?id=737110
2584 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
2586 * tests/check/gst/rtspserver.c:
2587 tests: Extend unit test timeout to accomodate for valgrind
2588 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
2590 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
2592 * gst/rtsp-server/rtsp-client.c:
2593 * gst/rtsp-server/rtsp-session.c:
2594 * gst/rtsp-server/rtsp-stream-transport.c:
2595 rtsp-*: Treat sending packets to clients as keepalive
2596 As long as gst-rtsp-server can successfully send RTP/RTCP data to
2597 clients then the client must be reading. This change makes the server
2598 timeout the connection if the client stops reading.
2599 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
2601 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
2603 * gst/rtsp-server/rtsp-client.c:
2604 rtsp-client: Allow backlog to grow while expiring session
2605 Allow the send backlog in the RTSP watch to grow to unlimited size while
2606 attempting to bring the media pipeline to NULL due to a session
2607 expiring. Without this change the appsink element cannot change state
2608 because it is blocked while rendering data in the new_sample callback.
2609 This callback will block until it has successfully put the data into the
2610 send backlog. There is a chance that the send backlog is full at this
2611 point which means that the callback may block for a long time, possibly
2612 forever. Therefore the media pipeline may also be prevented from
2613 changing state for a long time.
2614 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
2616 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
2618 * gst/rtsp-server/rtsp-client.c:
2619 rtsp-client: Make old compilers happy
2620 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
2621 Just in case that guint8 doesn't fit in a pointer. Just in case ...
2623 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
2625 * gst/rtsp-server/rtsp-client.c:
2626 client: raise the backlog limits before pausing
2627 We need to raise the backlog limits before pausing the pipeline or else
2628 the appsink might be blocking in the render method in wait_backlog() and
2629 we would deadlock waiting for paused.
2630 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
2632 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
2634 * gst/rtsp-server/rtsp-client.c:
2635 client: make define for the WATCH_BACKLOG
2636 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
2638 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
2640 * gst/rtsp-server/rtsp-client.c:
2641 client: simplify session transport handling
2642 link/unlink of the transport in a session was done to keep track of all
2643 TCP transports and to send RTP/RTCP data to the streams. We can simplify
2644 that by putting all the TCP transports in a hashtable indexed with the
2646 We also don't need to link/unlink the transports when we pause/resume
2647 the streams. The same effect is already achieved when we pause/play the
2648 media. Indeed, when we pause the media, the transport is removed from
2649 the media and the callbacks will not be called anymore.
2650 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
2652 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
2654 * gst/rtsp-server/rtsp-stream-transport.c:
2655 * gst/rtsp-server/rtsp-stream-transport.h:
2656 stream-transport: make method to handle received data
2657 Make a method to handle the data received on a channel. It sends the
2658 data to the stream of the transport on the RTP or RTCP pads based on
2661 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
2663 * examples/test-mp4.c:
2664 test: add example of dumping RTCP reports
2666 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
2668 * gst/rtsp-server/rtsp-media.c:
2669 * gst/rtsp-server/rtsp-stream.c:
2670 * gst/rtsp-server/rtsp-stream.h:
2671 rtsp-media: Make sure that sequence numbers are monotonic after pause
2672 The sequence number is not monotonic for RTP packets after pause. The
2673 reason is basepayloader generates a randon sequence number when the
2674 pipeline goes from ready to pause. With this fix generation of sequence
2675 number will be monotonic when going from pause to play request.
2676 https://bugzilla.gnome.org/show_bug.cgi?id=736017
2678 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
2680 * gst/rtsp-server/rtsp-client.c:
2681 rtsp-client: Protect saved clients watch with a mutex
2682 Fixes a crash when close() is called while merging clients
2683 in handle_tunnel(). In that case close() would destroy the
2684 watch while it is still being used in handle_tunnel().
2685 https://bugzilla.gnome.org/show_bug.cgi?id=735570
2687 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
2689 * gst/rtsp-server/rtsp-stream.c:
2690 rtsp-stream: Remove the multicast group udp sources when removing from the bin
2692 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
2694 * gst/rtsp-server/rtsp-media.c:
2695 * gst/rtsp-server/rtsp-stream.c:
2696 * gst/rtsp-server/rtsp-stream.h:
2697 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
2698 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
2699 seeking and will always continue counting the time. This leads to
2700 the NPT after a backwards seek to be something completely different
2701 to the actual seek position.
2702 https://bugzilla.gnome.org/show_bug.cgi?id=732644
2704 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
2706 * examples/test-appsrc.c:
2707 examples: fix another reference leak
2708 gst_rtsp_media_get_element() returns a new ref.
2710 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2712 * examples/test-appsrc.c:
2713 examples: unref element after usage
2714 gst_bin_get_by_name_recurse_up() returns an element
2715 reference that must be unreffed after usage.
2716 https://bugzilla.gnome.org/show_bug.cgi?id=734546
2718 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
2720 * gst/rtsp-server/rtsp-media.c:
2721 signals: Fix copy-pasto in target-state signal offset
2723 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
2727 Makefile: Add usage of build-checks step
2728 Allows building checks without running them
2730 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
2732 * gst/rtsp-server/rtsp-stream.c:
2733 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
2734 When a UDP multicast transport is used it is expected that the server listens
2735 for RTP and RTCP packets on the multicast group with the corresponding port.
2736 Without this we will never get RTCP packets from clients in multicast mode.
2737 https://bugzilla.gnome.org/show_bug.cgi?id=732238
2739 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
2744 === release 1.4.0 ===
2746 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
2752 * gst-rtsp-server.doap:
2755 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
2757 * gst/rtsp-server/rtsp-media.h:
2758 media: correct misspelled words in description
2759 https://bugzilla.gnome.org/show_bug.cgi?id=733244
2761 === release 1.3.91 ===
2763 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
2769 * gst-rtsp-server.doap:
2772 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
2774 * docs/libs/gst-rtsp-server-sections.txt:
2777 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
2779 * gst/rtsp-server/rtsp-server.c:
2780 server: implement client REMOVE filter
2782 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
2784 * gst/rtsp-server/rtsp-client.c:
2785 * gst/rtsp-server/rtsp-client.h:
2786 client: expose _close() method
2787 Expose a previously internal close method to close the client
2790 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
2792 * gst/rtsp-server/rtsp-session-pool.c:
2793 session-pool: signal session-removed outside of the lock
2794 Release the lock before emiting the session-removed signal.
2796 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
2798 * gst/rtsp-server/rtsp-client.c:
2799 * gst/rtsp-server/rtsp-server.c:
2800 * gst/rtsp-server/rtsp-session-pool.c:
2801 * gst/rtsp-server/rtsp-session.c:
2802 * gst/rtsp-server/rtsp-stream.c:
2803 filter: Release lock in filter functions
2804 Release the object lock before calling the filter functions. We need to
2805 keep a cookie to detect when the list changed during the filter
2806 callback. We also keep a hashtable to make sure we only call the filter
2807 function once for each object in case of concurrent modification.
2808 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
2810 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
2812 * gst/rtsp-server/rtsp-client.c:
2813 client: check if watch is set in handle_teardown()
2814 The unit tests run without a watch
2816 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
2818 * tests/check/gst/client.c:
2819 client tests: send teardown to cleanup session
2821 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
2823 * tests/check/gst/rtspserver.c:
2824 server tests: send teardown to cleanup session
2826 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
2828 * gst/rtsp-server/rtsp-client.c:
2829 client: keep ref to client for the session removed handler
2830 This extra ref will be dropped when all client sessions have been
2831 removed. A session is removed when a client sends teardown, closes its
2832 endpoint of the TCP connection or the sessions expires.
2833 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
2835 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
2837 * gst/rtsp-server/rtsp-client.c:
2838 * gst/rtsp-server/rtsp-session.c:
2839 * tests/check/gst/client.c:
2840 client: manage media in session as a last step
2841 Once we manage a media in a session, we can't unmanage it anymore
2842 without destroying it. Therefore, first check everything before we
2843 manage the media, otherwise if something is wrong we have no way to
2845 If we created a new session and something went wrong, remove the session
2846 again. Fixes a leak in the unit test.
2848 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
2850 * examples/test-mp4.c:
2851 * examples/test-ogg.c:
2852 examples: print 'stream ready at url' for mp4 and ogg example
2854 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
2856 * gst/rtsp-server/rtsp-client.c:
2857 * gst/rtsp-server/rtsp-sdp.c:
2858 rtsp: fix for MIKEY api change
2860 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
2862 * gst/rtsp-server/rtsp-client.c:
2863 client: free watch context only once
2864 The watch context is freed when the source is destroyed. Avoids
2865 a CRITICAL when we try to unref the context twice.
2867 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
2869 * gst/rtsp-server/rtsp-client.c:
2872 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
2874 * gst/rtsp-server/rtsp-client.c:
2875 client: protect sessions with lock
2876 Protect the list of sessions with the lock.
2877 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
2879 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
2881 * gst/rtsp-server/rtsp-client.c:
2882 Client: keep a ref to the session
2883 Don't just keep a weak ref to the session objects but use a hard ref. We
2884 will be notified when a session is removed from the pool (expired) with
2885 the new session-removed signal.
2886 Don't automatically close the RTSP connection when all the sessions of
2887 a client are removed, a client can continue to operate and it can create
2888 a new session if it wants. If you want to remove the client from the
2889 server, you have to use gst_rtsp_server_client_filter() now.
2890 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
2891 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
2893 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
2895 * gst/rtsp-server/rtsp-session-pool.c:
2896 * gst/rtsp-server/rtsp-session-pool.h:
2897 session-pool: add session-removed signal
2898 Add a signal to be notified when a session is removed from the pool.
2900 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
2902 * gst/rtsp-server/Makefile.am:
2903 * gst/rtsp-server/rtsp-server.h:
2904 Make rtsp-server.h a single-include header, use it for G-I
2905 https://bugzilla.gnome.org/show_bug.cgi?id=732411
2907 === release 1.3.90 ===
2909 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
2915 * gst-rtsp-server.doap:
2918 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
2920 * gst/rtsp-server/rtsp-stream.c:
2921 stream: crypto can be NULL
2923 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
2925 * gst/rtsp-server/rtsp-client.c:
2926 * gst/rtsp-server/rtsp-media.c:
2927 * gst/rtsp-server/rtsp-mount-points.c:
2928 introspection: add missing allow-none annotations
2929 https://bugzilla.gnome.org/show_bug.cgi?id=730952
2931 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
2933 * gst/rtsp-server/rtsp-address-pool.c:
2934 * gst/rtsp-server/rtsp-media.c:
2935 * gst/rtsp-server/rtsp-session-media.c:
2936 * gst/rtsp-server/rtsp-session-pool.c:
2937 * gst/rtsp-server/rtsp-stream-transport.c:
2938 * gst/rtsp-server/rtsp-stream.c:
2939 * gst/rtsp-server/rtsp-token.c:
2940 introspection: add (nullable) annotations to return values
2941 https://bugzilla.gnome.org/show_bug.cgi?id=730952
2943 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
2945 * gst/rtsp-server/rtsp-client.c:
2946 * gst/rtsp-server/rtsp-stream.c:
2947 gi: improve annotations
2948 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
2950 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
2952 * gst/rtsp-server/rtsp-client.c:
2953 * gst/rtsp-server/rtsp-media-factory.c:
2954 * gst/rtsp-server/rtsp-media.c:
2955 * gst/rtsp-server/rtsp-server.c:
2956 signals: use generic marshal function
2957 Use the generic C marshal function.
2958 Use more explicit type instead of G_TYPE_POINTER
2960 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
2962 * gst/rtsp-server/rtsp-context.h:
2963 context: add type macro
2965 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
2967 * gst/rtsp-server/rtsp-client.c:
2968 * gst/rtsp-server/rtsp-sdp.c:
2969 * gst/rtsp-server/rtsp-sdp.h:
2970 sdp: hide key length defines
2971 They don't have a namespace.
2973 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
2978 === release 1.3.3 ===
2980 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
2986 * gst-rtsp-server.doap:
2989 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2991 * gst/rtsp-server/rtsp-client.c:
2992 * gst/rtsp-server/rtsp-sdp.c:
2993 * gst/rtsp-server/rtsp-sdp.h:
2994 mikey: add different key length parameters
2995 Add encryption and authentication key length parameters to MIKEY. For
2996 the encoders, the key lengths are obtained from the cipher and auth
2997 algorithms set in the caps. For the decoders, they are obtained while
2998 parsing the key management from the client.
2999 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
3001 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
3003 * tests/check/gst/stream.c:
3004 stream tests: Make sure we get right multicast address from stream
3005 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
3007 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
3009 * gst/rtsp-server/rtsp-client.c:
3010 client: ref the context until rtsp watch is alive
3011 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
3013 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
3015 * gst/rtsp-server/rtsp-client.c:
3016 client: Destroy the rtsp watch after connection close
3018 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
3020 * gst/rtsp-server/rtsp-media.c:
3021 media: fix confusing comment
3023 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
3025 * gst/rtsp-server/rtsp-session.c:
3026 rtsp-session: Timeout in header.
3027 Adding the possbilty to always have timout in header.
3028 This is configurabe with setting "timeout-always-visible".
3029 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
3031 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
3036 === release 1.3.2 ===
3038 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
3045 * gst-rtsp-server.doap:
3048 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
3051 Automatic update of common submodule
3052 From 211fa5f to 1f5d3c3
3054 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
3056 * gst/rtsp-server/rtsp-client.c:
3057 client: store TCP ports in transport
3058 Store the TCP ports in the transport when we are doing RTSP over TCP.
3059 This way, we can easily get to the ports from the transport.
3060 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
3062 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
3064 * gst/rtsp-server/rtsp-stream.c:
3065 stream: add signals for new RTP/RTCP encoders
3066 New signals to allow the user to configure the dynamically created
3068 https://bugzilla.gnome.org/show_bug.cgi?id=730228
3070 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
3072 * gst/rtsp-server/rtsp-media.c:
3073 * gst/rtsp-server/rtsp-media.h:
3074 media: Make suspend()/unsuspend() virtual
3075 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
3077 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
3079 * gst/rtsp-server/rtsp-client.c:
3080 client: fix send-message signal marshaller
3081 Use generic marshalling for the send-message signal. It has
3082 two POINTER arguments, not just one.
3083 https://bugzilla.gnome.org/show_bug.cgi?id=729900
3085 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
3087 * tests/check/gst/media.c:
3088 tests: add and remove pads only once
3089 In this test we simulate a dynamic pad by watching the caps event.
3090 Because of renegotiation in the base payloader now, this caps is sent
3091 multiple times but we can only deal with 1 invocation, use a variable to
3092 only 'add and remove' the pad once.
3094 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
3096 * tests/check/gst/rtspserver.c:
3097 tests: add unit test for correct handling of Require headers
3098 https://bugzilla.gnome.org/show_bug.cgi?id=729426
3100 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
3102 * gst/rtsp-server/rtsp-client.c:
3103 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
3104 Servers must handle Require headers and must report a failure
3105 if they don't handle any of the Required options, see RFC 2326,
3106 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
3107 https://bugzilla.gnome.org/show_bug.cgi?id=729426
3109 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
3114 === release 1.3.1 ===
3116 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
3122 * gst-rtsp-server.doap:
3125 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
3128 Automatic update of common submodule
3129 From bcb1518 to 211fa5f
3131 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
3136 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
3138 * tests/check/gst/sessionmedia.c:
3139 tests: fix memory leak in sessionmedia unit test
3141 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
3143 * gst/rtsp-server/rtsp-client.c:
3144 client: emit a signal before sending a message
3145 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
3147 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
3149 * gst/rtsp-server/rtsp-client.c:
3150 client: pass context to send_message
3151 Pass the current context to send_message, we will need it later.
3153 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
3155 * gst/rtsp-server/rtsp-client.c:
3156 client: fix typo in comment
3158 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
3160 * gst/rtsp-server/rtsp-media.c:
3161 media: Do not stop thread twice if default_prepare() fails
3163 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
3165 * gst/rtsp-server/rtsp-client.c:
3166 client: set the watch to flushing before going to NULL
3167 First set the watch to flushing so that we unblock any current and
3168 future attempt to send data on the watch, Then set the pipeline to
3170 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
3172 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
3174 * gst/rtsp-server/rtsp-session-pool.c:
3175 * tests/check/gst/sessionpool.c:
3176 rtsp-session-pool: Fixes annotation
3177 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
3178 in the sessionpool test.
3179 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
3181 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
3183 * gst/rtsp-server/rtsp-media.c:
3184 * gst/rtsp-server/rtsp-media.h:
3185 media: make media_prepare virtual
3186 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
3188 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
3190 * gst/rtsp-server/rtsp-media.c:
3191 * tests/check/gst/media.c:
3192 media: stop the thread in more error cases
3194 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
3196 * gst/rtsp-server/rtsp-media.c:
3197 * tests/check/gst/media.c:
3198 media: allow NULL as the thread
3199 Use the default context whan passing a NULL thread.
3201 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
3203 * gst/rtsp-server/rtsp-client.c:
3204 rtsp-client: indent cleanup
3205 Coverity was moaning about unreachable code, and I think it was just
3206 confused by { being before the label. We'll see if it pops up again.
3209 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
3211 * gst/rtsp-server/rtsp-client.c:
3212 * gst/rtsp-server/rtsp-media.c:
3213 client: Add drop-backlog property
3214 When we have too many messages queued for a client (currently hardcoded
3215 to 100) we overflow and drop the messages. Add a drop-backlog property
3216 to control this behaviour. Setting this property to FALSE will retry
3217 to send the messages to the client by waiting for more room in the
3219 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
3221 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
3223 * gst/rtsp-server/rtsp-client.c:
3224 client: support for POST before GET when setting up a tunnel
3226 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
3228 * gst/rtsp-server/rtsp-client.c:
3229 client: remove watch of the second client after http tunnel setup
3230 The second client will be freed after the HTTP tunnel has been set up.
3231 Make sure it's RTSP watch is never dispatched again.
3232 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
3234 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
3236 * gst/rtsp-server/rtsp-media.c:
3237 * tests/check/gst/media.c:
3238 media: Make media_prepare() fail if port allocation fails
3239 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
3241 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
3243 * tests/check/gst/media.c:
3244 media test: cleanup the thread pool in tests
3246 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
3248 * gst/rtsp-server/rtsp-media.c:
3249 * tests/check/gst/media.c:
3250 rtsp-media: Unblock blocked streams in unprepare
3251 The streams will be blocked when a live media is prepared.
3252 The streams should be unblocked in gst_rtsp_media_unprepare.
3253 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
3255 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
3257 * gst/rtsp-server/rtsp-media.c:
3258 media: release the state lock when going to NULL
3259 Set our state to UNPREPARING and release the state-lock before
3260 setting the pipeline to the NULL state. This way, any pad-added
3261 callback will be able to take the state-lock and check that we are now
3262 unpreparing instead of deadlocking.
3263 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
3265 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
3267 * gst/rtsp-server/rtsp-media.c:
3268 media: protect status with lock
3269 Make sure we only update the status with the lock.
3271 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
3273 * gst/rtsp-server/rtsp-client.c:
3274 * gst/rtsp-server/rtsp-sdp.c:
3275 rtsp: update for MIKEY API changes
3277 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
3279 * gst/rtsp-server/rtsp-client.c:
3280 client: parse the mikey response from the client
3281 Parse the mikey response from the client and update the policy for
3284 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
3286 * gst/rtsp-server/rtsp-stream.c:
3287 * gst/rtsp-server/rtsp-stream.h:
3288 stream: add method to set crypto info
3289 Make a method to configure the crypto information of a stream.
3290 Set udpsrc in READY instead of PAUSED so that we can configure caps
3293 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
3295 * gst/rtsp-server/rtsp-client.c:
3296 client: cleanup error paths
3298 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
3300 * gst/rtsp-server/rtsp-media.c:
3303 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
3305 * examples/test-video.c:
3306 test: enable SRTP only on RTSPS
3307 We only want to enable SRTP when doing rtsp over TLS so that we can
3308 exchange the keys in a secure way.
3310 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
3312 * examples/test-video.c:
3313 test: print an error on failure
3315 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
3318 * examples/test-video.c:
3319 * gst/rtsp-server/rtsp-sdp.c:
3320 * gst/rtsp-server/rtsp-stream.c:
3321 * tests/check/Makefile.am:
3322 stream: add SRTP support
3323 Install srtp encoder and decoder elements in rtpbin
3326 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3328 * tests/check/Makefile.am:
3329 * tests/check/gst/sessionpool.c:
3330 tests: Add unit tests for sessionpool
3331 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
3333 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3335 * tests/check/gst/threadpool.c:
3336 tests: Improve code coverage of rtsp-threadpool tests
3337 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
3339 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3341 * tests/check/gst/sessionmedia.c:
3342 tests: Improve code coverage for rtsp-session-media
3343 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
3345 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3347 gobject-introspection: Add annotations to support language bindings
3348 In addition a few cosmetic changes:
3349 * Adjust the order of arguments
3350 * Fix typo: occured -> occurred
3351 * Fix indentation after Return:-clauses
3352 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
3354 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3356 * gst/rtsp-server/rtsp-stream.c:
3357 rtsp-stream: Don't mix IPv4 and IPv6 addresses
3358 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
3360 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
3362 * gst/rtsp-server/rtsp-stream.c:
3363 stream: take caps after the session manager
3364 Take the caps for the SDP after they leave the rtpbin so that we can
3365 also get the properties added by rtpbin elements.
3367 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
3369 * gst/rtsp-server/rtsp-stream.c:
3370 stream: release lock while pushing out packets
3371 Keep a cache of the transports and use this to iterate the transport
3372 while pushing packets. This allows us to release the lock early.
3373 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
3375 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
3377 * gst/rtsp-server/rtsp-client.c:
3378 * gst/rtsp-server/rtsp-client.h:
3379 rtsp-client: vmethod for modifying tunnel GET response
3380 Add a vmethod tunnel_http_response where the response to the HTTP GET
3381 for tunneled connections can be modified.
3382 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
3384 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
3386 * gst/rtsp-server/rtsp-sdp.c:
3387 sdp: make 1 media line per profile
3388 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
3389 line in the SDP for each profile. The client is then supposed to pick
3390 one of the profiles in the SETUP request. Because the m= lines have the
3391 same pt, the client also knows that only 1 option is possible.
3393 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
3395 * gst/rtsp-server/rtsp-media-factory.c:
3396 * gst/rtsp-server/rtsp-media-factory.h:
3397 * gst/rtsp-server/rtsp-media.c:
3398 factory: add profile property and pass to media and streams
3400 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
3402 * examples/test-multicast.c:
3403 * gst/rtsp-server/rtsp-sdp.c:
3404 sdp: pass multicast connection for multicast-only stream
3405 Pass the multicast address of the stream in the connection info in the
3406 SDP so that clients try a multicast connection first.
3407 Only allow multicast connections in the test-multicast example. Also
3408 increase the TTL a little.
3410 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3413 .gitignore: Ignore gcov intermediate files
3414 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
3416 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
3418 * gst/rtsp-server/rtsp-stream.c:
3419 stream: release some locks in error cases
3421 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3423 docs: Enable and fix gtk-doc warnings
3424 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
3425 * addresspool/mediafactory: Add missing annotation colon
3426 * stream: Annotate return value
3427 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
3429 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
3432 Automatic update of common submodule
3433 From fe1672e to bcb1518
3435 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
3438 Automatic update of common submodule
3439 From 1a07da9 to fe1672e
3441 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
3443 * examples/Makefile.am:
3444 examples: use LDADD for libs instead of LDFLAGS
3446 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
3449 configure: make sure releases are in .doap file
3451 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
3453 * examples/test-cgroups.c:
3454 examples: test-cgroups: don't put code with side effects into g_assert()
3455 The g_assert() might get compiled out with the right
3456 compiler/preprocessor flags.
3458 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
3460 * examples/.gitignore:
3461 examples: add cgroup test binary to .gitignore
3463 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
3465 * examples/test-cgroups.c:
3466 examples: fix cgroup test build
3467 Fixes build failure caused by compiler warning:
3468 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
3470 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
3473 .gitignore: ignore temp files created in the course of 'make check'
3475 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
3477 * gst/rtsp-server/rtsp-media.c:
3478 rtsp-media: don't loose frames handling new PLAY request
3479 If client supplied a range check if the range specifies the start point.
3480 If not, then do an accurate seek to the current position. If a start
3481 point was specified do do a key unit seek to make sure the streaming
3482 starts with decodeable frames.
3483 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
3485 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
3487 * gst/rtsp-server/rtsp-media.c:
3488 Revert "media: only flush when setting a new start position"
3489 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
3490 We need to do the flush in all cases, demuxer block currently for
3493 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
3495 * gst/rtsp-server/rtsp-media.c:
3496 media: only flush when setting a new start position
3497 Only flush the pipeline when we change the start position with
3499 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
3501 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
3503 * gst/rtsp-server/rtsp-stream.c:
3504 stream: set ttl-mc before adding the socket
3505 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
3506 never be set on socket.
3507 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
3509 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
3511 * gst/rtsp-server/rtsp-media.c:
3512 media: stop thread if media is already prepared
3513 in gst_rtsp_media_prepare() the thread is not used if media is already
3514 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
3516 https://bugzilla.gnome.org/show_bug.cgi?id=724182
3518 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
3521 build: Ship gst-rtsp-server.doap file
3523 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
3525 * tests/check/gst/rtspserver.c:
3526 tests: Fix another compiler warning with gcc
3528 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
3530 * gst/rtsp-server/rtsp-client.c:
3531 * gst/rtsp-server/rtsp-mount-points.c:
3532 * gst/rtsp-server/rtsp-stream.c:
3533 * tests/check/gst/client.c:
3534 rtsp-server: Fix lots of compiler warnings with clang
3536 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
3539 * gst-rtsp-server.doap:
3540 * tests/Makefile.am:
3541 configure: Synchronise with the configure scripts of the other modules
3543 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
3546 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
3548 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
3550 * gst/rtsp-server/rtsp-media.c:
3551 * gst/rtsp-server/rtsp-stream.c:
3552 Revert "rtsp-server: support build against last stable release"
3553 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
3554 Let us require 1.2.3 now, which is going to be released in a few
3557 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
3559 * gst/rtsp-server/rtsp-session-media.c:
3560 * gst/rtsp-server/rtsp-stream-transport.c:
3561 session: improve RTP-Info
3562 Ignore streams that can't generate RTP-Info instead of failing.
3563 Don't return the empty string when all streams are unconfigured but
3564 return NULL so that we don't generate and empty RTP-Info header.
3565 Improve docs a little.
3567 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
3569 * gst/rtsp-server/rtsp-session-media.c:
3570 Don't free rtpinfo GString when it is NULL
3571 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
3573 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
3575 * gst/rtsp-server/rtsp-media.c:
3576 media: only set keyframe flag when modifying start
3577 Only set the keyframe flag when we modify the start position. The
3578 keyframe flag should probably be ignored when no change is requested but
3579 until we can claim this is all documented properly and all demuxer
3580 implement this, avoid setting the flag.
3581 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
3583 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
3585 * gst/rtsp-server/rtsp-thread-pool.c:
3586 thread-pool: Unref source after mainloop has quit to avoid races in GLib
3587 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
3589 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
3591 * gst/rtsp-server/rtsp-stream.c:
3592 stream: handle NULL seqnum and rtptime arguments
3594 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
3596 * gst/rtsp-server/rtsp-thread-pool.c:
3597 * tests/check/gst/threadpool.c:
3598 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
3599 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
3601 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
3603 * gst/rtsp-server/rtsp-stream.c:
3604 stream: add fallback for missing stats property
3605 Use a fallback when the payloader does not have a stats property
3606 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
3608 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
3611 Automatic update of common submodule
3612 From f7bc1c3 to 1a07da9
3614 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
3616 * gst/rtsp-server/rtsp-stream.c:
3617 stream: don't leak stats structure
3618 Don't leak the stats structure and deal with NULL stats.
3620 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
3622 * gst/rtsp-server/rtsp-stream.c:
3623 stream: Get rtpinfo properties atomically from payloader
3624 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
3626 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
3628 * gst/rtsp-server/rtsp-media.c:
3629 media: refactor state change functions and signals
3630 Make functions to set the target state and the pipeline state and emit
3631 the signals from those functions.
3633 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
3635 * gst/rtsp-server/rtsp-media.c:
3636 * gst/rtsp-server/rtsp-media.h:
3637 media: add signal to notify of pending state changes
3639 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
3641 * gst/rtsp-server/rtsp-media.c:
3642 * gst/rtsp-server/rtsp-stream.c:
3643 rtsp-server: support build against last stable release
3644 Until 1.2.3 is out with the new get_type function and we
3647 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
3649 * gst/rtsp-server/rtsp-stream.c:
3650 stream: fix compilation
3652 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
3654 * gst/rtsp-server/rtsp-media.c:
3655 * gst/rtsp-server/rtsp-media.h:
3656 * gst/rtsp-server/rtsp-stream.c:
3657 * gst/rtsp-server/rtsp-stream.h:
3658 stream: add property to configure profiles
3660 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
3662 * gst/rtsp-server/rtsp-client.c:
3663 client: let stream check supported transport
3664 Delegate the check if a transport is allowed to the stream.
3665 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
3667 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
3669 * gst/rtsp-server/rtsp-stream.c:
3670 * gst/rtsp-server/rtsp-stream.h:
3671 stream: add method to check supported transport
3672 Add a method to check if a transport is supported
3674 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
3677 configure.ac: Only check for gstreamer-check, not check
3678 We include check in gstreamer-check since quite some time now.
3680 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
3682 * gst/rtsp-server/rtsp-session-media.c:
3683 * gst/rtsp-server/rtsp-stream-transport.c:
3684 * gst/rtsp-server/rtsp-stream.c:
3685 * gst/rtsp-server/rtsp-stream.h:
3686 stream: return clock-rate from get_rtpinfo
3687 And use it to correct the rtptime to the requested start-time.
3688 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
3690 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
3692 * gst/rtsp-server/rtsp-session-media.c:
3693 * gst/rtsp-server/rtsp-stream-transport.c:
3694 * gst/rtsp-server/rtsp-stream-transport.h:
3695 session-media: calculate start-time
3697 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
3699 * gst/rtsp-server/rtsp-stream-transport.c:
3700 * gst/rtsp-server/rtsp-stream.c:
3701 * gst/rtsp-server/rtsp-stream.h:
3702 stream: also return the running-time
3703 Return the running-time in the rtpinfo as well.
3705 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
3707 * gst/rtsp-server/rtsp-client.c:
3708 * gst/rtsp-server/rtsp-session-media.c:
3709 * gst/rtsp-server/rtsp-session-media.h:
3710 * gst/rtsp-server/rtsp-stream-transport.c:
3711 * gst/rtsp-server/rtsp-stream-transport.h:
3712 session-media: let the session-media make the RTPInfo
3713 Add method to create the RTPInfo for a stream-transport.
3714 Add method to create the RTPInfo for all stream-transports in a
3716 Use the session-media RTPInfo code in client. This allows us to refactor
3717 another method to link the TCP callbacks.
3719 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
3721 mount-points: sort sequence before g_sequence_lookup
3722 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
3723 sort sequence if dirty, otherwise lookup will fail.
3724 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
3726 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
3729 configure: rename package from gst-rtsp to gst-rtsp-server
3730 To match git module name and avoid confusion with the
3731 rtsp lib in gst-plugins-base and rtsp plugin in -good.
3733 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
3736 configure: bump core/base/good requirement to 1.2.0
3737 Bump to released stable version and make implicit
3738 requirements explicit.
3740 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
3745 Fix broken gettext setup which is not used anyway
3747 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
3750 Automatic update of common submodule
3751 From dbedaa0 to d48bed3
3753 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
3755 * gst/rtsp-server/rtsp-client.c:
3756 * gst/rtsp-server/rtsp-media.c:
3757 * gst/rtsp-server/rtsp-media.h:
3758 media: add setup_sdp vmethod
3759 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
3760 gst_rtsp_media_setup_sdp.
3761 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
3763 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
3765 * gst/rtsp-server/rtsp-stream.c:
3766 rtsp-stream: Check return value of sscanf
3767 streamid is only valid if sscanf matched something.
3769 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
3771 * gst/rtsp-server/rtsp-client.c:
3772 rtsp-client: Fix iteration
3773 Wouldn't even enter the code block otherwise (i++ was used as the check
3774 and not the postfix).
3776 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
3778 * gst/rtsp-server/rtsp-client.c:
3779 * gst/rtsp-server/rtsp-client.h:
3780 client: add vmethod to configure media and streams
3781 Implement a vmethod that can be used to configure the media and the
3782 streams based on the current context. Handle the blocksize handling in
3783 the default handler.
3784 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
3786 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
3789 Make git ignore more unit test binaries
3791 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
3793 * gst/rtsp-server/rtsp-address-pool.h:
3794 * gst/rtsp-server/rtsp-auth.h:
3795 * gst/rtsp-server/rtsp-client.h:
3796 * gst/rtsp-server/rtsp-context.h:
3797 * gst/rtsp-server/rtsp-media-factory-uri.h:
3798 * gst/rtsp-server/rtsp-media-factory.h:
3799 * gst/rtsp-server/rtsp-media.h:
3800 * gst/rtsp-server/rtsp-mount-points.h:
3801 * gst/rtsp-server/rtsp-server.h:
3802 * gst/rtsp-server/rtsp-session-media.h:
3803 * gst/rtsp-server/rtsp-session-pool.h:
3804 * gst/rtsp-server/rtsp-session.h:
3805 * gst/rtsp-server/rtsp-stream-transport.h:
3806 * gst/rtsp-server/rtsp-stream.h:
3807 * gst/rtsp-server/rtsp-thread-pool.h:
3808 * gst/rtsp-server/rtsp-token.h:
3809 rtsp-server: add padding to many public structures
3810 Not mini objects though, since they are not subclassable
3811 anyway, nor kept on the stack or inlined in a structure.
3813 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
3815 media: add new create_rtpbin vmethod
3816 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
3817 https://bugzilla.gnome.org/show_bug.cgi?id=719734
3819 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
3821 * tests/check/gst/media.c:
3822 tests: fix memory leak, free test's thread pool
3823 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
3825 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
3827 * gst/rtsp-server/rtsp-stream-transport.c:
3828 stream-transport: free url in finalize
3830 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
3832 * gst/rtsp-server/rtsp-media.c:
3833 media: also do state change in suspended state
3835 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
3837 * gst/rtsp-server/rtsp-client.c:
3838 * gst/rtsp-server/rtsp-media.c:
3839 media: also handle prepare and range in suspended state
3840 When we are suspended, we are already prepared.
3841 We can get the range in the suspended state.
3843 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
3845 * tests/check/Makefile.am:
3846 * tests/check/gst/sessionmedia.c:
3847 check: add test for uri in setup
3848 Added unit tests for the new functionality in GstRTSPStreamTransport.
3849 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
3851 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
3853 * gst/rtsp-server/rtsp-client.c:
3854 client: store setup uri and use in PLAY response
3855 Store the uri used when doing the setup and use that in the PLAY
3857 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
3859 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
3861 * gst/rtsp-server/rtsp-stream-transport.c:
3862 * gst/rtsp-server/rtsp-stream-transport.h:
3863 stream-transport: add method to get/set url
3865 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
3867 * gst/rtsp-server/rtsp-client.c:
3868 client: suspend after SDP and unsuspend before PLAYING
3869 Based on patches by Ognyan Tonchev <ognyan@axis.com>
3870 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
3872 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
3874 * gst/rtsp-server/rtsp-media-factory.c:
3875 * gst/rtsp-server/rtsp-media-factory.h:
3876 * gst/rtsp-server/rtsp-media.c:
3877 * gst/rtsp-server/rtsp-media.h:
3878 * gst/rtsp-server/rtsp-session-media.c:
3879 * gst/rtsp-server/rtsp-session.c:
3880 * tests/check/gst/media.c:
3881 * tests/check/gst/mediafactory.c:
3882 media: add suspend modes
3883 Add support for different suspend modes. The stream is suspended right after
3884 producing the SDP and after PAUSE. Different suspend modes are available that
3885 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
3886 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
3887 state and RESET will bring the pipeline to the NULL state.
3888 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
3889 this means that the pipeline needs to be prerolled again.
3890 Base on patches by Ognyan Tonchev <ognyan@axis.com>
3891 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3893 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
3895 * gst/rtsp-server/rtsp-media.c:
3896 media: start live streams in blocked state
3897 Start live streams in the blocked state and make them preroll using the
3898 messages. This ensure that no data is played by the sink until we explicitly
3899 unblock the stream right before going to PLAYING.
3900 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3902 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
3904 * gst/rtsp-server/rtsp-media.c:
3905 media: refactor starting and waiting for preroll
3906 Based on patches from Ognyan Tonchev <ognyan@axis.com>
3907 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3909 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
3911 * gst/rtsp-server/rtsp-stream.c:
3912 * gst/rtsp-server/rtsp-stream.h:
3913 stream: add API to block streams
3914 Add an API to block on the streams and make it post a message.
3915 Based on patch by Ognyan Tonchev <ognyan@axis.com>
3916 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3918 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
3920 * docs/libs/Makefile.am:
3921 docs: Specify the override file
3922 Even if it's empty (for now) it avoids make distcheck complaining
3924 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
3926 * gst/rtsp-server/rtsp-media.c:
3927 media: move default implementations to where they are used
3929 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
3931 * gst/rtsp-server/rtsp-media.c:
3932 media: take the right lock in gst_rtsp_media_set_pipeline_state()
3933 We need to take the state_lock when calling this method.
3935 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
3937 * gst/rtsp-server/rtsp-media.c:
3938 media: handle add-added on non-bins too
3939 Handle dynamic payloaders that are not bins, as used in the unit-test.
3941 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3943 * gst/rtsp-server/rtsp-media-factory.c:
3944 * gst/rtsp-server/rtsp-media-factory.h:
3945 * gst/rtsp-server/rtsp-media.c:
3946 rtsp-media/-factory: Fix request pad name comments
3947 These must be escaped for gtk-doc to parse the comments without warnings.
3949 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
3951 rtsp-media: remove transports if media is in error status
3952 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
3953 trying to change to GST_STATE_NULL and media is in error status, we
3954 remove all transports.
3955 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
3957 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
3959 * gst/rtsp-server/rtsp-media.c:
3960 rtsp-media: use element metadata to find payloader
3961 Use the element metadata to find the payloader instead of checking
3963 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
3965 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
3967 rtsp-stream: add getter for payload type
3968 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
3969 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
3970 element and create the stream with this one instead of the dynpay%d
3972 https://bugzilla.gnome.org/show_bug.cgi?id=712396
3974 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3976 * gst/rtsp-server/rtsp-client.c:
3977 * gst/rtsp-server/rtsp-context.h:
3978 * gst/rtsp-server/rtsp-media.c:
3979 * gst/rtsp-server/rtsp-mount-points.c:
3980 * gst/rtsp-server/rtsp-server.c:
3981 * gst/rtsp-server/rtsp-token.c:
3982 rtsp-*: Refer to NULL as a constant in comments
3984 https://bugzilla.gnome.org/show_bug.cgi?id=714988
3986 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3988 rtsp-*: Fix type name typos in comments
3989 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
3990 * rtsp-auth: Refer to part of constant name as text
3991 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
3992 * rtsp-session-media: Fix GstRTSPSessionMedia typo
3993 * rtsp-stream: Fix typo when refering to GstBin
3994 https://bugzilla.gnome.org/show_bug.cgi?id=714988
3996 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3999 * docs/libs/gst-rtsp-server-docs.sgml:
4000 * docs/libs/gst-rtsp-server-sections.txt:
4001 docs: Improve documentation
4002 * Include annotation-glossary to quiet gtk-doc
4003 * Rename remaining ClientState -> Context
4004 * Rename object hierarchy file
4005 * Remove stale chapter references
4006 * Add missing function and object references
4007 * Include missing GstRTSPAddressPoolResult
4008 https://bugzilla.gnome.org/show_bug.cgi?id=714988
4010 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
4012 * gst/rtsp-server/rtsp-client.c:
4013 * gst/rtsp-server/rtsp-server.c:
4014 * gst/rtsp-server/rtsp-session-pool.c:
4015 * gst/rtsp-server/rtsp-session.c:
4016 * gst/rtsp-server/rtsp-stream.c:
4017 rtsp-server: sprinkle some allow-none annotations for g-i
4019 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
4021 * gst/rtsp-server/rtsp-stream.c:
4022 * gst/rtsp-server/rtsp-stream.h:
4023 stream: add method to filter transports
4024 Add a method to safely iterate and collect the stream transports
4025 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
4027 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
4029 * gst/rtsp-server/rtsp-client.c:
4030 * gst/rtsp-server/rtsp-server.c:
4031 * gst/rtsp-server/rtsp-session-pool.c:
4032 * gst/rtsp-server/rtsp-session.c:
4033 rtsp: allow NULL func in filters
4034 Passing a null function make the filters return a list of
4037 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
4039 * gst/rtsp-server/rtsp-address-pool.c:
4040 * tests/check/gst/addresspool.c:
4041 address-pool: fix address increment
4042 Use a guint instead of guint8 to increment the address. It's still not
4043 completely correct because a guint might not be able to hold the complete
4044 address range, but that's an enhacement for later.
4045 Add unit test to test improved behaviour.
4046 https://bugzilla.gnome.org/show_bug.cgi?id=708237
4048 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
4050 * gst/rtsp-server/rtsp-client.c:
4051 * tests/check/gst/client.c:
4052 client: allow absolute path in requests
4053 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
4055 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
4057 * gst/rtsp-server/rtsp-client.c:
4058 * gst/rtsp-server/rtsp-client.h:
4059 client: make make_path_from_uri a vmethod
4061 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
4063 * docs/libs/gst-rtsp-server-sections.txt:
4064 * gst/rtsp-server/rtsp-stream.c:
4065 * gst/rtsp-server/rtsp-stream.h:
4066 * tests/check/Makefile.am:
4067 * tests/check/gst/stream.c:
4068 stream: Add functions to get rtp and rtcp sockets
4069 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
4071 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
4073 * gst/rtsp-server/rtsp-context.c:
4074 * gst/rtsp-server/rtsp-context.h:
4075 context: defing a GType for the context
4076 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
4078 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
4080 * gst/rtsp-server/Makefile.am:
4081 * gst/rtsp-server/rtsp-auth.c:
4082 * gst/rtsp-server/rtsp-context.c:
4083 * gst/rtsp-server/rtsp-media.c:
4084 * gst/rtsp-server/rtsp-mount-points.c:
4085 * gst/rtsp-server/rtsp-server.h:
4086 * gst/rtsp-server/rtsp-session-media.c:
4087 * gst/rtsp-server/rtsp-session.c:
4088 * gst/rtsp-server/rtsp-stream.c:
4089 Fixed several GIR warnings
4091 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
4093 * gst/rtsp-server/rtsp-auth.c:
4096 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
4098 * tests/check/Makefile.am:
4099 * tests/check/gst/token.c:
4100 tests: Add unit tests for token
4101 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
4103 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
4105 * gst/rtsp-server/rtsp-token.c:
4106 token: Validate args for gst_rtsp_token_is_allowed
4107 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
4109 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
4111 * gst/rtsp-server/rtsp-token.c:
4112 token: Fix bug when creating empty token
4113 We always want to have a valid GstStructure in the token.
4114 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
4116 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
4118 * gst/rtsp-server/rtsp-thread-pool.c:
4119 thread-pool: avoid race in shutdown
4120 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
4121 don't actually stop the mainloop ever. Solve this race by adding an idle source
4122 to the mainloop that calls the _quit. This way we immediately exit the mainloop
4123 if quit was called before we started it.
4125 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
4127 * tests/check/Makefile.am:
4128 * tests/check/gst/permissions.c:
4129 tests: Add unit tests for permissions
4130 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
4132 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
4134 * tests/check/gst/mediafactory.c:
4135 tests: Test mediafactory permissions
4136 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
4138 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
4140 * gst/rtsp-server/rtsp-permissions.c:
4141 permissions: Fix refcounting when adding/removing roles
4142 Previously a role that was removed was unreffed twice, and when
4143 replacing an existing role the replaced role was freed while still being
4144 referenced. Both bugs are now fixed.
4145 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
4147 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
4149 * tests/check/gst/media.c:
4150 * tests/check/gst/mediafactory.c:
4151 * tests/check/gst/rtspserver.c:
4152 tests: Check gst_rtsp_url_parse return value
4153 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
4155 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
4158 Automatic update of common submodule
4159 From 865aa20 to dbedaa0
4161 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
4163 * gst/rtsp-server/rtsp-server.c:
4164 rtsp-server: Fix socket leak
4165 https://bugzilla.gnome.org/show_bug.cgi?id=710088
4167 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
4169 * gst/rtsp-server/rtsp-session-pool.c:
4170 rtsp-session-pool: Make sure session IDs are properly URI-escaped
4171 https://bugzilla.gnome.org/show_bug.cgi?id=643812
4173 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4175 * examples/.gitignore:
4176 * examples/test-video.c:
4177 examples: fix compilation when WITH_AUTH is defined
4178 https://bugzilla.gnome.org/show_bug.cgi?id=710228
4180 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
4183 gitignore: Add new test binary
4185 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
4187 * tests/check/Makefile.am:
4188 * tests/check/gst/threadpool.c:
4189 thread-pool: Add unit test for the thread pools
4190 https://bugzilla.gnome.org/show_bug.cgi?id=710228
4192 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
4194 * gst/rtsp-server/rtsp-thread-pool.c:
4195 thread-pool: Fix thread leak when reusing threads
4196 https://bugzilla.gnome.org/show_bug.cgi?id=709730
4198 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
4200 * gst/rtsp-server/rtsp-server.c:
4201 * tests/check/gst/rtspserver.c:
4202 tests: fixed racy behavior in rtspserver tests
4203 https://bugzilla.gnome.org/show_bug.cgi?id=710078
4205 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
4207 * tests/check/gst/addresspool.c:
4208 tests: Improve address pool unit tests
4209 Add a range with mixed IPV4 and IPV6 addresses to pool.
4210 Get an IPV4 address from an IPV6-only pool.
4211 Get an IPV6 address from an IPV4-only pool.
4212 Reserve a IPV6 address from an IPV4-only pool.
4213 Check for unicast addresses in multicast-only pool.
4214 Check for unicast addresses in uni-/multicast-mixed pool.
4215 https://bugzilla.gnome.org/show_bug.cgi?id=710128
4217 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4219 * gst/rtsp-server/rtsp-client.c:
4220 client: append query string in PAUSE/PLAY/TEARDOWN as well
4222 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
4224 * gst/rtsp-server/rtsp-client.c:
4225 client: Add query to control path
4226 If the SETUP url contains a query it must be appended to the control
4227 path so that it matches any already created stream in the media. The
4228 query will also be appended to the session media path.
4230 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4232 * gst/rtsp-server/rtsp-media.c:
4233 rtsp-media: remove old line
4235 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
4237 * gst/rtsp-server/rtsp-stream.c:
4238 stream: Correct control comparison
4239 https://bugzilla.gnome.org/show_bug.cgi?id=709176
4241 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4243 * gst/rtsp-server/rtsp-media.c:
4244 media: Check dynamically if the pipeline supports seeking
4245 We should not depend on whether or not the pipeline state change
4246 returned NO_PREROLL or not. A media could dynamically change its
4247 element and switch from seekable to non seekable so it's best to test
4248 the seekable nature of the pipeline dynamically when we try to do a seek.
4250 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4252 * gst/rtsp-server/rtsp-media.c:
4253 media: Return FALSE if seeking is not supported
4255 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4257 * gst/rtsp-server/rtsp-media.c:
4258 rtsp-media: don't seek accurate by default
4259 Accurate seeking is perhaps a little overkill in the most common situation and
4260 causes some formats (mp3) over slow media to seek extremely slowly.
4262 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
4264 * tests/check/gst/rtspserver.c:
4265 tests: fix unit test
4266 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
4268 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
4270 * gst/rtsp-server/rtsp-client.c:
4271 client: Reply 400 if media cannot be constructed
4272 Reply 400 Bad Request instead of 503 Service Unavailable if media
4273 cannot be constructed in SETUP.
4274 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
4276 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
4278 * gst/rtsp-server/rtsp-client.c:
4279 client: Send setup reply once only
4280 If find_media() failed in handle_setup_request() two replies was sent.
4281 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
4283 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
4286 Automatic update of common submodule
4287 From 6b03ba7 to 865aa20
4289 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
4291 * gst/rtsp-server/rtsp-server.c:
4292 server: Emit client-connected signal earlier
4293 Emit client-connected before the client ref is given to a GSource,
4294 otherwise client-connected can be emitted after the client object has
4297 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
4299 * gst/rtsp-server/rtsp-address-pool.c:
4300 * gst/rtsp-server/rtsp-address-pool.h:
4301 * gst/rtsp-server/rtsp-stream.c:
4302 * tests/check/gst/addresspool.c:
4303 addresspool: return reason of failure
4304 Let gst_rtsp_address_pool_reserve_address() return the reason why
4305 the address could not be reserved.
4306 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
4308 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
4311 autogen.sh: Sync behaviour with other GStreamer modules
4312 Allows building from outside of tree amongst other things
4314 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
4317 Automatic update of common submodule
4318 From b613661 to 6b03ba7
4320 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
4323 Automatic update of common submodule
4324 From 74a6857 to b613661
4326 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
4329 Automatic update of common submodule
4330 From 01a7a46 to 74a6857
4332 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
4334 * gst/rtsp-server/rtsp-client.c:
4335 client: Do not read beyond end of path string
4336 If the setup was done without a control url, make sure we don't try to read the
4337 non-existing control string and crash.
4339 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4341 * gst/rtsp-server/rtsp-client.c:
4342 client: Fix RTPInfo header
4343 Refactor the method to make the content_base.
4344 Use the content-base and the control url to construct the RTPInfo
4347 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4349 * gst/rtsp-server/rtsp-client.c:
4350 client: map url to path only in describe
4351 Only map the request url to a path in the DESCRIBE method. The SDP then
4352 contains the base and control urls that should be used to SETUP/PAUSE/
4353 PLAY/TEARDOWN the media.
4355 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4357 * gst/rtsp-server/rtsp-client.c:
4358 Revert "client: map URL to path in requests"
4359 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
4360 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
4361 contains the base and control urls which are used in the SETUP, PLAY,
4362 PAUSE and TEARDOWN requests.
4364 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4366 * gst/rtsp-server/rtsp-client.c:
4367 client: map URL to path in requests
4369 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4371 * gst/rtsp-server/rtsp-client.c:
4372 * gst/rtsp-server/rtsp-mount-points.c:
4373 * gst/rtsp-server/rtsp-mount-points.h:
4374 mount-points: make vmethod to make path from uri
4375 Make a vmethod to transform an url into a path. The path is then used to lookup
4376 the factory. This makes it possible to also use other bits of the url, such as
4377 the query parameters, to locate the factory.
4379 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
4381 * gst/rtsp-server/rtsp-thread-pool.c:
4382 * gst/rtsp-server/rtsp-thread-pool.h:
4383 thread-pool: Add cleanup to wait for the threadpool to finish
4384 Also fix race condition if two threads are asking for the first
4385 thread from the thread pool at once. This would case two internal
4386 GThreadPools to be created.
4387 https://bugzilla.gnome.org/show_bug.cgi?id=707753
4389 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
4391 * gst/rtsp-server/rtsp-client.c:
4392 * tests/check/gst/client.c:
4393 client: free threadpool
4394 https://bugzilla.gnome.org/show_bug.cgi?id=707638
4396 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
4398 * tests/check/gst/mountpoints.c:
4399 mountpoints tests: unref matched factories
4400 https://bugzilla.gnome.org/show_bug.cgi?id=707638
4402 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
4404 * tests/check/gst/media.c:
4405 media tests: unref thread pool and caps
4406 https://bugzilla.gnome.org/show_bug.cgi?id=707638
4408 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
4410 * gst/rtsp-server/rtsp-auth.c:
4411 * gst/rtsp-server/rtsp-media-factory.c:
4412 * gst/rtsp-server/rtsp-media.c:
4413 auth, media, media-factory: unref permissions
4414 https://bugzilla.gnome.org/show_bug.cgi?id=707638
4416 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4418 * examples/Makefile.am:
4419 Makefile: add rule for appsrc example
4421 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4423 * examples/test-appsrc.c:
4424 tests: add appsrc example
4425 Add an example on how to use appsrc to feed the server pipeline with data.
4427 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
4429 * gst/rtsp-server/rtsp-client.c:
4430 rtsp-client: remove query part from content-base string
4431 Make sure that after the control url has been resolved, it's
4432 not a part of the query-string.
4433 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
4435 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4437 * gst/rtsp-server/rtsp-client.c:
4438 client: don't check url in response
4439 There is no url or method in the response to check
4441 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4443 * gst/rtsp-server/rtsp-client.c:
4444 * gst/rtsp-server/rtsp-client.h:
4445 Add handle-response signal for when we receive a GET_PARAMETER response
4447 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4449 * gst/rtsp-server/rtsp-server.c:
4450 Fix gst_rtsp_server_client_filter, using wrong variable type
4452 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
4454 * gst/rtsp-server/rtsp-media-factory-uri.c:
4455 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
4456 For AAC we need to check for framed=true instead of parsed=true.
4457 https://bugzilla.gnome.org/show_bug.cgi?id=701384
4459 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4461 * gst/rtsp-server/rtsp-stream.c:
4462 stream: optimize pipeline for protocols
4463 When TCP is not an allowed protocol for the stream, avoid creating the
4464 appsrc/appsink/queue and tee elements.
4466 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4468 * gst/rtsp-server/rtsp-media.c:
4469 media: set protocols on streams
4471 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4473 * gst/rtsp-server/rtsp-client.c:
4474 client: use protocols supported by stream
4476 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4478 * gst/rtsp-server/rtsp-media-factory.c:
4479 * gst/rtsp-server/rtsp-media.c:
4480 * gst/rtsp-server/rtsp-stream.c:
4481 media-factory: allow all protocols
4483 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4485 * gst/rtsp-server/rtsp-media.c:
4486 media: configure protocols in new streams
4488 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4490 * gst/rtsp-server/rtsp-stream.c:
4491 * gst/rtsp-server/rtsp-stream.h:
4492 stream: add protocols property
4494 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4496 * gst/rtsp-server/rtsp-media.c:
4497 rtsp-media: send state in "new-state" signal
4498 https://bugzilla.gnome.org/show_bug.cgi?id=705110
4500 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
4503 build: add subdir-objects to AM_INIT_AUTOMAKE
4504 Fixes warnings with automake 1.14
4505 https://bugzilla.gnome.org/show_bug.cgi?id=705350
4507 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4509 * docs/libs/gst-rtsp-server-sections.txt:
4510 * gst/rtsp-server/rtsp-client.c:
4511 * gst/rtsp-server/rtsp-server.c:
4512 * gst/rtsp-server/rtsp-server.h:
4513 server: add method to iterate clients of server
4515 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4517 * gst/rtsp-server/rtsp-media.c:
4518 * gst/rtsp-server/rtsp-media.h:
4519 Add vmethod for rtsp-media subclass to access rtpbin
4521 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4523 * gst/rtsp-server/rtsp-client.h:
4524 small documentation fix
4526 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4528 * gst/rtsp-server/rtsp-client.c:
4529 Do not take range header if range is invalid
4531 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4533 * docs/libs/gst-rtsp-server-sections.txt:
4534 * gst/rtsp-server/rtsp-media.c:
4535 media: add docs for new method
4537 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4539 * gst/rtsp-server/rtsp-media.c:
4540 * gst/rtsp-server/rtsp-media.h:
4541 Add API to rtsp-media set the pipeline's state
4543 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4545 * gst/rtsp-server/rtsp-media.c:
4546 Update current position/duration when gst_rtsp_media_get_range_string is called
4548 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4550 * examples/test-cgroups.c:
4551 tests: add some more docs
4553 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4555 * examples/test-cgroups.c:
4556 * gst/rtsp-server/Makefile.am:
4557 * gst/rtsp-server/rtsp-auth.c:
4558 * gst/rtsp-server/rtsp-auth.h:
4559 * gst/rtsp-server/rtsp-client.c:
4560 * gst/rtsp-server/rtsp-client.h:
4561 * gst/rtsp-server/rtsp-context.c:
4562 * gst/rtsp-server/rtsp-context.h:
4563 * gst/rtsp-server/rtsp-params.c:
4564 * gst/rtsp-server/rtsp-params.h:
4565 * gst/rtsp-server/rtsp-server.c:
4566 * gst/rtsp-server/rtsp-thread-pool.c:
4567 * gst/rtsp-server/rtsp-thread-pool.h:
4568 * tests/check/gst/client.c:
4569 ClientState -> Context
4570 Rename the clientstate to context and put the code in a separate file.
4572 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4574 * examples/test-auth.c:
4575 * gst/rtsp-server/rtsp-auth.c:
4576 * gst/rtsp-server/rtsp-auth.h:
4577 auth: add support for default token
4578 The default token is used when the user is not authenticated and can be used to
4579 give minimal permissions.
4581 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4583 * examples/test-auth.c:
4584 * gst/rtsp-server/rtsp-auth.c:
4585 auth: use defines when possible
4587 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4589 * gst/rtsp-server/rtsp-address-pool.c:
4590 address-pool: improve docs
4592 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4594 * gst/rtsp-server/rtsp-permissions.c:
4595 permissions: add the role to the copy
4597 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
4599 * gst/rtsp-server/rtsp-permissions.c:
4600 permissions: Also copy the roles
4602 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
4604 * gst/rtsp-server/rtsp-permissions.c:
4605 permissions: Make it build
4607 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4609 * gst/rtsp-server/rtsp-address-pool.h:
4612 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4614 * docs/libs/gst-rtsp-server-sections.txt:
4615 * gst/rtsp-server/rtsp-auth.c:
4616 * gst/rtsp-server/rtsp-auth.h:
4617 * gst/rtsp-server/rtsp-media.c:
4618 * gst/rtsp-server/rtsp-session-media.c:
4619 * gst/rtsp-server/rtsp-stream-transport.c:
4620 * gst/rtsp-server/rtsp-stream-transport.h:
4621 * gst/rtsp-server/rtsp-stream.c:
4622 * tests/check/gst/client.c:
4625 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4627 * docs/libs/gst-rtsp-server-sections.txt:
4628 * gst/rtsp-server/rtsp-address-pool.c:
4629 * gst/rtsp-server/rtsp-address-pool.h:
4630 * tests/check/gst/addresspool.c:
4631 * tests/check/gst/rtspserver.c:
4632 address-pool: cleanups
4633 Remove redundant method, improve docs.
4635 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4637 * docs/libs/gst-rtsp-server-sections.txt:
4638 * gst/rtsp-server/rtsp-auth.h:
4639 * gst/rtsp-server/rtsp-permissions.c:
4640 * gst/rtsp-server/rtsp-permissions.h:
4641 * gst/rtsp-server/rtsp-token.c:
4644 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4646 * gst/rtsp-server/rtsp-permissions.c:
4647 permissions: implement _remove_role
4649 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4651 * gst/rtsp-server/rtsp-permissions.c:
4652 permissions: update docs
4654 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4656 * tests/check/gst/client.c:
4657 tests: simplify tests
4658 Client settings are now disabled by default so we don't need an auth
4659 module to disable them.
4661 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4663 * gst/rtsp-server/rtsp-auth.c:
4664 auth: add default authorizations
4665 When no auth module is specified, use our table of defaults to look up the
4666 default value of the check instead of always allowing everything. This was
4667 we can disallow client settings by default.
4669 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4672 README: update readme
4674 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4676 * gst/rtsp-server/rtsp-thread-pool.c:
4677 * gst/rtsp-server/rtsp-thread-pool.h:
4678 thread-pool: add more docs
4680 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4682 * gst/rtsp-server/rtsp-thread-pool.c:
4683 * gst/rtsp-server/rtsp-thread-pool.h:
4684 thread-pool: fix race in thread reuse
4685 If we try to reuse a thread right after we made it stop, we end up using a
4686 stopped thread. Catch this case and only reuse threads that are not stopping.
4688 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4690 * gst/rtsp-server/rtsp-server.c:
4691 server: add small debug
4693 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4695 * tests/check/gst/client.c:
4697 Add some permissions to media so we can use the auth and enable
4700 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4702 * gst/rtsp-server/rtsp-client.c:
4703 client: support pushed context in handle_request
4704 If we already have a pushed state, reuse it and add our own things. This makes
4705 it easier to write tests.
4707 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4709 * gst/rtsp-server/rtsp-auth.c:
4710 auth: don't auth on methods
4711 Don't authorize on methods anymore but on the resources that we
4712 try to access, this is more flexible.
4713 Move the authorization checks to where they are needed and let the
4714 check return the response on error.
4716 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4718 * gst/rtsp-server/rtsp-mount-points.c:
4719 mount-points: add some debug
4721 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4723 * tests/check/gst/client.c:
4724 tests: almost fix test
4726 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4728 * gst/rtsp-server/rtsp-auth.c:
4729 * gst/rtsp-server/rtsp-auth.h:
4730 * gst/rtsp-server/rtsp-client.c:
4731 * gst/rtsp-server/rtsp-client.h:
4732 * gst/rtsp-server/rtsp-server.c:
4733 * gst/rtsp-server/rtsp-server.h:
4734 auth: let the auth module check client_settings
4735 Let the auth module decide if client settings are allowed for the
4738 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4740 * gst/rtsp-server/rtsp-token.c:
4741 * gst/rtsp-server/rtsp-token.h:
4742 token: add method to check boolean permission
4744 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4746 * examples/test-auth.c:
4747 * examples/test-cgroups.c:
4748 * gst/rtsp-server/rtsp-token.c:
4749 * gst/rtsp-server/rtsp-token.h:
4750 token: simplify token constructor
4751 Use variable arguments to make easier API.
4753 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4755 * examples/test-auth.c:
4756 * examples/test-cgroups.c:
4757 * gst/rtsp-server/rtsp-media-factory.c:
4758 * gst/rtsp-server/rtsp-media-factory.h:
4759 media-factory: add convenience API for factory
4761 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4763 * examples/test-auth.c:
4764 * examples/test-cgroups.c:
4765 * gst/rtsp-server/rtsp-permissions.c:
4766 * gst/rtsp-server/rtsp-permissions.h:
4767 permissions: simplify API a little
4768 Avoid passing GstStructure in the add_role method, use varargs instead
4769 to construct the structure behind the scenes. We can then also use the
4770 structure name as the role and simplify some more logic.
4772 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4774 * gst/rtsp-server/rtsp-auth.c:
4777 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4779 * gst/rtsp-server/rtsp-auth.c:
4780 * gst/rtsp-server/rtsp-auth.h:
4781 * gst/rtsp-server/rtsp-client.c:
4782 auth: handle unauthorized response
4783 Move handling of the unauthorized response to the auth module, it can add
4784 the appropriate headers to request authorization for the required method
4785 much better than the client.
4787 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4789 * gst/rtsp-server/rtsp-client.c:
4790 * gst/rtsp-server/rtsp-client.h:
4791 client: allow for sending any message, not only requests
4792 Change the _send_request() method to _send_message() so that we
4793 can both send requests and replies.
4795 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4797 * docs/libs/gst-rtsp-server-sections.txt:
4798 * gst/rtsp-server/rtsp-server.h:
4801 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4803 * examples/test-video.c:
4804 * gst/rtsp-server/rtsp-auth.c:
4805 * gst/rtsp-server/rtsp-auth.h:
4806 * gst/rtsp-server/rtsp-server.c:
4807 * gst/rtsp-server/rtsp-server.h:
4808 auth: move TLS handling to auth module
4809 Remove the TLS settings on the server and move it to the auth module because
4810 that is where security related bits go.
4812 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4814 * gst/rtsp-server/rtsp-client.c:
4815 * gst/rtsp-server/rtsp-client.h:
4816 client: add state push/pop
4818 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4820 * gst/rtsp-server/rtsp-client.c:
4821 * gst/rtsp-server/rtsp-client.h:
4822 client: add connection to state
4824 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4826 * gst/rtsp-server/rtsp-mount-points.c:
4827 mount-points: fix debug
4829 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4831 * tests/check/gst/media.c:
4832 tests: fix media test
4834 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4836 * gst/rtsp-server/rtsp-thread-pool.c:
4837 thread-pool: we don't require a state
4839 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4841 * gst/rtsp-server/rtsp-server.c:
4842 server: let context ref the server
4843 So that we don't risk losing the server object early anc crash.
4845 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4847 * tests/check/gst/client.c:
4848 tests: fix client test
4850 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4853 * docs/libs/gst-rtsp-server-docs.sgml:
4854 * docs/libs/gst-rtsp-server-sections.txt:
4855 * gst/rtsp-server/rtsp-address-pool.c:
4856 * gst/rtsp-server/rtsp-auth.c:
4857 * gst/rtsp-server/rtsp-client.c:
4858 * gst/rtsp-server/rtsp-client.h:
4859 * gst/rtsp-server/rtsp-media-factory-uri.c:
4860 * gst/rtsp-server/rtsp-media-factory.c:
4861 * gst/rtsp-server/rtsp-media-factory.h:
4862 * gst/rtsp-server/rtsp-media.c:
4863 * gst/rtsp-server/rtsp-mount-points.c:
4864 * gst/rtsp-server/rtsp-params.c:
4865 * gst/rtsp-server/rtsp-permissions.c:
4866 * gst/rtsp-server/rtsp-sdp.c:
4867 * gst/rtsp-server/rtsp-server.c:
4868 * gst/rtsp-server/rtsp-server.h:
4869 * gst/rtsp-server/rtsp-session-media.c:
4870 * gst/rtsp-server/rtsp-session-pool.c:
4871 * gst/rtsp-server/rtsp-session.c:
4872 * gst/rtsp-server/rtsp-stream-transport.c:
4873 * gst/rtsp-server/rtsp-stream.c:
4874 * gst/rtsp-server/rtsp-thread-pool.c:
4875 * gst/rtsp-server/rtsp-token.c:
4878 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4880 * gst/rtsp-server/rtsp-session-pool.c:
4881 * gst/rtsp-server/rtsp-session-pool.h:
4882 session-pool: make vmethod to create a session
4883 Make a vmethod to create a sessions so that subclasses can create
4884 custom session objects
4886 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4888 * gst/rtsp-server/rtsp-auth.c:
4889 * gst/rtsp-server/rtsp-media-factory.h:
4890 * gst/rtsp-server/rtsp-media.h:
4891 * gst/rtsp-server/rtsp-mount-points.h:
4892 * gst/rtsp-server/rtsp-session-pool.h:
4893 * gst/rtsp-server/rtsp-stream.h:
4896 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4898 * docs/libs/gst-rtsp-server-docs.sgml:
4899 * docs/libs/gst-rtsp-server-sections.txt:
4900 * gst/rtsp-server/rtsp-address-pool.c:
4901 * gst/rtsp-server/rtsp-address-pool.h:
4902 * gst/rtsp-server/rtsp-auth.c:
4903 * gst/rtsp-server/rtsp-client.h:
4904 * gst/rtsp-server/rtsp-media-factory.h:
4905 * gst/rtsp-server/rtsp-media.c:
4906 * gst/rtsp-server/rtsp-media.h:
4907 * gst/rtsp-server/rtsp-permissions.c:
4908 * gst/rtsp-server/rtsp-permissions.h:
4909 * gst/rtsp-server/rtsp-server.h:
4910 * gst/rtsp-server/rtsp-session-media.c:
4911 * gst/rtsp-server/rtsp-session-media.h:
4912 * gst/rtsp-server/rtsp-session-pool.h:
4913 * gst/rtsp-server/rtsp-session.h:
4914 * gst/rtsp-server/rtsp-stream-transport.h:
4915 * gst/rtsp-server/rtsp-stream.c:
4916 * gst/rtsp-server/rtsp-thread-pool.h:
4919 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4922 * examples/Makefile.am:
4923 configure: compile cgroup example conditionally
4924 Only compile the cgroup example when we have libcgroup
4926 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4929 * examples/Makefile.am:
4930 * examples/test-cgroups.c:
4931 examples: add cgroups example
4933 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4935 * tests/check/gst/rtspserver.c:
4936 tests: fix compilation
4938 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4940 * gst/rtsp-server/rtsp-thread-pool.c:
4941 thread-pool: fix vmethod invocation
4943 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4945 * gst/rtsp-server/rtsp-thread-pool.c:
4946 * gst/rtsp-server/rtsp-thread-pool.h:
4947 thread-pool: store thread type in thread
4949 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4951 * gst/rtsp-server/rtsp-client.c:
4952 client: pass thread from pool to media _prepare
4953 Get a thread from the configured threadpool and pass it to the prepare method of
4956 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4958 * gst/rtsp-server/rtsp-media.c:
4959 * gst/rtsp-server/rtsp-media.h:
4960 media: Accept a thread in _prepare
4961 Remove out own threadpool handling and use the provided thread and
4962 maincontext for the bus messages and the state changes.
4964 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4966 * gst/rtsp-server/rtsp-server.c:
4967 server: configure client thread pool
4969 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4971 * gst/rtsp-server/rtsp-client.c:
4972 * gst/rtsp-server/rtsp-client.h:
4973 client: add method to configure thread pool
4975 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4977 * gst/rtsp-server/rtsp-client.h:
4978 * gst/rtsp-server/rtsp-server.c:
4979 * gst/rtsp-server/rtsp-server.h:
4980 server: use thread pool
4981 Use the thread pool instead of doing our own thing.
4983 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4985 * gst/rtsp-server/Makefile.am:
4986 * gst/rtsp-server/rtsp-thread-pool.c:
4987 * gst/rtsp-server/rtsp-thread-pool.h:
4988 thread-pool: add object to manage threads
4989 Add an object to manage the client and media threads.
4991 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4993 * gst/rtsp-server/rtsp-auth.c:
4994 auth: debug authorization check
4996 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4998 * gst/rtsp-server/rtsp-media.c:
4999 media: start media pipeline in context
5000 Start the media pipeline in the provided context (or our default one
5001 when NULL). This makes sure that we run the bus thread in this context and that
5002 all media threads are children of this context.
5004 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5006 * gst/rtsp-server/rtsp-media-factory.c:
5007 factory: pass permissions to media by default
5009 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5011 * examples/test-auth.c:
5012 test: add permissions to auth test
5013 Ass some permissions to the media factory in the test.
5015 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5017 * gst/rtsp-server/rtsp-auth.c:
5018 * gst/rtsp-server/rtsp-auth.h:
5019 * gst/rtsp-server/rtsp-client.c:
5020 auth: simplify auth checks
5021 Remove client from methods, it's now in the state
5022 Perform the check specified by the string, use the information from the
5023 thread local context.
5025 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5027 * gst/rtsp-server/rtsp-client.c:
5028 * gst/rtsp-server/rtsp-client.h:
5029 client: add state to current thread
5030 Add the client to the ClientState object.
5031 Place the ClientState on the current thread.
5033 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5035 * gst/rtsp-server/rtsp-media-factory.c:
5036 * gst/rtsp-server/rtsp-media-factory.h:
5037 * gst/rtsp-server/rtsp-media.c:
5038 * gst/rtsp-server/rtsp-media.h:
5039 media: make it possible to set permissions
5040 Make it possible to set permissions on media and media factory objects
5042 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5044 * gst/rtsp-server/Makefile.am:
5045 * gst/rtsp-server/rtsp-permissions.c:
5046 * gst/rtsp-server/rtsp-permissions.h:
5047 permissions: add permissions object
5048 Add a mini object to store permissions based on a role.
5050 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5052 * examples/test-auth.c:
5053 * gst/rtsp-server/rtsp-auth.c:
5054 * gst/rtsp-server/rtsp-auth.h:
5055 * gst/rtsp-server/rtsp-client.c:
5056 auth: add auth checks
5057 Add an enum with auth checks and implement the checks in the auth object.
5058 Perform the checks from the client.
5060 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5062 * examples/test-auth.c:
5063 * gst/rtsp-server/rtsp-auth.c:
5064 * gst/rtsp-server/rtsp-auth.h:
5065 * gst/rtsp-server/rtsp-client.h:
5066 auth: use the token after authentication
5067 After we authenticated a user, keep the Token around in the state.
5069 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5071 * gst/rtsp-server/rtsp-client.c:
5072 * gst/rtsp-server/rtsp-media.c:
5073 * gst/rtsp-server/rtsp-media.h:
5074 * tests/check/gst/media.c:
5075 media: add optional context for bus messages
5076 Add an optional mainloop to _prepare that will handle the bus messages instead
5077 of always using the shared mainloop.
5079 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5081 * gst/rtsp-server/Makefile.am:
5082 * gst/rtsp-server/rtsp-token.c:
5083 * gst/rtsp-server/rtsp-token.h:
5084 token: add authorization token
5085 Add a simply miniobject that contains the authorizations. The object contains a
5086 GstStructure that hold all authorization fields. When a user is authenticated,
5087 the auth module will create a Token for the user. The token is then used to
5088 check what operations the user is allowed to do and various other configuration
5091 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5093 * examples/test-auth.c:
5094 * gst/rtsp-server/rtsp-auth.c:
5095 * gst/rtsp-server/rtsp-auth.h:
5096 * gst/rtsp-server/rtsp-client.c:
5097 * gst/rtsp-server/rtsp-client.h:
5098 * gst/rtsp-server/rtsp-media-factory.c:
5099 * gst/rtsp-server/rtsp-media-factory.h:
5100 * gst/rtsp-server/rtsp-media.c:
5101 * gst/rtsp-server/rtsp-media.h:
5102 auth: remove auth from media and factory
5103 Remove the auth object from media and factory. We want to have the RTSPClient
5104 authenticate and authorize resources, there is no need to place another auth
5105 manager on the media/factory.
5107 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5109 * examples/test-auth.c:
5110 * gst/rtsp-server/rtsp-auth.c:
5111 * gst/rtsp-server/rtsp-auth.h:
5112 * gst/rtsp-server/rtsp-client.h:
5113 auth: add support for multiple basic auth tokens
5114 Make it possible to add multiple basic authorisation tokens to one authorization
5115 object. Associate with each token an authorization group that will define what
5116 capabilities are allowed.
5118 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5120 * gst/rtsp-server/rtsp-client.c:
5121 client: error out on non-aggregate control
5122 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
5124 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5126 * gst/rtsp-server/rtsp-client.c:
5127 client: rework setup request a little
5128 Cache the media in DESCRIBE based on the longest matching path with the uri
5129 that we can find in the mount points.
5130 Rework the setup request a little to get the media from the session or from
5131 the longest matching path, this way we can derive the control string as
5132 everything after the path instead of hardcoding it.
5133 Find the stream based on the control string and only open a session when all
5136 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5138 * gst/rtsp-server/rtsp-media.c:
5139 * gst/rtsp-server/rtsp-media.h:
5140 media: add method to find a stream by control url
5142 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5144 * gst/rtsp-server/rtsp-stream.c:
5145 * gst/rtsp-server/rtsp-stream.h:
5146 stream: add method to check control url of stream
5148 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5150 * gst/rtsp-server/rtsp-client.c:
5151 * gst/rtsp-server/rtsp-session-media.c:
5152 * gst/rtsp-server/rtsp-session-media.h:
5153 * gst/rtsp-server/rtsp-session.c:
5154 * gst/rtsp-server/rtsp-session.h:
5155 session: use path matching for session media
5156 Use a path string instead of a uri to lookup session media in the sessions. Also
5157 use path matching to find the largest possible path that matches.
5159 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5161 * gst/rtsp-server/rtsp-client.c:
5162 * gst/rtsp-server/rtsp-mount-points.c:
5163 * gst/rtsp-server/rtsp-mount-points.h:
5164 * tests/check/gst/mountpoints.c:
5165 mount-points: remove useless vmethod
5166 Making lookups in the mount points should not be done with a URL, if there is a
5167 mapping to be done from URL to mount points, we'll need to do it somewhere
5170 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5172 * gst/rtsp-server/rtsp-mount-points.c:
5173 * gst/rtsp-server/rtsp-mount-points.h:
5174 * tests/check/gst/mountpoints.c:
5175 mount-points: improve mount point searching
5176 Use a GSequence to keep track of the mount points.
5177 Match a URL to the longest matching registered mount point. This should be the
5178 URL to perform aggreagate control and the remainder is the stream specific
5180 Add some unit tests for this.
5182 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
5184 * gst/rtsp-server/Makefile.am:
5185 rtsp-server: Allow building of static library
5187 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5189 * tests/check/gst/mediafactory.c:
5190 tests: fix compilation
5192 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5194 * gst/rtsp-server/rtsp-sdp.c:
5195 sdp: get control string from stream
5196 Use the control string as configured in the stream.
5198 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5200 * gst/rtsp-server/rtsp-stream.c:
5201 * gst/rtsp-server/rtsp-stream.h:
5202 stream: add methods and property to set control string
5204 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5206 * gst/rtsp-server/rtsp-client.c:
5208 Rename variables for clarity
5209 Keep media in state when we can
5211 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5213 * gst/rtsp-server/rtsp-client.c:
5214 * gst/rtsp-server/rtsp-stream.c:
5215 * gst/rtsp-server/rtsp-stream.h:
5216 stream: add more support for IPv6
5217 Rename _get_address to _get_multicast_address in GstRTSPStream to
5218 make it clear that this function only deals with multicast.
5219 Make it possible to have both an IPv4 and IPv6 multicast address on
5220 a stream. Give the client an IPv4 or IPv6 address depending on the
5221 address it used to connect to the server.
5222 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
5224 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5226 * gst/rtsp-server/rtsp-client.c:
5229 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5231 * gst/rtsp-server/rtsp-stream.c:
5232 stream: handle failed port allocation
5233 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
5234 can't allocate any family at all. Also keep track of what port families we
5236 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
5238 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5240 * gst/rtsp-server/rtsp-stream.c:
5241 stream: improve docs
5243 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5245 * gst/rtsp-server/rtsp-stream-transport.c:
5246 stream-transport: remove old if 0 block
5248 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
5250 * tests/check/gst/client.c:
5252 gst_rtsp_client_get_uri() has been removed
5253 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
5255 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5257 * gst/rtsp-server/rtsp-client.c:
5258 * gst/rtsp-server/rtsp-client.h:
5259 client: add method to filter managed sessions
5260 Add a method to filter the sessions managed by this client connection.
5261 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
5263 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5265 * gst/rtsp-server/rtsp-client.c:
5266 * gst/rtsp-server/rtsp-client.h:
5267 client: remove _get_uri() method
5268 Remove the get_uri() method on the client. A client has no uri, the uri
5269 property is an internal property to manage the last cached media for
5272 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5274 * gst/rtsp-server/rtsp-media-factory.h:
5275 media-factory: fix typo
5277 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
5279 * gst/rtsp-server/rtsp-media.c:
5280 rtsp-media: Do not leak the query in default_query_stop
5281 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
5283 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5285 * gst/rtsp-server/rtsp-media.c:
5286 media: don't unlock when conversion fails
5287 Don't unlock the state lock when conversion fails because it was not locked.
5289 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5291 * gst/rtsp-server/rtsp-media.c:
5292 * gst/rtsp-server/rtsp-media.h:
5293 Add query_position and query_stop vmethods to rtsp-media
5295 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5297 * gst/rtsp-server/rtsp-media.c:
5298 Fix typo in property install for rtsp-media's time-provider
5300 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5302 * gst/rtsp-server/rtsp-client.c:
5303 * gst/rtsp-server/rtsp-client.h:
5304 client: clean some variables
5305 Clean some variables and add some guards to _send_request()
5307 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5309 * gst/rtsp-server/rtsp-client.c:
5310 * gst/rtsp-server/rtsp-client.h:
5311 Add gst_rtsp_client_send_request API
5312 This makes it possible to send arbitrary messages to a client, such as
5313 SET_PARAMETER or GET_PARAMETER
5315 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5317 * gst/rtsp-server/rtsp-media.c:
5318 * gst/rtsp-server/rtsp-media.h:
5319 media: add _get_element() method
5320 Add method to get the element used when creating the media.
5321 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
5323 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5325 * gst/rtsp-server/rtsp-media.c:
5328 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
5330 * gst/rtsp-server/rtsp-stream.c:
5331 * gst/rtsp-server/rtsp-stream.h:
5332 stream: allow access to the rtp session
5333 https://bugzilla.gnome.org/show_bug.cgi?id=703004
5335 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
5337 * gst/rtsp-server/rtsp-stream.c:
5338 * gst/rtsp-server/rtsp-stream.h:
5339 dscp qos support in gst-rtsp-stream
5340 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
5342 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5344 * tests/check/gst/rtspserver.c:
5346 Actually do what the comment says. Also keep the old code around, not sure what
5347 should happen when you get a 454 from a TEARDOWN, does it close the connection?
5348 it currently doesn't.
5350 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5352 * gst/rtsp-server/rtsp-client.c:
5353 client: also watch newly created session
5354 When we newly created a session, start watching it immediately instead of
5355 on the next request.
5357 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
5359 * tests/check/gst/client.c:
5360 tests: add unit test for new-session
5361 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
5363 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5365 * gst/rtsp-server/rtsp-client.c:
5366 client: emit new-session when new session is created
5367 Only emit new-session when we created a new session for a client, not when a
5368 client picked up a previous session.
5369 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
5371 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
5373 * gst/rtsp-server/rtsp-client.c:
5374 client: handle asterisk as path in requests
5375 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
5377 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5379 * gst/rtsp-server/rtsp-media.c:
5380 media: handle segment query format mismatch
5381 It's possible that the segment query returns with a different format than what
5382 we asked for, handle this case also.
5384 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
5386 * gst/rtsp-server/rtsp-media.c:
5387 media: use segment stop in collect_media_stats
5388 Use segment stop instead of duration as range end point.
5389 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
5391 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
5393 * gst/rtsp-server/rtsp-media.c:
5394 * tests/check/gst/media.c:
5395 rtsp-media: Do not leak the element in take_pipeline
5396 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
5398 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
5400 * gst/rtsp-server/rtsp-client.c:
5401 * gst/rtsp-server/rtsp-client.h:
5402 rtsp-client: Make configure_client_transport virtual
5403 This patch makes configure_client_transport virtual. The functionality is
5404 needed to handle some weird clients sending multicast transport settings as url
5406 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
5408 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
5410 * gst/rtsp-server/rtsp-client.c:
5411 * gst/rtsp-server/rtsp-client.h:
5412 rtsp-client: Make param_set and param_get virtual
5413 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
5415 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
5417 * gst/rtsp-server/rtsp-client.c:
5418 * gst/rtsp-server/rtsp-media.c:
5419 * gst/rtsp-server/rtsp-media.h:
5420 media: convert_range replaces get_range_times
5421 get_range_times worked for handling UTC ranges for seeks, but we also
5422 need to convert back from NPT to the requested unit in
5423 get_range_string. convert_range is now used for both.
5424 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
5426 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5428 * gst/rtsp-server/rtsp-client.c:
5429 * gst/rtsp-server/rtsp-sdp.c:
5430 * gst/rtsp-server/rtsp-sdp.h:
5431 sdp: cleanup sdp info
5432 We don't need to pass the proto, we can more easily check a boolean.
5433 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
5435 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
5437 * gst/rtsp-server/rtsp-sdp.c:
5438 use 0.0.0.0 or :: for c= line instead of server address
5440 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
5442 * gst/rtsp-server/rtsp-client.c:
5443 use local address, not remote, in SDP
5444 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
5446 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5449 Automatic update of common submodule
5450 From 098c0d7 to 01a7a46
5452 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
5454 * gst/rtsp-server/rtsp-media.c:
5455 * gst/rtsp-server/rtsp-media.h:
5456 media: possibility to override range time conversion
5457 Make it possible to override the conversion from GstRTSPTimeRange to
5458 GstClockTimes, that is done before seeking on the media
5459 pipeline. Overriding can be useful for UTC ranges, where the default
5460 conversion gives nanoseconds since 1900.
5461 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
5463 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
5465 * gst/rtsp-server/rtsp-server.c:
5466 * gst/rtsp-server/rtsp-server.h:
5467 rtsp-server: Expose the use_client_settings API
5468 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
5470 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
5472 * gst/rtsp-server/rtsp-client.c:
5473 * gst/rtsp-server/rtsp-stream.c:
5474 * gst/rtsp-server/rtsp-stream.h:
5475 rtspstream: handle both ipv4 and ipv6 clients
5476 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
5478 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5480 * gst/rtsp-server/rtsp-sdp.c:
5481 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
5482 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
5483 We already have a way to place extra attributes in the SDP by using a string
5484 property with prefix x- or a- in the caps.
5486 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5488 * gst/rtsp-server/rtsp-sdp.c:
5489 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
5490 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
5491 We already have a way to place extra attributes in the SDP, just make a string
5492 property in the payloader with a- or x- prefix.
5494 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5496 * gst/rtsp-server/rtsp-sdp.c:
5497 rtsp: place a- and x- properties as attributes
5498 application/x-rtp has properties with a- and x- prefixes that should be
5499 placed as attributes in the SDP for the media instead of being added to the
5502 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5504 * examples/Makefile.am:
5505 * examples/test-video.c:
5506 example: add TLS example
5508 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5510 * gst/rtsp-server/rtsp-server.c:
5511 * gst/rtsp-server/rtsp-server.h:
5512 server: add support for TLS
5513 Add methods to set and get a TLS certificate.
5514 Add vmethod to configure a new connection. By default, configure the TLS
5515 certificate in a new connection if needed.
5517 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5519 * gst/rtsp-server/rtsp-server.c:
5520 * gst/rtsp-server/rtsp-server.h:
5521 server: remove accept_client vmethod
5522 This vmethod is not very useful so remove it.
5524 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5526 * gst/rtsp-server/rtsp-server.c:
5527 server: don't crash on NULL GError
5529 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
5531 * gst/rtsp-server/rtsp-session-pool.c:
5532 rtsp-session-pool: corrected session timeout detection
5533 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
5535 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5537 * gst/rtsp-server/rtsp-client.c:
5538 client: improve debug
5540 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5542 * gst/rtsp-server/rtsp-client.c:
5543 * gst/rtsp-server/rtsp-client.h:
5544 * gst/rtsp-server/rtsp-server.c:
5545 server: refactor connection setup
5546 Let the server accept the socket connection and construct a GstRTSPConnection
5547 from it. Remove the code from the client and let the client only deal with
5548 a fully configure GstRTSPConnection object.
5549 We will need this later when the server will configure the connection for
5552 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5554 * gst/rtsp-server/rtsp-stream.c:
5555 stream: keep the transport object alive
5556 Keep the transport object alive while we have it as qdata on the
5559 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
5561 * gst/rtsp-server/rtsp-client.c:
5562 * gst/rtsp-server/rtsp-server.c:
5563 rtsp-server: Do not crash on nmapping of server
5564 * generate error when gst_rtsp_connection_accept fails
5565 * do not stop accepting incoming connections because
5566 accepting a client fails
5567 https://bugzilla.gnome.org/show_bug.cgi?id=701072
5569 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
5571 * gst/rtsp-server/rtsp-client.c:
5572 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
5573 https://bugzilla.gnome.org/show_bug.cgi?id=700953
5575 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
5577 * gst/rtsp-server/rtsp-sdp.c:
5578 rtsp-sdp: Parse framerate caps field and set SDP attribute
5579 The SDP attribute and its format is described in RFC4566.
5580 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
5582 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
5584 * gst/rtsp-server/rtsp-sdp.c:
5585 rtsp-sdp: Parse width/height from caps and set SDP attribute
5586 The SDP attribute and its format is described in RFC6064.
5587 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
5589 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
5591 * gst/rtsp-server/rtsp-sdp.c:
5592 * tests/check/gst/client.c:
5593 rtsp-sdp: add bandwidth line
5594 https://bugzilla.gnome.org/show_bug.cgi?id=699220
5596 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5599 Automatic update of common submodule
5600 From 5edcd85 to 098c0d7
5602 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
5604 * tests/check/gst/media.c:
5605 tests: add dynamic payloader prepare/unprepare check
5607 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5609 * gst/rtsp-server/rtsp-media.c:
5610 media: release lock when removing fakesink
5612 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5614 * gst/rtsp-server/rtsp-stream.c:
5615 stream: set elements to NULL before removing
5616 When removing a stream, set the elements to NULL first. This avoids
5617 element-is-not-in-NULL-state errors when we dispose the elements.
5619 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
5622 Automatic update of common submodule
5623 From 3cb3d3c to 5edcd85
5625 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5627 * gst/rtsp-server/rtsp-media.c:
5628 * gst/rtsp-server/rtsp-media.h:
5629 media: listen to pad-removed signals
5630 Listen to the pad-removed signal and remove the stream associated with the
5632 Add signal to be notified of the removed pad.
5633 Remove the fakesink in unprepare()
5634 Fix signatures of the signal methods
5636 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5638 * examples/test-sdp.c:
5639 tests: add example of reusable pipelines
5641 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
5643 * gst/rtsp-server/rtsp-stream.c:
5644 * gst/rtsp-server/rtsp-stream.h:
5645 stream: add method to get the srcpad
5647 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
5649 * tests/check/gst/media.c:
5650 check: add media prepare/unprepare test
5651 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
5653 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
5655 * gst/rtsp-server/rtsp-media.c:
5656 media: disconnect from signal handlers in unprepare()
5657 We connected to the pad-added and no-more-pads signals in prepare() so
5658 we need to disconnect from them in unprepare().
5659 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
5661 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
5663 * gst/rtsp-server/rtsp-media.c:
5664 media: don't free streams array
5665 Don't free the streams array in the unprepare() method, they were not
5667 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
5669 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
5671 * gst/rtsp-server/rtsp-media.c:
5672 media: don't unref the pipeline in unprepare
5673 Unprepare() should undo what prepare() does. Because the pipeline is
5674 not created in prepare(), we should not unref it in unprepare()
5676 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
5678 * gst/rtsp-server/rtsp-stream.c:
5679 stream: clear session and caps for reuse
5680 Set the session and caps to NULL after unref otherwise we might unref
5682 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
5684 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
5686 * gst/rtsp-server/rtsp-client.c:
5687 client: send out teardown signal before tearing down
5688 The advantage is that in the signal handler you get direct access to
5689 information about what streams are about to get torn down (in the
5690 GstRTSPClientState).
5691 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
5693 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
5695 * gst/rtsp-server/rtsp-client.c:
5696 * gst/rtsp-server/rtsp-client.h:
5697 client: expose connection
5698 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
5700 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
5703 Automatic update of common submodule
5704 From aed87ae to 3cb3d3c
5706 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5708 * gst/rtsp-server/rtsp-media.c:
5709 * gst/rtsp-server/rtsp-media.h:
5710 * gst/rtsp-server/rtsp-session-media.c:
5711 * gst/rtsp-server/rtsp-session-media.h:
5712 media: add method to get the base_time of the pipeline
5713 Together with a shared clock, this base-time could eventually be sent to
5714 the client so that it can reconstruct the exact running-time of the clock
5717 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5719 * gst/rtsp-server/Makefile.am:
5720 * gst/rtsp-server/rtsp-media.c:
5721 * gst/rtsp-server/rtsp-media.h:
5722 * gst/rtsp-server/rtsp-sdp.c:
5723 media: add GstNetTimeProvider support
5724 Add a property to let the media provide a GstNetTimeProvider for its clock.
5725 Make methods to get the clock and nettimeprovider
5726 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
5727 provider and also the current time of the clock. This should make it possible
5728 for (GStreamer) clients to slave their clock to the server clock.
5730 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
5733 Automatic update of common submodule
5734 From 04c7a1e to aed87ae
5736 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5738 * gst/rtsp-server/rtsp-media.c:
5739 media: wait for buffering to complete
5740 Wait for buffering to complete before changing the state to the target state.
5742 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5744 * gst/rtsp-server/rtsp-media.c:
5745 media: small cleanup
5747 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
5749 * tests/check/gst/rtspserver.c:
5750 tests: remove extra unref in test_setup_non_existing_stream
5751 The unref is not needed anymore, teardown runs without it.
5752 https://bugzilla.gnome.org/show_bug.cgi?id=696542
5754 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
5756 * tests/check/gst/rtspserver.c:
5757 tests: GSocketService cleanup in test_bind_already_in_use
5758 Use g_socket_service_stop so the rtspserver test stops listening for
5759 incoming connections in test_bind_already_in_use.
5760 https://bugzilla.gnome.org/show_bug.cgi?id=696541
5762 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
5764 * gst/rtsp-server/rtsp-media-factory.c:
5765 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
5766 Instead use a GWeakRef which is safe to use
5767 This is a known GLib bug, see:
5768 https://bugzilla.gnome.org/show_bug.cgi?id=667145
5770 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
5772 * gst/rtsp-server/rtsp-client.c:
5773 * gst/rtsp-server/rtsp-media.c:
5774 * gst/rtsp-server/rtsp-media.h:
5775 * gst/rtsp-server/rtsp-sdp.c:
5776 * tests/check/gst/media.c:
5777 * tests/check/gst/rtspserver.c:
5778 rtsp-media/client: Reply to PLAY request with same type of Range
5779 Remember the type of Range from the PLAY request and use the same type for
5782 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
5784 * gst/rtsp-server/rtsp-client.c:
5785 * gst/rtsp-server/rtsp-client.h:
5786 * tests/check/gst/client.c:
5787 rtsp-client: expose uri
5789 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
5791 * tests/check/gst/mediafactory.c:
5792 tests: Hold ref while creating second media
5793 To test if the media aren't shared, make sure we keep the first one while creating a second
5794 otherwise the same memory address may be reused.
5796 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
5799 configure: remove out-of-date comment
5801 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
5804 .gitignore: ignore more build files
5806 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
5808 * tests/check/Makefile.am:
5809 tests: use right _LIBS variable for gst-plugins-base libs
5811 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5813 * tests/check/Makefile.am:
5814 check: add librtp to libs
5816 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
5818 * tests/check/gst/rtspserver.c:
5819 tests: Add test to check selecting a port the server will send from
5821 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
5823 * tests/check/gst/rtspserver.c:
5824 tests: Make sure packets are actually received
5826 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
5828 * gst/rtsp-server/rtsp-stream.c:
5829 stream: Select unicast address from pool if appropriate
5831 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
5833 * gst/rtsp-server/rtsp-stream.c:
5834 stream: Properties are always there in Gst 1.0
5836 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
5838 * tests/check/gst/addresspool.c:
5839 tests: Add tests for unicast addresses in pool
5841 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
5843 * gst/rtsp-server/rtsp-address-pool.c:
5844 * tests/check/gst/addresspool.c:
5845 address-pool: Verify that multicast addresses are used for multicast and vice-versa
5847 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
5849 * docs/libs/gst-rtsp-server-sections.txt:
5850 * gst/rtsp-server/rtsp-address-pool.c:
5851 * gst/rtsp-server/rtsp-address-pool.h:
5852 * gst/rtsp-server/rtsp-stream.c:
5853 * tests/check/gst/addresspool.c:
5854 address-pool: Add unicast addresses
5856 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
5859 * gst/rtsp-server/rtsp-server.c:
5860 * tests/check/gst/rtspserver.c:
5861 rtsp-server: Limit the number of threads per server instance
5862 If we exceed the maximum, just round robin the clients over the existing
5865 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
5867 * gst/rtsp-server/rtsp-server.c:
5868 rtsp-server: No need to store the GMainContext in the client context
5870 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
5872 * tests/check/gst/rtspserver.c:
5873 tests: Add test for client disconnection
5875 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
5877 * tests/check/gst/rtspserver.c:
5878 tests: Test client and session timeouts with multiple threads
5880 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
5882 * gst/rtsp-server/rtsp-address-pool.c:
5883 * gst/rtsp-server/rtsp-auth.c:
5884 * gst/rtsp-server/rtsp-client.c:
5885 * gst/rtsp-server/rtsp-media-factory-uri.c:
5886 * gst/rtsp-server/rtsp-media-factory.c:
5887 * gst/rtsp-server/rtsp-media.c:
5888 * gst/rtsp-server/rtsp-mount-points.c:
5889 * gst/rtsp-server/rtsp-server.c:
5890 * gst/rtsp-server/rtsp-session-media.c:
5891 * gst/rtsp-server/rtsp-session-pool.c:
5892 * gst/rtsp-server/rtsp-session.c:
5893 Document locking and its order
5895 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
5897 * tests/check/gst/rtspserver.c:
5898 tests: Test that slow DESCRIBE don't block other clients
5900 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
5902 * tests/check/gst/client.c:
5903 tests: Add tests for client-requested multicast address
5905 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
5907 * docs/libs/gst-rtsp-server-sections.txt:
5908 docs: Put the various functions in the right sections
5910 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
5912 * docs/libs/gst-rtsp-server-docs.sgml:
5913 * docs/libs/gst-rtsp-server-sections.txt:
5914 * gst/rtsp-server/rtsp-address-pool.c:
5915 * gst/rtsp-server/rtsp-address-pool.h:
5916 docs: Generate docs for GstRTSPAddressPool
5918 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
5920 * gst/rtsp-server/rtsp-client.c:
5921 * gst/rtsp-server/rtsp-stream.c:
5922 * gst/rtsp-server/rtsp-stream.h:
5923 client: Check client provided addresses against the address pool
5925 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
5927 * gst/rtsp-server/rtsp-address-pool.c:
5928 * gst/rtsp-server/rtsp-address-pool.h:
5929 * tests/check/gst/addresspool.c:
5930 address-pool: Add API to request a specific address from the pool
5931 Also add relevant unit tests.
5933 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
5935 * tests/check/gst/mediafactory.c:
5936 tests: Check the passing around of a RTSPAddressPool
5937 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
5938 way down to the stream.
5940 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
5942 * tests/check/gst/addresspool.c:
5943 tests: Add more tests for the address pool
5945 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
5947 * gst/rtsp-server/rtsp-address-pool.c:
5948 address-pool: Fix off by one error
5949 When splitting a port range, the port after a skip is not part of range.
5951 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
5954 Automatic update of common submodule
5955 From 2de221c to 04c7a1e
5957 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
5960 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
5961 AM_CONFIG_HEADER was removed in automake 1.13
5962 https://bugzilla.gnome.org/show_bug.cgi?id=693368
5964 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
5967 Automatic update of common submodule
5968 From a942293 to 2de221c
5970 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5972 * gst/rtsp-server/rtsp-client.c:
5973 client: make sure the watch exists while sending data
5974 Protect the send_func with a lock. This allows us to wait for sending
5975 to complete before changing the send_func and user_data. We add an
5976 extra ref to the watch to make sure that it remains valid during
5978 When closing the connection, set the send_func to NULL
5979 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
5981 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5983 * tests/check/Makefile.am:
5984 tests: use GST_*_1_0 environment variables everywhere
5985 The _1_0 suffixed environment variables override the
5986 non-suffixed ones, so if we're in an environment that
5987 sets the _1_0 suffixed ones, such as jhbuild, we need
5988 to set those to make sure ours actually always get
5991 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5994 Automatic update of common submodule
5995 From acb04d9 to a942293
5997 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5999 * gst/rtsp-server/rtsp-client.c:
6000 rtsp-client: set the client backlog
6001 Set the client backlog to a reasonable default
6003 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
6005 * gst/rtsp-server/rtsp-media.c:
6006 rtsp-media: Make the element a constructor parameter
6007 https://bugzilla.gnome.org/show_bug.cgi?id=689594
6009 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
6011 * docs/libs/Makefile.am:
6012 docs: Link with gcov library when gcov is enabled
6013 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
6015 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6017 * gst/rtsp-server/rtsp-media.c:
6018 media: match prepare with unprepare
6019 Really unprepare when there were an equal amount of prepare calls.
6021 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6023 * gst/rtsp-server/rtsp-media.c:
6024 media: media has to be unprepared in finalize
6025 Because unprepare takes away the last ref on the media.
6027 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6029 * gst/rtsp-server/rtsp-client.c:
6030 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
6031 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
6032 We can't use the refcount to trigger unprepare because it is the unprepare call
6033 that removes the last refcount after all messages are consumed. What we should
6034 probably do is make a prepared refcount and only unprepare when the refcount
6037 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6039 * gst/rtsp-server/rtsp-media.c:
6040 media: let the source unref the last media ref
6041 the last ref to the media is held by the source so we don't need to add more ref
6042 and unrefs, we simply destroy the media when the source is gone.
6044 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6046 * gst/rtsp-server/rtsp-media.c:
6047 media: improve debug
6049 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6051 * gst/rtsp-server/rtsp-media.c:
6053 Make sure we are in the right state when collecting the position and duration.
6054 Only make ourselves PREPARED when we were previously PREPARING.
6056 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6058 * gst/rtsp-server/rtsp-media.c:
6059 media: use g_object_ref/unref for GObjects
6061 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
6063 * gst/rtsp-server/rtsp-client.c:
6064 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
6065 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
6066 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
6067 isn't being used anymore.
6069 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
6071 * gst/rtsp-server/rtsp-media.c:
6072 Fix compiler warning
6074 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
6076 * gst/rtsp-server/rtsp-media-factory-uri.c:
6077 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
6079 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6081 * gst/rtsp-server/rtsp-session-media.h:
6084 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6086 * gst/rtsp-server/rtsp-media.c:
6087 * tests/check/gst/media.c:
6088 media: avoid element leak
6090 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6092 * gst/rtsp-server/rtsp-media.c:
6093 media: require an element in media constructor
6095 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6097 * gst/rtsp-server/rtsp-client.c:
6098 Revert "client: TEARDOWN brings that state to Init again"
6099 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
6100 The object is already disposed, there is no point in setting the state.
6102 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6104 * gst/rtsp-server/rtsp-client.c:
6105 client: TEARDOWN brings that state to Init again
6107 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6109 * docs/libs/gst-rtsp-server-sections.txt:
6110 * examples/test-auth.c:
6111 * gst/rtsp-server/rtsp-auth.c:
6112 * gst/rtsp-server/rtsp-auth.h:
6113 * gst/rtsp-server/rtsp-client.c:
6114 * gst/rtsp-server/rtsp-client.h:
6115 * gst/rtsp-server/rtsp-media-factory-uri.c:
6116 * gst/rtsp-server/rtsp-media-factory-uri.h:
6117 * gst/rtsp-server/rtsp-media-factory.c:
6118 * gst/rtsp-server/rtsp-media-factory.h:
6119 * gst/rtsp-server/rtsp-media.c:
6120 * gst/rtsp-server/rtsp-media.h:
6121 * gst/rtsp-server/rtsp-mount-points.c:
6122 * gst/rtsp-server/rtsp-mount-points.h:
6123 * gst/rtsp-server/rtsp-sdp.c:
6124 * gst/rtsp-server/rtsp-server.c:
6125 * gst/rtsp-server/rtsp-server.h:
6126 * gst/rtsp-server/rtsp-session-media.c:
6127 * gst/rtsp-server/rtsp-session-media.h:
6128 * gst/rtsp-server/rtsp-session-pool.c:
6129 * gst/rtsp-server/rtsp-session-pool.h:
6130 * gst/rtsp-server/rtsp-session.c:
6131 * gst/rtsp-server/rtsp-session.h:
6132 * gst/rtsp-server/rtsp-stream-transport.c:
6133 * gst/rtsp-server/rtsp-stream-transport.h:
6134 * gst/rtsp-server/rtsp-stream.c:
6135 * gst/rtsp-server/rtsp-stream.h:
6136 * tests/check/gst/media.c:
6137 rtsp: make object details private
6138 Make all object details private
6139 Add methods to access private bits
6141 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6143 * tests/check/Makefile.am:
6144 * tests/check/gst/media.c:
6145 tests: add media tests
6147 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6149 * gst/rtsp-server/rtsp-media.c:
6150 media: check if prepared for some methods
6151 Check that the media object is prepared before doing seek and getting the
6152 current position etc.
6153 Add some g_return checks.
6155 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6157 * tests/check/Makefile.am:
6158 * tests/check/gst/mediafactory.c:
6159 tests: add mediafactory test
6161 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6163 * gst/rtsp-server/rtsp-stream.c:
6164 stream: improve debug
6166 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6168 * gst/rtsp-server/rtsp-media.c:
6169 * gst/rtsp-server/rtsp-media.h:
6170 media: unref pipeline in finalize to avoid leaking it
6172 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6174 * gst/rtsp-server/rtsp-media-factory-uri.c:
6175 * gst/rtsp-server/rtsp-media.c:
6176 rtsp: use gst_object_unref on GstObjects
6178 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6180 * gst/rtsp-server/rtsp-media-factory.c:
6181 media-factory: require an url
6183 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6185 * examples/test-uri.c:
6186 examples: fix include
6188 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6190 * gst/rtsp-server/rtsp-server.h:
6191 server: remove unused include
6193 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6195 * tests/check/Makefile.am:
6196 * tests/check/gst/mountpoints.c:
6197 tests: add test for mountpoints
6199 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6201 * gst/rtsp-server/rtsp-client.c:
6202 client: fix factory leak
6203 Keep the factory in the state object only for authorization checks and make
6204 sure we unref it on failure. Also don't keep invalid objects in the state
6207 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6209 * gst/rtsp-server/rtsp-mount-points.c:
6210 mounts: add g_return_if guards
6212 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6214 * tests/check/gst/client.c:
6215 tests: add more tests
6217 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6219 * gst/rtsp-server/rtsp-client.c:
6220 client: improve debug
6222 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6224 * gst/rtsp-server/rtsp-client.c:
6225 client: improve debug and fix leaks
6226 Cleanup the uri and session when there is a bad request.
6228 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6233 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6235 * tests/check/gst/client.c:
6236 test: add test for session in options request
6238 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6240 * gst/rtsp-server/rtsp-client.c:
6241 client: use 454 when session can't be found
6242 We should use 454 when a session can't be found because there was no session
6243 pool configured in the server. This is not a server configuration problem
6244 because the server on which the request is done might not be the same one that
6245 will keep the sessions for us and so it does not need to support sessions.
6247 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6249 * gst/rtsp-server/rtsp-client.c:
6250 client: only free connection when there is one
6251 It's possible that the client doesn't have a connection when we try to free it.
6253 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6255 * tests/check/Makefile.am:
6256 * tests/check/gst/client.c:
6257 tests: add unit test for the client object
6259 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6261 * gst/rtsp-server/rtsp-client.c:
6262 client: small cleanup
6264 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6266 * gst/rtsp-server/rtsp-client.h:
6267 client: remove unused include
6269 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6271 * gst/rtsp-server/rtsp-client.c:
6272 client: fix compilation
6274 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6276 * gst/rtsp-server/rtsp-client.c:
6277 client: call destroy without the lock
6279 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6281 * gst/rtsp-server/rtsp-client.c:
6282 * gst/rtsp-server/rtsp-client.h:
6283 client: make the client usable without a socket
6284 Make a method to let the client handle a message and a callback when the client
6285 wants us to send a response message back. This makes it possible to also use the
6286 client object without the sockets, which should make it easier to test.
6288 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6290 * gst/rtsp-server/rtsp-client.c:
6291 * gst/rtsp-server/rtsp-client.h:
6292 client: small cleanup
6294 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6296 * docs/libs/gst-rtsp-server-sections.txt:
6297 * gst/rtsp-server/rtsp-client.c:
6298 * gst/rtsp-server/rtsp-client.h:
6299 * gst/rtsp-server/rtsp-server.c:
6300 client: remove reference to server
6301 We don't need to keep a ref to the server
6303 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6305 * gst/rtsp-server/rtsp-client.c:
6306 * gst/rtsp-server/rtsp-client.h:
6308 Also add some g_return_if()
6310 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6312 * gst/rtsp-server/rtsp-client.c:
6313 client: log more errors
6315 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6317 * gst/rtsp-server/rtsp-client.c:
6318 client: fix compilation
6320 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6322 * gst/rtsp-server/rtsp-client.c:
6323 * gst/rtsp-server/rtsp-client.h:
6324 client: add generic close-after-send support
6325 Add a property to send_response() to close the connection after the response has
6326 been sent to the client.
6328 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6331 * docs/libs/gst-rtsp-server-docs.sgml:
6332 * docs/libs/gst-rtsp-server-sections.txt:
6333 * docs/libs/gst-rtsp-server.types:
6334 * examples/test-auth.c:
6335 * examples/test-launch.c:
6336 * examples/test-mp4.c:
6337 * examples/test-multicast.c:
6338 * examples/test-multicast2.c:
6339 * examples/test-ogg.c:
6340 * examples/test-readme.c:
6341 * examples/test-sdp.c:
6342 * examples/test-uri.c:
6343 * examples/test-video.c:
6344 * gst/rtsp-server/Makefile.am:
6345 * gst/rtsp-server/rtsp-auth.h:
6346 * gst/rtsp-server/rtsp-client.c:
6347 * gst/rtsp-server/rtsp-client.h:
6348 * gst/rtsp-server/rtsp-media-mapping.c:
6349 * gst/rtsp-server/rtsp-media-mapping.h:
6350 * gst/rtsp-server/rtsp-mount-points.c:
6351 * gst/rtsp-server/rtsp-mount-points.h:
6352 * gst/rtsp-server/rtsp-server.c:
6353 * gst/rtsp-server/rtsp-server.h:
6354 * gst/rtsp-server/rtsp-session-media.c:
6355 * gst/rtsp-server/rtsp-session-pool.c:
6356 * gst/rtsp-server/rtsp-session-pool.h:
6357 * tests/check/gst/rtspserver.c:
6358 MediaMapping -> MountPoints
6359 Describes better what the object manages.
6361 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6364 configure: bump required version of -base
6366 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6368 * gst/rtsp-server/rtsp-media.c:
6371 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6373 * gst/rtsp-server/rtsp-media.c:
6374 * gst/rtsp-server/rtsp-media.h:
6375 media: support more Range formats
6376 Use the new -base methods to convert the Range string into a seek start and stop
6379 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6381 * examples/test-launch.c:
6382 examples: fix whitespace
6384 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6386 * examples/test-auth.c:
6387 test-auth: add example of how to remove sessions
6388 Add an example of the session filter api.
6390 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6392 * examples/test-uri.c:
6393 test-uri: remove mapping example
6395 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6397 * examples/test-uri.c:
6398 test-uri: fix callback signature
6400 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6402 * gst/rtsp-server/rtsp-media-factory.c:
6403 factory: keep ref to factory while media active
6404 While the media from a factory is alive, keep a ref to the factory.
6405 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
6407 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6409 * gst/rtsp-server/rtsp-media-factory-uri.c:
6410 factory-uri: add some debug
6412 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6414 * gst/rtsp-server/rtsp-stream.c:
6415 stream: set udp sources to PLAYING
6416 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
6417 so that it doesn't cause our pipeline to produce ASYNC-DONE.
6419 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6421 * gst/rtsp-server/rtsp-media-factory-uri.c:
6422 factory-uri: take ref to factory
6423 Take a ref to the factory that we place in our list.
6425 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6427 * tests/Makefile.am:
6428 * tests/test-reuse.c:
6429 test: add test for server reuse
6430 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
6432 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
6434 * gst/rtsp-server/rtsp-server.c:
6435 server: start and stop multiple times
6436 Stop listening on the RTSP port when the GSource is removed, so clients
6437 can't connect and the server can be started again.
6438 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
6440 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6442 * gst/rtsp-server/rtsp-server.c:
6443 server: fix small leak
6445 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6447 * gst/rtsp-server/rtsp-media.c:
6448 media: unref source in finish_unprepare
6449 The source is created in prepare, unref it in finish_unprepare.
6450 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
6452 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
6454 * gst/rtsp-server/rtsp-client.c:
6455 * gst/rtsp-server/rtsp-media.c:
6456 rtsp-media: remove bus watch before finalizing
6457 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
6458 * An extra media ref is added for the bus watch. This extra ref is unreffed by
6459 the GDestroyNotify function.
6460 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
6461 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
6462 gst_rtsp_media_unprepare before unreffing the media.
6463 This way, the bus watch will be removed before the media is finalized.
6464 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
6466 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
6468 * gst/rtsp-server/rtsp-client.c:
6469 * gst/rtsp-server/rtsp-client.h:
6470 client: wait until the TEARDOWN response is sent to close the connection
6471 Responses can be sent async so we need to wait until the TEARDOWN response has
6472 been written before we close the connection to the client. This avoids the risk
6473 of writing/polling closed sockets.
6474 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
6476 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
6478 * gst/rtsp-server/rtsp-stream.c:
6479 rtsp-stream: plug socket leak
6480 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
6482 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
6485 Automatic update of common submodule
6486 From 6bb6951 to a72faea
6488 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
6490 * gst/rtsp-server/rtsp-media-factory-uri.c:
6491 rtsp-server: don't use deprecated API
6493 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
6495 * gst/rtsp-server/rtsp-client.c:
6496 rtsp-client: fix unused-but-set-variable compiler warning
6497 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
6499 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6502 * docs/libs/gst-rtsp-server-sections.txt:
6503 * gst/rtsp-server/rtsp-client.c:
6506 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6508 * examples/Makefile.am:
6509 * examples/test-multicast2.c:
6510 examples: add another multicast example
6511 Add an example for how to configure separate multicast ranges for each media
6514 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6516 * examples/test-multicast.c:
6519 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6521 * gst/rtsp-server/rtsp-client.c:
6522 * gst/rtsp-server/rtsp-media.c:
6523 * gst/rtsp-server/rtsp-session-media.c:
6524 * gst/rtsp-server/rtsp-session-media.h:
6525 * gst/rtsp-server/rtsp-stream-transport.c:
6526 * gst/rtsp-server/rtsp-stream-transport.h:
6527 stream: use the address managed by the stream
6528 Use the address managed by the stream for multicast. This allows us to have 1
6529 multicast address for each stream.
6530 Because the address is now managed by the stream we don't have to pass it around
6532 Set the address pool on the streams.
6534 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6536 * gst/rtsp-server/rtsp-client.c:
6537 * gst/rtsp-server/rtsp-media.c:
6538 * gst/rtsp-server/rtsp-stream.c:
6541 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6543 * gst/rtsp-server/rtsp-media.c:
6544 * gst/rtsp-server/rtsp-media.h:
6545 media: add signal for new streams
6546 This allows applications to listen for new streams and configure properties on
6547 them, like the address pool.
6549 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6551 * gst/rtsp-server/rtsp-media.c:
6552 media: configure address pool in new streams
6554 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6556 * gst/rtsp-server/rtsp-stream.c:
6557 * gst/rtsp-server/rtsp-stream.h:
6558 stream: add methods to deal with address pool
6559 Add methods to get and set the address pool for the stream
6560 Add method to allocate and get the multicast addresses for this stream.
6562 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6564 * docs/libs/gst-rtsp-server-sections.txt:
6565 * gst/rtsp-server/rtsp-media.c:
6566 * gst/rtsp-server/rtsp-media.h:
6567 media: remove MTU property
6568 It is a stream property
6570 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6572 * gst/rtsp-server/rtsp-client.c:
6573 client: set blocksize only on stream
6574 Set the blocksize only on the current stream.
6576 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6578 * gst/rtsp-server/rtsp-stream.c:
6579 stream: share src and sink sockets
6580 the allocated socket is in the used-socket property, not socket.
6582 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6584 * gst/rtsp-server/rtsp-address-pool.c:
6585 * gst/rtsp-server/rtsp-address-pool.h:
6586 * gst/rtsp-server/rtsp-client.c:
6587 * gst/rtsp-server/rtsp-session-media.c:
6588 * gst/rtsp-server/rtsp-session-media.h:
6589 * gst/rtsp-server/rtsp-stream-transport.c:
6590 * gst/rtsp-server/rtsp-stream-transport.h:
6591 * tests/check/gst/addresspool.c:
6592 rtsp: make address-pool return an address object
6593 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
6594 store more info in the structure and allows us to more easily return the address
6595 to the right pool when no longer needed.
6596 Pass the address to the StreamTransport so that we can return it to the pool
6597 when the stream transport is freed or changed.
6599 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6601 * examples/Makefile.am:
6602 * examples/test-multicast.c:
6603 examples: add multicast example
6604 Show how to set up the multicast address pool so that media can be
6605 server with multicast.
6607 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6609 * gst/rtsp-server/rtsp-client.c:
6610 * gst/rtsp-server/rtsp-media-factory.c:
6611 * gst/rtsp-server/rtsp-media-factory.h:
6612 * gst/rtsp-server/rtsp-media.c:
6613 * gst/rtsp-server/rtsp-media.h:
6614 rtsp: use AddressPool
6615 Remove the multicast_group property.
6616 Use the configured addresspool to allocate multicast addresses.
6618 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6620 * gst/rtsp-server/rtsp-address-pool.c:
6621 * gst/rtsp-server/rtsp-address-pool.h:
6622 address-pool: add clear method
6624 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6626 * gst/rtsp-server/rtsp-address-pool.c:
6627 address-pool: small cleanups
6629 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6631 * tests/check/Makefile.am:
6632 * tests/check/gst/addresspool.c:
6633 tests: add addresspool unit test
6635 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6637 * gst/rtsp-server/Makefile.am:
6638 * gst/rtsp-server/rtsp-address-pool.c:
6639 * gst/rtsp-server/rtsp-address-pool.h:
6640 address-pool: add object to manage multicast addresses
6641 Make an object that can manage a rage of multicast addresses and ports.
6643 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6645 * gst/rtsp-server/rtsp-server.c:
6646 server: set default max-threads property
6648 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6650 * gst/rtsp-server/rtsp-media.c:
6651 media: wait for concurrent _prepare
6652 If a prepare is busy, wait for the result.
6654 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6656 * gst/rtsp-server/rtsp-media.c:
6657 media: add lock around message handler
6658 We don't want to dispatch messages while we are still processing the result of
6661 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6663 * gst/rtsp-server/rtsp-media.c:
6664 * gst/rtsp-server/rtsp-media.h:
6665 media: add lock to protect state changes
6667 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6669 * gst/rtsp-server/rtsp-stream.c:
6670 * gst/rtsp-server/rtsp-stream.h:
6673 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6675 * gst/rtsp-server/rtsp-stream-transport.c:
6676 * gst/rtsp-server/rtsp-stream-transport.h:
6677 * gst/rtsp-server/rtsp-stream.c:
6678 stream-transport: add keep-alive method
6680 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6682 * gst/rtsp-server/rtsp-stream-transport.c:
6683 * gst/rtsp-server/rtsp-stream-transport.h:
6684 * gst/rtsp-server/rtsp-stream.c:
6685 stream-transport: add method to handle RTP/RTCP
6686 Call new methods instead of poking into the structures directly.
6688 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6690 * gst/rtsp-server/rtsp-session-media.c:
6691 * gst/rtsp-server/rtsp-session-media.h:
6692 session-media: add locking
6694 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6696 * gst/rtsp-server/rtsp-session.c:
6697 * gst/rtsp-server/rtsp-session.h:
6698 session: add locking
6700 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6702 * gst/rtsp-server/rtsp-server.c:
6703 server: free old socket
6705 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6707 * gst/rtsp-server/rtsp-media-mapping.c:
6708 * gst/rtsp-server/rtsp-media-mapping.h:
6709 mapping: add locking
6711 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6713 * gst/rtsp-server/rtsp-media-factory.c:
6714 media-factory: add locking
6716 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6718 * gst/rtsp-server/rtsp-auth.c:
6719 * gst/rtsp-server/rtsp-auth.h:
6722 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6724 * gst/rtsp-server/rtsp-server.c:
6725 * gst/rtsp-server/rtsp-server.h:
6726 server: add max-thread property
6728 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6730 * gst/rtsp-server/rtsp-server.c:
6731 * gst/rtsp-server/rtsp-server.h:
6732 server: use a threadpool for the mainloops
6734 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6736 * gst/rtsp-server/rtsp-client.c:
6737 * gst/rtsp-server/rtsp-client.h:
6738 client: rename method
6739 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
6740 don't really create the client from the socket, we use the socket for the
6743 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6745 * gst/rtsp-server/rtsp-client.c:
6746 * gst/rtsp-server/rtsp-client.h:
6747 * gst/rtsp-server/rtsp-server.c:
6748 server: rework maincontext handling in clients
6749 Make a separate method to attach a client to a MainContext.
6750 Let the server decide in what GMainContext the client will operate and give this
6751 context to the client in attach. Then the server can later decide to use a
6752 separate thread for each client or just use the mainthread.
6754 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6756 * gst/rtsp-server/rtsp-client.c:
6757 * gst/rtsp-server/rtsp-session.c:
6758 * gst/rtsp-server/rtsp-session.h:
6759 session: move session header code in session object
6761 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
6765 * examples/test-auth.c:
6766 * examples/test-launch.c:
6767 * examples/test-mp4.c:
6768 * examples/test-ogg.c:
6769 * examples/test-readme.c:
6770 * examples/test-sdp.c:
6771 * examples/test-uri.c:
6772 * examples/test-video.c:
6773 * gst/rtsp-server/rtsp-auth.c:
6774 * gst/rtsp-server/rtsp-auth.h:
6775 * gst/rtsp-server/rtsp-client.c:
6776 * gst/rtsp-server/rtsp-client.h:
6777 * gst/rtsp-server/rtsp-media-factory-uri.c:
6778 * gst/rtsp-server/rtsp-media-factory-uri.h:
6779 * gst/rtsp-server/rtsp-media-factory.c:
6780 * gst/rtsp-server/rtsp-media-factory.h:
6781 * gst/rtsp-server/rtsp-media-mapping.c:
6782 * gst/rtsp-server/rtsp-media-mapping.h:
6783 * gst/rtsp-server/rtsp-media.c:
6784 * gst/rtsp-server/rtsp-media.h:
6785 * gst/rtsp-server/rtsp-params.c:
6786 * gst/rtsp-server/rtsp-params.h:
6787 * gst/rtsp-server/rtsp-sdp.c:
6788 * gst/rtsp-server/rtsp-sdp.h:
6789 * gst/rtsp-server/rtsp-server.c:
6790 * gst/rtsp-server/rtsp-server.h:
6791 * gst/rtsp-server/rtsp-session-media.c:
6792 * gst/rtsp-server/rtsp-session-media.h:
6793 * gst/rtsp-server/rtsp-session-pool.c:
6794 * gst/rtsp-server/rtsp-session-pool.h:
6795 * gst/rtsp-server/rtsp-session.c:
6796 * gst/rtsp-server/rtsp-session.h:
6797 * gst/rtsp-server/rtsp-stream-transport.c:
6798 * gst/rtsp-server/rtsp-stream-transport.h:
6799 * gst/rtsp-server/rtsp-stream.c:
6800 * gst/rtsp-server/rtsp-stream.h:
6801 * tests/check/gst/rtspserver.c:
6802 * tests/test-cleanup.c:
6805 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6807 * gst/rtsp-server/rtsp-media.c:
6808 * gst/rtsp-server/rtsp-session-media.c:
6809 * gst/rtsp-server/rtsp-session.c:
6810 rtsp-server: added annotations to indicate type of ownership transfer of return values
6811 https://bugzilla.gnome.org/show_bug.cgi?id=680777
6813 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
6816 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
6818 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
6821 * bindings/Makefile.am:
6822 * bindings/vala/Makefile.am:
6823 * bindings/vala/gst-rtsp-server-0.10.deps:
6824 * bindings/vala/gst-rtsp-server-0.10.vapi:
6825 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
6826 * bindings/vala/packages/gst-rtsp-server-0.10.files:
6827 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6828 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6829 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
6831 bindings: remove vala bindings
6832 They'll be reunited with the other GStreamer bindings
6833 https://bugzilla.gnome.org/show_bug.cgi?id=680777
6835 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6837 * gst/rtsp-server/rtsp-client.c:
6838 * gst/rtsp-server/rtsp-session-media.c:
6839 * gst/rtsp-server/rtsp-session-media.h:
6840 * gst/rtsp-server/rtsp-stream-transport.c:
6841 * gst/rtsp-server/rtsp-stream-transport.h:
6842 rtsp: only create transport when needed
6843 Only create the StreamTransport when configured.
6845 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6847 * gst/rtsp-server/rtsp-client.c:
6848 client: small cleanup
6850 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6852 * gst/rtsp-server/rtsp-client.c:
6853 * gst/rtsp-server/rtsp-client.h:
6854 * gst/rtsp-server/rtsp-stream-transport.c:
6855 * gst/rtsp-server/rtsp-stream-transport.h:
6856 rtsp: refactor configuration of transport
6857 Move the configuration of the transport to a place where it makes
6860 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6862 * gst/rtsp-server/rtsp-client.c:
6863 client: refactor transport parsing
6865 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6867 * gst/rtsp-server/rtsp-client.c:
6868 client: refuse to change the MTU on shared media
6869 If we change the MTU of chared media, it changes for all clients.
6870 We don't want to set the MTU to something large for clients that
6873 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6875 * examples/test-mp4.c:
6876 * gst/rtsp-server/rtsp-media.c:
6877 small fixes to docs and debug
6879 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6881 * gst/rtsp-server/rtsp-stream.c:
6882 stream: transports must already have been removed
6884 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6886 * gst/rtsp-server/rtsp-media.c:
6887 * gst/rtsp-server/rtsp-stream.c:
6888 * gst/rtsp-server/rtsp-stream.h:
6889 stream: improve join and leave of the pipeline
6891 Do the cleanup properly
6894 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6896 * gst/rtsp-server/rtsp-media.c:
6897 media: move unprepare below default implementation
6898 Makes it easier to find the default implementation
6900 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6902 * gst/rtsp-server/rtsp-media.c:
6903 media: signal unprepared when we actually finish
6905 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6907 * gst/rtsp-server/rtsp-media.c:
6908 media: no need to unlock, unprepare does that when needed
6910 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6912 * docs/libs/gst-rtsp-server-sections.txt:
6913 * gst/rtsp-server/rtsp-media-factory.h:
6914 * gst/rtsp-server/rtsp-media-mapping.c:
6915 * gst/rtsp-server/rtsp-media.h:
6916 * gst/rtsp-server/rtsp-params.c:
6917 * gst/rtsp-server/rtsp-server.c:
6918 * gst/rtsp-server/rtsp-session-pool.h:
6919 * gst/rtsp-server/rtsp-session.c:
6920 * gst/rtsp-server/rtsp-session.h:
6921 * gst/rtsp-server/rtsp-stream-transport.h:
6922 * gst/rtsp-server/rtsp-stream.h:
6925 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6927 * gst/rtsp-server/rtsp-client.c:
6928 * gst/rtsp-server/rtsp-media-mapping.h:
6929 * gst/rtsp-server/rtsp-media.c:
6930 * gst/rtsp-server/rtsp-media.h:
6931 * gst/rtsp-server/rtsp-server.h:
6932 * gst/rtsp-server/rtsp-stream.c:
6933 * gst/rtsp-server/rtsp-stream.h:
6934 rtsp: fix MTU setting
6935 Fix setting of the MTU. There is no need for a vmethod.
6937 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6942 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6945 configure: bump version number after refactoring
6947 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6949 * gst/rtsp-server/Makefile.am:
6950 * gst/rtsp-server/rtsp-client.c:
6951 * gst/rtsp-server/rtsp-client.h:
6952 * gst/rtsp-server/rtsp-media-factory-uri.c:
6953 * gst/rtsp-server/rtsp-media-factory.c:
6954 * gst/rtsp-server/rtsp-media-factory.h:
6955 * gst/rtsp-server/rtsp-media.c:
6956 * gst/rtsp-server/rtsp-media.h:
6957 * gst/rtsp-server/rtsp-sdp.c:
6958 * gst/rtsp-server/rtsp-session-media.c:
6959 * gst/rtsp-server/rtsp-session-media.h:
6960 * gst/rtsp-server/rtsp-session.c:
6961 * gst/rtsp-server/rtsp-session.h:
6962 * gst/rtsp-server/rtsp-stream-transport.c:
6963 * gst/rtsp-server/rtsp-stream-transport.h:
6964 * gst/rtsp-server/rtsp-stream.c:
6965 * gst/rtsp-server/rtsp-stream.h:
6966 rtsp: massive refactoring
6967 Make GObjects from the remaining simple structures.
6968 Remove GstRTSPSessionStream, it's not needed.
6969 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
6970 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
6971 a GstRTSPStream should be transported to a client.
6972 Rename GstRTSPMediaFactory::get_element -> create_element because that
6973 more accurately describes what it does.
6974 Make nice methods instead of poking in the structures.
6975 Move some methods inside the relevant object source code.
6976 Use GPtrArray to store objects instead of plain arrays, it is more
6977 natural and allows us to more easily clean up.
6978 Move the allocation of udp ports to the Stream object. The Stream object
6979 contains the elements needed to stream the media to a client.
6980 Improve the prepare and unprepare methods. Unprepare should now undo
6981 everything prepare did. Improve also async unprepare when doing EOS on
6982 shutdown. Make sure we always unprepare correctly.
6984 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
6986 * gst/rtsp-server/rtsp-client.c:
6987 rtsp-client: Unref server address clients connected to
6988 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
6990 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
6992 * gst/rtsp-server/rtsp-server.c:
6993 rtsp-server: don't ref server socket if it is NULL
6994 Fixes test_bind_already_in_use unit test again after commit 6a497440.
6995 https://bugzilla.gnome.org/show_bug.cgi?id=686644
6997 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
6999 * tests/check/Makefile.am:
7000 tests: Add libgio link dependency
7001 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
7003 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7005 * gst/rtsp-server/rtsp-media-mapping.c:
7006 * gst/rtsp-server/rtsp-media-mapping.h:
7007 rtsp-media-mapping: rename find_media vfunc to find_factory
7008 The virtual method and class method should have the same name
7009 so it is correctly represented in GIR file
7010 https://bugzilla.gnome.org/show_bug.cgi?id=680777
7012 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7014 * gst/rtsp-server/rtsp-auth.c:
7015 * gst/rtsp-server/rtsp-client.c:
7016 * gst/rtsp-server/rtsp-media-factory-uri.c:
7017 * gst/rtsp-server/rtsp-media-factory.c:
7018 * gst/rtsp-server/rtsp-media-mapping.c:
7019 * gst/rtsp-server/rtsp-media.c:
7020 * gst/rtsp-server/rtsp-server.c:
7021 * gst/rtsp-server/rtsp-session-pool.c:
7022 * gst/rtsp-server/rtsp-session.c:
7023 rtsp-server: fixed comments and GIR annotations
7024 https://bugzilla.gnome.org/show_bug.cgi?id=680777
7026 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
7028 * gst/rtsp-server/rtsp-media-mapping.c:
7029 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
7031 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
7033 * gst/rtsp-server/rtsp-server.c:
7034 rtsp-server: allow binding on port 0 (binds on a random port)
7036 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
7038 * gst/rtsp-server/rtsp-server.c:
7039 * gst/rtsp-server/rtsp-server.h:
7040 rtsp-server: add bound-port property
7041 bound-port can be used to retrieve the port number when the server is bound on
7042 port 0, which binds on a random port.
7044 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
7046 * gst/rtsp-server/rtsp-media-factory.c:
7047 * gst/rtsp-server/rtsp-media-factory.h:
7048 rtsp-media-factory: make ::get_element overridable by GI bindings
7049 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
7050 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
7051 as the invoker for ::get_element(), making it overridable by GI generated
7054 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
7056 * gst/rtsp-server/rtsp-media-factory-uri.c:
7057 rtsp-media-factory-uri: don't autoplug parsers in a loop
7058 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
7061 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
7063 * gst/rtsp-server/Makefile.am:
7064 Explicitly link against gio. Fix link error on mac.
7066 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
7068 * gst/rtsp-server/rtsp-session.c:
7069 session: add ttl to the transport header in SETUP
7070 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
7072 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
7074 * gst/rtsp-server/rtsp-client.c:
7075 * gst/rtsp-server/rtsp-client.h:
7076 * gst/rtsp-server/rtsp-media.c:
7077 client: Use client transport settings for multicast if allowed.
7078 This patch makes it possible for the client to send transport settings for
7079 multicast (destination && ttl). Client settings must be explicitly allowed or
7080 the server will use its own settings.
7081 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
7083 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
7086 Automatic update of common submodule
7087 From 6c0b52c to 6bb6951
7089 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
7091 * gst/rtsp-server/rtsp-client.c:
7092 rtsp-client: do not destroy the rtsp watch
7093 Don't destroy the client watch while dispatching. The rtsp watch is
7094 automatically destroyed after the rtsp watch function closed() has
7096 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
7098 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
7101 Automatic update of common submodule
7102 From 4f962f7 to 6c0b52c
7104 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
7106 * gst/rtsp-server/rtsp-media.c:
7107 media: fix check for seekability
7109 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7111 * gst/rtsp-server/rtsp-client.c:
7112 client: use more GIO
7113 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
7115 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7117 * gst/rtsp-server/rtsp-server.c:
7118 server: remove obsolete includes
7120 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
7122 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
7123 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
7124 be available in "on_new_ssrc". The transports are added in
7125 gst_rtsp_media_set_state when going to PLAYING state. However,
7126 "on_new_ssrc" might be called before this happens.
7127 https://bugzilla.gnome.org/show_bug.cgi?id=683304
7129 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
7131 * gst/rtsp-server/rtsp-client.c:
7132 * gst/rtsp-server/rtsp-client.h:
7133 rtsp-client: add signals for rtsp requests (fixes #683287)
7135 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
7137 * gst/rtsp-server/rtsp-client.c:
7138 * gst/rtsp-server/rtsp-client.h:
7139 add new-session signal to rtsp-client (fixes #683058)
7141 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
7144 Automatic update of common submodule
7145 From 668acee to 4f962f7
7147 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
7149 * gst/rtsp-server/rtsp-server.c:
7150 * tests/check/gst/rtspserver.c:
7151 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
7152 Do not assume that *error is set in g_socket_address_enumerator_next.
7153 Added test_bind_already_in_use unit-test.
7154 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
7156 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
7159 Automatic update of common submodule
7160 From 94ccf4c to 668acee
7162 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
7164 * gst/rtsp-server/rtsp-client.c:
7165 * gst/rtsp-server/rtsp-client.h:
7166 rtsp-client: make create_sdp virtual method
7167 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
7169 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7172 Automatic update of common submodule
7173 From 98e386f to 94ccf4c
7175 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7177 * gst/rtsp-server/rtsp-client.c:
7180 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
7182 * gst/rtsp-server/rtsp-client.c:
7183 * gst/rtsp-server/rtsp-client.h:
7184 * gst/rtsp-server/rtsp-server.c:
7185 * gst/rtsp-server/rtsp-server.h:
7186 rtsp-server: use an existing socket to establish HTTP tunnel
7187 Make it possible to transfer a socket from an HTTP server to be used as
7188 an RTSP over HTTP tunnel.
7190 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
7192 * gst/rtsp-server/rtsp-client.c:
7193 * gst/rtsp-server/rtsp-media.c:
7194 * gst/rtsp-server/rtsp-media.h:
7195 rtsp: Handle the blocksize parameter
7196 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
7198 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
7200 * tests/check/Makefile.am:
7201 * tests/check/gst/rtspserver.c:
7202 Have unit test get header from source dir, not installed dir
7203 This makes compilation of unit tests work in a build directory other
7204 than the source directory.
7205 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
7207 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
7209 * gst/rtsp-server/rtsp-media.c:
7210 rtsp-media: update for gst_element_make_from_uri() changes
7212 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
7215 * tests/Makefile.am:
7216 * tests/check/Makefile.am:
7217 * tests/check/gst/rtspserver.c:
7219 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
7221 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
7223 * gst/rtsp-server/rtsp-media.c:
7224 rtsp-media: don't collect media stats when going to NULL
7225 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
7227 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7229 * gst/rtsp-server/rtsp-client.c:
7230 client: don't leak transports
7232 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
7234 * gst/rtsp-server/rtsp-client.c:
7235 rtsp-client: free transport on no_stream in SETUP handler
7237 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
7239 * gst/rtsp-server/rtsp-client.c:
7240 rtsp-client: changed session media iteration
7241 In client_unlink_session: now don't iterate in session->medias
7242 list where items are removed by gst_rtsp_session_release_media.
7243 Instead, repeatedly remove the first item.
7245 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
7247 * gst/rtsp-server/rtsp-client.c:
7248 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
7249 GstRTSPSessionMedia is not a GObject type. When the
7250 GstRTSPSession is freed, it will free the media.
7252 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
7254 * gst/rtsp-server/rtsp-media-factory.c:
7255 factory: plug pad leak in collect_streams
7256 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
7257 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
7258 will take one reference, and the other reference will otherwise
7261 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
7264 configure: suppress some warnings when debug is disabled
7265 Warnings about unused variables should be suppressed if core has the
7266 debug system disabled.
7267 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
7269 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7271 * docs/libs/Makefile.am:
7272 docs: fix build in uninstalled setup
7273 Include gst-plugins-base libs properly.
7275 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
7277 * docs/libs/gst-rtsp-server.types:
7278 docs: include headers defining rtsp-server object types
7279 Fixes compiler warnings during docs build.
7280 https://bugzilla.gnome.org/show_bug.cgi?id=676824
7282 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
7285 configure: Add warning flags for compiler when configuring
7286 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
7288 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7291 Automatic update of common submodule
7292 From 03a0e57 to 98e386f
7294 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7297 Automatic update of common submodule
7298 From 1fab359 to 03a0e57
7300 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
7302 * gst/rtsp-server/rtsp-client.c:
7303 client: fix GSocketAddress leak in gst_rtsp_client_accept
7304 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
7306 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7309 Automatic update of common submodule
7310 From f1b5a96 to 1fab359
7312 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7315 Automatic update of common submodule
7316 From 92b7266 to f1b5a96
7318 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7321 Automatic update of common submodule
7322 From ec1c4a8 to 92b7266
7324 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7327 Automatic update of common submodule
7328 From 3429ba6 to ec1c4a8
7330 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
7332 * gst/rtsp-server/rtsp-auth.c:
7333 * gst/rtsp-server/rtsp-client.c:
7334 * gst/rtsp-server/rtsp-media-factory-uri.c:
7335 * gst/rtsp-server/rtsp-server.c:
7336 rtsp: fix compiler warnings
7337 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
7339 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7342 Automatic update of common submodule
7343 From dc70203 to 3429ba6
7345 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7347 * gst/rtsp-server/rtsp-client.c:
7348 * gst/rtsp-server/rtsp-media-factory.c:
7349 * gst/rtsp-server/rtsp-media-factory.h:
7350 * gst/rtsp-server/rtsp-media.c:
7351 * gst/rtsp-server/rtsp-media.h:
7352 * gst/rtsp-server/rtsp-server.c:
7353 * gst/rtsp-server/rtsp-server.h:
7354 * gst/rtsp-server/rtsp-session-pool.c:
7355 * gst/rtsp-server/rtsp-session-pool.h:
7356 rtsp-server: port to new thread API
7358 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7361 Automatic update of common submodule
7362 From 6db25be to dc70203
7364 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7366 * gst/rtsp-server/rtsp-auth.c:
7367 * gst/rtsp-server/rtsp-auth.h:
7368 * gst/rtsp-server/rtsp-client.c:
7369 rtsp-server: Fix compilation and compiler warnings
7371 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7375 * gst/rtsp-server/Makefile.am:
7376 configure: Modernize autotools setup a bit
7377 Also we now only create tar.bz2 and tar.xz tarballs.
7379 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7382 Automatic update of common submodule
7383 From 464fe15 to 6db25be
7385 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7388 Automatic update of common submodule
7389 From 7fda524 to 464fe15
7391 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7394 * docs/libs/Makefile.am:
7395 * docs/version.entities.in:
7397 * gst/rtsp-server/Makefile.am:
7398 * pkgconfig/Makefile.am:
7399 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
7400 * pkgconfig/gstreamer-rtsp-server.pc.in:
7401 * tests/Makefile.am:
7402 rtsp-server: Update versioning
7404 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7406 Merge remote-tracking branch 'origin/0.10'
7408 gst/rtsp-server/rtsp-session-pool.c
7410 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7412 * gst/rtsp-server/rtsp-session-pool.c:
7413 rtsp-server: Don't use deprecated GLib API
7415 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7417 Replace master with 0.11
7419 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7421 Merge branch 'master' into 0.11
7423 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7425 Merge branch 'master' into 0.11
7427 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
7430 A couple minor typo fixes
7432 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7434 * gst/rtsp-server/rtsp-media.c:
7435 media: fix state of the appqueue
7437 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7439 * gst/rtsp-server/rtsp-media-factory-uri.c:
7440 factory: use videoconvert
7442 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7444 * gst/rtsp-server/rtsp-media-factory-uri.c:
7445 factory: change to new style caps
7447 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7449 * gst/rtsp-server/rtsp-client.c:
7450 * gst/rtsp-server/rtsp-client.h:
7451 * gst/rtsp-server/rtsp-media-factory-uri.c:
7452 * gst/rtsp-server/rtsp-media.c:
7453 * gst/rtsp-server/rtsp-server.c:
7454 * gst/rtsp-server/rtsp-server.h:
7455 * gst/rtsp-server/rtsp-session-pool.c:
7456 rtsp-server: port to GIO
7459 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7462 configure: fix build
7464 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7467 docs: fix for gst_rtsp_server_set_port() -> _set_service()
7468 https://bugzilla.gnome.org/show_bug.cgi?id=666548
7470 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7473 * examples/Makefile.am:
7474 First rule of gst-rtsp-server club: don't talk about gst-phonon
7476 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7479 * pkgconfig/Makefile.am:
7480 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
7481 * pkgconfig/gstreamer-rtsp-server.pc.in:
7482 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
7483 For consistency with all other modules.
7485 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7487 * gst/rtsp-server/rtsp-client.c:
7488 rtsp-client: update for new map API
7490 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7493 * bindings/Makefile.am:
7494 * bindings/python/Makefile.am:
7495 * bindings/python/arg-types.py:
7496 * bindings/python/codegen/Makefile.am:
7497 * bindings/python/codegen/__init__.py:
7498 * bindings/python/codegen/argtypes.py:
7499 * bindings/python/codegen/code-coverage.py:
7500 * bindings/python/codegen/codegen.py:
7501 * bindings/python/codegen/definitions.py:
7502 * bindings/python/codegen/defsparser.py:
7503 * bindings/python/codegen/docextract.py:
7504 * bindings/python/codegen/docgen.py:
7505 * bindings/python/codegen/fileprefix.override:
7506 * bindings/python/codegen/fileprefixmodule.c:
7507 * bindings/python/codegen/h2def.py:
7508 * bindings/python/codegen/mergedefs.py:
7509 * bindings/python/codegen/mkskel.py:
7510 * bindings/python/codegen/override.py:
7511 * bindings/python/codegen/reversewrapper.py:
7512 * bindings/python/codegen/scmexpr.py:
7513 * bindings/python/rtspserver-types.defs:
7514 * bindings/python/rtspserver.defs:
7515 * bindings/python/rtspserver.override:
7516 * bindings/python/rtspservermodule.c:
7517 * bindings/python/test.py:
7519 python: remove pygst-based python bindings
7520 pygi is the future, apparently.
7522 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
7525 Automatic update of common submodule
7526 From c463bc0 to 7fda524
7528 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7531 Automatic update of common submodule
7532 From 2a59016 to c463bc0
7534 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7537 Automatic update of common submodule
7538 From 0807187 to 2a59016
7540 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7543 Automatic update of common submodule
7544 From 11f0cd5 to 0807187
7546 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7548 * examples/test-auth.c:
7549 example: update for new caps
7551 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7553 * examples/test-video.c:
7554 * gst/rtsp-server/rtsp-client.c:
7555 * gst/rtsp-server/rtsp-media-factory-uri.c:
7556 * gst/rtsp-server/rtsp-media.c:
7557 * gst/rtsp-server/rtsp-media.h:
7558 * gst/rtsp-server/rtsp-session.c:
7559 * gst/rtsp-server/rtsp-session.h:
7560 rtsp-server: port some more to 0.11
7562 Remove bufferlist stuff
7564 Add queue before appsink now that preroll-queue-len is gone.
7565 Update for request pad changes.
7567 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7569 Merge branch 'master' into 0.11
7571 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
7573 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7574 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
7575 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
7577 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
7579 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7580 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
7581 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
7583 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7585 Merge branch 'master' into 0.11
7587 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7589 * gst/rtsp-server/rtsp-media.c:
7590 * gst/rtsp-server/rtsp-media.h:
7591 media: add a seekable boolean
7592 Maintain the seekable state with a new variable instead of reusing the
7595 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
7597 * gst/rtsp-server/rtsp-media.c:
7598 Disallow seek in live media
7600 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7602 Merge branch 'master' into 0.11
7604 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
7606 * gst/rtsp-server/rtsp-server.c:
7607 #ifdef statements for windows socket creation were missing
7609 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
7612 Automatic update of common submodule
7613 From a39eb83 to 11f0cd5
7615 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
7618 Automatic update of common submodule
7619 From 605cd9a to a39eb83
7621 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7623 Merge branch 'master' into 0.11
7625 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7627 * gst/rtsp-server/rtsp-client.c:
7628 client: use method to access property
7630 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7632 * gst/rtsp-server/rtsp-media-factory.c:
7633 * gst/rtsp-server/rtsp-media-factory.h:
7634 media-factory: add protocols property
7635 Add a property to configure the allowed protocols in the media created from the
7638 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7640 * gst/rtsp-server/rtsp-media-factory.c:
7641 * gst/rtsp-server/rtsp-media-factory.h:
7642 media-factory: add media-configure signal
7643 Add signal to allow the application to configure the media after it was created
7646 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7648 * gst/rtsp-server/rtsp-client.c:
7649 client: use method to access property
7651 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7653 * gst/rtsp-server/rtsp-media-factory.c:
7654 * gst/rtsp-server/rtsp-media-factory.h:
7655 media-factory: add protocols property
7656 Add a property to configure the allowed protocols in the media created from the
7659 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7661 * gst/rtsp-server/rtsp-media-factory.c:
7662 * gst/rtsp-server/rtsp-media-factory.h:
7663 media-factory: add media-configure signal
7664 Add signal to allow the application to configure the media after it was created
7667 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7669 Merge branch 'master' into 0.11
7671 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7673 * gst/rtsp-server/rtsp-client.c:
7674 client: use media multicast group
7676 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7678 * gst/rtsp-server/rtsp-media-factory.h:
7679 * gst/rtsp-server/rtsp-server.h:
7680 * gst/rtsp-server/rtsp-session-pool.h:
7681 * gst/rtsp-server/rtsp-session.h:
7684 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
7686 * gst/rtsp-server/rtsp-client.c:
7687 * gst/rtsp-server/rtsp-sdp.h:
7688 sdp: copy and free the server ip address
7689 Copy and free the server ip address to make memory management easier later.
7691 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7693 * gst/rtsp-server/rtsp-media-factory.c:
7694 media-factory: configure multicast in media
7696 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7698 * gst/rtsp-server/rtsp-media.c:
7699 * gst/rtsp-server/rtsp-media.h:
7700 media: add property for multicast group
7701 Add a property to configure the multicast group in the media.
7702 Based on patches from Marc Leeman and Robert Krakora.
7704 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7706 * gst/rtsp-server/rtsp-media-factory.c:
7707 * gst/rtsp-server/rtsp-media-factory.h:
7708 media-factory: add property for multicast group
7709 Add a property to configure the multicast group in the media factory.
7710 Based on patches from Marc Leeman and Robert Krakora.
7712 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7714 * gst/rtsp-server/rtsp-client.c:
7715 client: do configuration of transport in one place
7716 Move the configuration of the transport destination address to where we also
7717 configure the other bits.
7719 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7721 * gst/rtsp-server/rtsp-client.c:
7722 client: use media multicast group
7724 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7726 * gst/rtsp-server/rtsp-media-factory.h:
7727 * gst/rtsp-server/rtsp-server.h:
7728 * gst/rtsp-server/rtsp-session-pool.h:
7729 * gst/rtsp-server/rtsp-session.h:
7732 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
7734 * gst/rtsp-server/rtsp-client.c:
7735 * gst/rtsp-server/rtsp-sdp.h:
7736 sdp: copy and free the server ip address
7737 Copy and free the server ip address to make memory management easier later.
7739 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7741 * gst/rtsp-server/rtsp-media-factory.c:
7742 media-factory: configure multicast in media
7744 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7746 * gst/rtsp-server/rtsp-media.c:
7747 * gst/rtsp-server/rtsp-media.h:
7748 media: add property for multicast group
7749 Add a property to configure the multicast group in the media.
7750 Based on patches from Marc Leeman and Robert Krakora.
7752 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7754 * gst/rtsp-server/rtsp-media-factory.c:
7755 * gst/rtsp-server/rtsp-media-factory.h:
7756 media-factory: add property for multicast group
7757 Add a property to configure the multicast group in the media factory.
7758 Based on patches from Marc Leeman and Robert Krakora.
7760 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7762 * gst/rtsp-server/rtsp-client.c:
7763 client: do configuration of transport in one place
7764 Move the configuration of the transport destination address to where we also
7765 configure the other bits.
7767 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7769 Merge branch 'master' into 0.11
7771 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
7773 * gst/rtsp-server/rtsp-client.c:
7774 client: destroy pipeline on client disconnect with no prior TEARDOWN.
7775 The problem occurs when the client abruptly closes the connection without
7776 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
7777 server is where the pipeline gets torn down. Since this handler is not called,
7778 the pipeline remains and is up and running. Subsequent clients get their own
7779 pipelines and if the do not issue TEARDOWNs then those pipelines will also
7780 remain up and running. This is a resource leak.
7782 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7784 Merge branch 'master' into 0.11
7786 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
7788 * gst/rtsp-server/rtsp-media-factory.c:
7789 * gst/rtsp-server/rtsp-media-factory.h:
7790 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
7791 For example, it can be used to retrieve source elements like appsrc, in a more
7792 convenient way than subclassing get_element.
7794 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7796 Merge branch 'master' into 0.11
7798 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
7800 * gst/rtsp-server/rtsp-server.c:
7801 rtsp-server: hold on to reference while using object
7803 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7805 * gst/rtsp-server/rtsp-media.c:
7808 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7811 configure: use unstable api
7813 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
7815 * gst/rtsp-server/rtsp-client.c:
7816 client: fix reference counting
7818 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
7820 * gst/rtsp-server/rtsp-client.c:
7821 * gst/rtsp-server/rtsp-media.c:
7822 fix compiler warnings about unused variables
7824 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
7826 * examples/test-launch.c:
7827 * examples/test-readme.c:
7828 * examples/test-uri.c:
7829 * examples/test-video.c:
7830 examples: tell rtsp uri when ready
7832 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
7835 Automatic update of common submodule
7836 From 69b981f to 605cd9a
7838 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7840 * gst/rtsp-server/rtsp-client.c:
7841 client: update for buffer API change
7843 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7845 * gst/rtsp-server/Makefile.am:
7846 Makefile.am: 0.10 => @GST_MAJORMINOR@
7848 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7850 * gst/rtsp-server/rtsp-media-factory-uri.c:
7851 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
7853 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7855 * gst/rtsp-server/.gitignore:
7856 .gitignore: 0.10 => 0.11
7858 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7860 * gst/rtsp-server/Makefile.am:
7861 Makefile.am: 0.10 => @GST_MAJORMINOR@
7863 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7865 Merge branch 'master' into 0.11
7867 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
7870 Automatic update of common submodule
7871 From 9e5bbd5 to 69b981f
7873 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
7876 Automatic update of common submodule
7877 From fd35073 to 9e5bbd5
7879 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
7882 Automatic update of common submodule
7883 From 46dfcea to fd35073
7885 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7887 * gst/rtsp-server/rtsp-media-factory-uri.c:
7888 * gst/rtsp-server/rtsp-media.c:
7889 media: port to new caps API
7891 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7893 Merge branch 'master' into 0.11
7895 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
7897 * bindings/vala/gst-rtsp-server-0.10.vapi:
7898 Updated Vala bindings.
7899 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
7901 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
7903 * gst/rtsp-server/rtsp-server.c:
7904 * gst/rtsp-server/rtsp-server.h:
7905 Add a signal for newly connected clients.
7906 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
7908 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
7910 * bindings/python/rtspserver.override:
7911 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
7913 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7915 * gst/rtsp-server/Makefile.am:
7916 * gst/rtsp-server/rtsp-client.c:
7917 * gst/rtsp-server/rtsp-funnel.c:
7918 * gst/rtsp-server/rtsp-funnel.h:
7919 * gst/rtsp-server/rtsp-media.c:
7920 rtsp-server: port to 0.11
7922 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7927 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7929 Merge branch 'master' into 0.11
7934 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7937 Automatic update of common submodule
7938 From c3cafe1 to 46dfcea
7940 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
7942 * bindings/python/Makefile.am:
7943 * bindings/python/rtspserver.defs:
7944 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
7946 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
7948 * bindings/python/arg-types.py:
7949 python bindings: add GstRTSPUrlParam
7950 Needed to implement MediaFactory virtual proxies
7952 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
7954 * bindings/python/arg-types.py:
7955 python bindings: fix returning GstRTSPUrl types
7957 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
7959 * bindings/python/arg-types.py:
7960 python bindings: add arg type for GstRTSPUrl
7962 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
7964 * bindings/python/rtspserver.defs:
7965 python bindings: fix the definition of MediaFactory.collect_stream
7967 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
7970 Automatic update of common submodule
7971 From 1ccbe09 to c3cafe1
7973 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7976 Automatic update of common submodule
7977 From 193b717 to 1ccbe09
7979 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
7982 Automatic update of common submodule
7983 From b77e2bf to 193b717
7985 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7988 build: Include lcov.mak to allow test coverage report generation
7990 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7993 Automatic update of common submodule
7994 From d8814b6 to b77e2bf
7996 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7999 Automatic update of common submodule
8000 From 6aaa286 to d8814b6
8002 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
8005 Automatic update of common submodule
8006 From 6aec6b9 to 6aaa286
8008 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
8011 autogen: wingo signed comment
8013 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
8015 * gst/rtsp-server/rtsp-session-pool.c:
8016 session: use full charset for RTSP session ID
8017 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
8018 session ID more difficult.
8019 https://bugzilla.gnome.org/show_bug.cgi?id=643812
8021 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8023 * gst/rtsp-server/Makefile.am:
8024 rtsp-server: Don't install the funnel header
8026 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
8029 Automatic update of common submodule
8030 From 1de7f6a to 6aec6b9
8032 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8035 configure: require core/base 0.10.31
8036 Needed at least for gst_plugin_feature_rank_compare_func().
8038 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
8041 Automatic update of common submodule
8042 From f94d739 to 1de7f6a
8044 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8046 * gst/rtsp-server/rtsp-media.c:
8047 media: remove more unused code
8049 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8051 * gst/rtsp-server/rtsp-media.c:
8052 * gst/rtsp-server/rtsp-media.h:
8053 media: remove duplicate filtering
8054 Remove the duplicate filtering code now that we have a released -good version.
8055 Give a warning instead.
8057 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8059 * gst/rtsp-server/rtsp-media-factory.c:
8060 * gst/rtsp-server/rtsp-media.c:
8061 media: fix default buffer size
8063 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8065 * gst/rtsp-server/rtsp-media-factory.c:
8066 * gst/rtsp-server/rtsp-media-factory.h:
8067 media-factory: add property to configure the buffer-size
8068 Add a property to configure the kernel UDP buffer size.
8070 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8072 * gst/rtsp-server/rtsp-media.c:
8073 * gst/rtsp-server/rtsp-media.h:
8074 media: add property to configure kernel buffer sizes
8075 Add a property to configure the kernel UDP buffer size.
8077 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8080 configure: set PYGOBJECT_REQ before using it
8081 https://bugzilla.gnome.org/show_bug.cgi?id=640641
8083 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8086 docs: recursive into sub-directories on 'make upload'
8088 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8090 * docs/libs/gst-rtsp-server-docs.sgml:
8091 * docs/version.entities.in:
8092 docs: mention full version these docs are for, not just major-minor
8094 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8099 === release 0.10.8 ===
8101 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8106 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8108 * gst/rtsp-server/rtsp-server.c:
8109 rtsp-server: clarify docs a little
8111 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8113 * gst/rtsp-server/rtsp-media.c:
8114 media: init debug category before starting thread
8116 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8118 * gst/rtsp-server/rtsp-auth.c:
8119 auth: add realm to make it more spec compliant
8121 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8123 * gst/rtsp-server/rtsp-server.c:
8124 * gst/rtsp-server/rtsp-server.h:
8127 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8129 * examples/test-video.c:
8130 example: improve example docs a little
8132 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8134 * gst/rtsp-server/rtsp-server.c:
8135 server: ensure the watch has a ref to the server
8137 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8139 * gst/rtsp-server/rtsp-server.c:
8140 server: simpify channel function
8142 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8144 * gst/rtsp-server/rtsp-server.c:
8145 * gst/rtsp-server/rtsp-server.h:
8146 server: simplify management of channel and source
8147 We don't need to keep around the channel and source objects. Let the mainloop
8148 and the source manage the source and channel respectively.
8150 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8156 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8159 * tests/Makefile.am:
8160 * tests/test-cleanup.c:
8161 tests: add tests directory and cleanup test
8163 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8165 * gst/rtsp-server/rtsp-media-factory-uri.c:
8166 * gst/rtsp-server/rtsp-media-factory.c:
8167 * gst/rtsp-server/rtsp-media-mapping.c:
8168 * gst/rtsp-server/rtsp-media.c:
8169 * gst/rtsp-server/rtsp-session-pool.c:
8170 * gst/rtsp-server/rtsp-session.c:
8171 server: improve debugging in various objects
8173 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8175 * gst/rtsp-server/rtsp-server.c:
8176 server: chain up to the parent finalize
8178 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
8180 * bindings/python/rtspserver-types.defs:
8181 * bindings/python/rtspserver.defs:
8182 * bindings/python/rtspserver.override:
8183 * bindings/python/test.py:
8184 gst-rtsp-server: update python bindings
8186 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8188 * gst/rtsp-server/rtsp-client.c:
8189 client: use the response from the clientstate
8190 Create the response object only once and store in the client state.
8191 Make all methods use the state response,
8193 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8195 * gst/rtsp-server/rtsp-server.c:
8196 server: use signal to keep track of clients
8197 Keep track of all the clients that the server creates and remove them when they
8198 fire the 'closed' signal.
8200 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8202 * gst/rtsp-server/rtsp-client.c:
8203 * gst/rtsp-server/rtsp-client.h:
8204 client: emit signal when closing
8206 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8208 * examples/.gitignore:
8209 * examples/Makefile.am:
8210 * examples/test-auth.c:
8211 * examples/test-video.c:
8212 * gst/rtsp-server/rtsp-auth.c:
8213 * gst/rtsp-server/rtsp-auth.h:
8214 * gst/rtsp-server/rtsp-client.c:
8215 * gst/rtsp-server/rtsp-media-factory.c:
8216 * gst/rtsp-server/rtsp-media.c:
8217 * gst/rtsp-server/rtsp-media.h:
8218 * gst/rtsp-server/rtsp-session-pool.h:
8219 * gst/rtsp-server/rtsp-session.h:
8220 media: enable per factory authorisations
8221 Allow for adding a GstRTSPAuth on the factory and media level and check
8222 permissions when accessing the factory.
8223 Add hints to the auth methods for future more fine grained authorisation.
8224 Add example application for per factory authentication.
8226 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8228 * gst/rtsp-server/rtsp-auth.c:
8229 * gst/rtsp-server/rtsp-auth.h:
8230 * gst/rtsp-server/rtsp-client.c:
8231 * gst/rtsp-server/rtsp-client.h:
8232 * gst/rtsp-server/rtsp-params.c:
8233 * gst/rtsp-server/rtsp-params.h:
8234 rtsp-server: Pass ClientState structure arround
8235 Pass the collected information for the ongoing request in a GstRTSPClientState
8236 structure that we can then pass around to simplify the method arguments. This
8237 will also be handy when we implement logging functionality.
8239 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8241 * gst/rtsp-server/rtsp-media-factory.c:
8242 * gst/rtsp-server/rtsp-media-factory.h:
8243 media-factory: add methods to configure authorisation
8245 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8247 * gst/rtsp-server/rtsp-client.c:
8248 client: unref auth in finalize
8250 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8252 * gst/rtsp-server/rtsp-server.c:
8253 server: unref auth in finalize
8255 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8257 * docs/libs/gst-rtsp-server-docs.sgml:
8258 * docs/libs/gst-rtsp-server-sections.txt:
8259 * docs/libs/gst-rtsp-server.types:
8262 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8264 * gst/rtsp-server/rtsp-server.c:
8265 * gst/rtsp-server/rtsp-server.h:
8266 server: separate create and accept
8267 Create separate create and accept methods so that subclasses can create custom
8269 Configure the server in the client object and prepare for keeping track of
8272 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8274 * gst/rtsp-server/rtsp-client.c:
8275 * gst/rtsp-server/rtsp-client.h:
8276 client: add support for setting the server.
8277 Add support for keeping a ref to the server that started this client
8280 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8282 * gst/rtsp-server/rtsp-auth.c:
8283 auth: fix memleak and add some docs
8284 Fix a memleak of the basic auth token.
8285 Add docs for the helper function
8287 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8289 * gst/rtsp-server/rtsp-auth.c:
8290 * gst/rtsp-server/rtsp-auth.h:
8291 * gst/rtsp-server/rtsp-client.c:
8292 client: delegate setup of auth to the manager
8293 Delegate the configuration of the authentication tokens to the manager object
8296 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8298 * examples/test-video.c:
8299 * gst/rtsp-server/Makefile.am:
8300 * gst/rtsp-server/rtsp-auth.c:
8301 * gst/rtsp-server/rtsp-auth.h:
8302 * gst/rtsp-server/rtsp-client.c:
8303 * gst/rtsp-server/rtsp-client.h:
8304 * gst/rtsp-server/rtsp-server.c:
8305 * gst/rtsp-server/rtsp-server.h:
8306 auth: add authentication object
8307 Add an object that can check the authorization of requests.
8308 Implement basic authentication.
8309 Add example authentication to test-video
8311 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8313 * gst/rtsp-server/rtsp-server.c:
8314 * gst/rtsp-server/rtsp-server.h:
8315 server: move includes back
8316 the includes are needed for sockaddr_in.
8318 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8320 * gst/rtsp-server/rtsp-client.c:
8321 * gst/rtsp-server/rtsp-client.h:
8322 * gst/rtsp-server/rtsp-server.c:
8323 * gst/rtsp-server/rtsp-server.h:
8324 rtsp: move network includes where they are needed
8326 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
8328 * gst/rtsp-server/rtsp-media.h:
8329 rtsp-media.h: Minor corrections in comments.
8332 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
8335 Automatic update of common submodule
8336 From e572c87 to f94d739
8338 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8342 * docs/libs/.gitignore:
8343 * examples/.gitignore:
8344 * gst/rtsp-server/.gitignore:
8347 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8349 * docs/libs/Makefile.am:
8350 docs: We don't build ps/pdf for API reference docs
8352 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8355 Automatic update of common submodule
8356 From ccbaa85 to e572c87
8358 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8361 Automatic update of common submodule
8362 From 46445ad to ccbaa85
8364 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8366 * gst/rtsp-server/Makefile.am:
8367 * gst/rtsp-server/rtsp-funnel.c:
8368 * gst/rtsp-server/rtsp-funnel.h:
8369 * gst/rtsp-server/rtsp-media.c:
8370 funnel: rename fsfunnel to rtspfunnel
8371 Rename the funnel to avoid conflicts with the farsight one.
8373 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8375 * gst/rtsp-server/Makefile.am:
8376 * gst/rtsp-server/fs-funnel.c:
8377 * gst/rtsp-server/fs-funnel.h:
8378 * gst/rtsp-server/rtsp-media.c:
8379 rtsp-media: add and use fsfunnel
8380 Add a copy of fsfunnel to the build because input-selector removed the (broken)
8381 select-all property that we need.
8383 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8385 * gst/rtsp-server/Makefile.am:
8386 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
8387 Use PKG_CONFIG_PATH specified at configure time (if any) as well
8388 for the g-ir-compiler, rather than just assuming the env var has
8391 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8398 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
8400 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8403 * gst/rtsp-server/Makefile.am:
8404 gobject-introspection: fix g-i build for uninstalled setup
8405 Requires gst-plugins-base git (> 0.10.31.2).
8407 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8409 * examples/test-uri.c:
8410 examples: add some more options and comments
8412 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8414 * gst/rtsp-server/rtsp-media-factory-uri.c:
8415 factory-uri: use right property type
8417 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8419 * gst/rtsp-server/rtsp-media-factory-uri.c:
8420 factory-uri: attempt to configure buffer-lists
8421 Attempt to configure buffer lists in the payloader for improved performance.
8423 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8425 * gst/rtsp-server/rtsp-media.c:
8426 media: attempt to configure bigger UDP buffers
8427 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
8428 send buffers with high bitrate streams.
8430 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
8432 * gst/rtsp-server/rtsp-client.c:
8433 client: use the socket length from getsockname
8434 Use the length returned by getsockname to perform the getnameinfo call because
8435 the size can depend on the socket type and platform.
8438 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8440 * docs/libs/gst-rtsp-server-docs.sgml:
8441 * docs/libs/gst-rtsp-server-sections.txt:
8442 docs: add uri factory to the docs
8444 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8446 * gst/rtsp-server/rtsp-client.c:
8447 * gst/rtsp-server/rtsp-media.h:
8450 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8452 * gst/rtsp-server/rtsp-client.c:
8453 * gst/rtsp-server/rtsp-media.c:
8454 * gst/rtsp-server/rtsp-media.h:
8455 * gst/rtsp-server/rtsp-session.c:
8456 * gst/rtsp-server/rtsp-session.h:
8457 rtsp-server: add support for buffer lists
8458 Add support for sending bufferlists received from appsink.
8461 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8463 * gst/rtsp-server/rtsp-client.c:
8464 * gst/rtsp-server/rtsp-media.c:
8465 * gst/rtsp-server/rtsp-media.h:
8466 * gst/rtsp-server/rtsp-sdp.c:
8467 media: make method to retrieve the play range
8468 Make a method to retrieve the playback range so that we can conditionally create
8469 a different range for the SDP and the PLAY requests.
8471 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8473 * gst/rtsp-server/rtsp-media.c:
8474 * gst/rtsp-server/rtsp-media.h:
8475 media: add signal to notify of state changes
8477 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8479 * gst/rtsp-server/rtsp-client.h:
8480 client: cleanup headers
8482 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8484 * gst/rtsp-server/rtsp-client.c:
8487 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8489 * gst/rtsp-server/rtsp-media-factory-uri.c:
8490 * gst/rtsp-server/rtsp-media-factory-uri.h:
8491 factory-uri: add support for gstpay
8492 Add an option to prefer gstpay over decoder + raw payloader.
8494 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8496 * gst/rtsp-server/rtsp-media-factory-uri.c:
8497 * gst/rtsp-server/rtsp-media-factory-uri.h:
8498 factory-uri: rework the autoplugger.
8499 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
8502 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8504 * gst/rtsp-server/rtsp-media-factory-uri.c:
8505 factory-uri: use better factory filter
8506 Make better payloader filter based on autoplug rank and RTP use case.
8508 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8511 Automatic update of common submodule
8512 From 169462a to 46445ad
8514 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8516 * gst/rtsp-server/rtsp-server.c:
8517 server: set SO_REUSEADDR before bind
8518 Set the SO_REUSEADDR _before_ bind() to make it actually work.
8520 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8522 * gst/rtsp-server/rtsp-media.c:
8523 * gst/rtsp-server/rtsp-media.h:
8524 media: emit prepared signal when prepared
8525 Make a 'prepared' signal and emit it when we successfully prepared the element.
8526 This signal can be used to configure the media object after it has been prepared
8529 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
8532 Automatic update of common submodule
8533 From 011bcc8 to 169462a
8535 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
8537 python an optional dependency
8538 * configure.ac: Move up valgrind and g-i checks. Make the python
8539 dependency optional, as it was before.
8541 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8543 Merge branch 'master' into 0.11
8548 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8550 * gst/rtsp-server/rtsp-media.c:
8551 media: update range when active clients changed
8552 When we changed the number of active clients, update the current range
8553 information because we want the second client connecting to a shared resource
8554 continue from where the stream currently.
8556 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8558 * gst/rtsp-server/rtsp-media-factory-uri.c:
8559 * gst/rtsp-server/rtsp-media-factory-uri.h:
8560 factory-uri: add colorspace and fix pt
8561 Rework the way we pass data to the autoplugger.
8562 When we have raw caps, plug a converter element to make pluggin to raw
8563 payloaders more successful.
8564 Make sure all dynamically plugged payloaders have a unique payload types.
8566 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8568 * examples/Makefile.am:
8569 * examples/test-uri.c:
8570 example: add example of the uri factory
8572 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8574 * gst/rtsp-server/Makefile.am:
8575 * gst/rtsp-server/rtsp-media-factory-uri.c:
8576 * gst/rtsp-server/rtsp-media-factory-uri.h:
8577 * gst/rtsp-server/rtsp-server.h:
8578 factory-uri: add a factory to stream any URI
8579 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
8582 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8584 * gst/rtsp-server/rtsp-media.c:
8585 * gst/rtsp-server/rtsp-media.h:
8586 media: ignore spurious ASYNC_DONE messages
8587 When we are dynamically adding pads, the addition of the udpsrc elements will
8588 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
8589 the real ASYNC_DONE when everything is prerolled.
8591 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8593 * gst/rtsp-server/rtsp-media-factory.c:
8594 * gst/rtsp-server/rtsp-media-factory.h:
8595 media-factory: make lock macro
8597 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
8599 * gst/rtsp-server/rtsp-client.c:
8600 rtsp-server: Remove unused variable and dead assignment
8602 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
8604 * examples/test-launch.c:
8605 * examples/test-mp4.c:
8606 * examples/test-ogg.c:
8607 * examples/test-readme.c:
8608 * examples/test-sdp.c:
8609 * examples/test-video.c:
8610 examples: Run gst-indent
8612 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
8614 * gst/rtsp-server/rtsp-client.c:
8615 * gst/rtsp-server/rtsp-media-factory.c:
8616 * gst/rtsp-server/rtsp-media-mapping.c:
8617 * gst/rtsp-server/rtsp-media.c:
8618 * gst/rtsp-server/rtsp-params.c:
8619 * gst/rtsp-server/rtsp-sdp.c:
8620 * gst/rtsp-server/rtsp-server.c:
8621 * gst/rtsp-server/rtsp-session-pool.c:
8622 * gst/rtsp-server/rtsp-session.c:
8623 rtsp-server: Run gst-indent
8624 Since it wasn't using the upstream common previously, there was no
8625 indentation check before commiting.
8627 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
8629 * gst/rtsp-server/rtsp-media-mapping.h:
8630 * gst/rtsp-server/rtsp-media.c:
8631 * gst/rtsp-server/rtsp-media.h:
8632 * gst/rtsp-server/rtsp-sdp.c:
8633 * gst/rtsp-server/rtsp-session-pool.h:
8634 * gst/rtsp-server/rtsp-session.c:
8635 * gst/rtsp-server/rtsp-session.h:
8636 rtsp-server: Some more doc fixups
8638 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8641 Makefile: Add cruft-cleaning support
8643 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8648 * docs/libs/Makefile.am:
8649 * docs/libs/gst-rtsp-server-docs.sgml:
8650 * docs/libs/gst-rtsp-server-sections.txt:
8651 * docs/libs/gst-rtsp-server.types:
8652 * docs/version.entities.in:
8653 docs: Add gtk-doc build system
8655 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8657 * gst/rtsp-server/Makefile.am:
8658 Makefile.am: Use standard GIR make behaviour
8660 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8664 autogen/configure: Bring more in sync to standard gst module behaviour
8666 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8668 * gst/rtsp-server/rtsp-media.c:
8669 media: warn and fail when gstrtpbin is not found
8671 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8674 configure: open 0.11 branch
8676 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
8680 Add common submodule
8682 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
8685 * common/Makefile.am:
8686 * common/c-to-xml.py:
8688 * common/coverage/coverage-report-entry.pl:
8689 * common/coverage/coverage-report.pl:
8690 * common/coverage/coverage-report.xsl:
8691 * common/coverage/lcov.mak:
8692 * common/gettext.patch:
8693 * common/glib-gen.mak:
8694 * common/gst-autogen.sh:
8695 * common/gst-xmlinspect.py:
8697 * common/gstdoc-scangobj:
8698 * common/gtk-doc-plugins.mak:
8699 * common/gtk-doc.mak:
8700 * common/m4/.gitignore:
8701 * common/m4/Makefile.am:
8703 * common/m4/as-ac-expand.m4:
8704 * common/m4/as-auto-alt.m4:
8705 * common/m4/as-compiler-flag.m4:
8706 * common/m4/as-compiler.m4:
8707 * common/m4/as-docbook.m4:
8708 * common/m4/as-libtool-tags.m4:
8709 * common/m4/as-libtool.m4:
8710 * common/m4/as-python.m4:
8711 * common/m4/as-scrub-include.m4:
8712 * common/m4/as-version.m4:
8713 * common/m4/ax_create_stdint_h.m4:
8714 * common/m4/check.m4:
8715 * common/m4/glib-gettext.m4:
8716 * common/m4/gst-arch.m4:
8717 * common/m4/gst-args.m4:
8718 * common/m4/gst-check.m4:
8719 * common/m4/gst-debuginfo.m4:
8720 * common/m4/gst-default.m4:
8721 * common/m4/gst-doc.m4:
8722 * common/m4/gst-error.m4:
8723 * common/m4/gst-feature.m4:
8724 * common/m4/gst-function.m4:
8725 * common/m4/gst-gettext.m4:
8726 * common/m4/gst-glib2.m4:
8727 * common/m4/gst-libxml2.m4:
8728 * common/m4/gst-plugindir.m4:
8729 * common/m4/gst-valgrind.m4:
8730 * common/m4/gtk-doc.m4:
8731 * common/m4/introspection.m4:
8733 * common/mangle-tmpl.py:
8734 * common/plugins.xsl:
8736 * common/release.mak:
8737 * common/scangobj-merge.py:
8738 * common/upload.mak:
8739 common: Remove static version
8741 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
8743 * common/m4/introspection.m4:
8744 Update introspection.m4 to match usage
8746 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8750 Remove old stuff from the README
8752 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8757 === release 0.10.7 ===
8759 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8764 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8766 * examples/test-ogg.c:
8767 test-ogg: remove parsers
8768 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
8769 buffers with timestamps. Using the parsers also seems to break things.
8771 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8773 * bindings/vala/gst-rtsp-server-0.10.vapi:
8774 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8775 Updated Vala bindings
8777 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8779 * common/m4/introspection.m4:
8781 * gst/rtsp-server/Makefile.am:
8782 Added initial gobject-introspection support
8784 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8786 * gst/rtsp-server/rtsp-media-factory.c:
8787 media-factory: don't use host for shared hash key
8788 When we generate the key to share made between connections, don't include the
8789 host used to connect so that we can share media even if between clients that
8790 connected with localhost and ones with the ip address.
8792 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8794 * bindings/vala/Makefile.am:
8795 build: fix distcheck
8797 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8799 * bindings/vala/gst-rtsp-server-0.10.vapi:
8800 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8801 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8802 Update Vala bindings
8804 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8806 * bindings/vala/Makefile.am:
8808 Fix configure checks and installation location for Vala bindings
8811 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8816 === release 0.10.6 ===
8818 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8821 configure: release 0.10.6
8823 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8825 * gst/rtsp-server/rtsp-media.c:
8826 media: help the compiler a little
8828 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8830 * gst/rtsp-server/rtsp-media.c:
8831 * gst/rtsp-server/rtsp-media.h:
8832 * gst/rtsp-server/rtsp-session.c:
8833 media: cleanup media transport before freeing
8834 Cleanup the media transport data before freeing. In particular, remove the qdata
8835 from the rtpsource object.
8837 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8839 * gst/rtsp-server/rtsp-media-factory.c:
8840 * gst/rtsp-server/rtsp-media-factory.h:
8841 * gst/rtsp-server/rtsp-media.c:
8842 * gst/rtsp-server/rtsp-media.h:
8843 media-factory: add eos-shutdown property
8844 Add an eos-shutdown property that will send an EOS to the pipeline before
8845 shutting it down. This allows for nice cleanup in case of a muxer.
8848 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8850 * gst/rtsp-server/rtsp-media.c:
8851 * gst/rtsp-server/rtsp-media.h:
8852 media: use multiudpsink send-duplicates when we can
8853 If we have a new enough multiudpsink with the send-duplicates property, use this
8854 instead of doing our own filtering. Our custom filtering code should eventually
8855 be removed when we can depend on a released -good.
8857 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8859 * gst/rtsp-server/rtsp-media.c:
8860 media: don't leak destinations
8861 Refactor and cleanup the destinations array when the stream is destroyed.
8863 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8865 * gst/rtsp-server/rtsp-media.c:
8866 * gst/rtsp-server/rtsp-media.h:
8867 media: don't add udp addresses multiple times
8868 Keep track of the udp addresses we added to udpsink and never add the same udp
8869 destination twice. This avoids duplicate packets when using multicast.
8871 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8873 * gst/rtsp-server/rtsp-server.c:
8874 server: disable use of SO_LINGER
8875 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
8876 server close()s the connection.
8878 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8880 * gst/rtsp-server/rtsp-server.c:
8881 server: use 5 second linger period in SO_LINGER
8882 Wait 5 seconds before clearing the send buffers and reseting the connection with
8883 the client when we do a close. This should be enough time to get the message to
8887 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
8889 * gst/rtsp-server/rtsp-server.c:
8890 server: use SO_LINGER
8891 SO_LINGER on the socket will make sure that any pending data on the socket is
8892 flushed ASAP and that the socket connection is reset. This makes sure that the
8893 socket can be reused immediately.
8896 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8899 README: add blurb about shared media factories
8901 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
8903 * gst/rtsp-server/rtsp-media.c:
8904 Add stdlib.h for atoi()
8906 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8908 * bindings/python/Makefile.am:
8909 * bindings/vala/Makefile.am:
8910 build: distcheck fixes
8911 Fix 'make distcheck', somewhat (it still fails because it tries to
8912 install files into /usr/share/vala/vapi/ irrespective of the
8915 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8918 configure: bump core/base requirements to released version
8919 Makes things less confusing for people.
8921 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8924 configure: fail if GStreamer core/base requirements are not met
8926 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8928 * gst/rtsp-server/rtsp-client.c:
8929 client: improve client cleanups
8930 Make sure the session does not timeout when using TCP. We need to do this
8931 because quicktime player does not send RTCP for some reason in tunneled
8933 Refactor some cleanup code.
8936 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8938 * gst/rtsp-server/rtsp-session.c:
8939 * gst/rtsp-server/rtsp-session.h:
8940 session: add support for prevent session timeouts
8941 Add an atomix counter to prevent session timeouts when we are, for example,
8944 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8946 * gst/rtsp-server/rtsp-client.c:
8947 client: fix unlink on session timeouts
8948 When our session times out, make sure we unlink all streams in this
8950 Remove the tunnelid when closing the connection.
8952 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8954 * gst/rtsp-server/rtsp-session.c:
8955 session: small cleanups
8957 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8959 * gst/rtsp-server/rtsp-client.c:
8960 client: handle lost_tunnel callbacks
8961 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
8962 hashtable so that we can reuse it for when the client reopens the POST
8964 Close the connection after a TEARDOWN.
8965 Make sure or watchid is cleared when the watch is removed.
8968 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8970 * gst/rtsp-server/rtsp-client.c:
8971 * gst/rtsp-server/rtsp-media.c:
8972 * gst/rtsp-server/rtsp-sdp.c:
8973 rtsp-server: add more support for multicast
8975 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8978 * gst/rtsp-server/rtsp-media.c:
8979 * gst/rtsp-server/rtsp-media.h:
8980 media: allow configuration of allowed lower transport
8982 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8984 * gst/rtsp-server/rtsp-client.h:
8985 * gst/rtsp-server/rtsp-media.c:
8986 * gst/rtsp-server/rtsp-media.h:
8987 * gst/rtsp-server/rtsp-sdp.c:
8988 * gst/rtsp-server/rtsp-sdp.h:
8989 * gst/rtsp-server/rtsp-server.c:
8990 rtsp: keep track of server ip and ipv6
8991 Keep track of how the client connected to the server and setup the udp ports
8992 with the same protocol.
8993 Copy the server ip address in the SDP so that clients can send RTCP back to
8996 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8998 * gst/rtsp-server/rtsp-session.c:
9001 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9003 * gst/rtsp-server/rtsp-client.c:
9004 client: use right size for malloc
9006 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9008 * gst/rtsp-server/rtsp-server.c:
9009 server: comment ipv6 server listening address
9011 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9013 * gst/rtsp-server/rtsp-media.c:
9014 media: allow for ipv6 sockets
9016 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9018 * gst/rtsp-server/rtsp-server.c:
9019 * gst/rtsp-server/rtsp-server.h:
9020 server: rework server part
9021 Allow setting a bind address, make sure we can deal with ipv6.
9022 Remove the port property and change with the service property.
9024 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9026 * gst/rtsp-server/rtsp-media.h:
9027 media: update comments a little
9029 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9031 * gst/rtsp-server/rtsp-client.c:
9032 client: make content-base better
9033 Use the URI formatting functions to make a content-base. Also make sure that
9034 there is a trailing / at the end.
9036 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9038 * gst/rtsp-server/rtsp-client.c:
9039 client: guard against invalid paths
9041 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9043 * examples/test-video.c:
9044 test: catch server bind errors
9046 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
9048 * gst/rtsp-server/rtsp-media.c:
9049 rtspmedia: emit "unprepared" if _prepare fails.
9050 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
9051 media object is removed from its factory's cache.
9053 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9055 * gst/rtsp-server/rtsp-media.c:
9056 media: collect media position when seek completes
9058 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
9060 * gst/rtsp-server/rtsp-client.c:
9061 client: call unlink_streams in client finalize
9064 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9066 * gst/rtsp-server/rtsp-media.c:
9067 media: limit the time to wait to something huge
9068 Avoid waiting forever but limit the timeout to 20 seconds.
9070 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9072 * gst/rtsp-server/rtsp-sdp.c:
9073 sdp: reindent and check for prepared status
9075 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9077 * gst/rtsp-server/rtsp-media.c:
9078 * gst/rtsp-server/rtsp-media.h:
9079 * gst/rtsp-server/rtsp-session.c:
9080 media: avoid doing _get_state() for state changes
9081 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
9082 until the media is prerolled or in error. This avoids doing a blocking call of
9083 gst_element_get_state() that can cause lockups when there is an error.
9086 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9088 * gst/rtsp-server/rtsp-media.c:
9091 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9093 * gst/rtsp-server/rtsp-media-factory.c:
9094 media-factory: better error handling
9095 Improve the error handling a bit.
9097 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9099 * gst/rtsp-server/rtsp-client.c:
9100 client: rework transport parsing
9101 Rework the transport parsing code so that we can ignore transports we don't
9102 support instead of just picking the first one we can parse.
9103 Configure a (for now hardcoded) destination for multicast transports.
9105 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9107 * gst/rtsp-server/rtsp-media.c:
9108 media: set multicast sink parameters
9109 Disable loop and automatic multicast join on the udpsink elements.
9110 Add some more debug info.
9111 Reset some state variables in the right place.
9112 Use the right port numbers for multicast.
9114 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9116 * gst/rtsp-server/rtsp-session.c:
9117 session: handle transport setup correctly
9118 Handle UDP, MCAST and TCP transport negotiation more correctly.
9119 Store the server session SSRC in the transport.
9121 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9123 * gst/rtsp-server/rtsp-client.c:
9124 rtsp-client: implement error_full
9125 Implement error_full to avoid some segfaults when the rtspconnection calls it.
9128 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9131 * gst/rtsp-server/rtsp-client.c:
9132 * gst/rtsp-server/rtsp-server.c:
9133 docs: update docs and comments
9135 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
9137 * gst/rtsp-server/rtsp-sdp.c:
9138 sdp: make server work better when behind a proxy
9140 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9142 * gst/rtsp-server/rtsp-client.c:
9143 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
9145 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9147 * gst/rtsp-server/rtsp-client.c:
9148 * gst/rtsp-server/rtsp-media-factory.c:
9149 * gst/rtsp-server/rtsp-media-mapping.c:
9150 * gst/rtsp-server/rtsp-media.c:
9151 * gst/rtsp-server/rtsp-server.c:
9152 * gst/rtsp-server/rtsp-session-pool.c:
9153 * gst/rtsp-server/rtsp-session.c:
9154 Use GStreamer's debugging subsystem
9156 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9158 * gst/rtsp-server/rtsp-media-factory.c:
9159 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
9161 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9166 === release 0.10.5 ===
9168 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9173 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9176 configure: bump required versions
9178 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
9180 * gst/rtsp-server/rtsp-client.c:
9181 client: call weak-unref on client->sessions from finalize
9184 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9186 * gst/rtsp-server/rtsp-media.c:
9187 media: Fixed crasher where caps got unref'ed too often
9189 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9192 * pkgconfig/.gitignore:
9193 * pkgconfig/Makefile.am:
9194 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
9195 Added pkg-config file to use gst-rtsp-server uninstalled
9197 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9199 * gst/rtsp-server/rtsp-media.c:
9200 media: add some docs
9202 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
9204 * gst/rtsp-server/rtsp-client.c:
9205 rtsp: Use gst_rtsp_watch_send_message().
9206 Use gst_rtsp_watch_send_message() since the old API which used
9207 gst_rtsp_watch_queue_message() has been deprecated.
9209 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9214 === release 0.10.4 ===
9216 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9221 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9223 * gst/rtsp-server/rtsp-client.c:
9224 * gst/rtsp-server/rtsp-session.c:
9225 * gst/rtsp-server/rtsp-session.h:
9226 rtsp: allocate channels in TCP mode
9227 When the client does not provide us with channels in TCP mode, allocate channels
9230 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9232 * gst/rtsp-server/rtsp-client.c:
9233 client: don't crash when tunnelid is missing
9234 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
9235 don't crash but return an error response to the client.
9238 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9240 * bindings/vala/gst-rtsp-server-0.10.vapi:
9241 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9242 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9243 bindings: update vala bindings with new method
9245 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9247 * gst/rtsp-server/rtsp-session-pool.c:
9248 * gst/rtsp-server/rtsp-session-pool.h:
9249 sessionpool: add function to filter sessions
9250 Add generic function to retrieve/remove sessions.
9252 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9255 configure: bump core/base requirements to release
9257 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9259 * gst/rtsp-server/rtsp-media.c:
9260 media: fix indentation
9262 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9264 * gst/rtsp-server/rtsp-media.c:
9265 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
9267 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9269 * gst/rtsp-server/rtsp-media.c:
9270 set state and remove elements of media in for loop
9272 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
9274 * bindings/vala/gst-rtsp-server-0.10.vapi:
9275 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9276 Added gst_rtsp_media_remove_elements function to Vala bindings
9278 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
9280 * gst/rtsp-server/rtsp-media.c:
9281 * gst/rtsp-server/rtsp-media.h:
9282 Added gst_rtsp_media_remove_elements function
9284 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
9286 * gst/rtsp-server/rtsp-media.c:
9287 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
9289 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9291 * bindings/vala/gst-rtsp-server-0.10.vapi:
9292 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9293 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9294 Updated Vala bindings
9296 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9298 * gst/rtsp-server/rtsp-media.c:
9299 * gst/rtsp-server/rtsp-media.h:
9300 Added vmethod unprepare to GstRTSPMedia
9301 The default implementation sets the state of the pipeline to GST_STATE_NULL
9303 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9305 * gst/rtsp-server/rtsp-media-factory.c:
9306 * gst/rtsp-server/rtsp-media-factory.h:
9307 Made collect_streams function public
9309 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9311 * gst/rtsp-server/rtsp-media-factory.c:
9312 * gst/rtsp-server/rtsp-media-factory.h:
9313 * gst/rtsp-server/rtsp-media.c:
9314 Added vmethod create_pipeline to GstRTSPMediaFactory
9315 The pipeline is created in this method and the GstRTSPMedia's element is added to it
9317 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9319 * gst/rtsp-server/rtsp-client.c:
9320 client: use g_source_destroy()
9321 We need to use g_source_destroy() because we might have added the source to a
9322 different main context than the default one.
9324 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9326 * gst/rtsp-server/Makefile.am:
9327 * gst/rtsp-server/rtsp-client.c:
9328 * gst/rtsp-server/rtsp-params.c:
9329 * gst/rtsp-server/rtsp-params.h:
9330 rtsp: prepare for handling GET/SET_PARAMETER
9331 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
9333 Fix return codes of handlers.
9335 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9337 * gst/rtsp-server/rtsp-media.c:
9338 media: don't leak session pads
9340 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9342 * gst/rtsp-server/rtsp-media.c:
9343 media: clean up the messages a bit
9345 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9347 * gst/rtsp-server/rtsp-sdp.c:
9348 sdp: warn and skip streams without media
9350 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9352 * bindings/vala/gst-rtsp-server-0.10.vapi:
9353 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9354 vala: Fixed typo in header file of RTSPMediaStream
9356 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9358 * gst/rtsp-server/rtsp-media.c:
9361 Make dumping RTCP stats configurable
9363 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9365 * gst/rtsp-server/rtsp-media.c:
9366 media: be less verbose and leak less
9368 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9370 * gst/rtsp-server/rtsp-media.c:
9371 media: don't leak the destination address
9373 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9375 * gst/rtsp-server/rtsp-client.c:
9376 * gst/rtsp-server/rtsp-media.c:
9377 * gst/rtsp-server/rtsp-media.h:
9378 * gst/rtsp-server/rtsp-session.c:
9379 * gst/rtsp-server/rtsp-session.h:
9380 rtsp: use RTCP to keep the session alive
9381 Use the RTCP rtcp-from stats field to find the associated session and use this
9382 to keep the session alive.
9384 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9386 * gst/rtsp-server/rtsp-session.c:
9387 session: add 5sec to the real session timeout
9388 Allow the session to live 5sec longer before really timing out. This should give
9389 clients some extra time to keep the session active.
9391 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9393 * gst/rtsp-server/rtsp-client.c:
9394 client: replay OK to GET/SET_PARAMETER
9395 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
9396 so that we return OK for those requests.
9398 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9400 * gst/rtsp-server/rtsp-media.c:
9401 * gst/rtsp-server/rtsp-media.h:
9402 media: keep track of active transports
9403 Keep track of which transport is active to avoid closing the connection too
9405 Remove the destination transport also when going to NULL.
9406 Print some stats about the SDES and other RTCP messages we receive from the
9409 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9411 * examples/.gitignore:
9412 * examples/Makefile.am:
9413 * examples/test-sdp.c:
9414 example: add SDP relay example
9416 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9418 * gst/rtsp-server/rtsp-media.c:
9419 media: also count active TCP connections
9421 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9423 * gst/rtsp-server/rtsp-media-factory.c:
9424 * gst/rtsp-server/rtsp-media.c:
9425 * gst/rtsp-server/rtsp-media.h:
9426 rtsp: add support for dynamic elements
9427 Add support for dynamic elements.
9428 Don't set live pipelines back to paused.
9430 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9432 * gst/rtsp-server/rtsp-sdp.c:
9433 sdp: don't add encoding name when absent in caps
9435 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9437 * gst/rtsp-server/rtsp-client.c:
9438 client: warn when we can't do RTP-Info
9440 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9442 * gst/rtsp-server/rtsp-media-factory.c:
9443 factory: factor out the stream construction
9445 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9447 * gst/rtsp-server/rtsp-client.c:
9448 client: only add RTP-Info when we have the info
9449 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
9452 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9457 === release 0.10.3 ===
9459 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9463 - Fixes a bug where it put the wrong verion in pkgconfig
9464 - Link RTP and RTCP sources
9466 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9468 * gst/rtsp-server/rtsp-media.c:
9469 * gst/rtsp-server/rtsp-media.h:
9470 media: link the RTP udpsrc to the session manager
9471 Link the RTP udpsrc and the appsrc to the session manager so that they don't
9472 shut down when the client sends a packet to open firewalls.
9474 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9476 * pkgconfig/gst-rtsp-server.pc.in:
9477 Don't use hard-coded version number in pkg-config file
9479 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9484 === release 0.10.2 ===
9486 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9491 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9494 * common/m4/.gitignore:
9495 * examples/.gitignore:
9496 * pkgconfig/.gitignore:
9497 add some .gitignore files
9499 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9501 * gst/rtsp-server/rtsp-media.c:
9502 media: seek to key frames
9504 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9506 * gst/rtsp-server/rtsp-media.c:
9507 media: emit the unprepared signal by id
9508 Emit the unprepared signal by id instead of name and set the media as
9511 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9513 * gst/rtsp-server/rtsp-media.c:
9514 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
9516 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9518 * gst/rtsp-server/rtsp-server.c:
9519 Added finalize function to GstRTPSPServer to unref session pool and media mapping
9521 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9523 * bindings/vala/gst-rtsp-server-0.10.vapi:
9524 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9525 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9526 Updated vala bindings
9528 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9530 * gst/rtsp-server/Makefile.am:
9531 * gst/rtsp-server/rtsp-client.c:
9532 * gst/rtsp-server/rtsp-media.c:
9533 server: use appsink and appsrc with the API
9534 Use the appsink/appsrc API instead of the signals for higher
9537 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9539 * examples/test-ogg.c:
9540 tests: set the payload type correctly
9542 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9544 * gst/rtsp-server/rtsp-media-factory.c:
9545 factory: connect to the unprepare signal
9546 Connect to the unprepare signal for non-reusable media so that we can remove
9547 them from the cache.
9549 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9551 * gst/rtsp-server/rtsp-media.c:
9552 * gst/rtsp-server/rtsp-media.h:
9553 media: add signal to notify of unprepare
9555 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9557 * gst/rtsp-server/rtsp-media.c:
9558 * gst/rtsp-server/rtsp-media.h:
9559 media: more work on making the media shared
9560 Add a reusable flag to medias, indicating that they can be reused after a state
9564 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9566 * examples/test-readme.c:
9567 examples: mark the example as shared for testing
9569 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9571 * gst/rtsp-server/rtsp-media.c:
9572 * gst/rtsp-server/rtsp-media.h:
9573 client: support shared media
9574 Always perform the state actions even if the target state of the pipeline is
9575 already correct, we still want to add/remove the transports when we are dealing
9577 Keep a counter of the number of active transports for a media so that we can use
9578 this to perform a state change when needed.
9579 Perform a state change of the pipeline only when the first transport was added
9580 or when there are no active transports.
9582 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9584 * gst/rtsp-server/rtsp-client.c:
9585 client: fix refcounting crasher
9586 Don't need to remove the weak refs in the finalize methods, they are already
9587 removed in the dispose.
9588 Don't register the callback with a DestroyNofity.
9590 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9592 * gst/rtsp-server/rtsp-client.c:
9593 Fix rtsp client refcount management in TCP mode.
9594 Don't unref a client ref we never had. Fixes an unref
9595 of an already-free client object after a client
9596 teardown request for me.
9598 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9600 * gst/rtsp-server/rtsp-session.c:
9601 docs: fix typo in API docs
9603 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9605 * gst/rtsp-server/rtsp-media.c:
9607 Keep the udp sources in playing even if we go to paused. unlock the sources when
9609 Add some more debug info.
9610 Only seek when we need to.
9611 Keep track of the position when we go to paused.
9613 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9615 * gst/rtsp-server/rtsp-client.c:
9616 * gst/rtsp-server/rtsp-media.c:
9617 * gst/rtsp-server/rtsp-media.h:
9618 Add beginnings of seeking.
9619 Parse the Range header and perform a seek on the pipeline for the requested
9620 position. It's disabled currently until I figure out what's going wrong.
9622 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9624 * gst/rtsp-server/rtsp-client.c:
9625 allow pause requests for now.
9628 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9630 * gst/rtsp-server/rtsp-client.c:
9631 Remove weak ref on the session in teardown
9632 We need to remove our weakref from the session when we do a teardown because
9633 else we close the TCP connection prematurely.
9635 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9637 * gst/rtsp-server/rtsp-client.c:
9638 * gst/rtsp-server/rtsp-client.h:
9639 * gst/rtsp-server/rtsp-session-pool.c:
9640 Do some more session cleanup
9641 Make session timeout kill the TCP connection that currently watches the
9643 Remove the client timeout property.
9645 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9647 * gst/rtsp-server/rtsp-client.c:
9648 * gst/rtsp-server/rtsp-client.h:
9649 * gst/rtsp-server/rtsp-media.c:
9650 * gst/rtsp-server/rtsp-media.h:
9651 * gst/rtsp-server/rtsp-server.c:
9652 * gst/rtsp-server/rtsp-session.c:
9653 * gst/rtsp-server/rtsp-session.h:
9655 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
9658 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9660 * examples/Makefile.am:
9661 * examples/test-launch.c:
9662 Add example server that takes launch lines
9663 Add an example server that streams any -launch line.
9665 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9667 * examples/test-readme.c:
9668 * gst/rtsp-server/rtsp-client.c:
9669 * gst/rtsp-server/rtsp-media.c:
9670 * gst/rtsp-server/rtsp-media.h:
9671 Add support for live streams
9672 Add support for live streams and ranges
9673 Start on handling TCP data transfer.
9675 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9677 * gst/rtsp-server/rtsp-media.c:
9678 Free the pipeline before other things
9681 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9683 * gst/rtsp-server/rtsp-client.c:
9684 Only free the pending tunnel if there is one
9687 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9689 * gst/rtsp-server/rtsp-client.c:
9690 * gst/rtsp-server/rtsp-client.h:
9691 * gst/rtsp-server/rtsp-media.c:
9692 rtsp-server: Add support for tunneling
9693 Add support for tunneling over HTTP.
9694 Use new connection methods to retrieve the url.
9695 Dispatch messages based on the message type instead of blindly
9696 assuming it's always a request.
9697 Keep track of the watch id so that we can remove it later.
9698 Set the media pipeline to NULL before unreffing the pipeline.
9700 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9702 * gst/rtsp-server/rtsp-client.c:
9703 * gst/rtsp-server/rtsp-client.h:
9704 Fix for channel -> watch rename in gstreamer
9705 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
9707 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9709 * gst/rtsp-server/rtsp-client.c:
9710 * gst/rtsp-server/rtsp-client.h:
9712 Use the async RTSP channels instead of spawning a new thread for each client.
9713 If a sessionid is specified in a request, fail if we don't have the session.
9715 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9717 * gst/rtsp-server/rtsp-media.c:
9718 Add better debug info
9719 Add some better debug info.
9721 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9723 * examples/test-video.c:
9725 Add support for session timeouts in the example.
9727 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9729 * gst/rtsp-server/rtsp-session-pool.c:
9730 * gst/rtsp-server/rtsp-session-pool.h:
9731 Pass GTimeVal around for performance reasons
9732 Get the current time only once and pass it around so that sessions don't have to
9733 get the current time anymore.
9734 Add experimental support for a GSource that dispatches when the session needs to
9737 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9739 * gst/rtsp-server/rtsp-session.c:
9740 * gst/rtsp-server/rtsp-session.h:
9741 Add better support for session timeouts
9742 Add a method to request the number of milliseconds when a session will timeout.
9744 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9746 * gst/rtsp-server/rtsp-media.c:
9747 * gst/rtsp-server/rtsp-media.h:
9748 Add suport for RTP manager monitoring
9749 Add the first stage in monitoring the rtp manager.
9750 Make sure we don't update the state to something we don't want.
9752 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9754 * gst/rtsp-server/rtsp-client.c:
9755 Add support for session keepalive
9756 Get and update the session timeout for all requests. get the session as early as
9759 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9761 * gst/rtsp-server/rtsp-media-factory.h:
9762 * gst/rtsp-server/rtsp-media.c:
9763 * gst/rtsp-server/rtsp-media.h:
9764 Handle media bus messages
9765 Handle media bus messages in a custom mainloop and dispatch them to the
9766 RTSPMedia objects. Let the default implementation handle some common messages.
9768 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9770 * gst/rtsp-server/rtsp-client.c:
9771 * gst/rtsp-server/rtsp-session-pool.c:
9772 * gst/rtsp-server/rtsp-session.c:
9773 Some more session timeout handling
9774 Move the session header setting code to a central place so that we always add
9775 the timeout parameter too.
9776 Handle timeouts by running the session cleanup code.
9777 Stop media before cleaning up.
9779 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9781 * gst/rtsp-server/rtsp-client.c:
9782 * gst/rtsp-server/rtsp-client.h:
9783 Add timeout property
9784 Add a timeout property ot the client and make the other properties into GObject
9787 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9789 * gst/rtsp-server/rtsp-session-pool.c:
9790 Use getters and setters in property code
9791 Use the getters and setters for the timeout property instead of locking
9794 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9796 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
9798 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9800 * gst/rtsp-server/rtsp-session-pool.c:
9801 * gst/rtsp-server/rtsp-session-pool.h:
9802 * gst/rtsp-server/rtsp-session.c:
9803 * gst/rtsp-server/rtsp-session.h:
9804 Add more timeout stuff
9805 Add method to check if a session is expired.
9806 Add method to perform cleanup on a session pool.
9808 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9810 * gst/rtsp-server/rtsp-client.c:
9811 * gst/rtsp-server/rtsp-session-pool.c:
9812 * gst/rtsp-server/rtsp-session-pool.h:
9813 * gst/rtsp-server/rtsp-session.c:
9814 * gst/rtsp-server/rtsp-session.h:
9815 Add beginnings of session timeouts and limits
9816 Add the timeout value to the Session header for unusual timeout values.
9817 Allow us to configure a limit to the amount of active sessions in a pool. Set a
9818 limit on the amount of retry we do after a sessionid collision.
9819 Add properties to the sessionid and the timeout of a session. Keep track of
9820 creation time and last access time for sessions.
9822 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9824 * gst/rtsp-server/rtsp-client.c:
9825 * gst/rtsp-server/rtsp-media.c:
9826 * gst/rtsp-server/rtsp-media.h:
9827 * gst/rtsp-server/rtsp-sdp.c:
9828 * gst/rtsp-server/rtsp-session-pool.c:
9829 * gst/rtsp-server/rtsp-session.c:
9830 * gst/rtsp-server/rtsp-session.h:
9831 Cleanup of sessions and more
9832 Fix the refcounting of media and sessions in the client. Properly clean up the
9833 session data when the client performs a teardown.
9834 Add Server header to responses.
9835 Allow for multiple uri setups in one session.
9836 Add Range header to the PLAY response and add the range attribute to the SDP
9838 Fix the session pool remove method, it used the wrong key in the hashtable. Also
9839 give the ownership of the sessionid to the session object.
9841 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9843 * gst/rtsp-server/rtsp-server.c:
9844 * gst/rtsp-server/rtsp-server.h:
9846 Rename the 'server_port' variable to simply 'port'.
9848 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9851 * gst/rtsp-server/rtsp-client.c:
9852 * gst/rtsp-server/rtsp-media.c:
9853 * gst/rtsp-server/rtsp-media.h:
9854 * gst/rtsp-server/rtsp-session.c:
9855 * gst/rtsp-server/rtsp-session.h:
9856 Rework the way we handle transports for streams
9857 Make the media accept an array of transports for the streams that we have
9858 configured for the play/pause requests.
9859 Implement server states for a client and its media.
9860 Require 0.10.22.1 (git HEAD) of gstreamer.
9862 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9864 * gst/rtsp-server/rtsp-client.c:
9865 * gst/rtsp-server/rtsp-media-factory.c:
9866 Drop const from functions dealing with urls
9867 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
9868 have the right const in them.
9870 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9872 * gst/rtsp-server/rtsp-client.c:
9873 * gst/rtsp-server/rtsp-media.c:
9874 * gst/rtsp-server/rtsp-sdp.c:
9878 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9880 * gst/rtsp-server/rtsp-client.c:
9881 * gst/rtsp-server/rtsp-media-factory.c:
9882 * gst/rtsp-server/rtsp-media.c:
9883 * gst/rtsp-server/rtsp-media.h:
9885 Don't keep a reference to the GstRTSPMedia in the stream.
9886 Free more things when freeing the GstRTSPMedia.
9888 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9891 * gst/rtsp-server/rtsp-media-factory.c:
9892 * gst/rtsp-server/rtsp-media-factory.h:
9893 * gst/rtsp-server/rtsp-media.c:
9894 * gst/rtsp-server/rtsp-media.h:
9895 * gst/rtsp-server/rtsp-server.c:
9896 * gst/rtsp-server/rtsp-server.h:
9897 More docs and small cleanups
9898 Add some more docs and update the README
9899 Cleanup some method names.
9900 Remove an unneeded idx field in the GstRTSPMediaStream
9902 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9905 * examples/Makefile.am:
9906 * examples/test-readme.c:
9907 Add a README and more example code
9908 Add a README file that contains a small introduction on how to use the server
9909 along with the example code explained in the readme.
9911 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9913 * gst/rtsp-server/rtsp-media.c:
9914 * gst/rtsp-server/rtsp-server.c:
9915 Fix some leaks and change default port
9916 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
9917 we finished the initial preroll. If we keep them locked, setting the pipeline to
9918 NULL will not stop and clean up the sources correctly.
9919 Change the default RTSP port to 8554 aka the official alternative RTSP port.
9921 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9923 * gst/rtsp-server/rtsp-session.c:
9924 * gst/rtsp-server/rtsp-session.h:
9925 Cleanups to the session object
9926 Remove some unneeded variables in the session state of a stream such as the
9927 owner media and the server transport.
9928 Get the configuration of a media stream in a session based on the media_stream
9929 in the original object instead of our cached index.
9930 Free more data in the finalize method.
9932 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9934 * gst/rtsp-server/rtsp-client.c:
9935 * gst/rtsp-server/rtsp-client.h:
9936 Cleanups and reuse media from DESCRIBE
9937 Handle thread create errors.
9938 Rename some internal methods to better match what they actually do.
9939 Handle misconfiguration of session_pool and media_mapping gracefully.
9940 Cache the DESCRIBE media and uri in the client connection and reuse them when
9941 we receive a SETUP request in the same connection for the same uri.
9942 Cleanup the client connection object.
9944 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9946 * gst/rtsp-server/rtsp-media-factory.c:
9947 * gst/rtsp-server/rtsp-media-factory.h:
9948 * gst/rtsp-server/rtsp-media.c:
9949 * gst/rtsp-server/rtsp-media.h:
9950 Add shared properties to media and factory
9951 Add the shared property to media.
9952 Implement some simple caching in the factory depending on if the media is shared
9955 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9957 * gst/rtsp-server/rtsp-client.c:
9958 Add a little comment
9959 Add some comment about the content-base header.
9961 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9963 * examples/Makefile.am:
9964 * examples/test-mp4.c:
9965 * examples/test-ogg.c:
9966 * examples/test-video.c:
9967 * gst/rtsp-server/Makefile.am:
9968 * gst/rtsp-server/rtsp-client.c:
9969 * gst/rtsp-server/rtsp-client.h:
9970 * gst/rtsp-server/rtsp-media-factory.c:
9971 * gst/rtsp-server/rtsp-media-factory.h:
9972 * gst/rtsp-server/rtsp-media.c:
9973 * gst/rtsp-server/rtsp-media.h:
9974 * gst/rtsp-server/rtsp-sdp.c:
9975 * gst/rtsp-server/rtsp-sdp.h:
9976 * gst/rtsp-server/rtsp-server.c:
9977 * gst/rtsp-server/rtsp-server.h:
9978 * gst/rtsp-server/rtsp-session.c:
9979 * gst/rtsp-server/rtsp-session.h:
9980 Reorganize things, prepare for media sharing
9981 Added various other test server examples
9982 Move the SDP message generation to a separate helper.
9983 Refactor common code for finding the session.
9984 Add content-base for realplayer compatibility
9985 Clean up request uris before processing for better vlc compatibility.
9986 Move prerolling and pipeline construction to the RTSPMedia object.
9987 Use multiudpsink for future pipeline reuse.
9989 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9995 === release 0.10.1 ===
9997 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10000 Make 0.10.1 release
10003 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10005 * bindings/vala/Makefile.am:
10007 Add more directories and files to the dist.
10009 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10011 * bindings/python/Makefile.am:
10012 * bindings/python/rtspserver.override:
10013 Fixed compile error of python bindings
10015 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10017 * bindings/vala/gst-rtsp-server-0.10.vapi:
10018 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
10019 Marked values as nullable accordingly
10021 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10023 * bindings/vala/gst-rtsp-server-0.10.vapi:
10024 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
10025 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
10026 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
10027 Updated Vala bindings
10029 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10031 * gst/rtsp-server/rtsp-client.c:
10032 * gst/rtsp-server/rtsp-media-mapping.c:
10033 * gst/rtsp-server/rtsp-media-mapping.h:
10034 * gst/rtsp-server/rtsp-media.h:
10035 * gst/rtsp-server/rtsp-session-pool.h:
10036 Cleanups and doc updates
10037 Add some more documentation and do some minor cleanups here and there.
10039 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10041 * gst/rtsp-server/rtsp-client.c:
10042 * gst/rtsp-server/rtsp-media-factory.c:
10043 * gst/rtsp-server/rtsp-media-factory.h:
10044 * gst/rtsp-server/rtsp-media.c:
10045 * gst/rtsp-server/rtsp-media.h:
10046 * gst/rtsp-server/rtsp-session.c:
10047 * gst/rtsp-server/rtsp-session.h:
10049 Rename GstRTSPMediaBin to GstRTSPMedia
10050 Parse the request url into a GstRTSPUri object and pass this object to the
10051 various handlers and methods that require the uri.
10053 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10057 Add some more docs and remove some old code from the example.
10059 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10061 * gst/rtsp-server/rtsp-client.c:
10062 Handle state change failures better
10063 Handle state change failures better when changing the state of the pipeline to
10066 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10068 * gst/rtsp-server/rtsp-media-factory.c:
10069 * gst/rtsp-server/rtsp-media-factory.h:
10070 Make element creation more extendible
10071 Add get_element vmethod to the default MediaFactory so that subclasses can just
10072 override that method and still use the default logic for making a MediaBin from
10075 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10078 * gst/rtsp-server/Makefile.am:
10079 * gst/rtsp-server/rtsp-client.c:
10080 * gst/rtsp-server/rtsp-client.h:
10081 * gst/rtsp-server/rtsp-media-factory.c:
10082 * gst/rtsp-server/rtsp-media-factory.h:
10083 * gst/rtsp-server/rtsp-media-mapping.c:
10084 * gst/rtsp-server/rtsp-media-mapping.h:
10085 * gst/rtsp-server/rtsp-media.c:
10086 * gst/rtsp-server/rtsp-media.h:
10087 * gst/rtsp-server/rtsp-server.c:
10088 * gst/rtsp-server/rtsp-server.h:
10089 * gst/rtsp-server/rtsp-session.c:
10090 * gst/rtsp-server/rtsp-session.h:
10091 Make the server handle arbitrary pipelines
10092 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
10093 The GstMediaBin object has a handle to a bin with elements and to a list of
10094 GstMediaStream objects that this bin produces.
10095 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
10096 with methods to register and remove those mappings.
10097 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
10098 used by the server instance.
10099 Modify the example application so that it shows how to create custom pipelines
10100 attached to a specific mount point.
10101 Various misc cleanps.
10103 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10105 * gst/rtsp-server/rtsp-server.c:
10106 * gst/rtsp-server/rtsp-server.h:
10107 Allow setting a custom media factory for a server
10109 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10111 * gst/rtsp-server/rtsp-client.c:
10112 * gst/rtsp-server/rtsp-client.h:
10113 Allow setting a custom media factory for a client.
10115 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10117 * gst/rtsp-server/Makefile.am:
10118 Add Makefile entry for the media factory
10120 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10122 * gst/rtsp-server/rtsp-media-factory.c:
10123 * gst/rtsp-server/rtsp-media-factory.h:
10124 Add media factory to map urls to media pipeline objects.
10126 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10128 * gst/rtsp-server/rtsp-media.c:
10129 * gst/rtsp-server/rtsp-media.h:
10130 Add comments. Remove unused field
10132 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10134 * gst/rtsp-server/rtsp-session-pool.c:
10135 * gst/rtsp-server/rtsp-session-pool.h:
10136 Allow custom session pools to override the session id allocation algorithms Add some comments.
10138 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10140 * gst/rtsp-server/rtsp-session.h:
10143 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10145 * gst/rtsp-server/rtsp-client.c:
10146 * gst/rtsp-server/rtsp-client.h:
10147 Move the connection code in one place Add some comments
10149 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10151 * gst/rtsp-server/rtsp-server.c:
10152 * gst/rtsp-server/rtsp-server.h:
10153 Make vmethod to create and accept new clients. Add some docs.
10155 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10157 * gst/rtsp-server/rtsp-server.c:
10158 * gst/rtsp-server/rtsp-server.h:
10159 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
10161 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10163 * gst/rtsp-server/rtsp-client.c:
10164 * gst/rtsp-server/rtsp-client.h:
10165 Name the parameters more appropriately.
10167 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10169 * gst/rtsp-server/rtsp-session-pool.c:
10170 Do some more cleanup of the session pool.
10172 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10174 * gst/rtsp-server/Makefile.am:
10175 * gst/rtsp-server/rtsp-client.c:
10176 Check if return value of gst_rtsp_session_get_media is not NULL
10178 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10180 * gst/rtsp-server/Makefile.am:
10181 Install rtsp-session and rtsp-session-pool headers
10183 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10188 * bindings/python/Makefile.am:
10189 * bindings/python/arg-types.py:
10190 * bindings/python/codegen/Makefile.am:
10191 * bindings/python/codegen/__init__.py:
10192 * bindings/python/codegen/argtypes.py:
10193 * bindings/python/codegen/code-coverage.py:
10194 * bindings/python/codegen/codegen.py:
10195 * bindings/python/codegen/definitions.py:
10196 * bindings/python/codegen/defsparser.py:
10197 * bindings/python/codegen/docextract.py:
10198 * bindings/python/codegen/docgen.py:
10199 * bindings/python/codegen/fileprefix.override:
10200 * bindings/python/codegen/fileprefixmodule.c:
10201 * bindings/python/codegen/h2def.py:
10202 * bindings/python/codegen/mergedefs.py:
10203 * bindings/python/codegen/mkskel.py:
10204 * bindings/python/codegen/override.py:
10205 * bindings/python/codegen/reversewrapper.py:
10206 * bindings/python/codegen/scmexpr.py:
10207 * bindings/python/rtspserver-types.defs:
10208 * bindings/python/rtspserver.defs:
10209 * bindings/python/rtspserver.override:
10210 * bindings/python/rtspservermodule.c:
10212 Add python bindings.
10214 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10216 * bindings/Makefile.am:
10218 Don't go into python dir when requirements for python bindings are missing
10220 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10222 * bindings/Makefile.am:
10223 * bindings/vala/Makefile.am:
10225 Install Vala bindings if vala is available
10227 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10229 * bindings/vala/gst-rtsp-server-0.10.deps:
10230 * bindings/vala/gst-rtsp-server-0.10.vapi:
10231 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
10232 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
10233 * bindings/vala/packages/gst-rtsp-server-0.10.files:
10234 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
10235 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
10236 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
10237 Regenerated Vala bindings
10239 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10241 * bindings/vala/gst-rtsp-server.vapi:
10242 * bindings/vala/packages/gst-rtsp-server.metadata:
10243 Fixed typo in included headers for vala bindings
10245 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10249 * pkgconfig/Makefile.am:
10250 * pkgconfig/gst-rtsp-server.pc.in:
10251 Added pkgconfig file
10253 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
10255 * bindings/vala/gst-rtsp-server.vapi:
10256 * bindings/vala/packages/gst-rtsp-server.excludes:
10257 * bindings/vala/packages/gst-rtsp-server.gi:
10258 * bindings/vala/packages/gst-rtsp-server.metadata:
10259 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
10261 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
10263 * bindings/vala/gst-rtsp-server.vapi:
10264 * bindings/vala/packages/gst-rtsp-server.deps:
10265 * bindings/vala/packages/gst-rtsp-server.files:
10266 * bindings/vala/packages/gst-rtsp-server.gi:
10267 * bindings/vala/packages/gst-rtsp-server.metadata:
10268 * bindings/vala/packages/gst-rtsp-server.namespace:
10269 Added Vala bindings
10271 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
10273 * gst/rtsp-server/rtsp-session.c:
10274 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
10276 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
10278 * examples/Makefile.am:
10279 * gst/rtsp-server/Makefile.am:
10280 Put GStreamer version in library name
10282 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10284 * examples/Makefile.am:
10285 * gst/rtsp-server/Makefile.am:
10286 Fix some issues to pass distcheck
10288 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10290 * gst/rtsp-server/rtsp-server.c:
10291 Added port property to GstRTSPServer class.
10293 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10298 * examples/Makefile.am:
10301 * gst/rtsp-server/Makefile.am:
10302 * gst/rtsp-server/rtsp-client.c:
10303 * gst/rtsp-server/rtsp-client.h:
10304 * gst/rtsp-server/rtsp-media.c:
10305 * gst/rtsp-server/rtsp-media.h:
10306 * gst/rtsp-server/rtsp-server.c:
10307 * gst/rtsp-server/rtsp-server.h:
10308 * gst/rtsp-server/rtsp-session-pool.c:
10309 * gst/rtsp-server/rtsp-session-pool.h:
10310 * gst/rtsp-server/rtsp-session.c:
10311 * gst/rtsp-server/rtsp-session.h:
10313 Split in library and example program
10315 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
10317 * src/rtsp-client.h:
10318 Removed obsolete variable
10320 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
10322 * src/rtsp-client.c:
10323 * src/rtsp-client.h:
10324 Removed pipeline variable GstRTSPClient, because it's only used in one function
10326 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10328 * src/rtsp-media.c:
10329 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
10331 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
10333 * src/rtsp-session.c:
10334 Initialize some more vars.
10336 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
10338 * src/rtsp-session.c:
10339 Initialize variable to avoid compiler warning.
10341 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
10344 Add a reasonable generic .gitignore