3 2017-07-14 Sebastian Dröge <slomo@coaxion.net>
10 2017-06-20 12:08:05 +0300 Sebastian Dröge <sebastian@centricular.com>
16 * gst-rtsp-server.doap:
20 === release 1.12.0 ===
22 2017-05-04 15:40:46 +0300 Sebastian Dröge <sebastian@centricular.com>
28 * gst-rtsp-server.doap:
32 === release 1.11.91 ===
34 2017-04-27 17:42:02 +0300 Sebastian Dröge <sebastian@centricular.com>
40 * gst-rtsp-server.doap:
44 2017-04-24 20:30:37 +0100 Tim-Philipp Müller <tim@centricular.com>
47 Automatic update of common submodule
48 From 60aeef6 to 48a5d85
50 2017-04-13 14:20:10 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
52 * gst/rtsp-server/rtsp-media-factory.c:
53 * gst/rtsp-server/rtsp-media.c:
54 * gst/rtsp-server/rtsp-session.c:
55 * gst/rtsp-server/rtsp-stream.c:
56 gi: Fix some annotations and docstrings
58 2017-04-13 13:52:26 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
60 * gst/rtsp-server/meson.build:
65 2017-04-10 23:51:12 +0100 Tim-Philipp Müller <tim@centricular.com>
69 Automatic update of common submodule
70 From 39ac2f5 to 60aeef6
72 === release 1.11.90 ===
74 2017-04-07 16:35:03 +0300 Sebastian Dröge <sebastian@centricular.com>
80 * gst-rtsp-server.doap:
84 2017-03-27 18:19:33 +0100 Tim-Philipp Müller <tim@centricular.com>
86 * examples/test-launch.c:
87 examples: make test-launch pipeline shared by default as well
89 2017-02-27 19:10:44 +0200 Sebastian Dröge <sebastian@centricular.com>
91 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
92 gstreamer-rtsp-server: Add both srcdir and builddir to the include path
93 Just the build dir is not going to work for srcdir!=builddir.
95 2017-02-24 15:59:54 +0200 Sebastian Dröge <sebastian@centricular.com>
100 2017-02-24 15:37:49 +0200 Sebastian Dröge <sebastian@centricular.com>
105 === release 1.11.2 ===
107 2017-02-24 15:10:07 +0200 Sebastian Dröge <sebastian@centricular.com>
113 * gst-rtsp-server.doap:
116 2017-02-14 20:40:26 +0000 Tim-Philipp Müller <tim@centricular.com>
119 meson: dist meson build files
120 Ship meson build files in tarballs, so people who use tarballs
121 in their builds can start playing with meson already.
123 2017-02-07 23:39:37 +1100 Jan Schmidt <jan@centricular.com>
125 * examples/test-record.c:
126 examples/test-record: Add extra line to initial printout
127 Add an example line of how to deliver a stream to the
130 2017-01-19 14:57:19 +0200 Sebastian Dröge <sebastian@centricular.com>
132 * gst/rtsp-server/rtsp-client.c:
133 rtsp-client: Also handle the (S|G)ET_PARAMETER case of size==0 || !data as keep-alive
134 If there is no Content-Length header, no body would be allocated and the
135 '\0' would also not be appended to the body.
137 2017-01-19 14:24:07 +0200 Sebastian Dröge <sebastian@centricular.com>
139 * gst/rtsp-server/rtsp-client.c:
140 rtsp-client: Fix handling of keep-alive GET_PARAMETER/SET_PARAMETER
141 While they logically have 0 bytes length, GstRTSPConnection is appending
142 a '\0' to everything making the size be 1 instead.
144 2017-01-13 12:39:36 +0000 Tim-Philipp Müller <tim@centricular.com>
149 2017-01-12 19:04:23 +0200 Sebastian Dröge <sebastian@centricular.com>
151 * gst/rtsp-server/rtsp-session.c:
152 rtsp-session: Only remove deprecated API if requested to do so, not just when disabling
153 gst_rtsp_session_is_expired() and gst_rtsp_session_next_timeout() were
156 2017-01-12 16:32:59 +0200 Sebastian Dröge <sebastian@centricular.com>
161 === release 1.11.1 ===
163 2017-01-12 16:14:46 +0200 Sebastian Dröge <sebastian@centricular.com>
169 * gst-rtsp-server.doap:
170 * win32/common/libgstrtspserver.def:
173 2017-01-10 08:34:50 +0100 Patricia Muscalu <patricia@axis.com>
175 * gst/rtsp-server/rtsp-stream.c:
176 rtsp-stream: corrected if-statement in _get_server_port()
177 This bug was accidentally introduced while fixing a segfault
178 in _get_server_port() function.
179 https://bugzilla.gnome.org/show_bug.cgi?id=776345
181 2017-01-09 14:12:05 +0100 Patricia Muscalu <patricia@axis.com>
183 * gst/rtsp-server/rtsp-stream.c:
184 * tests/check/gst/stream.c:
185 rtsp-stream: fixed segmenation fault in _get_server_port()
186 Calling function gst_rtsp_stream_get_server_port() results in
187 segmenation fault in the RTP/RTSP/TCP case.
188 Port that the server will use to receive RTCP makes only
189 sense in the UDP case, however the function should handle
190 the TCP case in a nicer way.
191 https://bugzilla.gnome.org/show_bug.cgi?id=776345
193 2017-01-09 12:22:40 +0300 Aleksandr Slobodeniuk <alenuke@yandex.ru>
195 * gst/rtsp-server/rtsp-media-factory.c:
196 dosc: Fix a little typo
197 https://bugzilla.gnome.org/show_bug.cgi?id=777037
199 2017-01-04 16:20:54 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
201 * pkgconfig/Makefile.am:
202 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
203 * pkgconfig/meson.build:
204 meson: generate pkg-config -uninstalled pc files
205 Generating those files is useful for users building the GStreamer stack
206 using meson and having to link it to another project which is still
208 https://bugzilla.gnome.org/show_bug.cgi?id=776810
210 2017-01-04 16:11:08 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
212 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
213 pkgconfig: fix -uninstalled pc file
214 pcfiledir was never defined so the paths were wrong.
215 https://bugzilla.gnome.org/show_bug.cgi?id=776867
217 2016-12-21 13:41:50 +0100 Patricia Muscalu <patricia@axis.com>
219 * gst/rtsp-server/rtsp-stream.c:
220 * tests/check/gst/rtspserver.c:
221 rtsp-stream: Fixed TCP transport case
222 Make sure that the appsink element is actually added to
223 the bin before trying to link it with the elements in it.
224 https://bugzilla.gnome.org/show_bug.cgi?id=776343
226 2016-12-16 17:26:04 +0000 Tim-Philipp Müller <tim@centricular.com>
232 Remove generated .spec file
233 Likely extremely bitrotten, and we should not ship this anyway.
235 2016-12-03 08:21:02 +0100 Edward Hervey <bilboed@bilboed.com>
238 Automatic update of common submodule
239 From f980fd9 to 39ac2f5
241 2016-12-02 15:40:09 +0100 Edward Hervey <edward@centricular.com>
243 * gst/rtsp-server/rtsp-media.c:
244 media: Fix pt map caps
245 Since decryption is handled within rtpbin, all outcoming stream
246 caps will be application/x-rtp (i.e. regular rtp)
247 Fixes RECORD with SRTP streams
249 2016-12-02 15:38:04 +0100 Edward Hervey <edward@centricular.com>
251 * gst/rtsp-server/rtsp-media-factory.c:
252 media-factory: Create media objects with the proper transport mode
253 The function called immediately afterwards (collect_streams()) will
254 need it to work properly
256 2016-12-02 14:36:50 +0200 Sebastian Dröge <sebastian@centricular.com>
258 * gst/rtsp-server/rtsp-auth.c:
259 rtsp-auth: Don't remove digest-auth nonces that already/still have a client connected
261 2016-12-01 18:04:34 +0200 Sebastian Dröge <sebastian@centricular.com>
263 * gst/rtsp-server/rtsp-media-factory.c:
264 rtsp-media-factory: Don't create a pipeline for the media pipeline string
265 We're going to put a pipeline into a pipeline otherwise, which is not
268 2016-10-25 15:41:28 +0300 Kseniia Vasilchuk <vasilchukkseniia@gmail.com>
270 * gst/rtsp-server/rtsp-media.c:
271 media: Fix race condition around finish_unprepare() if called multiple time
272 https://bugzilla.gnome.org/show_bug.cgi?id=755329
274 2016-11-30 14:06:36 +1100 Jan Schmidt <jan@centricular.com>
276 * gst/rtsp-sink/gstrtspclientsink.c:
277 rtspclientsink: Don't leave stale pointer after unref
278 Fix a warning on shutdown - don't keep a pointer to an
279 alread-unreffed object.
281 2016-11-26 11:24:50 +0000 Tim-Philipp Müller <tim@centricular.com>
284 common: use https protocol for common submodule
285 https://bugzilla.gnome.org/show_bug.cgi?id=775110
287 2016-11-21 23:29:56 +1100 Matthew Waters <matthew@centricular.com>
289 * gst/rtsp-server/rtsp-stream.c:
290 stream: block the output of rtpbin instead of the source pipeline
291 85c52e194bcb81928b96614be0ae47d59eccb1ce introduced a more correct
292 detection of the srtp rollover counter to add to the SDP.
293 Unfortunately, it was incomplete for live pipelines where the logic
294 blocks the source bin before creating the SDP and thus would never have
295 the necessary informaiton to create a correct SDP with srtp encryption.
296 Move the pad blocks to rtpbin's output pads instead so that the
297 necessary information can be created before we need the information for
299 https://bugzilla.gnome.org/show_bug.cgi?id=770239
301 2016-11-21 16:02:39 +0100 Dag Gullberg <dagg@axis.com>
303 * gst/rtsp-server/rtsp-client.c:
304 rtsp-client: add IDLE timeout, before session exists
305 The RTSP server will not timeout an idle RTSP connection
306 (note this is different from doing timeout on a RTSP
308 At least for Apache this is a problem when running RTSP over
309 HTTPS since it uses one of the threads (there is a rather
310 limited number) that are available for handling requests.
311 https://bugzilla.gnome.org/show_bug.cgi?id=771830
313 2016-11-23 09:45:08 +0000 Tim-Philipp Müller <tim@centricular.com>
318 2016-11-21 13:05:50 +0100 Göran Jönsson <goranjn@axis.com>
320 * gst/rtsp-server/rtsp-stream.c:
321 rtsp-stream: Set close-socket FALSE on UDP src:es
322 With this RTSP server can use the sockets independent on the udpsrc
324 When the udp src is finalized it will unref socket and when g_socket
325 is finalized the socket will be closed.
326 https://bugzilla.gnome.org/show_bug.cgi?id=765673
328 2016-11-18 17:47:13 +0200 Sebastian Dröge <sebastian@centricular.com>
330 * gst/rtsp-sink/gstrtspclientsink.c:
331 rtspclientsink: Move to new helper function to parse authentication responses
332 https://bugzilla.gnome.org/show_bug.cgi?id=774416
334 2016-11-16 08:42:24 +0200 Sebastian Dröge <sebastian@centricular.com>
336 * examples/Makefile.am:
337 * examples/test-auth-digest.c:
338 * gst/rtsp-server/rtsp-auth.c:
339 * gst/rtsp-server/rtsp-auth.h:
340 * win32/common/libgstrtspserver.def:
341 rtsp-auth: Add support for Digest authentication
342 https://bugzilla.gnome.org/show_bug.cgi?id=774416
344 2016-11-17 09:41:53 -0800 Scott D Phillips <scott.d.phillips@intel.com>
347 * gst/rtsp-server/meson.build:
349 * tests/check/meson.build:
351 * win32/common/libgstrtspserver.def:
352 Enable building with MSVC
353 https://bugzilla.gnome.org/show_bug.cgi?id=774640
355 2016-11-18 20:23:14 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
358 meson: gstreamer gst_check_dep does not exist on windows
360 2016-11-17 09:43:37 -0800 Scott D Phillips <scott.d.phillips@intel.com>
362 * gst/rtsp-server/rtsp-client.c:
363 client: update do_send_message to match type GstRTSPClientSendFunc
364 This type mismatch fails building with MSVC
365 https://bugzilla.gnome.org/show_bug.cgi?id=774640
367 2016-11-11 14:42:08 +0200 Sebastian Dröge <sebastian@centricular.com>
369 * gst/rtsp-server/rtsp-sdp.c:
370 rtsp-sdp: Fix indentation
372 2016-11-10 05:16:00 +0000 Neha Arora <arora.neha@samsung.com>
374 * gst/rtsp-server/rtsp-media.c:
375 rtsp-media: Only signal "new-state" if the state has actually changed
376 https://bugzilla.gnome.org/show_bug.cgi?id=774173
378 2016-08-24 11:39:13 +0200 Branko Subasic <branko@axis.com>
380 * gst/rtsp-server/rtsp-client.c:
381 * gst/rtsp-server/rtsp-client.h:
382 client: emit signal in the beginning of each rtsp request
383 These signals let the application validate the requests, configure the
384 media/stream in a certain way and also generate error status code in
385 case of error or bad request.
386 https://bugzilla.gnome.org/show_bug.cgi?id=758062
388 2016-11-01 18:10:35 +0000 Tim-Philipp Müller <tim@centricular.com>
391 meson: update version
393 === release 1.11.0 ===
395 2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com>
400 === release 1.10.0 ===
402 2016-11-01 18:06:46 +0200 Sebastian Dröge <sebastian@centricular.com>
408 * gst-rtsp-server.doap:
411 2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com>
413 * tests/check/gst/rtspserver.c:
414 * tests/check/gst/stream.c:
415 tests: try to avoid using the same ports in different tests
416 Causes problems with client multicast tests otherwise if
417 tests are run in parallel.
418 https://bugzilla.gnome.org/show_bug.cgi?id=773640
420 2016-10-28 17:50:59 +0100 Tim-Philipp Müller <tim@centricular.com>
422 * tests/check/gst/client.c:
423 tests: client: use fail_unless_equals_foo() for better failure reporting
425 2016-09-26 11:16:04 +0200 Göran Jönsson <goranjn@axis.com>
427 * gst/rtsp-server/rtsp-client.c:
428 rtsp-client: Session filter in unwatch session
429 Call session filter with filter_session_media as paramer in
430 client_unwatch_session if using drop_backlog = FALSE.
431 In client_unwatch_session its allowed to grow the watchs backlog.
432 If using drop_backlog = FALSE and the backlog is full it will cause
433 a deadlock when setting session media state to NULL
434 if the backlog is not allowed to grow.
435 https://bugzilla.gnome.org/show_bug.cgi?id=771983
437 2016-10-20 21:40:18 +0100 Tim-Philipp Müller <tim@centricular.com>
440 meson: add fallbacks for gst modules
443 2016-09-14 17:48:39 +0300 Nikita Bobkov <NikitaDBobkov@gmail.com>
445 * gst/rtsp-server/rtsp-client.c:
446 rtsp-client: Fix factory leaking in find_media() in error cases
447 https://bugzilla.gnome.org/show_bug.cgi?id=771488
449 2016-10-06 11:47:50 -0400 Xavier Claessens <xavier.claessens@collabora.com>
451 * gst/rtsp-server/rtsp-stream.c:
452 stream: Fix randomly missing streams from SDP with dynamic elements
453 When using dynamic elements, gst_rtsp_stream_join_bin() is called from
454 "pad-added" signal. In that case priv->srcpad could already have its caps,
455 and they'll be sent to priv->send_src[0] pad. That means that when it
456 connects "notify::caps" signal, that pad could already have received its
457 caps and the signal won't be emitted anymore.
458 In that case priv->caps stay to NULL and when building the SDP that stream
459 gets ignored. Leading to missing video or audio when playing in client side.
460 https://bugzilla.gnome.org/show_bug.cgi?id=772478
462 2016-09-30 11:42:08 +0100 Tim-Philipp Müller <tim@centricular.com>
465 meson: update version
467 === release 1.9.90 ===
469 2016-09-30 13:04:12 +0300 Sebastian Dröge <sebastian@centricular.com>
475 * gst-rtsp-server.doap:
478 2016-09-17 13:17:19 +0100 Ian Jamison <ian.dev@arkver.com>
480 * gst/rtsp-server/rtsp-media-factory.c:
481 * gst/rtsp-server/rtsp-media.c:
482 * gst/rtsp-server/rtsp-stream.c:
483 rtsp-server: Hint that set_multicast_iface expects the name of the interface
484 To prevent any possibly confusion with IPs or anything else.
485 https://bugzilla.gnome.org/show_bug.cgi?id=771530
487 2016-09-18 09:58:55 -0400 Sebastian Dröge <sebastian@centricular.com>
489 * gst/rtsp-server/rtsp-media-factory.c:
490 * gst/rtsp-server/rtsp-media.c:
491 rtsp-media: Call g_free() instead of g_object_unref() on multicast-iface strings
492 https://bugzilla.gnome.org/show_bug.cgi?id=763000#c5
494 2016-09-14 11:31:15 +0200 Sebastian Dröge <sebastian@centricular.com>
497 configure: Depend on gstreamer 1.9.2.1
499 2016-09-10 20:52:31 +1000 Jan Schmidt <jan@centricular.com>
503 Automatic update of common submodule
504 From b18d820 to f980fd9
506 2016-09-10 09:58:31 +1000 Jan Schmidt <jan@centricular.com>
510 Automatic update of common submodule
511 From 6f2d209 to b18d820
513 2016-09-07 18:44:34 +0300 Sebastian Dröge <sebastian@centricular.com>
515 * gst/rtsp-server/rtsp-stream.c:
516 rtsp-stream: Remove unused _locked() variant of a function
517 It was added during refactoring.
519 2016-09-07 10:21:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
521 * gst/rtsp-server/rtsp-stream.c:
522 stream: cosmetic cleanup
523 https://bugzilla.gnome.org/show_bug.cgi?id=766612
525 2016-09-07 10:16:19 -0400 Xavier Claessens <xavier.claessens@collabora.com>
527 * gst/rtsp-server/rtsp-stream.c:
528 stream: Compare IP addresses case insensitive in more places
529 https://bugzilla.gnome.org/show_bug.cgi?id=766612
531 2016-09-07 10:12:18 -0400 Xavier Claessens <xavier.claessens@collabora.com>
534 * gst/rtsp-server/rtsp-stream.c:
535 stream: Fix leaked joined_bin
536 There is no need to keep a strong ref on it, and _leave_bin() was
537 setting it to NULL before calling g_clear_object() so it was leaked.
538 https://bugzilla.gnome.org/show_bug.cgi?id=766612
540 2016-09-06 19:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
542 * gst/rtsp-server/rtsp-stream.c:
543 rtsp-stream: Compare IP address strings case insensitive
544 Otherwise IPv6 addresses might fail this comparision.
546 2016-09-06 19:10:21 +0300 Sebastian Dröge <sebastian@centricular.com>
548 * gst/rtsp-server/rtsp-stream.c:
549 rtsp-stream: Bind multicast sockets to ANY as before
550 https://bugzilla.gnome.org/show_bug.cgi?id=766612#c48
552 2016-09-05 18:31:36 +0300 Kseniia <vasilchukkseniia@gmail.com>
554 * gst/rtsp-server/rtsp-session.c:
555 rtsp-session: Fix segfault when doing keep-alive after removing the session
556 If keep-alive happens after removing the session but before finalizing the
557 stream transport, we would segfault.
558 https://bugzilla.gnome.org/show_bug.cgi?id=750544
560 2016-09-05 18:04:50 +0300 Sebastian Dröge <sebastian@centricular.com>
562 * gst/rtsp-server/rtsp-stream.c:
563 rtsp-stream: Always create multicast UDP elements if the protocol flag is set
564 Adding them later will cause deadlocks due to
565 1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
566 2) adding the multicast sink
567 3) waiting for it to get data to preroll again
568 3) never happens because the queues after the tee are full.
570 2016-09-05 16:32:57 +0300 Sebastian Dröge <sebastian@centricular.com>
572 * gst/rtsp-server/rtsp-stream.c:
573 rtsp-stream: Fix up various multicast related issues
575 2016-09-05 13:40:59 +0300 Sebastian Dröge <sebastian@centricular.com>
577 * tests/check/gst/stream.c:
578 tests: Fix compilation
580 2016-07-28 15:33:05 -0400 Xavier Claessens <xavier.claessens@collabora.com>
582 * gst/rtsp-server/rtsp-client.c:
583 * gst/rtsp-server/rtsp-stream.c:
584 * tests/check/gst/stream.c:
585 stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast
586 This is basically reverting changes introduced in commit f62a9a7,
587 because it was introducing various regressions:
588 - It introduces a leak of udpsrc elements that got wrongly fixed by adding
589 an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
590 ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
591 - If a mcast client connects, it creates a new socket in SETUP to try to respect
592 the destination/port given by the client in the transport, and overrides the
593 socket already set on the udpsink element. That means that if we already had a
594 client connected, the source address on the udp packets it receives suddenly
596 - If a 2nd mcast client connects, the destination/port in its transport is
597 ignored but its transport wasn't updated.
598 What this patch does:
599 - Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
600 - Always have a tee+queue when udp is enabled. This could be optimized
601 again in a later patch, but is more complicated. If no unicast clients
602 connects then those elements are useless, this could be also optimized
604 - When mcast transport is added, it creates a new set of udpsrc/udpsink,
605 seperated from those for unicast clients. Since we already support only
606 one mcast address, we also create only one set of elements.
607 https://bugzilla.gnome.org/show_bug.cgi?id=766612
609 2016-07-28 15:20:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
611 * gst/rtsp-server/rtsp-stream.c:
612 stream: factor our plug_src function
613 https://bugzilla.gnome.org/show_bug.cgi?id=766612
615 2016-07-21 21:46:16 -0400 Xavier Claessens <xavier.claessens@collabora.com>
617 * gst/rtsp-server/rtsp-stream.c:
618 stream: factor out plug_sink function
619 https://bugzilla.gnome.org/show_bug.cgi?id=766612
621 2016-07-20 23:05:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
623 * gst/rtsp-server/rtsp-stream.c:
624 stream: small documentation clarification
625 https://bugzilla.gnome.org/show_bug.cgi?id=766612
627 2016-07-20 15:35:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
629 * gst/rtsp-server/rtsp-stream.c:
630 stream: rename addr_v4/6 to mcast_addr_v4/6 for clarity
631 https://bugzilla.gnome.org/show_bug.cgi?id=766612
633 2016-07-14 11:10:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
635 * gst/rtsp-server/rtsp-stream.c:
636 stream: Keep a ref on joined bin
637 https://bugzilla.gnome.org/show_bug.cgi?id=766612
639 2016-07-20 15:11:32 -0400 Xavier Claessens <xavier.claessens@collabora.com>
641 * gst/rtsp-server/rtsp-stream.c:
643 https://bugzilla.gnome.org/show_bug.cgi?id=766612
645 2016-07-20 23:18:23 -0400 Xavier Claessens <xavier.claessens@collabora.com>
647 * gst/rtsp-server/rtsp-stream.c:
648 stream: small fix in error code path
649 https://bugzilla.gnome.org/show_bug.cgi?id=766612
651 2016-07-20 20:09:57 -0400 Xavier Claessens <xavier.claessens@collabora.com>
653 * gst/rtsp-server/rtsp-stream.c:
654 Revert "rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc"
655 This partly reverts commit cba045e1b19fad6e689e10206f57903e15f1229a,
656 but keeps unit tests.
657 https://bugzilla.gnome.org/show_bug.cgi?id=766612
659 2016-09-01 12:33:00 +0300 Sebastian Dröge <sebastian@centricular.com>
664 === release 1.9.2 ===
666 2016-09-01 12:32:51 +0300 Sebastian Dröge <sebastian@centricular.com>
672 * gst-rtsp-server.doap:
675 2016-01-27 01:03:52 +0000 Tim-Philipp Müller <tim@centricular.com>
678 * examples/meson.build:
680 * gst/rtsp-server/meson.build:
681 * gst/rtsp-sink/meson.build:
683 * pkgconfig/meson.build:
684 * tests/check/meson.build:
686 Add support for Meson as alternative/parallel build system
687 https://github.com/mesonbuild/meson
689 2016-08-26 21:56:13 +0200 Josep Torra <n770galaxy@gmail.com>
692 * tests/check/Makefile.am:
693 build: silence error about pthread for 'make check' in osx
694 Fixes "clang: error: argument unused during compilation: '-pthread'"
696 2015-09-25 15:04:00 +0000 Nikita Bobkov <NikitaDBobkov@gmail.com>
698 * gst/rtsp-server/rtsp-client.c:
699 rtsp-client: Fix leaking of media in error cases
700 With additional fixes by Kseniya Vasilchuk <vasilchukkseniia@gmail.com>
701 and myself to make the media refcounting a bit easier to follow.
702 https://bugzilla.gnome.org/show_bug.cgi?id=755632
704 2016-08-02 15:08:22 +0300 Sebastian Dröge <sebastian@centricular.com>
706 * gst/rtsp-server/rtsp-client.c:
707 rtsp-client: Fix leaking of session in error cases
708 https://bugzilla.gnome.org/show_bug.cgi?id=755632
710 2016-07-11 21:16:04 +0200 Stefan Sauer <ensonic@users.sf.net>
713 Automatic update of common submodule
714 From f363b32 to f49c55e
716 2016-07-06 13:51:15 +0300 Sebastian Dröge <sebastian@centricular.com>
721 === release 1.9.1 ===
723 2016-07-06 13:28:12 +0300 Sebastian Dröge <sebastian@centricular.com>
729 * gst-rtsp-server.doap:
732 2016-06-24 02:02:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
735 configure: Need to add -DGST_STATIC_COMPILATION when building only statically
736 https://bugzilla.gnome.org/show_bug.cgi?id=767463
738 2016-06-21 11:49:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
741 Automatic update of common submodule
742 From ac2f647 to f363b32
744 2016-04-14 22:56:11 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
746 * gst/rtsp-server/rtsp-sdp.c:
747 * gst/rtsp-server/rtsp-sdp.h:
748 * gst/rtsp-server/rtsp-stream.c:
749 * gst/rtsp-server/rtsp-stream.h:
750 sdp: add rollover counters for all sender SSRC
751 We add different crypto sessions in MIKEY, one for each sender
752 SSRC. Currently, all of them will have the same security policy, 0.
753 The rollover counters are obtained from the srtpenc element using the
755 https://bugzilla.gnome.org/show_bug.cgi?id=730539
757 2016-06-07 20:44:42 +0100 Tim-Philipp Müller <tim@centricular.com>
759 * gst/rtsp-server/rtsp-media-factory.h:
760 * gst/rtsp-server/rtsp-server.h:
763 2016-05-25 10:28:43 +0100 Tim-Philipp Müller <tim@centricular.com>
765 * gst/rtsp-server/Makefile.am:
766 g-i: pass compiler env to g-ir-scanner
767 It's what introspection.mak does as well. Should
768 fix spurious build failures on gnome-continuous
769 (caused by g-ir-scanner getting compiler details
770 via python which is broken in some environments
771 so passing the compiler details bypasses that).
773 2016-05-18 16:48:44 +0100 Ian <ian.arkver.dev@gmail.com>
775 * gst/rtsp-server/rtsp-session.c:
776 rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header
777 This works with rtspsrc and live555, but fails with e.g. ffmpeg.
778 https://bugzilla.gnome.org/show_bug.cgi?id=766619
780 2016-03-07 14:48:38 +0100 Edward Hervey <bilboed@bilboed.com>
782 * gst/rtsp-sink/gstrtspclientsink.c:
783 rtspclientsink: Check return value of sscanf
784 And just make sure we always have 0/0 if we have an error
787 2016-04-25 08:55:25 -0400 Jake Foytik <jake.foytik@ipconfigure.com>
789 * gst/rtsp-server/rtsp-stream.c:
790 * tests/check/gst/rtspserver.c:
791 * tests/check/gst/stream.c:
792 rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
793 - Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
794 - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
795 - Create unit test for shared media.
796 https://bugzilla.gnome.org/show_bug.cgi?id=764744
798 2016-04-11 10:55:23 +0300 Sebastian Dröge <sebastian@centricular.com>
800 * gst/rtsp-server/rtsp-stream.c:
801 rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
802 For IPv6 addresses, binding to a multicast group does not work on Linux
803 either. Always bind to ANY and then later join the multicast group.
804 https://bugzilla.gnome.org/show_bug.cgi?id=764679
806 2016-04-14 10:05:02 +0100 Julien Isorce <j.isorce@samsung.com>
809 Automatic update of common submodule
810 From 6f2d209 to ac2f647
812 2016-04-06 10:09:46 +0200 Patricia Muscalu <patricia@axis.com>
814 * gst/rtsp-server/rtsp-thread-pool.c:
815 rtsp-thread-pool: explained why GSource is a part of ThreadImpl
816 Clarified why it is necessary to add source information to
817 GstRTSPThreadImpl. See the reported bug in GLib:
818 https://bugzilla.gnome.org/show_bug.cgi?id=720186
819 for more information.
820 https://bugzilla.gnome.org/show_bug.cgi?id=761702
822 2016-04-04 12:58:38 +0300 Sebastian Dröge <sebastian@centricular.com>
824 * examples/Makefile.am:
825 examples: Clean up CFLAGS/LDADD even more
826 The internal .la should come first and is part of LDADD, as is
829 2016-04-04 12:39:39 +0300 Sebastian Dröge <sebastian@centricular.com>
831 * examples/Makefile.am:
832 examples: Clean up CFLAGS/LDADD to link with the correct versions of all libraries
834 2016-04-03 12:06:29 +0300 Sebastian Dröge <sebastian@centricular.com>
836 * gst/rtsp-server/Makefile.am:
837 rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS)
839 2015-12-30 18:39:05 +0200 Sebastian Dröge <sebastian@centricular.com>
841 * gst/rtsp-server/rtsp-client.c:
842 * gst/rtsp-server/rtsp-media-factory.c:
843 * gst/rtsp-server/rtsp-media-factory.h:
844 * gst/rtsp-server/rtsp-media.c:
845 * gst/rtsp-server/rtsp-media.h:
846 * gst/rtsp-server/rtsp-sdp.c:
847 * gst/rtsp-server/rtsp-stream.c:
848 * gst/rtsp-server/rtsp-stream.h:
849 rtsp-server: Implement clock signalling according to RFC7273
850 For NTP and PTP clocks we signal the actual clock that is used and signal
851 the direct media clock offset.
852 For all other clocks we at least signal that it's the local sender clock.
853 This allows receivers to know which clock was used to generate the media and
854 its RTP timestamps. Receivers can then implement network synchronization,
855 either absolute or at least relative by getting the sender clock rate directly
856 via NTP/PTP instead of estimating it from RTP timestamps and packet receive
858 https://bugzilla.gnome.org/show_bug.cgi?id=760005
860 2016-03-02 19:42:58 +0200 Sebastian Dröge <sebastian@centricular.com>
862 * gst/rtsp-sink/gstrtspclientsink.c:
863 rtspclientsink: Add support for setting the multicast interface
864 https://bugzilla.gnome.org/show_bug.cgi?id=763000
866 2016-03-02 19:42:13 +0200 Sebastian Dröge <sebastian@centricular.com>
868 * gst/rtsp-server/rtsp-media-factory.c:
869 * gst/rtsp-server/rtsp-media-factory.h:
870 * gst/rtsp-server/rtsp-media.c:
871 * gst/rtsp-server/rtsp-media.h:
872 * gst/rtsp-server/rtsp-stream.c:
873 * gst/rtsp-server/rtsp-stream.h:
874 rtsp-media: Add support for setting the multicast interface
875 https://bugzilla.gnome.org/show_bug.cgi?id=763000
877 2016-03-07 08:50:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
879 * gst/rtsp-sink/gstrtspclientsink.c:
880 rtspclientsink: use new gst_element_class_add_static_pad_template()
881 https://bugzilla.gnome.org/show_bug.cgi?id=763196
883 2016-03-24 13:33:43 +0200 Sebastian Dröge <sebastian@centricular.com>
888 === release 1.8.0 ===
890 2016-03-24 13:00:35 +0200 Sebastian Dröge <sebastian@centricular.com>
896 * gst-rtsp-server.doap:
899 2016-03-16 23:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
901 * gst/rtsp-server/rtsp-stream.c:
902 rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
903 This would get us NO_PREROLL in the bin again and break seeking.
904 Thanks to Carlos Rafael Giani for helping to debug this!
905 https://bugzilla.gnome.org/show_bug.cgi?id=740509
907 === release 1.7.91 ===
909 2016-03-15 12:26:13 +0200 Sebastian Dröge <sebastian@centricular.com>
915 * gst-rtsp-server.doap:
918 2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
920 * gst/rtsp-server/rtsp-stream.c:
921 rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
922 Without this, RECORD pipelines are broken because
923 a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
924 added later. Previously it was there earlier and due to NO_PREROLL caused the
925 pipeline to preroll immediately
926 b) the udpsrc for the pipeline is added later and never set to PLAYING state,
927 as the corresponding code previously was only for PLAY pipelines.
928 https://bugzilla.gnome.org/show_bug.cgi?id=763281
930 2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
932 * gst/rtsp-server/rtsp-stream.c:
933 rtsp-stream: Fix typo in the docstring
934 gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
936 2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
938 * gst/rtsp-server/rtsp-stream.c:
939 rtsp-stream: Disable multicast loopback for all our sockets
940 On Windows this is a receiver-side setting, on Linux a sender-side setting. As
941 we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
942 loopback setting on the socket... while udpsink does which unfortunately has
943 no effect here on Windows but on Linux.
944 https://bugzilla.gnome.org/show_bug.cgi?id=757488
946 2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
948 * tests/check/gst/stream.c:
949 stream tests: added new tests
950 Test a case when the address pool only contains multicast addresses
951 and the client is requesting unicast udp.
952 Added tests for multicast ports allocation.
953 https://bugzilla.gnome.org/show_bug.cgi?id=757488
955 2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
957 * gst/rtsp-server/rtsp-stream.c:
958 rtsp-stream: Only bind multicast sockets to ANY on Windows
959 On Linux it is still needed to bind to the multicast address
960 to filter out random other packets, while on Windows binding
961 to multicast addresses just fails.
963 2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
965 * gst/rtsp-server/rtsp-stream.c:
966 rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
967 Otherwise we fail to allocate UDP ports if the pool only contains multicast
968 addresses, which is something that used to work before. For unicast addresses
969 if the pool contains none, we just allocate them as if there is no pool at
971 https://bugzilla.gnome.org/show_bug.cgi?id=757488
973 2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
975 * gst/rtsp-server/rtsp-client.c:
976 * gst/rtsp-server/rtsp-stream.c:
977 rtsp-server: Fix indentation
979 2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
981 * gst/rtsp-server/rtsp-stream.c:
982 rtsp-stream: Don't bind the sockets to multicast addresses
983 This works on Linux but fails completely on Windows. You're supposed
984 to bind to ANY and then join the multicast group.
985 https://bugzilla.gnome.org/show_bug.cgi?id=757488
987 === release 1.7.90 ===
989 2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
995 * gst-rtsp-server.doap:
998 2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
1001 Automatic update of common submodule
1002 From b64f03f to 6f2d209
1004 2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
1006 * gst/rtsp-sink/gstrtspclientsink.c:
1007 * tests/check/gst/rtspclientsink.c:
1008 rtspsink: Fix some leaks in rtspclientsink and the unit test.
1009 https://bugzilla.gnome.org/show_bug.cgi?id=762525
1011 2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
1013 * tests/check/gst/media.c:
1014 * tests/check/gst/rtspclientsink.c:
1015 * tests/check/gst/rtspserver.c:
1016 * tests/check/gst/stream.c:
1017 tests: unit test fixes
1018 Removed port allocation test from the media suite.
1019 The port allocation failure is now in the stream suite.
1021 Make sure that the media is suspended after the DESCRIBE request
1022 before reconfiguring the UDP sinks.
1024 In the RECORD case we have to set async property to false
1025 for the appsink element in the test in order to make sure
1026 that the media pipeline doesn't hang in start_preroll().
1027 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1029 2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
1031 * gst/rtsp-server/rtsp-client.c:
1032 * gst/rtsp-server/rtsp-stream.c:
1033 * gst/rtsp-server/rtsp-stream.h:
1034 rtsp-stream: postpone UDP socket allocation until SETUP
1035 Postpone the allocation of the UDP sockets until we know
1036 what transport has been chosen by the client.
1037 Both unicast and multicast UDP sources are created in one
1039 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1041 2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
1043 * gst/rtsp-server/rtsp-stream.c:
1044 rtsp-stream: postpone the creation of the UDP sources
1045 Code refactoring: allocate the UDP ports after the sender and
1046 the reciver parts have been created.
1047 We postpone the creation of the UDP sources until the UDP
1048 ports have been allocated.
1049 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1051 2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
1053 * gst/rtsp-server/rtsp-stream.c:
1054 rtsp-stream: added function for setting UDP sources to PLAYING state
1055 Code refactoring: Introduced a function for setting UDP sources
1057 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1059 2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
1061 * gst/rtsp-server/rtsp-stream.c:
1062 rtsp-stream: added function for creating and configuring UDP sources
1063 Code refactoring: create and configure UDP sources in a separate function.
1064 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1066 2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
1068 * gst/rtsp-server/rtsp-stream.c:
1069 rtsp-stream: added function for RTP/RTCP socket configuration
1070 Code refactoring: configure RTP and RTCP sockets for UDP sinks
1071 in a separate function.
1072 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1074 2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
1076 * gst/rtsp-server/rtsp-stream.c:
1077 rtsp-stream: added function for creating and configuring UDP sinks
1078 Code refactoring: create and configure UDP sinks in a separate function.
1079 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1081 2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
1083 * gst/rtsp-server/rtsp-stream.c:
1084 rtsp-stream: added helper function for creating the sender/receiver parts
1085 Code refactoring: introduced helper function for creating
1086 the receiver and the sender parts of the streaming pipeline.
1087 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1089 2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
1094 === release 1.7.2 ===
1096 2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
1102 * gst-rtsp-server.doap:
1105 2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
1107 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
1108 uninstalled.pc: add support for non libtool build systems
1109 Currently the .la path is provided which requires to use libtool as
1110 mentioned in the GStreamer manual section-helloworld-compilerun.html.
1111 It is fine as long as the application is built using libtool.
1112 So currently it is not possible to compile a GStreamer application
1113 within gst-uninstalled with CMake or other build system different
1115 This patch allows to do the following in gst-uninstalled env:
1116 gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
1117 gstreamer-rtsp-server-1.0)
1118 Previously it required to prepend libtool --mode=link
1119 https://bugzilla.gnome.org/show_bug.cgi?id=720778
1121 2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
1123 * gst/rtsp-sink/gstrtspclientsink.c:
1124 rtspclientsink: remove check for impossible condition
1125 Goto error label checks stream to see if it needs to be unreferenced before
1126 returning, but this goto jumps happens before the stream is ever set, so it
1127 will always be NULL in this error label.
1130 2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
1132 * gst/rtsp-sink/gstrtspclientsink.c:
1133 rtspclientsink: clean switch statements
1134 Coverity demands for fallthrough statements to be clearly commented,
1135 to distinguish from accidental fall throughs. And it also needs all
1136 cases to finish with a break, even if the break is never going to be
1137 executed like in the case of a continue jump.
1141 2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
1143 * tests/check/Makefile.am:
1144 tests: extend the AM_TESTS_ENVIRONMENT from check.mak
1145 To get the CK_DEFAULT_TIMEOUT defined for all tests
1146 Also removes a 120 seconds timeout that was set as default
1147 explicitly in this module
1148 https://bugzilla.gnome.org/show_bug.cgi?id=761472
1150 2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
1154 Automatic update of common submodule
1155 From 86e4663 to b64f03f
1157 2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
1159 * gst/rtsp-server/rtsp-media.c:
1160 rtsp-media: fix state_lock not locked again when preroll fails
1161 https://bugzilla.gnome.org/show_bug.cgi?id=761399
1163 2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
1166 configure: Move plugin specific flags below all the others
1167 They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
1168 -no-undefined. And -no-undefined is required on Windows to build DLLs.
1170 2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
1172 * gst/rtsp-sink/gstrtspclientsink.c:
1173 rtspclientsink: Simplify slightly using new -base API
1174 Use the new Mikey and SDP API in the base plugins libs
1175 to simplify some code.
1176 https://bugzilla.gnome.org/show_bug.cgi?id=758180
1178 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1183 * gst/rtsp-sink/Makefile.am:
1184 * gst/rtsp-sink/gstrtspclientsink.c:
1185 * gst/rtsp-sink/gstrtspclientsink.h:
1186 * gst/rtsp-sink/plugin.c:
1187 * tests/check/Makefile.am:
1188 * tests/check/gst/rtspclientsink.c:
1189 rtspsink: Add rtspclientsink element
1190 Add an rtspclientsink element that accepts streams for which
1191 there is a registered payloader and sends them to
1192 an RTSP server using RECORD.
1193 Sending is synchronised to the pipeline clock. Payload-types
1194 are automatically selected. The 'new-payloader' signal is fired
1195 for custom configuration of payloaders when they are created.
1196 Can now stream a movie like this:
1198 ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
1199 decodebin name=depay1 ! audioconvert ! autoaudiosink )"
1201 gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
1202 queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
1203 https://bugzilla.gnome.org/show_bug.cgi?id=758180
1205 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1207 * gst/rtsp-server/rtsp-stream.c:
1208 * gst/rtsp-server/rtsp-stream.h:
1209 rtsp-stream: Add functions for using rtsp-stream from the client
1210 Add a boolean to indicate that the rtsp-stream is running on the
1211 'client' side of an RTSP connection, for sending streams via
1212 RECORD. In that case, the roles of the client/server ports
1213 in transport setup are swapped.
1214 https://bugzilla.gnome.org/show_bug.cgi?id=758180
1216 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1218 * gst/rtsp-server/rtsp-sdp.c:
1219 * gst/rtsp-server/rtsp-sdp.h:
1220 rtsp-sdp: Add gst_rtsp_sdp_from_stream()
1221 A new function that adds info from a GstRTSPStream into an SDP message.
1222 https://bugzilla.gnome.org/show_bug.cgi?id=758180
1224 2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
1226 * gst/rtsp-server/rtsp-media.c:
1227 rtsp-media: Fix mutex beeing unlocked while they should be locked
1228 https://bugzilla.gnome.org/show_bug.cgi?id=761226
1230 2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
1232 * gst/rtsp-server/rtsp-media-factory.c:
1233 rtsp-media-factory: add missing break in "clock" property setter
1236 2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
1238 * gst/rtsp-server/rtsp-stream.c:
1239 rtsp-stream: fixed assert during update transport
1240 When RTSP server trying update transport during multicast, it throws an
1241 assert. The assert is thrown because it is trying to get the parent of
1242 an non-existing funnel element.
1243 https://bugzilla.gnome.org/show_bug.cgi?id=760150
1245 2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
1247 * gst/rtsp-server/rtsp-permissions.h:
1248 * gst/rtsp-server/rtsp-thread-pool.h:
1249 * gst/rtsp-server/rtsp-token.h:
1250 docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
1251 gtk-doc can handle static inline functions just fine these days,
1252 there's no need for this stuff any more.
1254 2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1256 * gst/rtsp-server/rtsp-media.c:
1257 * gst/rtsp-server/rtsp-sdp.c:
1258 sdp: replace duplicated codes to call new base sdp apis
1259 https://bugzilla.gnome.org/show_bug.cgi?id=745880
1261 2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
1263 * examples/test-netclock.c:
1264 test-netclock: Use the new API to configure a clock directly
1266 2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
1268 * gst/rtsp-server/rtsp-media-factory.c:
1269 * gst/rtsp-server/rtsp-media-factory.h:
1270 * gst/rtsp-server/rtsp-media.c:
1271 * gst/rtsp-server/rtsp-media.h:
1272 rtsp-media: Add API to directly configure a clock on the media pipelines
1274 2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
1276 * gst/rtsp-server/rtsp-media.c:
1277 rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
1279 2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
1281 * gst/rtsp-server/rtsp-media-factory.c:
1282 rtsp-media-factory: Add FIXME for 2.0
1284 2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
1286 * gst/rtsp-server/rtsp-stream.c:
1287 rtsp-stream: Fix indentation
1289 2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1291 * gst/rtsp-server/rtsp-media.c:
1292 rtsp-media: Do not prepare media after media times out
1293 Deferred calls to start_prepare() can be deferred past the point until
1294 which wait_preroll() and by proxy gst_rtsp_media_get_status() is
1295 prepared to wait. Previously there was no lock and no check for this
1296 situation. This meant that a media could be prepared and unprepared
1297 simultaneously by two different threads. Now a lock is in place and a
1298 suitable check is done.
1299 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
1301 2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
1303 * gst/rtsp-server/rtsp-client.c:
1304 * gst/rtsp-server/rtsp-media-factory.c:
1305 * gst/rtsp-server/rtsp-media-factory.h:
1306 * gst/rtsp-server/rtsp-media.c:
1307 * gst/rtsp-server/rtsp-media.h:
1308 rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
1309 Without TEARDOWN it might be desireable to keep the media running and continue
1310 sending data to the client, even if the RTSP connection itself is
1312 Only do this for session medias that have only UDP transports. If there's at
1313 least on TCP transport, it will stop working and cause problems when the
1314 connection is disconnected.
1315 https://bugzilla.gnome.org/show_bug.cgi?id=758999
1317 2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
1322 === release 1.7.1 ===
1324 2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
1330 * gst-rtsp-server.doap:
1333 2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
1336 configure: Make -Bsymbolic check work with clang.
1337 Update the -Bsymbolic check with the version glib has. This version
1339 https://bugzilla.gnome.org/show_bug.cgi?id=759713
1341 2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
1343 * gst/rtsp-server/rtsp-session-pool.c:
1344 rtsp-session-pool: Avoid dollar sign ($) in session ids
1345 Live555 in VLC strips off dollar signs and then gets very confused,
1346 we don't loose too much entropy by just skipping it.
1348 2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
1350 * gst/rtsp-server/rtsp-address-pool.h:
1351 * gst/rtsp-server/rtsp-auth.h:
1352 * gst/rtsp-server/rtsp-client.h:
1353 * gst/rtsp-server/rtsp-media-factory-uri.h:
1354 * gst/rtsp-server/rtsp-media-factory.h:
1355 * gst/rtsp-server/rtsp-media.h:
1356 * gst/rtsp-server/rtsp-mount-points.h:
1357 * gst/rtsp-server/rtsp-permissions.h:
1358 * gst/rtsp-server/rtsp-server.h:
1359 * gst/rtsp-server/rtsp-session-media.h:
1360 * gst/rtsp-server/rtsp-session-pool.h:
1361 * gst/rtsp-server/rtsp-session.h:
1362 * gst/rtsp-server/rtsp-stream-transport.h:
1363 * gst/rtsp-server/rtsp-stream.h:
1364 * gst/rtsp-server/rtsp-thread-pool.h:
1365 * gst/rtsp-server/rtsp-token.h:
1366 rtsp-server: Add g_autoptr() support to all types
1367 https://bugzilla.gnome.org/show_bug.cgi?id=754464
1369 2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
1371 * gst/rtsp-server/rtsp-stream.c:
1372 rtsp-stream: fixed valgrind error
1373 Fixed the valgrind error in unit test. The UDP source created during
1374 gst_rtsp_stream_join_bin() was not released while destroying the rtp
1376 https://bugzilla.gnome.org/show_bug.cgi?id=759010
1378 2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1382 Automatic update of common submodule
1383 From b319909 to 86e4663
1385 2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
1387 * gst/rtsp-server/rtsp-client.c:
1388 rtsp-client: suspend media during setup request
1389 SETUP request from clients needs to suspend the media to clear the
1390 prerolled buffers. Otherwise it will not affect the prerolled buffer
1391 and the prerolled buffers will be incorrect (for example block-size
1392 from setup request will not affect the prerolled buffer unless the
1393 media is suspended).
1394 https://bugzilla.gnome.org/show_bug.cgi?id=758268
1396 2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
1398 * gst/rtsp-server/rtsp-stream.c:
1399 rtsp-stream: create stream pipeline based on transport
1400 Based on the protocol, create the rtsp stream pipeline. If only TCP or
1401 only UDP is set as the transport protocol, it will not add the extra tee
1402 or queue element to the pipeline. Both these elements will be added, if
1403 it supports both TCP and UDP protocols. This improves the pipeline
1404 performance when one protocol is present.
1405 https://bugzilla.gnome.org/show_bug.cgi?id=758179
1407 2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
1409 * gst/rtsp-server/rtsp-stream.c:
1410 rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
1411 Adding them when not needed will start some logic inside rtpbin that might be
1412 problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
1413 would start up a rtpjitterbuffer and behave in weird ways.
1414 We still set up the UDP sources for RTP receiving for a sender media to be
1415 able to receive any packets sent by the client for NAT traversal. They will
1416 all go to a fakesink though.
1417 Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
1418 NO_PREROLL, which will cause deadlocks when seeking the media as it will never
1419 receive ASYNC_DONE after a seek.
1420 https://bugzilla.gnome.org/show_bug.cgi?id=758319
1422 2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
1424 * gst/rtsp-server/rtsp-stream.c:
1425 rtsp-stream: Disable multicast loopback for the multicast udp sources too
1426 On POSIX this setting is for sender sockets, on Windows for receiver sockets.
1427 Previously we were only setting this for sender sockets, which caused looped
1428 back packets to be received on Windows if a multicast transport was used.
1430 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1432 * examples/test-record-auth.c:
1433 * examples/test-record.c:
1434 examples: Actually use the provided port in the record examples
1436 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1438 * examples/test-record-auth.c:
1439 test-record-auth: Add the option to build in TLS support
1441 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1443 * examples/test-auth.c:
1444 test-auth: Use an 'anonymous' user for unauthenticated default
1445 There's a comment on one of the resources that 'user' and 'admin'
1446 shouldn't even be able to see it, but they can if the default
1447 token is 'admin2', since that gives them access anyway.
1449 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1451 * examples/.gitignore:
1452 * examples/Makefile.am:
1453 * examples/test-record-auth.c:
1454 Add test-record-auth example
1456 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1458 * gst/rtsp-server/rtsp-client.c:
1459 * tests/check/gst/client.c:
1460 rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
1462 2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
1464 * gst/rtsp-server/rtsp-server.c:
1465 rtsp-server: Change the logic so we don't pop a NULL context
1466 When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
1467 will sometimes fail. This call is made before any context is pushed
1468 resulting in an attempt to pop a NULL context.
1469 https://bugzilla.gnome.org/show_bug.cgi?id=757949
1471 2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
1473 * tests/check/gst/rtspserver.c:
1474 rtspserver: Add udp-mcast transport SETUP test
1475 Refactor utility functions in the test file so they can handle
1476 more than UDP and TCP as lower transport.
1477 https://bugzilla.gnome.org/show_bug.cgi?id=756969
1479 2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
1481 * gst/rtsp-server/rtsp-stream.c:
1482 rtsp-stream: Always unref return value of gst_object_get_parent()
1483 Fixes a leak of a GstBin in the udp-mcast case.
1484 https://bugzilla.gnome.org/show_bug.cgi?id=756968
1486 2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
1489 Automatic update of common submodule
1490 From b99800a to b319909
1492 2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
1495 Use new GST_ENABLE_EXTRA_CHECKS #define
1496 https://bugzilla.gnome.org/show_bug.cgi?id=756870
1498 2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
1501 Automatic update of common submodule
1502 From 6babecd to b99800a
1504 2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
1507 Update GLib dependency to 2.40.0
1509 2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1511 * examples/test-mp4.c:
1512 * gst/rtsp-server/rtsp-stream.c:
1513 stream: listen to sender ssrc signals
1514 https://bugzilla.gnome.org/show_bug.cgi?id=746747
1516 2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
1519 common: update for new suppression
1520 Makes check-valgrind pass with glib 2.46
1522 2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1524 * gst/rtsp-server/rtsp-media.c:
1525 rtsp-media: Take reference to media that will be prepared
1526 default_prepare() takes a transfer-none reference GstRTSPMedia object.
1527 Later on a g_idle_source_new() is created and a pointer to the media
1528 object is passed as user data. If the media is freed before the idle
1529 source is dispatched the media object pointer is invalid, but the idle
1530 source callback expects it to still be valid. To fix this a reference to
1531 the media object is taken when registering the source callback function
1532 and a corresponding release of the reference is done when the souce is
1534 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
1536 2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
1538 * examples/test-launch.c:
1539 * examples/test-mp4.c:
1540 * examples/test-ogg.c:
1541 * examples/test-record.c:
1542 * examples/test-uri.c:
1543 rtsp-server: Fix memory leaks when context parse fails
1544 When g_option_context_parse fails, context and error variables are not getting free'd
1545 which results in memory leaks. Free'ing the same.
1546 And replacing g_error_free with g_clear_error, which checks if the error being passed
1547 is not NULL and sets the variable to NULL on free'ing.
1548 https://bugzilla.gnome.org/show_bug.cgi?id=753863
1550 2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
1555 === release 1.6.0 ===
1557 2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
1563 * gst-rtsp-server.doap:
1566 === release 1.5.91 ===
1568 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
1574 * gst-rtsp-server.doap:
1577 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
1579 * docs/libs/gst-rtsp-server-sections.txt:
1580 * gst/rtsp-server/rtsp-stream.c:
1581 stream: fix docs for recently-added get/set_buffer_size API
1582 https://bugzilla.gnome.org/show_bug.cgi?id=749095
1584 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
1586 * gst/rtsp-server/rtsp-media.c:
1587 rtsp-media: Don't crash on encrypted RTX SDP
1588 In parse_keymgmt(), don't mutate the input string that's been passed
1589 as const, especially since we might need the original value again if
1590 the same key info applies to multiple streams (RTX, for example).
1591 https://bugzilla.gnome.org/show_bug.cgi?id=754753
1593 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
1595 * examples/test-mp4.c:
1596 test-mp4: Support filenames with spaces in them. Error out on too few arguments
1598 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
1600 * examples/test-record.c:
1601 test-record: Check parameter count and print out help
1602 If no launch pipeline was supplied, print out some help
1604 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
1606 * gst/rtsp-server/rtsp-media.c:
1607 * gst/rtsp-server/rtsp-stream.c:
1608 * gst/rtsp-server/rtsp-stream.h:
1609 rtsp-stream: Implement UDP buffer size setting.
1610 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
1612 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
1613 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
1615 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
1617 * gst/rtsp-server/rtsp-media.h:
1618 rtsp-media: Fix small typo causing gtk-doc to complain
1620 === release 1.5.90 ===
1622 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
1628 * gst-rtsp-server.doap:
1631 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1633 * gst/rtsp-server/rtsp-media-factory.c:
1634 media-factory: get port number through gst_rtsp_url_get_port
1635 https://bugzilla.gnome.org/show_bug.cgi?id=753473
1637 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
1639 * tests/check/gst/media.c:
1640 media-test: Removing unnecessary assertion
1641 https://bugzilla.gnome.org/show_bug.cgi?id=753385
1643 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1645 * gst/rtsp-server/rtsp-server.c:
1646 Document that source keeps a ref on server until it's destroyed
1647 https://bugzilla.gnome.org/show_bug.cgi?id=749227
1649 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1651 * tests/check/gst/media.c:
1652 media-test: Test for multiple dynamic payload
1653 https://bugzilla.gnome.org/show_bug.cgi?id=753385
1655 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1657 * gst/rtsp-server/rtsp-media.c:
1658 media: Only add fakesink once per pipeline
1659 The intention is to prevent going PLAYING state before pads are created.
1660 If there was mutilple dynamic payload, it would leak few fakesink and
1661 actually prevent from ever reaching playing state.
1662 https://bugzilla.gnome.org/show_bug.cgi?id=753385
1664 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1666 * gst/rtsp-server/rtsp-media.c:
1667 Revert "rtsp-media: Only add 1 fakesink per pipeline"
1668 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
1670 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1672 * gst/rtsp-server/rtsp-media.c:
1673 rtsp-media: Only add 1 fakesink per pipeline
1674 There should be only one fakesink per pipeline, not per dynpay. This
1675 would lead to element naming clash.
1677 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
1679 * gst/rtsp-server/rtsp-media.c:
1680 rtsp-media: assertion error due to wrong condition check
1681 In media to caps function, reserved_keys array is being used for variable i,
1682 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
1683 changed it to variable j
1684 https://bugzilla.gnome.org/show_bug.cgi?id=753009
1686 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
1688 * gst/rtsp-server/rtsp-media.c:
1689 rtsp-media: Strip keys from the fmtp that we use internally in our caps
1690 Skip keys from the fmtp, which we already use ourselves for the
1691 caps. Some software is adding random things like clock-rate into
1692 the fmtp, and we would otherwise here set a string-typed clock-rate
1693 in the caps... and thus fail to create valid RTP caps
1694 https://bugzilla.gnome.org/show_bug.cgi?id=753009
1696 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1698 * gst/rtsp-server/rtsp-thread-pool.c:
1699 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
1700 https://bugzilla.gnome.org/show_bug.cgi?id=752640
1702 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
1705 Automatic update of common submodule
1706 From f74b2df to 9aed1d7
1708 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
1713 === release 1.5.2 ===
1715 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
1721 * gst-rtsp-server.doap:
1724 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
1726 * gst/rtsp-server/rtsp-client.c:
1727 * gst/rtsp-server/rtsp-client.h:
1728 * tests/check/gst/client.c:
1729 rtsp-client: allow application to decide what requirements are supported
1730 Add "check-requirements" signal and vfunc to allow application
1731 (and subclasses) to check the requirements.
1732 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
1733 https://bugzilla.gnome.org/show_bug.cgi?id=749417
1735 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1738 Automatic update of common submodule
1739 From 6015d26 to f74b2df
1741 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
1743 * gst/rtsp-server/rtsp-media.c:
1744 rtsp-media: Always use real payloader when creating streams
1745 A bin that contains the real payloader might be used as payloader. In this
1746 case we have to get the real payloader for the various properties it provides.
1747 Example use cases for this are bins that payload some media and then have
1748 additional elements that add metadata or RTP extension headers to the stream.
1749 https://bugzilla.gnome.org/show_bug.cgi?id=750800
1751 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
1753 * examples/test-netclock-client.c:
1754 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
1756 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
1758 * examples/test-netclock-client.c:
1759 * examples/test-netclock.c:
1760 test-netclock: Use new ntp-time-source property on rtpbin
1761 Select the clock time to be used as NTP time source. This allows proper
1762 synchronization between receivers, independent of sharing base times, and just
1763 requires them to use the same clock.
1765 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
1767 * examples/test-netclock-client.c:
1768 * examples/test-netclock.c:
1769 test-netclock: Setting the same base time on sender and receiver is not necessary
1770 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
1772 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1774 * gst/rtsp-server/rtsp-stream.c:
1775 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
1776 https://bugzilla.gnome.org/show_bug.cgi?id=750764
1778 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1780 * docs/libs/gst-rtsp-server.types:
1781 docs: add missing types
1782 https://bugzilla.gnome.org/show_bug.cgi?id=750764
1784 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1786 * docs/libs/gst-rtsp-server-sections.txt:
1787 docs: add missing apis
1788 https://bugzilla.gnome.org/show_bug.cgi?id=750764
1790 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
1792 * examples/test-netclock-client.c:
1793 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
1795 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1797 * docs/libs/gst-rtsp-server-sections.txt:
1798 * gst/rtsp-server/rtsp-auth.c:
1799 * gst/rtsp-server/rtsp-auth.h:
1800 GstRTSPAuth: Add client certificate authentication support
1801 https://bugzilla.gnome.org/show_bug.cgi?id=750471
1803 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
1805 * examples/test-netclock-client.c:
1806 test-netclock-client: Use new GstClock API to wait for clock synchronization
1808 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
1810 * examples/test-netclock-client.c:
1811 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
1812 A mainloop is needed to get glimagesink to display something on OSX, and
1813 the source-setup signal just makes things a little bit easier.
1815 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
1818 Automatic update of common submodule
1819 From d9a3353 to 6015d26
1821 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
1824 Automatic update of common submodule
1825 From d37af32 to d9a3353
1827 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
1830 Automatic update of common submodule
1831 From 21ba2e5 to d37af32
1833 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
1836 Automatic update of common submodule
1837 From c408583 to 21ba2e5
1839 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
1841 * docs/libs/Makefile.am:
1842 docs: remove variables that we define in the snippet from common
1843 This is syncing our Makefile.am with upstream gtkdoc.
1845 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
1848 Automatic update of common submodule
1849 From 44a3517 to c408583
1851 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
1856 === release 1.5.1 ===
1858 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
1864 * gst-rtsp-server.doap:
1867 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
1869 * gst/rtsp-server/rtsp-client.c:
1870 rtsp-client: No flush during Teardown.
1871 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
1872 backlog is empty it can happen that just a part of a message will be
1873 sent and rest is in backlog queue. If then flush during teardown
1874 just a part of message will be sent.This can lead to client miss
1875 teardown response since it expect to get the last part of message.
1876 The flushing during teardown was introduced to fix a deadlock that now
1877 is fixed more generally in handle_request by temporary setting backlog
1879 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
1881 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
1883 * tests/check/Makefile.am:
1884 tests: Use AM_TESTS_ENVIRONMENT
1885 Needed by the new automake test runner and the
1886 current version of the common submodule.
1888 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
1890 * gst/rtsp-server/rtsp-media.h:
1891 * gst/rtsp-server/rtsp-stream.h:
1892 rtsp-server: Use single-include rtsp header to make sure we get all definitions
1894 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
1896 * gst/rtsp-server/rtsp-media.c:
1897 rtsp-media: Mark some more functions static
1899 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
1901 * gst/rtsp-server/rtsp-media.c:
1902 rtsp-media: Only unblock the media in suspend() when actually changing the state
1903 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
1905 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
1907 * examples/test-video-rtx.c:
1908 examples: Use AVPF profile for the RTX example
1910 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
1912 * gst/rtsp-server/rtsp-sdp.c:
1913 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
1915 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1917 * gst/rtsp-server/rtsp-stream.c:
1918 rtsp-stream: get valid clock-rate from last-sample
1919 clock-rate in last-sample's caps is integer, not unsigned.
1920 To get this value properly, variable needs to be type-casted to int.
1921 https://bugzilla.gnome.org/show_bug.cgi?id=747614
1923 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
1927 autogen.sh: only run autopoint if gettext requested in configure.ac
1928 Not just because there happens to be a po directory.
1929 https://bugzilla.gnome.org/show_bug.cgi?id=748058
1931 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
1934 Revert "configure.ac: uncomment gettext version setup"
1935 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
1936 We don't need a gettext setup here and there's no po
1937 directory either, so no reason why autopoint would be
1938 run in the first place.
1939 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
1941 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
1943 * examples/test-multicast.c:
1944 * examples/test-multicast2.c:
1945 * examples/test-sdp.c:
1946 * examples/test-video-rtx.c:
1947 * examples/test-video.c:
1948 * tests/test-cleanup.c:
1949 * tests/test-reuse.c:
1950 Fix timeout function signatures across tests and examples
1952 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
1954 * tests/check/Makefile.am:
1955 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
1956 Make sure the test environment is set up.
1957 https://bugzilla.gnome.org//show_bug.cgi?id=747624
1959 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
1962 configure: bump automake requirement to 1.14 and autoconf to 2.69
1963 This is only required for builds from git, people can still
1964 build tarballs if they only have older autotools.
1965 https://bugzilla.gnome.org//show_bug.cgi?id=747624
1967 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
1970 configure.ac: uncomment gettext version setup
1971 Fixes autogen.sh. It would run autopoint, which would complain
1972 that it could not find the gettext version in configure.ac.
1973 https://bugzilla.gnome.org/show_bug.cgi?id=748058
1975 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1977 * examples/test-video-rtx.c:
1978 test-video-rtx: set exact payload type to PCMA payloader
1979 Setting wrong payload type causes failure to do retransmission through audio stream
1980 https://bugzilla.gnome.org/show_bug.cgi?id=747839
1982 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1984 * gst/rtsp-server/rtsp-media.c:
1985 * gst/rtsp-server/rtsp-stream.c:
1986 * gst/rtsp-server/rtsp-stream.h:
1987 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
1988 Because of duplicated g_signal_connect for request-aux-sender signal,
1989 wrong stream pointer is passed to the signal handler.
1990 Instead of passing each stream, pass stream array and get the relevant stream.
1991 https://bugzilla.gnome.org/show_bug.cgi?id=747839
1993 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
1997 Update autogen.sh to latest version from common
1998 Fixes build after aclocal_check etc. helpers have been removed.
2000 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
2003 Automatic update of common submodule
2004 From bc76a8b to c8fb372
2006 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
2008 * gst/rtsp-server/rtsp-stream.c:
2009 rtsp-stream: Limit the queues to 1 buffer
2010 We only need them to be able to pre-roll, queueing up more data here
2011 is only going to harm latency and memory usage.
2013 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
2015 * gst/rtsp-server/rtsp-stream.c:
2016 rtsp-stream: Update comment and ASCII art to the latest code
2017 We have a queue in front of the udpsink too to prevent the pipeline from
2020 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
2022 * gst/rtsp-server/rtsp-stream.c:
2023 rtsp-media: Properly return first rtptime
2024 Instead we where returning first GstBuffer timestamp. This would result
2025 in clock skew and unwanted behaviour in RTSP playback.
2026 https://bugzilla.gnome.org/show_bug.cgi?id=746479
2028 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
2030 * gst/rtsp-server/rtsp-stream.c:
2031 rtsp-stream: Don't leave buffer mapped
2032 If the seq is NULL, the RTP buffer was left mapped. We should always
2035 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
2040 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
2042 * gst/rtsp-server/rtsp-media-factory.c:
2043 * tests/check/gst/client.c:
2044 Fix double semicolons
2046 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
2048 * gst/rtsp-server/rtsp-stream.c:
2049 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
2050 This gives more accurate values than asking the payloader. There might be
2051 queueing happening between the payloader and the sink.
2052 https://bugzilla.gnome.org/show_bug.cgi?id=745704
2054 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
2056 * gst/rtsp-server/rtsp-media.c:
2057 rtsp-media: Don't seek for PLAY if the position will not change
2058 https://bugzilla.gnome.org/show_bug.cgi?id=745704
2060 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
2062 * gst/rtsp-server/rtsp-media.c:
2063 rtsp-media: Don't include payload type in the caps for framesize
2064 When the sdp media attribute framesize are converted to caps
2065 the <payload> should not be included.
2066 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
2067 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
2069 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
2071 * gst/rtsp-server/rtsp-sdp.c:
2072 rtsp-sdp: add payload type to the sdp framesize attribute
2073 The sdp framesize attribute is desribed in RFC6064. It is specified
2074 for payloading of H263 and has the following form
2075 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
2076 should be added to the caps in a payloader and the <payload type> should
2077 be added by the rtsp-server.
2078 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
2080 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
2082 * examples/test-uri.c:
2083 examples: test-uri: fix tainted variable
2084 Insignificant but this keeps Coverity happy.
2087 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
2089 * examples/.gitignore:
2090 * examples/Makefile.am:
2091 * examples/test-netclock-client.c:
2092 * examples/test-netclock.c:
2093 examples: Add a simple example of network synch for live streams.
2094 An example server and client that works for synchronising live streams
2095 only - as it can't support pause/play.
2097 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
2099 * gst/rtsp-server/rtsp-media-factory.c:
2100 * gst/rtsp-server/rtsp-media-factory.h:
2101 rtsp-media-factory: Add functions to set/get the media gtype
2102 Allow specifying the GType of a GstRtspMedia subclass to create
2103 as a simpler way to get the factory to create a custom
2104 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
2106 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
2108 * gst/rtsp-server/rtsp-media.c:
2109 rtsp-media: fix double unlock in _get_buffer_size()
2110 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
2111 because of double g_mutex_unlock () usage.
2112 https://bugzilla.gnome.org/show_bug.cgi?id=745434
2114 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
2116 * gst/rtsp-server/rtsp-session-pool.c:
2117 * gst/rtsp-server/rtsp-session.c:
2118 * gst/rtsp-server/rtsp-session.h:
2119 rtsp-session: Use monotonic time for RTSP session timeout
2120 Changed RTSP session timeout handling to monotonic time
2121 and deprecating the API for current system time.
2122 This fixes timeouts when the system time changes.
2123 https://bugzilla.gnome.org/show_bug.cgi?id=743346
2125 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
2127 * gst/rtsp-server/rtsp-client.c:
2128 * gst/rtsp-server/rtsp-media.c:
2129 rtsp-client: Only error out in PLAY if seeking actually failed
2130 If the media was just not seekable, we continue from whatever position we are
2131 and let the client decide if that is what is wanted or not.
2132 Only if the actual seek failed, we can't really recover and should error out.
2134 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
2136 * gst/rtsp-server/rtsp-stream.c:
2137 rtsp-stream: Add necessary queues between tee and multiudpsink
2138 https://bugzilla.gnome.org/show_bug.cgi?id=744379
2140 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
2142 * gst/rtsp-server/rtsp-client.c:
2143 * gst/rtsp-server/rtsp-media.c:
2144 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
2145 Instead error out properly the same way as if the SEEKING query already
2148 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
2150 * gst/rtsp-server/rtsp-stream.h:
2151 rtsp-stream: minor code formatting fix
2153 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
2155 * gst/rtsp-server/rtsp-media.c:
2156 rtsp-media: fix logic for collect_streams
2157 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
2158 all streams it knows if it got any, and can check if the transport mode is OK.
2161 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
2163 * gst/rtsp-server/rtsp-media.c:
2164 rtsp-media: Don't set the transport mode based on what elements we find
2165 Just print a warning if the one that was set before disagrees with what
2166 elements we found. It must already be set to something before as this
2167 function is called after we received the SDP from ANNOUNCE in RECORD mode,
2168 and we would reject ANNOUNCE if the RECORD flag was not set.
2170 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
2172 * tests/check/gst/rtspserver.c:
2173 tests: rtspserver: rename shadowed variable
2174 We have two different 'sink' variables here,
2175 rename one of them for clarity.
2177 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
2179 * gst/rtsp-server/rtsp-client.c:
2180 rtsp-client: fix awkward if clause
2182 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
2184 * examples/test-uri.c:
2185 examples: test-uri: improve uri argument handling and accept file names
2186 Print an error if the argument passed is not a URI and can't
2187 be converted into one, or no arguments have been provided.
2189 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
2191 * examples/test-uri.c:
2192 examples: test-uri: don't remove mount point after 10 seconds
2193 It's very irritating when trying to test stuff repeatedly
2194 and serves no real purpose other than showing that it can
2197 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
2199 * examples/.gitignore:
2200 examples: add new test-record to .gitignore
2202 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
2204 * examples/test-record.c:
2205 * gst/rtsp-server/rtsp-client.c:
2206 * gst/rtsp-server/rtsp-media-factory.c:
2207 * gst/rtsp-server/rtsp-media-factory.h:
2208 * gst/rtsp-server/rtsp-media.c:
2209 * gst/rtsp-server/rtsp-media.h:
2210 * tests/check/gst/rtspserver.c:
2211 rtsp-media: Use flags to distinguish between PLAY and RECORD media
2213 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
2215 * examples/test-record.c:
2216 test-record: Set latency for playback-style example to 2s instead of 200ms
2218 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
2220 * tests/check/gst/rtspserver.c:
2221 tests: add some unit tests for ANNOUNCE and RECORD
2222 https://bugzilla.gnome.org/show_bug.cgi?id=743175
2224 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
2226 * gst/rtsp-server/rtsp-client.c:
2227 rtsp-client: fix a couple of leaks in handle_announce
2229 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
2231 * gst/rtsp-server/rtsp-media-factory.c:
2232 * gst/rtsp-server/rtsp-media-factory.h:
2233 * gst/rtsp-server/rtsp-media.c:
2234 * gst/rtsp-server/rtsp-media.h:
2235 rtsp-media: Expose latency setting for setting the rtpbin latency
2237 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
2239 * examples/test-record.c:
2240 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
2242 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
2244 * gst/rtsp-server/rtsp-stream.c:
2245 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
2247 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
2249 * examples/Makefile.am:
2250 * examples/test-record.c:
2251 * gst/rtsp-server/rtsp-client.c:
2252 * gst/rtsp-server/rtsp-client.h:
2253 * gst/rtsp-server/rtsp-media-factory.c:
2254 * gst/rtsp-server/rtsp-media-factory.h:
2255 * gst/rtsp-server/rtsp-media.c:
2256 * gst/rtsp-server/rtsp-media.h:
2257 * gst/rtsp-server/rtsp-session-media.c:
2258 * gst/rtsp-server/rtsp-stream.c:
2259 * gst/rtsp-server/rtsp-stream.h:
2260 Add initial support for RECORD
2261 We currently only support media that is RECORD or PLAY only, not both at once.
2262 https://bugzilla.gnome.org/show_bug.cgi?id=743175
2264 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
2266 * gst/rtsp-server/rtsp-stream.c:
2267 rtsp-stream: RTCP and RTP transport cache cookies seperated
2268 RTCP packets were not sent because the same tr_cache_cookie was used for
2269 both RTP and RTCP. So only one of the tr_cache lists were populated
2270 depending on which one was sent first. If the tr_cache list is not
2271 populated then no packets can be sent. Most often this happened to be
2272 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
2273 resulted in both the tr_cache_lists to be populated regardless of which
2275 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
2277 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
2279 * gst/rtsp-server/rtsp-stream.c:
2280 rtsp-stream: fix false compiler warning
2281 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
2283 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
2285 * gst/rtsp-server/rtsp-client.c:
2286 rtsp-client: log interleaved data received
2288 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
2290 * gst/rtsp-server/rtsp-client.c:
2291 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
2293 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
2295 * gst/rtsp-server/rtsp-client.c:
2296 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
2298 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
2300 * gst/rtsp-server/rtsp-client.c:
2301 rtsp-client: Use a random session ID in the SDP
2302 RFC4566 Section 5.2 says that it should make the username, session id,
2303 nettype, addrtype and unicast address tuple globally unique. Always using
2304 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
2305 Instead let's create a 64 bit random number, which at least brings us
2306 closer to the goal of global uniqueness.
2307 https://tools.ietf.org/html/rfc4566#section-5.2
2309 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
2311 * examples/test-launch.c:
2312 * examples/test-mp4.c:
2313 * examples/test-ogg.c:
2314 * examples/test-uri.c:
2315 examples: Don't call gst_init() and gst_get_option_group()
2316 The latter calls the former at the appropriate time.
2318 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
2320 * gst/rtsp-server/rtsp-client.c:
2321 rtsp-client: Drop trailing \0 of RTSP DATA messages
2322 We add a trailing \0 in GstRTSPConnection to make parsing of
2323 string message bodies easier (e.g. the SDP from DESCRIBE) but
2324 for actual data this means we have to drop it or otherwise
2325 create invalid data.
2327 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
2329 * gst/rtsp-server/rtsp-stream.c:
2330 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
2331 Fixes crash when two threads access handle_new_sample() at the same
2332 time, one for RTP, one for RTCP.
2333 Otherwise, when iterating over the transports cache, it might be modified by
2334 another thread at the same time if the transports cookie has changed.
2335 https://bugzilla.gnome.org/show_bug.cgi?id=742954
2337 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
2339 * gst/rtsp-server/rtsp-stream.c:
2340 rtsp-stream: Set format=TIME on our app sources for TCP
2342 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
2344 * gst/rtsp-server/rtsp-session-pool.c:
2345 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
2346 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
2347 RFC 2326 states that session IDs may consist of alphanumeric as well as
2348 the safe characters $-_.+ -- N.B. the percent character is not allowed.
2349 Previously the session ID was URI-escaped, this meant that any character
2350 which was not alphanumeric or any of the characters +-._~ would be
2351 percent encoded. While the RFC (surprisingly) mentions that linear white
2352 space in session IDs should be URI-escaped, it does not say anything
2353 about other characters. Moreover no white space is allowed in the
2354 session ID. Finally the percent character which is the result of
2355 URI-escaping is not allowed in a session ID.
2356 So there is no reason to do any URI-escaping, and now it is removed.
2357 https://bugzilla.gnome.org/show_bug.cgi?id=742869
2359 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
2362 Automatic update of common submodule
2363 From f2c6b95 to bc76a8b
2365 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
2368 Fix 'make check' from top-level directory
2370 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2372 * examples/test-launch.c:
2373 * examples/test-mp4.c:
2374 * examples/test-ogg.c:
2375 * examples/test-uri.c:
2376 examples: Add command-line parsing and take a 'port' argument
2377 This allows users to run multiple servers on different ports for testing.
2378 Only done for examples that actually take arguments and hence are capable of
2379 outputting different streams for each instance on each port.
2380 https://bugzilla.gnome.org/show_bug.cgi?id=742115
2382 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
2384 * gst/rtsp-server/rtsp-client.c:
2385 * gst/rtsp-server/rtsp-client.h:
2386 rtsp-client: Add a send_message default signal handler
2387 This allows subclasses to easily hook into the response sending
2388 mechanism without doing everything from a signal, which seems
2389 awkward from subclasses.
2391 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
2394 Automatic update of common submodule
2395 From ef1ffdc to f2c6b95
2397 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2401 configure: add --disable-examples switch
2402 https://bugzilla.gnome.org/show_bug.cgi?id=741678
2404 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
2406 * examples/.gitignore:
2407 * examples/Makefile.am:
2408 * examples/test-video-rtx.c:
2409 examples: add a retransmisison example implementing RFC4588
2410 Currently only SSRC-multiplexed rtx streams are supported
2412 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
2414 * gst/rtsp-server/rtsp-stream.c:
2415 rtsp-stream: Fix some minor memory leaks
2417 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
2419 * gst/rtsp-server/rtsp-media.c:
2420 rtsp-media: Some minor cleanup
2422 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
2424 * gst/rtsp-server/rtsp-stream.c:
2425 rtsp-stream: Fix compiler warnings
2426 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
2427 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
2429 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
2430 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
2433 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
2435 * docs/libs/gst-rtsp-server-sections.txt:
2436 * gst/rtsp-server/rtsp-media-factory.c:
2437 * gst/rtsp-server/rtsp-media-factory.h:
2438 * gst/rtsp-server/rtsp-media.c:
2439 * gst/rtsp-server/rtsp-media.h:
2440 * gst/rtsp-server/rtsp-sdp.c:
2441 * gst/rtsp-server/rtsp-stream.c:
2442 * gst/rtsp-server/rtsp-stream.h:
2443 media: implement ssrc-multiplexed retransmission support
2444 based off RFC 4588 and the server-rtpaux example in -good
2446 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
2448 * gst/rtsp-server/rtsp-client.c:
2449 * gst/rtsp-server/rtsp-stream-transport.c:
2450 * gst/rtsp-server/rtsp-stream.c:
2451 rtsp: Ref transports in hash table.
2452 Also ref streams for transports.
2453 This solves a crash when reciving a rtcp after teardown but before
2455 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
2457 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
2460 Automatic update of common submodule
2461 From 7bb2bce to ef1ffdc
2463 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
2465 * gst/rtsp-server/rtsp-client.c:
2466 client: refactor cleanup of cached media
2468 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
2470 * tests/check/gst/client.c:
2472 The session leak is now fixed, lets remove those FIXME comments.
2474 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
2476 * tests/check/gst/rtspserver.c:
2477 tests: Test to setup two sessions on one connection
2478 https://bugzilla.gnome.org/show_bug.cgi?id=739112
2480 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
2482 * tests/check/gst/rtspserver.c:
2483 tests: Test setup with tcp transport
2484 https://bugzilla.gnome.org/show_bug.cgi?id=739112
2486 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
2488 * gst/rtsp-server/rtsp-client.c:
2489 client: Configure transport after creating session media
2490 The default implementation of configure_client_transport() in
2491 rtsp-client uses the session media when it chooses channels for
2492 interleaved traffic.
2493 https://bugzilla.gnome.org/show_bug.cgi?id=739112
2495 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
2497 * gst/rtsp-server/rtsp-client.c:
2498 * gst/rtsp-server/rtsp-session-media.c:
2499 client: Stop caching media in client when doing setup
2500 If the media has been managed by a session media, it should not be
2501 cached in the client any longer. The GstRTSPSessionMedia object is now
2502 responsible for unpreparing the GstRTSPMedia object using
2503 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
2505 https://bugzilla.gnome.org/show_bug.cgi?id=739112
2507 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2509 * gst/rtsp-server/rtsp-stream.c:
2510 rtsp-stream: unref srtp decoder when leaving bin
2511 https://bugzilla.gnome.org/show_bug.cgi?id=739481
2513 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2515 * gst/rtsp-server/rtsp-client.c:
2516 rtsp-client: mikey memory leaks
2517 https://bugzilla.gnome.org/show_bug.cgi?id=739383
2519 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
2522 Automatic update of common submodule
2523 From 84d06cd to 7bb2bce
2525 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
2528 Parallelise 'make check-valgrind'
2530 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
2533 Automatic update of common submodule
2534 From a8c8939 to 84d06cd
2536 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
2539 Automatic update of common submodule
2540 From 36388a1 to a8c8939
2542 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
2544 * gst/rtsp-server/rtsp-media.c:
2545 rtsp-media: deactivate media when shutting down from paused
2546 This was only done when going directly from playing.
2547 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
2549 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2551 * gst/rtsp-server/rtsp-client.c:
2552 * gst/rtsp-server/rtsp-context.h:
2553 rtsp-client: add stream transport to context
2554 We add the stream transport to the context so we can get the configured
2555 client stream transport in the setup request signal.
2556 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
2558 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2560 * gst/rtsp-server/rtsp-stream.c:
2561 stream: release lock even not all transports have been removed
2562 We don't want to keep the lock even we return FALSE because not all the
2563 transports have been removed. This could lead into a deadlock.
2564 https://bugzilla.gnome.org/show_bug.cgi?id=737797
2566 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
2568 * gst/rtsp-server/rtsp-sdp.c:
2569 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
2570 These were renamed in GstRTPBasePayload in 1.0
2572 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2574 * gst/rtsp-server/rtsp-client.c:
2575 client: set session media to NULL without the lock
2576 We need to set session medias to NULL without the client lock otherwise
2577 we can end up in a deadlock if another thread is waiting for the lock
2578 and media unprepare is also waiting for that thread to end.
2579 https://bugzilla.gnome.org/show_bug.cgi?id=737690
2581 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
2583 * gst/rtsp-server/rtsp-media.c:
2584 rtsp-media: Set state to UNPREPARING in all cases
2586 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
2588 * gst/rtsp-server/rtsp-media.c:
2589 media: set state to unpreparing when unprepare is initiated
2590 https://bugzilla.gnome.org/show_bug.cgi?id=737675
2592 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
2594 * gst/rtsp-server/rtsp-client.c:
2595 rtsp-client: Remove backlog limit while processings requests
2596 If the backlog limit is kept two cases of deadlocks may be
2597 encountered when streaming over TCP. Without the backlog
2598 limit this deadlocks can not happen, at the expence of
2600 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
2602 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
2604 * gst/rtsp-server/rtsp-client.c:
2605 rtsp-client: do not free main context before rtsp watch
2606 https://bugzilla.gnome.org/show_bug.cgi?id=737110
2608 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
2610 * tests/check/gst/rtspserver.c:
2611 tests: Extend unit test timeout to accomodate for valgrind
2612 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
2614 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
2616 * gst/rtsp-server/rtsp-client.c:
2617 * gst/rtsp-server/rtsp-session.c:
2618 * gst/rtsp-server/rtsp-stream-transport.c:
2619 rtsp-*: Treat sending packets to clients as keepalive
2620 As long as gst-rtsp-server can successfully send RTP/RTCP data to
2621 clients then the client must be reading. This change makes the server
2622 timeout the connection if the client stops reading.
2623 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
2625 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
2627 * gst/rtsp-server/rtsp-client.c:
2628 rtsp-client: Allow backlog to grow while expiring session
2629 Allow the send backlog in the RTSP watch to grow to unlimited size while
2630 attempting to bring the media pipeline to NULL due to a session
2631 expiring. Without this change the appsink element cannot change state
2632 because it is blocked while rendering data in the new_sample callback.
2633 This callback will block until it has successfully put the data into the
2634 send backlog. There is a chance that the send backlog is full at this
2635 point which means that the callback may block for a long time, possibly
2636 forever. Therefore the media pipeline may also be prevented from
2637 changing state for a long time.
2638 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
2640 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
2642 * gst/rtsp-server/rtsp-client.c:
2643 rtsp-client: Make old compilers happy
2644 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
2645 Just in case that guint8 doesn't fit in a pointer. Just in case ...
2647 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
2649 * gst/rtsp-server/rtsp-client.c:
2650 client: raise the backlog limits before pausing
2651 We need to raise the backlog limits before pausing the pipeline or else
2652 the appsink might be blocking in the render method in wait_backlog() and
2653 we would deadlock waiting for paused.
2654 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
2656 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
2658 * gst/rtsp-server/rtsp-client.c:
2659 client: make define for the WATCH_BACKLOG
2660 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
2662 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
2664 * gst/rtsp-server/rtsp-client.c:
2665 client: simplify session transport handling
2666 link/unlink of the transport in a session was done to keep track of all
2667 TCP transports and to send RTP/RTCP data to the streams. We can simplify
2668 that by putting all the TCP transports in a hashtable indexed with the
2670 We also don't need to link/unlink the transports when we pause/resume
2671 the streams. The same effect is already achieved when we pause/play the
2672 media. Indeed, when we pause the media, the transport is removed from
2673 the media and the callbacks will not be called anymore.
2674 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
2676 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
2678 * gst/rtsp-server/rtsp-stream-transport.c:
2679 * gst/rtsp-server/rtsp-stream-transport.h:
2680 stream-transport: make method to handle received data
2681 Make a method to handle the data received on a channel. It sends the
2682 data to the stream of the transport on the RTP or RTCP pads based on
2685 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
2687 * examples/test-mp4.c:
2688 test: add example of dumping RTCP reports
2690 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
2692 * gst/rtsp-server/rtsp-media.c:
2693 * gst/rtsp-server/rtsp-stream.c:
2694 * gst/rtsp-server/rtsp-stream.h:
2695 rtsp-media: Make sure that sequence numbers are monotonic after pause
2696 The sequence number is not monotonic for RTP packets after pause. The
2697 reason is basepayloader generates a randon sequence number when the
2698 pipeline goes from ready to pause. With this fix generation of sequence
2699 number will be monotonic when going from pause to play request.
2700 https://bugzilla.gnome.org/show_bug.cgi?id=736017
2702 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
2704 * gst/rtsp-server/rtsp-client.c:
2705 rtsp-client: Protect saved clients watch with a mutex
2706 Fixes a crash when close() is called while merging clients
2707 in handle_tunnel(). In that case close() would destroy the
2708 watch while it is still being used in handle_tunnel().
2709 https://bugzilla.gnome.org/show_bug.cgi?id=735570
2711 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
2713 * gst/rtsp-server/rtsp-stream.c:
2714 rtsp-stream: Remove the multicast group udp sources when removing from the bin
2716 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
2718 * gst/rtsp-server/rtsp-media.c:
2719 * gst/rtsp-server/rtsp-stream.c:
2720 * gst/rtsp-server/rtsp-stream.h:
2721 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
2722 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
2723 seeking and will always continue counting the time. This leads to
2724 the NPT after a backwards seek to be something completely different
2725 to the actual seek position.
2726 https://bugzilla.gnome.org/show_bug.cgi?id=732644
2728 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
2730 * examples/test-appsrc.c:
2731 examples: fix another reference leak
2732 gst_rtsp_media_get_element() returns a new ref.
2734 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2736 * examples/test-appsrc.c:
2737 examples: unref element after usage
2738 gst_bin_get_by_name_recurse_up() returns an element
2739 reference that must be unreffed after usage.
2740 https://bugzilla.gnome.org/show_bug.cgi?id=734546
2742 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
2744 * gst/rtsp-server/rtsp-media.c:
2745 signals: Fix copy-pasto in target-state signal offset
2747 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
2751 Makefile: Add usage of build-checks step
2752 Allows building checks without running them
2754 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
2756 * gst/rtsp-server/rtsp-stream.c:
2757 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
2758 When a UDP multicast transport is used it is expected that the server listens
2759 for RTP and RTCP packets on the multicast group with the corresponding port.
2760 Without this we will never get RTCP packets from clients in multicast mode.
2761 https://bugzilla.gnome.org/show_bug.cgi?id=732238
2763 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
2768 === release 1.4.0 ===
2770 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
2776 * gst-rtsp-server.doap:
2779 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
2781 * gst/rtsp-server/rtsp-media.h:
2782 media: correct misspelled words in description
2783 https://bugzilla.gnome.org/show_bug.cgi?id=733244
2785 === release 1.3.91 ===
2787 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
2793 * gst-rtsp-server.doap:
2796 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
2798 * docs/libs/gst-rtsp-server-sections.txt:
2801 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
2803 * gst/rtsp-server/rtsp-server.c:
2804 server: implement client REMOVE filter
2806 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
2808 * gst/rtsp-server/rtsp-client.c:
2809 * gst/rtsp-server/rtsp-client.h:
2810 client: expose _close() method
2811 Expose a previously internal close method to close the client
2814 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
2816 * gst/rtsp-server/rtsp-session-pool.c:
2817 session-pool: signal session-removed outside of the lock
2818 Release the lock before emiting the session-removed signal.
2820 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
2822 * gst/rtsp-server/rtsp-client.c:
2823 * gst/rtsp-server/rtsp-server.c:
2824 * gst/rtsp-server/rtsp-session-pool.c:
2825 * gst/rtsp-server/rtsp-session.c:
2826 * gst/rtsp-server/rtsp-stream.c:
2827 filter: Release lock in filter functions
2828 Release the object lock before calling the filter functions. We need to
2829 keep a cookie to detect when the list changed during the filter
2830 callback. We also keep a hashtable to make sure we only call the filter
2831 function once for each object in case of concurrent modification.
2832 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
2834 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
2836 * gst/rtsp-server/rtsp-client.c:
2837 client: check if watch is set in handle_teardown()
2838 The unit tests run without a watch
2840 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
2842 * tests/check/gst/client.c:
2843 client tests: send teardown to cleanup session
2845 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
2847 * tests/check/gst/rtspserver.c:
2848 server tests: send teardown to cleanup session
2850 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
2852 * gst/rtsp-server/rtsp-client.c:
2853 client: keep ref to client for the session removed handler
2854 This extra ref will be dropped when all client sessions have been
2855 removed. A session is removed when a client sends teardown, closes its
2856 endpoint of the TCP connection or the sessions expires.
2857 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
2859 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
2861 * gst/rtsp-server/rtsp-client.c:
2862 * gst/rtsp-server/rtsp-session.c:
2863 * tests/check/gst/client.c:
2864 client: manage media in session as a last step
2865 Once we manage a media in a session, we can't unmanage it anymore
2866 without destroying it. Therefore, first check everything before we
2867 manage the media, otherwise if something is wrong we have no way to
2869 If we created a new session and something went wrong, remove the session
2870 again. Fixes a leak in the unit test.
2872 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
2874 * examples/test-mp4.c:
2875 * examples/test-ogg.c:
2876 examples: print 'stream ready at url' for mp4 and ogg example
2878 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
2880 * gst/rtsp-server/rtsp-client.c:
2881 * gst/rtsp-server/rtsp-sdp.c:
2882 rtsp: fix for MIKEY api change
2884 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
2886 * gst/rtsp-server/rtsp-client.c:
2887 client: free watch context only once
2888 The watch context is freed when the source is destroyed. Avoids
2889 a CRITICAL when we try to unref the context twice.
2891 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
2893 * gst/rtsp-server/rtsp-client.c:
2896 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
2898 * gst/rtsp-server/rtsp-client.c:
2899 client: protect sessions with lock
2900 Protect the list of sessions with the lock.
2901 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
2903 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
2905 * gst/rtsp-server/rtsp-client.c:
2906 Client: keep a ref to the session
2907 Don't just keep a weak ref to the session objects but use a hard ref. We
2908 will be notified when a session is removed from the pool (expired) with
2909 the new session-removed signal.
2910 Don't automatically close the RTSP connection when all the sessions of
2911 a client are removed, a client can continue to operate and it can create
2912 a new session if it wants. If you want to remove the client from the
2913 server, you have to use gst_rtsp_server_client_filter() now.
2914 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
2915 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
2917 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
2919 * gst/rtsp-server/rtsp-session-pool.c:
2920 * gst/rtsp-server/rtsp-session-pool.h:
2921 session-pool: add session-removed signal
2922 Add a signal to be notified when a session is removed from the pool.
2924 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
2926 * gst/rtsp-server/Makefile.am:
2927 * gst/rtsp-server/rtsp-server.h:
2928 Make rtsp-server.h a single-include header, use it for G-I
2929 https://bugzilla.gnome.org/show_bug.cgi?id=732411
2931 === release 1.3.90 ===
2933 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
2939 * gst-rtsp-server.doap:
2942 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
2944 * gst/rtsp-server/rtsp-stream.c:
2945 stream: crypto can be NULL
2947 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
2949 * gst/rtsp-server/rtsp-client.c:
2950 * gst/rtsp-server/rtsp-media.c:
2951 * gst/rtsp-server/rtsp-mount-points.c:
2952 introspection: add missing allow-none annotations
2953 https://bugzilla.gnome.org/show_bug.cgi?id=730952
2955 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
2957 * gst/rtsp-server/rtsp-address-pool.c:
2958 * gst/rtsp-server/rtsp-media.c:
2959 * gst/rtsp-server/rtsp-session-media.c:
2960 * gst/rtsp-server/rtsp-session-pool.c:
2961 * gst/rtsp-server/rtsp-stream-transport.c:
2962 * gst/rtsp-server/rtsp-stream.c:
2963 * gst/rtsp-server/rtsp-token.c:
2964 introspection: add (nullable) annotations to return values
2965 https://bugzilla.gnome.org/show_bug.cgi?id=730952
2967 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
2969 * gst/rtsp-server/rtsp-client.c:
2970 * gst/rtsp-server/rtsp-stream.c:
2971 gi: improve annotations
2972 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
2974 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
2976 * gst/rtsp-server/rtsp-client.c:
2977 * gst/rtsp-server/rtsp-media-factory.c:
2978 * gst/rtsp-server/rtsp-media.c:
2979 * gst/rtsp-server/rtsp-server.c:
2980 signals: use generic marshal function
2981 Use the generic C marshal function.
2982 Use more explicit type instead of G_TYPE_POINTER
2984 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
2986 * gst/rtsp-server/rtsp-context.h:
2987 context: add type macro
2989 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
2991 * gst/rtsp-server/rtsp-client.c:
2992 * gst/rtsp-server/rtsp-sdp.c:
2993 * gst/rtsp-server/rtsp-sdp.h:
2994 sdp: hide key length defines
2995 They don't have a namespace.
2997 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
3002 === release 1.3.3 ===
3004 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
3010 * gst-rtsp-server.doap:
3013 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
3015 * gst/rtsp-server/rtsp-client.c:
3016 * gst/rtsp-server/rtsp-sdp.c:
3017 * gst/rtsp-server/rtsp-sdp.h:
3018 mikey: add different key length parameters
3019 Add encryption and authentication key length parameters to MIKEY. For
3020 the encoders, the key lengths are obtained from the cipher and auth
3021 algorithms set in the caps. For the decoders, they are obtained while
3022 parsing the key management from the client.
3023 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
3025 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
3027 * tests/check/gst/stream.c:
3028 stream tests: Make sure we get right multicast address from stream
3029 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
3031 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
3033 * gst/rtsp-server/rtsp-client.c:
3034 client: ref the context until rtsp watch is alive
3035 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
3037 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
3039 * gst/rtsp-server/rtsp-client.c:
3040 client: Destroy the rtsp watch after connection close
3042 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
3044 * gst/rtsp-server/rtsp-media.c:
3045 media: fix confusing comment
3047 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
3049 * gst/rtsp-server/rtsp-session.c:
3050 rtsp-session: Timeout in header.
3051 Adding the possbilty to always have timout in header.
3052 This is configurabe with setting "timeout-always-visible".
3053 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
3055 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
3060 === release 1.3.2 ===
3062 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
3069 * gst-rtsp-server.doap:
3072 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
3075 Automatic update of common submodule
3076 From 211fa5f to 1f5d3c3
3078 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
3080 * gst/rtsp-server/rtsp-client.c:
3081 client: store TCP ports in transport
3082 Store the TCP ports in the transport when we are doing RTSP over TCP.
3083 This way, we can easily get to the ports from the transport.
3084 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
3086 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
3088 * gst/rtsp-server/rtsp-stream.c:
3089 stream: add signals for new RTP/RTCP encoders
3090 New signals to allow the user to configure the dynamically created
3092 https://bugzilla.gnome.org/show_bug.cgi?id=730228
3094 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
3096 * gst/rtsp-server/rtsp-media.c:
3097 * gst/rtsp-server/rtsp-media.h:
3098 media: Make suspend()/unsuspend() virtual
3099 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
3101 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
3103 * gst/rtsp-server/rtsp-client.c:
3104 client: fix send-message signal marshaller
3105 Use generic marshalling for the send-message signal. It has
3106 two POINTER arguments, not just one.
3107 https://bugzilla.gnome.org/show_bug.cgi?id=729900
3109 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
3111 * tests/check/gst/media.c:
3112 tests: add and remove pads only once
3113 In this test we simulate a dynamic pad by watching the caps event.
3114 Because of renegotiation in the base payloader now, this caps is sent
3115 multiple times but we can only deal with 1 invocation, use a variable to
3116 only 'add and remove' the pad once.
3118 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
3120 * tests/check/gst/rtspserver.c:
3121 tests: add unit test for correct handling of Require headers
3122 https://bugzilla.gnome.org/show_bug.cgi?id=729426
3124 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
3126 * gst/rtsp-server/rtsp-client.c:
3127 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
3128 Servers must handle Require headers and must report a failure
3129 if they don't handle any of the Required options, see RFC 2326,
3130 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
3131 https://bugzilla.gnome.org/show_bug.cgi?id=729426
3133 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
3138 === release 1.3.1 ===
3140 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
3146 * gst-rtsp-server.doap:
3149 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
3152 Automatic update of common submodule
3153 From bcb1518 to 211fa5f
3155 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
3160 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
3162 * tests/check/gst/sessionmedia.c:
3163 tests: fix memory leak in sessionmedia unit test
3165 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
3167 * gst/rtsp-server/rtsp-client.c:
3168 client: emit a signal before sending a message
3169 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
3171 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
3173 * gst/rtsp-server/rtsp-client.c:
3174 client: pass context to send_message
3175 Pass the current context to send_message, we will need it later.
3177 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
3179 * gst/rtsp-server/rtsp-client.c:
3180 client: fix typo in comment
3182 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
3184 * gst/rtsp-server/rtsp-media.c:
3185 media: Do not stop thread twice if default_prepare() fails
3187 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
3189 * gst/rtsp-server/rtsp-client.c:
3190 client: set the watch to flushing before going to NULL
3191 First set the watch to flushing so that we unblock any current and
3192 future attempt to send data on the watch, Then set the pipeline to
3194 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
3196 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
3198 * gst/rtsp-server/rtsp-session-pool.c:
3199 * tests/check/gst/sessionpool.c:
3200 rtsp-session-pool: Fixes annotation
3201 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
3202 in the sessionpool test.
3203 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
3205 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
3207 * gst/rtsp-server/rtsp-media.c:
3208 * gst/rtsp-server/rtsp-media.h:
3209 media: make media_prepare virtual
3210 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
3212 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
3214 * gst/rtsp-server/rtsp-media.c:
3215 * tests/check/gst/media.c:
3216 media: stop the thread in more error cases
3218 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
3220 * gst/rtsp-server/rtsp-media.c:
3221 * tests/check/gst/media.c:
3222 media: allow NULL as the thread
3223 Use the default context whan passing a NULL thread.
3225 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
3227 * gst/rtsp-server/rtsp-client.c:
3228 rtsp-client: indent cleanup
3229 Coverity was moaning about unreachable code, and I think it was just
3230 confused by { being before the label. We'll see if it pops up again.
3233 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
3235 * gst/rtsp-server/rtsp-client.c:
3236 * gst/rtsp-server/rtsp-media.c:
3237 client: Add drop-backlog property
3238 When we have too many messages queued for a client (currently hardcoded
3239 to 100) we overflow and drop the messages. Add a drop-backlog property
3240 to control this behaviour. Setting this property to FALSE will retry
3241 to send the messages to the client by waiting for more room in the
3243 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
3245 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
3247 * gst/rtsp-server/rtsp-client.c:
3248 client: support for POST before GET when setting up a tunnel
3250 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
3252 * gst/rtsp-server/rtsp-client.c:
3253 client: remove watch of the second client after http tunnel setup
3254 The second client will be freed after the HTTP tunnel has been set up.
3255 Make sure it's RTSP watch is never dispatched again.
3256 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
3258 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
3260 * gst/rtsp-server/rtsp-media.c:
3261 * tests/check/gst/media.c:
3262 media: Make media_prepare() fail if port allocation fails
3263 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
3265 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
3267 * tests/check/gst/media.c:
3268 media test: cleanup the thread pool in tests
3270 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
3272 * gst/rtsp-server/rtsp-media.c:
3273 * tests/check/gst/media.c:
3274 rtsp-media: Unblock blocked streams in unprepare
3275 The streams will be blocked when a live media is prepared.
3276 The streams should be unblocked in gst_rtsp_media_unprepare.
3277 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
3279 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
3281 * gst/rtsp-server/rtsp-media.c:
3282 media: release the state lock when going to NULL
3283 Set our state to UNPREPARING and release the state-lock before
3284 setting the pipeline to the NULL state. This way, any pad-added
3285 callback will be able to take the state-lock and check that we are now
3286 unpreparing instead of deadlocking.
3287 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
3289 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
3291 * gst/rtsp-server/rtsp-media.c:
3292 media: protect status with lock
3293 Make sure we only update the status with the lock.
3295 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
3297 * gst/rtsp-server/rtsp-client.c:
3298 * gst/rtsp-server/rtsp-sdp.c:
3299 rtsp: update for MIKEY API changes
3301 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
3303 * gst/rtsp-server/rtsp-client.c:
3304 client: parse the mikey response from the client
3305 Parse the mikey response from the client and update the policy for
3308 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
3310 * gst/rtsp-server/rtsp-stream.c:
3311 * gst/rtsp-server/rtsp-stream.h:
3312 stream: add method to set crypto info
3313 Make a method to configure the crypto information of a stream.
3314 Set udpsrc in READY instead of PAUSED so that we can configure caps
3317 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
3319 * gst/rtsp-server/rtsp-client.c:
3320 client: cleanup error paths
3322 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
3324 * gst/rtsp-server/rtsp-media.c:
3327 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
3329 * examples/test-video.c:
3330 test: enable SRTP only on RTSPS
3331 We only want to enable SRTP when doing rtsp over TLS so that we can
3332 exchange the keys in a secure way.
3334 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
3336 * examples/test-video.c:
3337 test: print an error on failure
3339 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
3342 * examples/test-video.c:
3343 * gst/rtsp-server/rtsp-sdp.c:
3344 * gst/rtsp-server/rtsp-stream.c:
3345 * tests/check/Makefile.am:
3346 stream: add SRTP support
3347 Install srtp encoder and decoder elements in rtpbin
3350 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3352 * tests/check/Makefile.am:
3353 * tests/check/gst/sessionpool.c:
3354 tests: Add unit tests for sessionpool
3355 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
3357 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3359 * tests/check/gst/threadpool.c:
3360 tests: Improve code coverage of rtsp-threadpool tests
3361 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
3363 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3365 * tests/check/gst/sessionmedia.c:
3366 tests: Improve code coverage for rtsp-session-media
3367 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
3369 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3371 gobject-introspection: Add annotations to support language bindings
3372 In addition a few cosmetic changes:
3373 * Adjust the order of arguments
3374 * Fix typo: occured -> occurred
3375 * Fix indentation after Return:-clauses
3376 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
3378 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3380 * gst/rtsp-server/rtsp-stream.c:
3381 rtsp-stream: Don't mix IPv4 and IPv6 addresses
3382 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
3384 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
3386 * gst/rtsp-server/rtsp-stream.c:
3387 stream: take caps after the session manager
3388 Take the caps for the SDP after they leave the rtpbin so that we can
3389 also get the properties added by rtpbin elements.
3391 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
3393 * gst/rtsp-server/rtsp-stream.c:
3394 stream: release lock while pushing out packets
3395 Keep a cache of the transports and use this to iterate the transport
3396 while pushing packets. This allows us to release the lock early.
3397 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
3399 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
3401 * gst/rtsp-server/rtsp-client.c:
3402 * gst/rtsp-server/rtsp-client.h:
3403 rtsp-client: vmethod for modifying tunnel GET response
3404 Add a vmethod tunnel_http_response where the response to the HTTP GET
3405 for tunneled connections can be modified.
3406 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
3408 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
3410 * gst/rtsp-server/rtsp-sdp.c:
3411 sdp: make 1 media line per profile
3412 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
3413 line in the SDP for each profile. The client is then supposed to pick
3414 one of the profiles in the SETUP request. Because the m= lines have the
3415 same pt, the client also knows that only 1 option is possible.
3417 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
3419 * gst/rtsp-server/rtsp-media-factory.c:
3420 * gst/rtsp-server/rtsp-media-factory.h:
3421 * gst/rtsp-server/rtsp-media.c:
3422 factory: add profile property and pass to media and streams
3424 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
3426 * examples/test-multicast.c:
3427 * gst/rtsp-server/rtsp-sdp.c:
3428 sdp: pass multicast connection for multicast-only stream
3429 Pass the multicast address of the stream in the connection info in the
3430 SDP so that clients try a multicast connection first.
3431 Only allow multicast connections in the test-multicast example. Also
3432 increase the TTL a little.
3434 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3437 .gitignore: Ignore gcov intermediate files
3438 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
3440 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
3442 * gst/rtsp-server/rtsp-stream.c:
3443 stream: release some locks in error cases
3445 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3447 docs: Enable and fix gtk-doc warnings
3448 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
3449 * addresspool/mediafactory: Add missing annotation colon
3450 * stream: Annotate return value
3451 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
3453 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
3456 Automatic update of common submodule
3457 From fe1672e to bcb1518
3459 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
3462 Automatic update of common submodule
3463 From 1a07da9 to fe1672e
3465 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
3467 * examples/Makefile.am:
3468 examples: use LDADD for libs instead of LDFLAGS
3470 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
3473 configure: make sure releases are in .doap file
3475 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
3477 * examples/test-cgroups.c:
3478 examples: test-cgroups: don't put code with side effects into g_assert()
3479 The g_assert() might get compiled out with the right
3480 compiler/preprocessor flags.
3482 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
3484 * examples/.gitignore:
3485 examples: add cgroup test binary to .gitignore
3487 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
3489 * examples/test-cgroups.c:
3490 examples: fix cgroup test build
3491 Fixes build failure caused by compiler warning:
3492 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
3494 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
3497 .gitignore: ignore temp files created in the course of 'make check'
3499 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
3501 * gst/rtsp-server/rtsp-media.c:
3502 rtsp-media: don't loose frames handling new PLAY request
3503 If client supplied a range check if the range specifies the start point.
3504 If not, then do an accurate seek to the current position. If a start
3505 point was specified do do a key unit seek to make sure the streaming
3506 starts with decodeable frames.
3507 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
3509 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
3511 * gst/rtsp-server/rtsp-media.c:
3512 Revert "media: only flush when setting a new start position"
3513 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
3514 We need to do the flush in all cases, demuxer block currently for
3517 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
3519 * gst/rtsp-server/rtsp-media.c:
3520 media: only flush when setting a new start position
3521 Only flush the pipeline when we change the start position with
3523 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
3525 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
3527 * gst/rtsp-server/rtsp-stream.c:
3528 stream: set ttl-mc before adding the socket
3529 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
3530 never be set on socket.
3531 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
3533 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
3535 * gst/rtsp-server/rtsp-media.c:
3536 media: stop thread if media is already prepared
3537 in gst_rtsp_media_prepare() the thread is not used if media is already
3538 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
3540 https://bugzilla.gnome.org/show_bug.cgi?id=724182
3542 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
3545 build: Ship gst-rtsp-server.doap file
3547 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
3549 * tests/check/gst/rtspserver.c:
3550 tests: Fix another compiler warning with gcc
3552 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
3554 * gst/rtsp-server/rtsp-client.c:
3555 * gst/rtsp-server/rtsp-mount-points.c:
3556 * gst/rtsp-server/rtsp-stream.c:
3557 * tests/check/gst/client.c:
3558 rtsp-server: Fix lots of compiler warnings with clang
3560 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
3563 * gst-rtsp-server.doap:
3564 * tests/Makefile.am:
3565 configure: Synchronise with the configure scripts of the other modules
3567 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
3570 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
3572 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
3574 * gst/rtsp-server/rtsp-media.c:
3575 * gst/rtsp-server/rtsp-stream.c:
3576 Revert "rtsp-server: support build against last stable release"
3577 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
3578 Let us require 1.2.3 now, which is going to be released in a few
3581 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
3583 * gst/rtsp-server/rtsp-session-media.c:
3584 * gst/rtsp-server/rtsp-stream-transport.c:
3585 session: improve RTP-Info
3586 Ignore streams that can't generate RTP-Info instead of failing.
3587 Don't return the empty string when all streams are unconfigured but
3588 return NULL so that we don't generate and empty RTP-Info header.
3589 Improve docs a little.
3591 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
3593 * gst/rtsp-server/rtsp-session-media.c:
3594 Don't free rtpinfo GString when it is NULL
3595 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
3597 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
3599 * gst/rtsp-server/rtsp-media.c:
3600 media: only set keyframe flag when modifying start
3601 Only set the keyframe flag when we modify the start position. The
3602 keyframe flag should probably be ignored when no change is requested but
3603 until we can claim this is all documented properly and all demuxer
3604 implement this, avoid setting the flag.
3605 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
3607 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
3609 * gst/rtsp-server/rtsp-thread-pool.c:
3610 thread-pool: Unref source after mainloop has quit to avoid races in GLib
3611 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
3613 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
3615 * gst/rtsp-server/rtsp-stream.c:
3616 stream: handle NULL seqnum and rtptime arguments
3618 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
3620 * gst/rtsp-server/rtsp-thread-pool.c:
3621 * tests/check/gst/threadpool.c:
3622 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
3623 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
3625 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
3627 * gst/rtsp-server/rtsp-stream.c:
3628 stream: add fallback for missing stats property
3629 Use a fallback when the payloader does not have a stats property
3630 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
3632 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
3635 Automatic update of common submodule
3636 From f7bc1c3 to 1a07da9
3638 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
3640 * gst/rtsp-server/rtsp-stream.c:
3641 stream: don't leak stats structure
3642 Don't leak the stats structure and deal with NULL stats.
3644 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
3646 * gst/rtsp-server/rtsp-stream.c:
3647 stream: Get rtpinfo properties atomically from payloader
3648 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
3650 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
3652 * gst/rtsp-server/rtsp-media.c:
3653 media: refactor state change functions and signals
3654 Make functions to set the target state and the pipeline state and emit
3655 the signals from those functions.
3657 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
3659 * gst/rtsp-server/rtsp-media.c:
3660 * gst/rtsp-server/rtsp-media.h:
3661 media: add signal to notify of pending state changes
3663 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
3665 * gst/rtsp-server/rtsp-media.c:
3666 * gst/rtsp-server/rtsp-stream.c:
3667 rtsp-server: support build against last stable release
3668 Until 1.2.3 is out with the new get_type function and we
3671 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
3673 * gst/rtsp-server/rtsp-stream.c:
3674 stream: fix compilation
3676 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
3678 * gst/rtsp-server/rtsp-media.c:
3679 * gst/rtsp-server/rtsp-media.h:
3680 * gst/rtsp-server/rtsp-stream.c:
3681 * gst/rtsp-server/rtsp-stream.h:
3682 stream: add property to configure profiles
3684 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
3686 * gst/rtsp-server/rtsp-client.c:
3687 client: let stream check supported transport
3688 Delegate the check if a transport is allowed to the stream.
3689 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
3691 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
3693 * gst/rtsp-server/rtsp-stream.c:
3694 * gst/rtsp-server/rtsp-stream.h:
3695 stream: add method to check supported transport
3696 Add a method to check if a transport is supported
3698 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
3701 configure.ac: Only check for gstreamer-check, not check
3702 We include check in gstreamer-check since quite some time now.
3704 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
3706 * gst/rtsp-server/rtsp-session-media.c:
3707 * gst/rtsp-server/rtsp-stream-transport.c:
3708 * gst/rtsp-server/rtsp-stream.c:
3709 * gst/rtsp-server/rtsp-stream.h:
3710 stream: return clock-rate from get_rtpinfo
3711 And use it to correct the rtptime to the requested start-time.
3712 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
3714 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
3716 * gst/rtsp-server/rtsp-session-media.c:
3717 * gst/rtsp-server/rtsp-stream-transport.c:
3718 * gst/rtsp-server/rtsp-stream-transport.h:
3719 session-media: calculate start-time
3721 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
3723 * gst/rtsp-server/rtsp-stream-transport.c:
3724 * gst/rtsp-server/rtsp-stream.c:
3725 * gst/rtsp-server/rtsp-stream.h:
3726 stream: also return the running-time
3727 Return the running-time in the rtpinfo as well.
3729 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
3731 * gst/rtsp-server/rtsp-client.c:
3732 * gst/rtsp-server/rtsp-session-media.c:
3733 * gst/rtsp-server/rtsp-session-media.h:
3734 * gst/rtsp-server/rtsp-stream-transport.c:
3735 * gst/rtsp-server/rtsp-stream-transport.h:
3736 session-media: let the session-media make the RTPInfo
3737 Add method to create the RTPInfo for a stream-transport.
3738 Add method to create the RTPInfo for all stream-transports in a
3740 Use the session-media RTPInfo code in client. This allows us to refactor
3741 another method to link the TCP callbacks.
3743 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
3745 mount-points: sort sequence before g_sequence_lookup
3746 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
3747 sort sequence if dirty, otherwise lookup will fail.
3748 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
3750 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
3753 configure: rename package from gst-rtsp to gst-rtsp-server
3754 To match git module name and avoid confusion with the
3755 rtsp lib in gst-plugins-base and rtsp plugin in -good.
3757 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
3760 configure: bump core/base/good requirement to 1.2.0
3761 Bump to released stable version and make implicit
3762 requirements explicit.
3764 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
3769 Fix broken gettext setup which is not used anyway
3771 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
3774 Automatic update of common submodule
3775 From dbedaa0 to d48bed3
3777 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
3779 * gst/rtsp-server/rtsp-client.c:
3780 * gst/rtsp-server/rtsp-media.c:
3781 * gst/rtsp-server/rtsp-media.h:
3782 media: add setup_sdp vmethod
3783 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
3784 gst_rtsp_media_setup_sdp.
3785 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
3787 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
3789 * gst/rtsp-server/rtsp-stream.c:
3790 rtsp-stream: Check return value of sscanf
3791 streamid is only valid if sscanf matched something.
3793 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
3795 * gst/rtsp-server/rtsp-client.c:
3796 rtsp-client: Fix iteration
3797 Wouldn't even enter the code block otherwise (i++ was used as the check
3798 and not the postfix).
3800 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
3802 * gst/rtsp-server/rtsp-client.c:
3803 * gst/rtsp-server/rtsp-client.h:
3804 client: add vmethod to configure media and streams
3805 Implement a vmethod that can be used to configure the media and the
3806 streams based on the current context. Handle the blocksize handling in
3807 the default handler.
3808 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
3810 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
3813 Make git ignore more unit test binaries
3815 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
3817 * gst/rtsp-server/rtsp-address-pool.h:
3818 * gst/rtsp-server/rtsp-auth.h:
3819 * gst/rtsp-server/rtsp-client.h:
3820 * gst/rtsp-server/rtsp-context.h:
3821 * gst/rtsp-server/rtsp-media-factory-uri.h:
3822 * gst/rtsp-server/rtsp-media-factory.h:
3823 * gst/rtsp-server/rtsp-media.h:
3824 * gst/rtsp-server/rtsp-mount-points.h:
3825 * gst/rtsp-server/rtsp-server.h:
3826 * gst/rtsp-server/rtsp-session-media.h:
3827 * gst/rtsp-server/rtsp-session-pool.h:
3828 * gst/rtsp-server/rtsp-session.h:
3829 * gst/rtsp-server/rtsp-stream-transport.h:
3830 * gst/rtsp-server/rtsp-stream.h:
3831 * gst/rtsp-server/rtsp-thread-pool.h:
3832 * gst/rtsp-server/rtsp-token.h:
3833 rtsp-server: add padding to many public structures
3834 Not mini objects though, since they are not subclassable
3835 anyway, nor kept on the stack or inlined in a structure.
3837 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
3839 media: add new create_rtpbin vmethod
3840 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
3841 https://bugzilla.gnome.org/show_bug.cgi?id=719734
3843 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
3845 * tests/check/gst/media.c:
3846 tests: fix memory leak, free test's thread pool
3847 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
3849 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
3851 * gst/rtsp-server/rtsp-stream-transport.c:
3852 stream-transport: free url in finalize
3854 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
3856 * gst/rtsp-server/rtsp-media.c:
3857 media: also do state change in suspended state
3859 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
3861 * gst/rtsp-server/rtsp-client.c:
3862 * gst/rtsp-server/rtsp-media.c:
3863 media: also handle prepare and range in suspended state
3864 When we are suspended, we are already prepared.
3865 We can get the range in the suspended state.
3867 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
3869 * tests/check/Makefile.am:
3870 * tests/check/gst/sessionmedia.c:
3871 check: add test for uri in setup
3872 Added unit tests for the new functionality in GstRTSPStreamTransport.
3873 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
3875 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
3877 * gst/rtsp-server/rtsp-client.c:
3878 client: store setup uri and use in PLAY response
3879 Store the uri used when doing the setup and use that in the PLAY
3881 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
3883 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
3885 * gst/rtsp-server/rtsp-stream-transport.c:
3886 * gst/rtsp-server/rtsp-stream-transport.h:
3887 stream-transport: add method to get/set url
3889 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
3891 * gst/rtsp-server/rtsp-client.c:
3892 client: suspend after SDP and unsuspend before PLAYING
3893 Based on patches by Ognyan Tonchev <ognyan@axis.com>
3894 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
3896 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
3898 * gst/rtsp-server/rtsp-media-factory.c:
3899 * gst/rtsp-server/rtsp-media-factory.h:
3900 * gst/rtsp-server/rtsp-media.c:
3901 * gst/rtsp-server/rtsp-media.h:
3902 * gst/rtsp-server/rtsp-session-media.c:
3903 * gst/rtsp-server/rtsp-session.c:
3904 * tests/check/gst/media.c:
3905 * tests/check/gst/mediafactory.c:
3906 media: add suspend modes
3907 Add support for different suspend modes. The stream is suspended right after
3908 producing the SDP and after PAUSE. Different suspend modes are available that
3909 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
3910 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
3911 state and RESET will bring the pipeline to the NULL state.
3912 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
3913 this means that the pipeline needs to be prerolled again.
3914 Base on patches by Ognyan Tonchev <ognyan@axis.com>
3915 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3917 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
3919 * gst/rtsp-server/rtsp-media.c:
3920 media: start live streams in blocked state
3921 Start live streams in the blocked state and make them preroll using the
3922 messages. This ensure that no data is played by the sink until we explicitly
3923 unblock the stream right before going to PLAYING.
3924 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3926 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
3928 * gst/rtsp-server/rtsp-media.c:
3929 media: refactor starting and waiting for preroll
3930 Based on patches from Ognyan Tonchev <ognyan@axis.com>
3931 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3933 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
3935 * gst/rtsp-server/rtsp-stream.c:
3936 * gst/rtsp-server/rtsp-stream.h:
3937 stream: add API to block streams
3938 Add an API to block on the streams and make it post a message.
3939 Based on patch by Ognyan Tonchev <ognyan@axis.com>
3940 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3942 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
3944 * docs/libs/Makefile.am:
3945 docs: Specify the override file
3946 Even if it's empty (for now) it avoids make distcheck complaining
3948 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
3950 * gst/rtsp-server/rtsp-media.c:
3951 media: move default implementations to where they are used
3953 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
3955 * gst/rtsp-server/rtsp-media.c:
3956 media: take the right lock in gst_rtsp_media_set_pipeline_state()
3957 We need to take the state_lock when calling this method.
3959 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
3961 * gst/rtsp-server/rtsp-media.c:
3962 media: handle add-added on non-bins too
3963 Handle dynamic payloaders that are not bins, as used in the unit-test.
3965 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3967 * gst/rtsp-server/rtsp-media-factory.c:
3968 * gst/rtsp-server/rtsp-media-factory.h:
3969 * gst/rtsp-server/rtsp-media.c:
3970 rtsp-media/-factory: Fix request pad name comments
3971 These must be escaped for gtk-doc to parse the comments without warnings.
3973 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
3975 rtsp-media: remove transports if media is in error status
3976 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
3977 trying to change to GST_STATE_NULL and media is in error status, we
3978 remove all transports.
3979 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
3981 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
3983 * gst/rtsp-server/rtsp-media.c:
3984 rtsp-media: use element metadata to find payloader
3985 Use the element metadata to find the payloader instead of checking
3987 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
3989 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
3991 rtsp-stream: add getter for payload type
3992 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
3993 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
3994 element and create the stream with this one instead of the dynpay%d
3996 https://bugzilla.gnome.org/show_bug.cgi?id=712396
3998 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4000 * gst/rtsp-server/rtsp-client.c:
4001 * gst/rtsp-server/rtsp-context.h:
4002 * gst/rtsp-server/rtsp-media.c:
4003 * gst/rtsp-server/rtsp-mount-points.c:
4004 * gst/rtsp-server/rtsp-server.c:
4005 * gst/rtsp-server/rtsp-token.c:
4006 rtsp-*: Refer to NULL as a constant in comments
4008 https://bugzilla.gnome.org/show_bug.cgi?id=714988
4010 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4012 rtsp-*: Fix type name typos in comments
4013 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
4014 * rtsp-auth: Refer to part of constant name as text
4015 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
4016 * rtsp-session-media: Fix GstRTSPSessionMedia typo
4017 * rtsp-stream: Fix typo when refering to GstBin
4018 https://bugzilla.gnome.org/show_bug.cgi?id=714988
4020 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4023 * docs/libs/gst-rtsp-server-docs.sgml:
4024 * docs/libs/gst-rtsp-server-sections.txt:
4025 docs: Improve documentation
4026 * Include annotation-glossary to quiet gtk-doc
4027 * Rename remaining ClientState -> Context
4028 * Rename object hierarchy file
4029 * Remove stale chapter references
4030 * Add missing function and object references
4031 * Include missing GstRTSPAddressPoolResult
4032 https://bugzilla.gnome.org/show_bug.cgi?id=714988
4034 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
4036 * gst/rtsp-server/rtsp-client.c:
4037 * gst/rtsp-server/rtsp-server.c:
4038 * gst/rtsp-server/rtsp-session-pool.c:
4039 * gst/rtsp-server/rtsp-session.c:
4040 * gst/rtsp-server/rtsp-stream.c:
4041 rtsp-server: sprinkle some allow-none annotations for g-i
4043 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
4045 * gst/rtsp-server/rtsp-stream.c:
4046 * gst/rtsp-server/rtsp-stream.h:
4047 stream: add method to filter transports
4048 Add a method to safely iterate and collect the stream transports
4049 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
4051 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
4053 * gst/rtsp-server/rtsp-client.c:
4054 * gst/rtsp-server/rtsp-server.c:
4055 * gst/rtsp-server/rtsp-session-pool.c:
4056 * gst/rtsp-server/rtsp-session.c:
4057 rtsp: allow NULL func in filters
4058 Passing a null function make the filters return a list of
4061 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
4063 * gst/rtsp-server/rtsp-address-pool.c:
4064 * tests/check/gst/addresspool.c:
4065 address-pool: fix address increment
4066 Use a guint instead of guint8 to increment the address. It's still not
4067 completely correct because a guint might not be able to hold the complete
4068 address range, but that's an enhacement for later.
4069 Add unit test to test improved behaviour.
4070 https://bugzilla.gnome.org/show_bug.cgi?id=708237
4072 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
4074 * gst/rtsp-server/rtsp-client.c:
4075 * tests/check/gst/client.c:
4076 client: allow absolute path in requests
4077 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
4079 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
4081 * gst/rtsp-server/rtsp-client.c:
4082 * gst/rtsp-server/rtsp-client.h:
4083 client: make make_path_from_uri a vmethod
4085 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
4087 * docs/libs/gst-rtsp-server-sections.txt:
4088 * gst/rtsp-server/rtsp-stream.c:
4089 * gst/rtsp-server/rtsp-stream.h:
4090 * tests/check/Makefile.am:
4091 * tests/check/gst/stream.c:
4092 stream: Add functions to get rtp and rtcp sockets
4093 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
4095 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
4097 * gst/rtsp-server/rtsp-context.c:
4098 * gst/rtsp-server/rtsp-context.h:
4099 context: defing a GType for the context
4100 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
4102 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
4104 * gst/rtsp-server/Makefile.am:
4105 * gst/rtsp-server/rtsp-auth.c:
4106 * gst/rtsp-server/rtsp-context.c:
4107 * gst/rtsp-server/rtsp-media.c:
4108 * gst/rtsp-server/rtsp-mount-points.c:
4109 * gst/rtsp-server/rtsp-server.h:
4110 * gst/rtsp-server/rtsp-session-media.c:
4111 * gst/rtsp-server/rtsp-session.c:
4112 * gst/rtsp-server/rtsp-stream.c:
4113 Fixed several GIR warnings
4115 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
4117 * gst/rtsp-server/rtsp-auth.c:
4120 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
4122 * tests/check/Makefile.am:
4123 * tests/check/gst/token.c:
4124 tests: Add unit tests for token
4125 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
4127 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
4129 * gst/rtsp-server/rtsp-token.c:
4130 token: Validate args for gst_rtsp_token_is_allowed
4131 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
4133 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
4135 * gst/rtsp-server/rtsp-token.c:
4136 token: Fix bug when creating empty token
4137 We always want to have a valid GstStructure in the token.
4138 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
4140 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
4142 * gst/rtsp-server/rtsp-thread-pool.c:
4143 thread-pool: avoid race in shutdown
4144 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
4145 don't actually stop the mainloop ever. Solve this race by adding an idle source
4146 to the mainloop that calls the _quit. This way we immediately exit the mainloop
4147 if quit was called before we started it.
4149 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
4151 * tests/check/Makefile.am:
4152 * tests/check/gst/permissions.c:
4153 tests: Add unit tests for permissions
4154 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
4156 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
4158 * tests/check/gst/mediafactory.c:
4159 tests: Test mediafactory permissions
4160 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
4162 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
4164 * gst/rtsp-server/rtsp-permissions.c:
4165 permissions: Fix refcounting when adding/removing roles
4166 Previously a role that was removed was unreffed twice, and when
4167 replacing an existing role the replaced role was freed while still being
4168 referenced. Both bugs are now fixed.
4169 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
4171 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
4173 * tests/check/gst/media.c:
4174 * tests/check/gst/mediafactory.c:
4175 * tests/check/gst/rtspserver.c:
4176 tests: Check gst_rtsp_url_parse return value
4177 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
4179 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
4182 Automatic update of common submodule
4183 From 865aa20 to dbedaa0
4185 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
4187 * gst/rtsp-server/rtsp-server.c:
4188 rtsp-server: Fix socket leak
4189 https://bugzilla.gnome.org/show_bug.cgi?id=710088
4191 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
4193 * gst/rtsp-server/rtsp-session-pool.c:
4194 rtsp-session-pool: Make sure session IDs are properly URI-escaped
4195 https://bugzilla.gnome.org/show_bug.cgi?id=643812
4197 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4199 * examples/.gitignore:
4200 * examples/test-video.c:
4201 examples: fix compilation when WITH_AUTH is defined
4202 https://bugzilla.gnome.org/show_bug.cgi?id=710228
4204 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
4207 gitignore: Add new test binary
4209 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
4211 * tests/check/Makefile.am:
4212 * tests/check/gst/threadpool.c:
4213 thread-pool: Add unit test for the thread pools
4214 https://bugzilla.gnome.org/show_bug.cgi?id=710228
4216 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
4218 * gst/rtsp-server/rtsp-thread-pool.c:
4219 thread-pool: Fix thread leak when reusing threads
4220 https://bugzilla.gnome.org/show_bug.cgi?id=709730
4222 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
4224 * gst/rtsp-server/rtsp-server.c:
4225 * tests/check/gst/rtspserver.c:
4226 tests: fixed racy behavior in rtspserver tests
4227 https://bugzilla.gnome.org/show_bug.cgi?id=710078
4229 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
4231 * tests/check/gst/addresspool.c:
4232 tests: Improve address pool unit tests
4233 Add a range with mixed IPV4 and IPV6 addresses to pool.
4234 Get an IPV4 address from an IPV6-only pool.
4235 Get an IPV6 address from an IPV4-only pool.
4236 Reserve a IPV6 address from an IPV4-only pool.
4237 Check for unicast addresses in multicast-only pool.
4238 Check for unicast addresses in uni-/multicast-mixed pool.
4239 https://bugzilla.gnome.org/show_bug.cgi?id=710128
4241 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4243 * gst/rtsp-server/rtsp-client.c:
4244 client: append query string in PAUSE/PLAY/TEARDOWN as well
4246 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
4248 * gst/rtsp-server/rtsp-client.c:
4249 client: Add query to control path
4250 If the SETUP url contains a query it must be appended to the control
4251 path so that it matches any already created stream in the media. The
4252 query will also be appended to the session media path.
4254 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4256 * gst/rtsp-server/rtsp-media.c:
4257 rtsp-media: remove old line
4259 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
4261 * gst/rtsp-server/rtsp-stream.c:
4262 stream: Correct control comparison
4263 https://bugzilla.gnome.org/show_bug.cgi?id=709176
4265 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4267 * gst/rtsp-server/rtsp-media.c:
4268 media: Check dynamically if the pipeline supports seeking
4269 We should not depend on whether or not the pipeline state change
4270 returned NO_PREROLL or not. A media could dynamically change its
4271 element and switch from seekable to non seekable so it's best to test
4272 the seekable nature of the pipeline dynamically when we try to do a seek.
4274 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4276 * gst/rtsp-server/rtsp-media.c:
4277 media: Return FALSE if seeking is not supported
4279 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4281 * gst/rtsp-server/rtsp-media.c:
4282 rtsp-media: don't seek accurate by default
4283 Accurate seeking is perhaps a little overkill in the most common situation and
4284 causes some formats (mp3) over slow media to seek extremely slowly.
4286 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
4288 * tests/check/gst/rtspserver.c:
4289 tests: fix unit test
4290 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
4292 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
4294 * gst/rtsp-server/rtsp-client.c:
4295 client: Reply 400 if media cannot be constructed
4296 Reply 400 Bad Request instead of 503 Service Unavailable if media
4297 cannot be constructed in SETUP.
4298 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
4300 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
4302 * gst/rtsp-server/rtsp-client.c:
4303 client: Send setup reply once only
4304 If find_media() failed in handle_setup_request() two replies was sent.
4305 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
4307 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
4310 Automatic update of common submodule
4311 From 6b03ba7 to 865aa20
4313 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
4315 * gst/rtsp-server/rtsp-server.c:
4316 server: Emit client-connected signal earlier
4317 Emit client-connected before the client ref is given to a GSource,
4318 otherwise client-connected can be emitted after the client object has
4321 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
4323 * gst/rtsp-server/rtsp-address-pool.c:
4324 * gst/rtsp-server/rtsp-address-pool.h:
4325 * gst/rtsp-server/rtsp-stream.c:
4326 * tests/check/gst/addresspool.c:
4327 addresspool: return reason of failure
4328 Let gst_rtsp_address_pool_reserve_address() return the reason why
4329 the address could not be reserved.
4330 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
4332 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
4335 autogen.sh: Sync behaviour with other GStreamer modules
4336 Allows building from outside of tree amongst other things
4338 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
4341 Automatic update of common submodule
4342 From b613661 to 6b03ba7
4344 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
4347 Automatic update of common submodule
4348 From 74a6857 to b613661
4350 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
4353 Automatic update of common submodule
4354 From 01a7a46 to 74a6857
4356 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
4358 * gst/rtsp-server/rtsp-client.c:
4359 client: Do not read beyond end of path string
4360 If the setup was done without a control url, make sure we don't try to read the
4361 non-existing control string and crash.
4363 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4365 * gst/rtsp-server/rtsp-client.c:
4366 client: Fix RTPInfo header
4367 Refactor the method to make the content_base.
4368 Use the content-base and the control url to construct the RTPInfo
4371 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4373 * gst/rtsp-server/rtsp-client.c:
4374 client: map url to path only in describe
4375 Only map the request url to a path in the DESCRIBE method. The SDP then
4376 contains the base and control urls that should be used to SETUP/PAUSE/
4377 PLAY/TEARDOWN the media.
4379 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4381 * gst/rtsp-server/rtsp-client.c:
4382 Revert "client: map URL to path in requests"
4383 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
4384 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
4385 contains the base and control urls which are used in the SETUP, PLAY,
4386 PAUSE and TEARDOWN requests.
4388 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4390 * gst/rtsp-server/rtsp-client.c:
4391 client: map URL to path in requests
4393 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4395 * gst/rtsp-server/rtsp-client.c:
4396 * gst/rtsp-server/rtsp-mount-points.c:
4397 * gst/rtsp-server/rtsp-mount-points.h:
4398 mount-points: make vmethod to make path from uri
4399 Make a vmethod to transform an url into a path. The path is then used to lookup
4400 the factory. This makes it possible to also use other bits of the url, such as
4401 the query parameters, to locate the factory.
4403 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
4405 * gst/rtsp-server/rtsp-thread-pool.c:
4406 * gst/rtsp-server/rtsp-thread-pool.h:
4407 thread-pool: Add cleanup to wait for the threadpool to finish
4408 Also fix race condition if two threads are asking for the first
4409 thread from the thread pool at once. This would case two internal
4410 GThreadPools to be created.
4411 https://bugzilla.gnome.org/show_bug.cgi?id=707753
4413 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
4415 * gst/rtsp-server/rtsp-client.c:
4416 * tests/check/gst/client.c:
4417 client: free threadpool
4418 https://bugzilla.gnome.org/show_bug.cgi?id=707638
4420 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
4422 * tests/check/gst/mountpoints.c:
4423 mountpoints tests: unref matched factories
4424 https://bugzilla.gnome.org/show_bug.cgi?id=707638
4426 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
4428 * tests/check/gst/media.c:
4429 media tests: unref thread pool and caps
4430 https://bugzilla.gnome.org/show_bug.cgi?id=707638
4432 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
4434 * gst/rtsp-server/rtsp-auth.c:
4435 * gst/rtsp-server/rtsp-media-factory.c:
4436 * gst/rtsp-server/rtsp-media.c:
4437 auth, media, media-factory: unref permissions
4438 https://bugzilla.gnome.org/show_bug.cgi?id=707638
4440 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4442 * examples/Makefile.am:
4443 Makefile: add rule for appsrc example
4445 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4447 * examples/test-appsrc.c:
4448 tests: add appsrc example
4449 Add an example on how to use appsrc to feed the server pipeline with data.
4451 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
4453 * gst/rtsp-server/rtsp-client.c:
4454 rtsp-client: remove query part from content-base string
4455 Make sure that after the control url has been resolved, it's
4456 not a part of the query-string.
4457 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
4459 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4461 * gst/rtsp-server/rtsp-client.c:
4462 client: don't check url in response
4463 There is no url or method in the response to check
4465 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4467 * gst/rtsp-server/rtsp-client.c:
4468 * gst/rtsp-server/rtsp-client.h:
4469 Add handle-response signal for when we receive a GET_PARAMETER response
4471 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4473 * gst/rtsp-server/rtsp-server.c:
4474 Fix gst_rtsp_server_client_filter, using wrong variable type
4476 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
4478 * gst/rtsp-server/rtsp-media-factory-uri.c:
4479 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
4480 For AAC we need to check for framed=true instead of parsed=true.
4481 https://bugzilla.gnome.org/show_bug.cgi?id=701384
4483 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4485 * gst/rtsp-server/rtsp-stream.c:
4486 stream: optimize pipeline for protocols
4487 When TCP is not an allowed protocol for the stream, avoid creating the
4488 appsrc/appsink/queue and tee elements.
4490 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4492 * gst/rtsp-server/rtsp-media.c:
4493 media: set protocols on streams
4495 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4497 * gst/rtsp-server/rtsp-client.c:
4498 client: use protocols supported by stream
4500 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4502 * gst/rtsp-server/rtsp-media-factory.c:
4503 * gst/rtsp-server/rtsp-media.c:
4504 * gst/rtsp-server/rtsp-stream.c:
4505 media-factory: allow all protocols
4507 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4509 * gst/rtsp-server/rtsp-media.c:
4510 media: configure protocols in new streams
4512 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4514 * gst/rtsp-server/rtsp-stream.c:
4515 * gst/rtsp-server/rtsp-stream.h:
4516 stream: add protocols property
4518 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4520 * gst/rtsp-server/rtsp-media.c:
4521 rtsp-media: send state in "new-state" signal
4522 https://bugzilla.gnome.org/show_bug.cgi?id=705110
4524 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
4527 build: add subdir-objects to AM_INIT_AUTOMAKE
4528 Fixes warnings with automake 1.14
4529 https://bugzilla.gnome.org/show_bug.cgi?id=705350
4531 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4533 * docs/libs/gst-rtsp-server-sections.txt:
4534 * gst/rtsp-server/rtsp-client.c:
4535 * gst/rtsp-server/rtsp-server.c:
4536 * gst/rtsp-server/rtsp-server.h:
4537 server: add method to iterate clients of server
4539 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4541 * gst/rtsp-server/rtsp-media.c:
4542 * gst/rtsp-server/rtsp-media.h:
4543 Add vmethod for rtsp-media subclass to access rtpbin
4545 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4547 * gst/rtsp-server/rtsp-client.h:
4548 small documentation fix
4550 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4552 * gst/rtsp-server/rtsp-client.c:
4553 Do not take range header if range is invalid
4555 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4557 * docs/libs/gst-rtsp-server-sections.txt:
4558 * gst/rtsp-server/rtsp-media.c:
4559 media: add docs for new method
4561 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4563 * gst/rtsp-server/rtsp-media.c:
4564 * gst/rtsp-server/rtsp-media.h:
4565 Add API to rtsp-media set the pipeline's state
4567 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4569 * gst/rtsp-server/rtsp-media.c:
4570 Update current position/duration when gst_rtsp_media_get_range_string is called
4572 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4574 * examples/test-cgroups.c:
4575 tests: add some more docs
4577 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4579 * examples/test-cgroups.c:
4580 * gst/rtsp-server/Makefile.am:
4581 * gst/rtsp-server/rtsp-auth.c:
4582 * gst/rtsp-server/rtsp-auth.h:
4583 * gst/rtsp-server/rtsp-client.c:
4584 * gst/rtsp-server/rtsp-client.h:
4585 * gst/rtsp-server/rtsp-context.c:
4586 * gst/rtsp-server/rtsp-context.h:
4587 * gst/rtsp-server/rtsp-params.c:
4588 * gst/rtsp-server/rtsp-params.h:
4589 * gst/rtsp-server/rtsp-server.c:
4590 * gst/rtsp-server/rtsp-thread-pool.c:
4591 * gst/rtsp-server/rtsp-thread-pool.h:
4592 * tests/check/gst/client.c:
4593 ClientState -> Context
4594 Rename the clientstate to context and put the code in a separate file.
4596 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4598 * examples/test-auth.c:
4599 * gst/rtsp-server/rtsp-auth.c:
4600 * gst/rtsp-server/rtsp-auth.h:
4601 auth: add support for default token
4602 The default token is used when the user is not authenticated and can be used to
4603 give minimal permissions.
4605 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4607 * examples/test-auth.c:
4608 * gst/rtsp-server/rtsp-auth.c:
4609 auth: use defines when possible
4611 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4613 * gst/rtsp-server/rtsp-address-pool.c:
4614 address-pool: improve docs
4616 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4618 * gst/rtsp-server/rtsp-permissions.c:
4619 permissions: add the role to the copy
4621 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
4623 * gst/rtsp-server/rtsp-permissions.c:
4624 permissions: Also copy the roles
4626 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
4628 * gst/rtsp-server/rtsp-permissions.c:
4629 permissions: Make it build
4631 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4633 * gst/rtsp-server/rtsp-address-pool.h:
4636 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4638 * docs/libs/gst-rtsp-server-sections.txt:
4639 * gst/rtsp-server/rtsp-auth.c:
4640 * gst/rtsp-server/rtsp-auth.h:
4641 * gst/rtsp-server/rtsp-media.c:
4642 * gst/rtsp-server/rtsp-session-media.c:
4643 * gst/rtsp-server/rtsp-stream-transport.c:
4644 * gst/rtsp-server/rtsp-stream-transport.h:
4645 * gst/rtsp-server/rtsp-stream.c:
4646 * tests/check/gst/client.c:
4649 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4651 * docs/libs/gst-rtsp-server-sections.txt:
4652 * gst/rtsp-server/rtsp-address-pool.c:
4653 * gst/rtsp-server/rtsp-address-pool.h:
4654 * tests/check/gst/addresspool.c:
4655 * tests/check/gst/rtspserver.c:
4656 address-pool: cleanups
4657 Remove redundant method, improve docs.
4659 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4661 * docs/libs/gst-rtsp-server-sections.txt:
4662 * gst/rtsp-server/rtsp-auth.h:
4663 * gst/rtsp-server/rtsp-permissions.c:
4664 * gst/rtsp-server/rtsp-permissions.h:
4665 * gst/rtsp-server/rtsp-token.c:
4668 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4670 * gst/rtsp-server/rtsp-permissions.c:
4671 permissions: implement _remove_role
4673 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4675 * gst/rtsp-server/rtsp-permissions.c:
4676 permissions: update docs
4678 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4680 * tests/check/gst/client.c:
4681 tests: simplify tests
4682 Client settings are now disabled by default so we don't need an auth
4683 module to disable them.
4685 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4687 * gst/rtsp-server/rtsp-auth.c:
4688 auth: add default authorizations
4689 When no auth module is specified, use our table of defaults to look up the
4690 default value of the check instead of always allowing everything. This was
4691 we can disallow client settings by default.
4693 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4696 README: update readme
4698 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4700 * gst/rtsp-server/rtsp-thread-pool.c:
4701 * gst/rtsp-server/rtsp-thread-pool.h:
4702 thread-pool: add more docs
4704 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4706 * gst/rtsp-server/rtsp-thread-pool.c:
4707 * gst/rtsp-server/rtsp-thread-pool.h:
4708 thread-pool: fix race in thread reuse
4709 If we try to reuse a thread right after we made it stop, we end up using a
4710 stopped thread. Catch this case and only reuse threads that are not stopping.
4712 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4714 * gst/rtsp-server/rtsp-server.c:
4715 server: add small debug
4717 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4719 * tests/check/gst/client.c:
4721 Add some permissions to media so we can use the auth and enable
4724 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4726 * gst/rtsp-server/rtsp-client.c:
4727 client: support pushed context in handle_request
4728 If we already have a pushed state, reuse it and add our own things. This makes
4729 it easier to write tests.
4731 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4733 * gst/rtsp-server/rtsp-auth.c:
4734 auth: don't auth on methods
4735 Don't authorize on methods anymore but on the resources that we
4736 try to access, this is more flexible.
4737 Move the authorization checks to where they are needed and let the
4738 check return the response on error.
4740 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4742 * gst/rtsp-server/rtsp-mount-points.c:
4743 mount-points: add some debug
4745 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4747 * tests/check/gst/client.c:
4748 tests: almost fix test
4750 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4752 * gst/rtsp-server/rtsp-auth.c:
4753 * gst/rtsp-server/rtsp-auth.h:
4754 * gst/rtsp-server/rtsp-client.c:
4755 * gst/rtsp-server/rtsp-client.h:
4756 * gst/rtsp-server/rtsp-server.c:
4757 * gst/rtsp-server/rtsp-server.h:
4758 auth: let the auth module check client_settings
4759 Let the auth module decide if client settings are allowed for the
4762 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4764 * gst/rtsp-server/rtsp-token.c:
4765 * gst/rtsp-server/rtsp-token.h:
4766 token: add method to check boolean permission
4768 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4770 * examples/test-auth.c:
4771 * examples/test-cgroups.c:
4772 * gst/rtsp-server/rtsp-token.c:
4773 * gst/rtsp-server/rtsp-token.h:
4774 token: simplify token constructor
4775 Use variable arguments to make easier API.
4777 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4779 * examples/test-auth.c:
4780 * examples/test-cgroups.c:
4781 * gst/rtsp-server/rtsp-media-factory.c:
4782 * gst/rtsp-server/rtsp-media-factory.h:
4783 media-factory: add convenience API for factory
4785 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4787 * examples/test-auth.c:
4788 * examples/test-cgroups.c:
4789 * gst/rtsp-server/rtsp-permissions.c:
4790 * gst/rtsp-server/rtsp-permissions.h:
4791 permissions: simplify API a little
4792 Avoid passing GstStructure in the add_role method, use varargs instead
4793 to construct the structure behind the scenes. We can then also use the
4794 structure name as the role and simplify some more logic.
4796 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4798 * gst/rtsp-server/rtsp-auth.c:
4801 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4803 * gst/rtsp-server/rtsp-auth.c:
4804 * gst/rtsp-server/rtsp-auth.h:
4805 * gst/rtsp-server/rtsp-client.c:
4806 auth: handle unauthorized response
4807 Move handling of the unauthorized response to the auth module, it can add
4808 the appropriate headers to request authorization for the required method
4809 much better than the client.
4811 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4813 * gst/rtsp-server/rtsp-client.c:
4814 * gst/rtsp-server/rtsp-client.h:
4815 client: allow for sending any message, not only requests
4816 Change the _send_request() method to _send_message() so that we
4817 can both send requests and replies.
4819 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4821 * docs/libs/gst-rtsp-server-sections.txt:
4822 * gst/rtsp-server/rtsp-server.h:
4825 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4827 * examples/test-video.c:
4828 * gst/rtsp-server/rtsp-auth.c:
4829 * gst/rtsp-server/rtsp-auth.h:
4830 * gst/rtsp-server/rtsp-server.c:
4831 * gst/rtsp-server/rtsp-server.h:
4832 auth: move TLS handling to auth module
4833 Remove the TLS settings on the server and move it to the auth module because
4834 that is where security related bits go.
4836 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4838 * gst/rtsp-server/rtsp-client.c:
4839 * gst/rtsp-server/rtsp-client.h:
4840 client: add state push/pop
4842 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4844 * gst/rtsp-server/rtsp-client.c:
4845 * gst/rtsp-server/rtsp-client.h:
4846 client: add connection to state
4848 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4850 * gst/rtsp-server/rtsp-mount-points.c:
4851 mount-points: fix debug
4853 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4855 * tests/check/gst/media.c:
4856 tests: fix media test
4858 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4860 * gst/rtsp-server/rtsp-thread-pool.c:
4861 thread-pool: we don't require a state
4863 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4865 * gst/rtsp-server/rtsp-server.c:
4866 server: let context ref the server
4867 So that we don't risk losing the server object early anc crash.
4869 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4871 * tests/check/gst/client.c:
4872 tests: fix client test
4874 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4877 * docs/libs/gst-rtsp-server-docs.sgml:
4878 * docs/libs/gst-rtsp-server-sections.txt:
4879 * gst/rtsp-server/rtsp-address-pool.c:
4880 * gst/rtsp-server/rtsp-auth.c:
4881 * gst/rtsp-server/rtsp-client.c:
4882 * gst/rtsp-server/rtsp-client.h:
4883 * gst/rtsp-server/rtsp-media-factory-uri.c:
4884 * gst/rtsp-server/rtsp-media-factory.c:
4885 * gst/rtsp-server/rtsp-media-factory.h:
4886 * gst/rtsp-server/rtsp-media.c:
4887 * gst/rtsp-server/rtsp-mount-points.c:
4888 * gst/rtsp-server/rtsp-params.c:
4889 * gst/rtsp-server/rtsp-permissions.c:
4890 * gst/rtsp-server/rtsp-sdp.c:
4891 * gst/rtsp-server/rtsp-server.c:
4892 * gst/rtsp-server/rtsp-server.h:
4893 * gst/rtsp-server/rtsp-session-media.c:
4894 * gst/rtsp-server/rtsp-session-pool.c:
4895 * gst/rtsp-server/rtsp-session.c:
4896 * gst/rtsp-server/rtsp-stream-transport.c:
4897 * gst/rtsp-server/rtsp-stream.c:
4898 * gst/rtsp-server/rtsp-thread-pool.c:
4899 * gst/rtsp-server/rtsp-token.c:
4902 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4904 * gst/rtsp-server/rtsp-session-pool.c:
4905 * gst/rtsp-server/rtsp-session-pool.h:
4906 session-pool: make vmethod to create a session
4907 Make a vmethod to create a sessions so that subclasses can create
4908 custom session objects
4910 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4912 * gst/rtsp-server/rtsp-auth.c:
4913 * gst/rtsp-server/rtsp-media-factory.h:
4914 * gst/rtsp-server/rtsp-media.h:
4915 * gst/rtsp-server/rtsp-mount-points.h:
4916 * gst/rtsp-server/rtsp-session-pool.h:
4917 * gst/rtsp-server/rtsp-stream.h:
4920 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4922 * docs/libs/gst-rtsp-server-docs.sgml:
4923 * docs/libs/gst-rtsp-server-sections.txt:
4924 * gst/rtsp-server/rtsp-address-pool.c:
4925 * gst/rtsp-server/rtsp-address-pool.h:
4926 * gst/rtsp-server/rtsp-auth.c:
4927 * gst/rtsp-server/rtsp-client.h:
4928 * gst/rtsp-server/rtsp-media-factory.h:
4929 * gst/rtsp-server/rtsp-media.c:
4930 * gst/rtsp-server/rtsp-media.h:
4931 * gst/rtsp-server/rtsp-permissions.c:
4932 * gst/rtsp-server/rtsp-permissions.h:
4933 * gst/rtsp-server/rtsp-server.h:
4934 * gst/rtsp-server/rtsp-session-media.c:
4935 * gst/rtsp-server/rtsp-session-media.h:
4936 * gst/rtsp-server/rtsp-session-pool.h:
4937 * gst/rtsp-server/rtsp-session.h:
4938 * gst/rtsp-server/rtsp-stream-transport.h:
4939 * gst/rtsp-server/rtsp-stream.c:
4940 * gst/rtsp-server/rtsp-thread-pool.h:
4943 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4946 * examples/Makefile.am:
4947 configure: compile cgroup example conditionally
4948 Only compile the cgroup example when we have libcgroup
4950 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4953 * examples/Makefile.am:
4954 * examples/test-cgroups.c:
4955 examples: add cgroups example
4957 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4959 * tests/check/gst/rtspserver.c:
4960 tests: fix compilation
4962 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4964 * gst/rtsp-server/rtsp-thread-pool.c:
4965 thread-pool: fix vmethod invocation
4967 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4969 * gst/rtsp-server/rtsp-thread-pool.c:
4970 * gst/rtsp-server/rtsp-thread-pool.h:
4971 thread-pool: store thread type in thread
4973 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4975 * gst/rtsp-server/rtsp-client.c:
4976 client: pass thread from pool to media _prepare
4977 Get a thread from the configured threadpool and pass it to the prepare method of
4980 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4982 * gst/rtsp-server/rtsp-media.c:
4983 * gst/rtsp-server/rtsp-media.h:
4984 media: Accept a thread in _prepare
4985 Remove out own threadpool handling and use the provided thread and
4986 maincontext for the bus messages and the state changes.
4988 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4990 * gst/rtsp-server/rtsp-server.c:
4991 server: configure client thread pool
4993 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4995 * gst/rtsp-server/rtsp-client.c:
4996 * gst/rtsp-server/rtsp-client.h:
4997 client: add method to configure thread pool
4999 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5001 * gst/rtsp-server/rtsp-client.h:
5002 * gst/rtsp-server/rtsp-server.c:
5003 * gst/rtsp-server/rtsp-server.h:
5004 server: use thread pool
5005 Use the thread pool instead of doing our own thing.
5007 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5009 * gst/rtsp-server/Makefile.am:
5010 * gst/rtsp-server/rtsp-thread-pool.c:
5011 * gst/rtsp-server/rtsp-thread-pool.h:
5012 thread-pool: add object to manage threads
5013 Add an object to manage the client and media threads.
5015 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5017 * gst/rtsp-server/rtsp-auth.c:
5018 auth: debug authorization check
5020 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5022 * gst/rtsp-server/rtsp-media.c:
5023 media: start media pipeline in context
5024 Start the media pipeline in the provided context (or our default one
5025 when NULL). This makes sure that we run the bus thread in this context and that
5026 all media threads are children of this context.
5028 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5030 * gst/rtsp-server/rtsp-media-factory.c:
5031 factory: pass permissions to media by default
5033 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5035 * examples/test-auth.c:
5036 test: add permissions to auth test
5037 Ass some permissions to the media factory in the test.
5039 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5041 * gst/rtsp-server/rtsp-auth.c:
5042 * gst/rtsp-server/rtsp-auth.h:
5043 * gst/rtsp-server/rtsp-client.c:
5044 auth: simplify auth checks
5045 Remove client from methods, it's now in the state
5046 Perform the check specified by the string, use the information from the
5047 thread local context.
5049 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5051 * gst/rtsp-server/rtsp-client.c:
5052 * gst/rtsp-server/rtsp-client.h:
5053 client: add state to current thread
5054 Add the client to the ClientState object.
5055 Place the ClientState on the current thread.
5057 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5059 * gst/rtsp-server/rtsp-media-factory.c:
5060 * gst/rtsp-server/rtsp-media-factory.h:
5061 * gst/rtsp-server/rtsp-media.c:
5062 * gst/rtsp-server/rtsp-media.h:
5063 media: make it possible to set permissions
5064 Make it possible to set permissions on media and media factory objects
5066 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5068 * gst/rtsp-server/Makefile.am:
5069 * gst/rtsp-server/rtsp-permissions.c:
5070 * gst/rtsp-server/rtsp-permissions.h:
5071 permissions: add permissions object
5072 Add a mini object to store permissions based on a role.
5074 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5076 * examples/test-auth.c:
5077 * gst/rtsp-server/rtsp-auth.c:
5078 * gst/rtsp-server/rtsp-auth.h:
5079 * gst/rtsp-server/rtsp-client.c:
5080 auth: add auth checks
5081 Add an enum with auth checks and implement the checks in the auth object.
5082 Perform the checks from the client.
5084 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5086 * examples/test-auth.c:
5087 * gst/rtsp-server/rtsp-auth.c:
5088 * gst/rtsp-server/rtsp-auth.h:
5089 * gst/rtsp-server/rtsp-client.h:
5090 auth: use the token after authentication
5091 After we authenticated a user, keep the Token around in the state.
5093 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5095 * gst/rtsp-server/rtsp-client.c:
5096 * gst/rtsp-server/rtsp-media.c:
5097 * gst/rtsp-server/rtsp-media.h:
5098 * tests/check/gst/media.c:
5099 media: add optional context for bus messages
5100 Add an optional mainloop to _prepare that will handle the bus messages instead
5101 of always using the shared mainloop.
5103 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5105 * gst/rtsp-server/Makefile.am:
5106 * gst/rtsp-server/rtsp-token.c:
5107 * gst/rtsp-server/rtsp-token.h:
5108 token: add authorization token
5109 Add a simply miniobject that contains the authorizations. The object contains a
5110 GstStructure that hold all authorization fields. When a user is authenticated,
5111 the auth module will create a Token for the user. The token is then used to
5112 check what operations the user is allowed to do and various other configuration
5115 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5117 * examples/test-auth.c:
5118 * gst/rtsp-server/rtsp-auth.c:
5119 * gst/rtsp-server/rtsp-auth.h:
5120 * gst/rtsp-server/rtsp-client.c:
5121 * gst/rtsp-server/rtsp-client.h:
5122 * gst/rtsp-server/rtsp-media-factory.c:
5123 * gst/rtsp-server/rtsp-media-factory.h:
5124 * gst/rtsp-server/rtsp-media.c:
5125 * gst/rtsp-server/rtsp-media.h:
5126 auth: remove auth from media and factory
5127 Remove the auth object from media and factory. We want to have the RTSPClient
5128 authenticate and authorize resources, there is no need to place another auth
5129 manager on the media/factory.
5131 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5133 * examples/test-auth.c:
5134 * gst/rtsp-server/rtsp-auth.c:
5135 * gst/rtsp-server/rtsp-auth.h:
5136 * gst/rtsp-server/rtsp-client.h:
5137 auth: add support for multiple basic auth tokens
5138 Make it possible to add multiple basic authorisation tokens to one authorization
5139 object. Associate with each token an authorization group that will define what
5140 capabilities are allowed.
5142 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5144 * gst/rtsp-server/rtsp-client.c:
5145 client: error out on non-aggregate control
5146 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
5148 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5150 * gst/rtsp-server/rtsp-client.c:
5151 client: rework setup request a little
5152 Cache the media in DESCRIBE based on the longest matching path with the uri
5153 that we can find in the mount points.
5154 Rework the setup request a little to get the media from the session or from
5155 the longest matching path, this way we can derive the control string as
5156 everything after the path instead of hardcoding it.
5157 Find the stream based on the control string and only open a session when all
5160 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5162 * gst/rtsp-server/rtsp-media.c:
5163 * gst/rtsp-server/rtsp-media.h:
5164 media: add method to find a stream by control url
5166 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5168 * gst/rtsp-server/rtsp-stream.c:
5169 * gst/rtsp-server/rtsp-stream.h:
5170 stream: add method to check control url of stream
5172 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5174 * gst/rtsp-server/rtsp-client.c:
5175 * gst/rtsp-server/rtsp-session-media.c:
5176 * gst/rtsp-server/rtsp-session-media.h:
5177 * gst/rtsp-server/rtsp-session.c:
5178 * gst/rtsp-server/rtsp-session.h:
5179 session: use path matching for session media
5180 Use a path string instead of a uri to lookup session media in the sessions. Also
5181 use path matching to find the largest possible path that matches.
5183 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5185 * gst/rtsp-server/rtsp-client.c:
5186 * gst/rtsp-server/rtsp-mount-points.c:
5187 * gst/rtsp-server/rtsp-mount-points.h:
5188 * tests/check/gst/mountpoints.c:
5189 mount-points: remove useless vmethod
5190 Making lookups in the mount points should not be done with a URL, if there is a
5191 mapping to be done from URL to mount points, we'll need to do it somewhere
5194 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5196 * gst/rtsp-server/rtsp-mount-points.c:
5197 * gst/rtsp-server/rtsp-mount-points.h:
5198 * tests/check/gst/mountpoints.c:
5199 mount-points: improve mount point searching
5200 Use a GSequence to keep track of the mount points.
5201 Match a URL to the longest matching registered mount point. This should be the
5202 URL to perform aggreagate control and the remainder is the stream specific
5204 Add some unit tests for this.
5206 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
5208 * gst/rtsp-server/Makefile.am:
5209 rtsp-server: Allow building of static library
5211 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5213 * tests/check/gst/mediafactory.c:
5214 tests: fix compilation
5216 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5218 * gst/rtsp-server/rtsp-sdp.c:
5219 sdp: get control string from stream
5220 Use the control string as configured in the stream.
5222 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5224 * gst/rtsp-server/rtsp-stream.c:
5225 * gst/rtsp-server/rtsp-stream.h:
5226 stream: add methods and property to set control string
5228 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5230 * gst/rtsp-server/rtsp-client.c:
5232 Rename variables for clarity
5233 Keep media in state when we can
5235 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5237 * gst/rtsp-server/rtsp-client.c:
5238 * gst/rtsp-server/rtsp-stream.c:
5239 * gst/rtsp-server/rtsp-stream.h:
5240 stream: add more support for IPv6
5241 Rename _get_address to _get_multicast_address in GstRTSPStream to
5242 make it clear that this function only deals with multicast.
5243 Make it possible to have both an IPv4 and IPv6 multicast address on
5244 a stream. Give the client an IPv4 or IPv6 address depending on the
5245 address it used to connect to the server.
5246 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
5248 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5250 * gst/rtsp-server/rtsp-client.c:
5253 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5255 * gst/rtsp-server/rtsp-stream.c:
5256 stream: handle failed port allocation
5257 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
5258 can't allocate any family at all. Also keep track of what port families we
5260 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
5262 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5264 * gst/rtsp-server/rtsp-stream.c:
5265 stream: improve docs
5267 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5269 * gst/rtsp-server/rtsp-stream-transport.c:
5270 stream-transport: remove old if 0 block
5272 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
5274 * tests/check/gst/client.c:
5276 gst_rtsp_client_get_uri() has been removed
5277 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
5279 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5281 * gst/rtsp-server/rtsp-client.c:
5282 * gst/rtsp-server/rtsp-client.h:
5283 client: add method to filter managed sessions
5284 Add a method to filter the sessions managed by this client connection.
5285 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
5287 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5289 * gst/rtsp-server/rtsp-client.c:
5290 * gst/rtsp-server/rtsp-client.h:
5291 client: remove _get_uri() method
5292 Remove the get_uri() method on the client. A client has no uri, the uri
5293 property is an internal property to manage the last cached media for
5296 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5298 * gst/rtsp-server/rtsp-media-factory.h:
5299 media-factory: fix typo
5301 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
5303 * gst/rtsp-server/rtsp-media.c:
5304 rtsp-media: Do not leak the query in default_query_stop
5305 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
5307 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5309 * gst/rtsp-server/rtsp-media.c:
5310 media: don't unlock when conversion fails
5311 Don't unlock the state lock when conversion fails because it was not locked.
5313 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5315 * gst/rtsp-server/rtsp-media.c:
5316 * gst/rtsp-server/rtsp-media.h:
5317 Add query_position and query_stop vmethods to rtsp-media
5319 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5321 * gst/rtsp-server/rtsp-media.c:
5322 Fix typo in property install for rtsp-media's time-provider
5324 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5326 * gst/rtsp-server/rtsp-client.c:
5327 * gst/rtsp-server/rtsp-client.h:
5328 client: clean some variables
5329 Clean some variables and add some guards to _send_request()
5331 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5333 * gst/rtsp-server/rtsp-client.c:
5334 * gst/rtsp-server/rtsp-client.h:
5335 Add gst_rtsp_client_send_request API
5336 This makes it possible to send arbitrary messages to a client, such as
5337 SET_PARAMETER or GET_PARAMETER
5339 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5341 * gst/rtsp-server/rtsp-media.c:
5342 * gst/rtsp-server/rtsp-media.h:
5343 media: add _get_element() method
5344 Add method to get the element used when creating the media.
5345 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
5347 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5349 * gst/rtsp-server/rtsp-media.c:
5352 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
5354 * gst/rtsp-server/rtsp-stream.c:
5355 * gst/rtsp-server/rtsp-stream.h:
5356 stream: allow access to the rtp session
5357 https://bugzilla.gnome.org/show_bug.cgi?id=703004
5359 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
5361 * gst/rtsp-server/rtsp-stream.c:
5362 * gst/rtsp-server/rtsp-stream.h:
5363 dscp qos support in gst-rtsp-stream
5364 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
5366 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5368 * tests/check/gst/rtspserver.c:
5370 Actually do what the comment says. Also keep the old code around, not sure what
5371 should happen when you get a 454 from a TEARDOWN, does it close the connection?
5372 it currently doesn't.
5374 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5376 * gst/rtsp-server/rtsp-client.c:
5377 client: also watch newly created session
5378 When we newly created a session, start watching it immediately instead of
5379 on the next request.
5381 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
5383 * tests/check/gst/client.c:
5384 tests: add unit test for new-session
5385 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
5387 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5389 * gst/rtsp-server/rtsp-client.c:
5390 client: emit new-session when new session is created
5391 Only emit new-session when we created a new session for a client, not when a
5392 client picked up a previous session.
5393 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
5395 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
5397 * gst/rtsp-server/rtsp-client.c:
5398 client: handle asterisk as path in requests
5399 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
5401 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5403 * gst/rtsp-server/rtsp-media.c:
5404 media: handle segment query format mismatch
5405 It's possible that the segment query returns with a different format than what
5406 we asked for, handle this case also.
5408 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
5410 * gst/rtsp-server/rtsp-media.c:
5411 media: use segment stop in collect_media_stats
5412 Use segment stop instead of duration as range end point.
5413 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
5415 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
5417 * gst/rtsp-server/rtsp-media.c:
5418 * tests/check/gst/media.c:
5419 rtsp-media: Do not leak the element in take_pipeline
5420 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
5422 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
5424 * gst/rtsp-server/rtsp-client.c:
5425 * gst/rtsp-server/rtsp-client.h:
5426 rtsp-client: Make configure_client_transport virtual
5427 This patch makes configure_client_transport virtual. The functionality is
5428 needed to handle some weird clients sending multicast transport settings as url
5430 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
5432 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
5434 * gst/rtsp-server/rtsp-client.c:
5435 * gst/rtsp-server/rtsp-client.h:
5436 rtsp-client: Make param_set and param_get virtual
5437 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
5439 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
5441 * gst/rtsp-server/rtsp-client.c:
5442 * gst/rtsp-server/rtsp-media.c:
5443 * gst/rtsp-server/rtsp-media.h:
5444 media: convert_range replaces get_range_times
5445 get_range_times worked for handling UTC ranges for seeks, but we also
5446 need to convert back from NPT to the requested unit in
5447 get_range_string. convert_range is now used for both.
5448 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
5450 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5452 * gst/rtsp-server/rtsp-client.c:
5453 * gst/rtsp-server/rtsp-sdp.c:
5454 * gst/rtsp-server/rtsp-sdp.h:
5455 sdp: cleanup sdp info
5456 We don't need to pass the proto, we can more easily check a boolean.
5457 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
5459 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
5461 * gst/rtsp-server/rtsp-sdp.c:
5462 use 0.0.0.0 or :: for c= line instead of server address
5464 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
5466 * gst/rtsp-server/rtsp-client.c:
5467 use local address, not remote, in SDP
5468 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
5470 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5473 Automatic update of common submodule
5474 From 098c0d7 to 01a7a46
5476 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
5478 * gst/rtsp-server/rtsp-media.c:
5479 * gst/rtsp-server/rtsp-media.h:
5480 media: possibility to override range time conversion
5481 Make it possible to override the conversion from GstRTSPTimeRange to
5482 GstClockTimes, that is done before seeking on the media
5483 pipeline. Overriding can be useful for UTC ranges, where the default
5484 conversion gives nanoseconds since 1900.
5485 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
5487 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
5489 * gst/rtsp-server/rtsp-server.c:
5490 * gst/rtsp-server/rtsp-server.h:
5491 rtsp-server: Expose the use_client_settings API
5492 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
5494 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
5496 * gst/rtsp-server/rtsp-client.c:
5497 * gst/rtsp-server/rtsp-stream.c:
5498 * gst/rtsp-server/rtsp-stream.h:
5499 rtspstream: handle both ipv4 and ipv6 clients
5500 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
5502 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5504 * gst/rtsp-server/rtsp-sdp.c:
5505 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
5506 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
5507 We already have a way to place extra attributes in the SDP by using a string
5508 property with prefix x- or a- in the caps.
5510 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5512 * gst/rtsp-server/rtsp-sdp.c:
5513 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
5514 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
5515 We already have a way to place extra attributes in the SDP, just make a string
5516 property in the payloader with a- or x- prefix.
5518 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5520 * gst/rtsp-server/rtsp-sdp.c:
5521 rtsp: place a- and x- properties as attributes
5522 application/x-rtp has properties with a- and x- prefixes that should be
5523 placed as attributes in the SDP for the media instead of being added to the
5526 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5528 * examples/Makefile.am:
5529 * examples/test-video.c:
5530 example: add TLS example
5532 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5534 * gst/rtsp-server/rtsp-server.c:
5535 * gst/rtsp-server/rtsp-server.h:
5536 server: add support for TLS
5537 Add methods to set and get a TLS certificate.
5538 Add vmethod to configure a new connection. By default, configure the TLS
5539 certificate in a new connection if needed.
5541 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5543 * gst/rtsp-server/rtsp-server.c:
5544 * gst/rtsp-server/rtsp-server.h:
5545 server: remove accept_client vmethod
5546 This vmethod is not very useful so remove it.
5548 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5550 * gst/rtsp-server/rtsp-server.c:
5551 server: don't crash on NULL GError
5553 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
5555 * gst/rtsp-server/rtsp-session-pool.c:
5556 rtsp-session-pool: corrected session timeout detection
5557 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
5559 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5561 * gst/rtsp-server/rtsp-client.c:
5562 client: improve debug
5564 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5566 * gst/rtsp-server/rtsp-client.c:
5567 * gst/rtsp-server/rtsp-client.h:
5568 * gst/rtsp-server/rtsp-server.c:
5569 server: refactor connection setup
5570 Let the server accept the socket connection and construct a GstRTSPConnection
5571 from it. Remove the code from the client and let the client only deal with
5572 a fully configure GstRTSPConnection object.
5573 We will need this later when the server will configure the connection for
5576 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5578 * gst/rtsp-server/rtsp-stream.c:
5579 stream: keep the transport object alive
5580 Keep the transport object alive while we have it as qdata on the
5583 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
5585 * gst/rtsp-server/rtsp-client.c:
5586 * gst/rtsp-server/rtsp-server.c:
5587 rtsp-server: Do not crash on nmapping of server
5588 * generate error when gst_rtsp_connection_accept fails
5589 * do not stop accepting incoming connections because
5590 accepting a client fails
5591 https://bugzilla.gnome.org/show_bug.cgi?id=701072
5593 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
5595 * gst/rtsp-server/rtsp-client.c:
5596 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
5597 https://bugzilla.gnome.org/show_bug.cgi?id=700953
5599 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
5601 * gst/rtsp-server/rtsp-sdp.c:
5602 rtsp-sdp: Parse framerate caps field and set SDP attribute
5603 The SDP attribute and its format is described in RFC4566.
5604 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
5606 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
5608 * gst/rtsp-server/rtsp-sdp.c:
5609 rtsp-sdp: Parse width/height from caps and set SDP attribute
5610 The SDP attribute and its format is described in RFC6064.
5611 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
5613 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
5615 * gst/rtsp-server/rtsp-sdp.c:
5616 * tests/check/gst/client.c:
5617 rtsp-sdp: add bandwidth line
5618 https://bugzilla.gnome.org/show_bug.cgi?id=699220
5620 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5623 Automatic update of common submodule
5624 From 5edcd85 to 098c0d7
5626 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
5628 * tests/check/gst/media.c:
5629 tests: add dynamic payloader prepare/unprepare check
5631 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5633 * gst/rtsp-server/rtsp-media.c:
5634 media: release lock when removing fakesink
5636 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5638 * gst/rtsp-server/rtsp-stream.c:
5639 stream: set elements to NULL before removing
5640 When removing a stream, set the elements to NULL first. This avoids
5641 element-is-not-in-NULL-state errors when we dispose the elements.
5643 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
5646 Automatic update of common submodule
5647 From 3cb3d3c to 5edcd85
5649 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5651 * gst/rtsp-server/rtsp-media.c:
5652 * gst/rtsp-server/rtsp-media.h:
5653 media: listen to pad-removed signals
5654 Listen to the pad-removed signal and remove the stream associated with the
5656 Add signal to be notified of the removed pad.
5657 Remove the fakesink in unprepare()
5658 Fix signatures of the signal methods
5660 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5662 * examples/test-sdp.c:
5663 tests: add example of reusable pipelines
5665 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
5667 * gst/rtsp-server/rtsp-stream.c:
5668 * gst/rtsp-server/rtsp-stream.h:
5669 stream: add method to get the srcpad
5671 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
5673 * tests/check/gst/media.c:
5674 check: add media prepare/unprepare test
5675 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
5677 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
5679 * gst/rtsp-server/rtsp-media.c:
5680 media: disconnect from signal handlers in unprepare()
5681 We connected to the pad-added and no-more-pads signals in prepare() so
5682 we need to disconnect from them in unprepare().
5683 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
5685 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
5687 * gst/rtsp-server/rtsp-media.c:
5688 media: don't free streams array
5689 Don't free the streams array in the unprepare() method, they were not
5691 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
5693 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
5695 * gst/rtsp-server/rtsp-media.c:
5696 media: don't unref the pipeline in unprepare
5697 Unprepare() should undo what prepare() does. Because the pipeline is
5698 not created in prepare(), we should not unref it in unprepare()
5700 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
5702 * gst/rtsp-server/rtsp-stream.c:
5703 stream: clear session and caps for reuse
5704 Set the session and caps to NULL after unref otherwise we might unref
5706 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
5708 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
5710 * gst/rtsp-server/rtsp-client.c:
5711 client: send out teardown signal before tearing down
5712 The advantage is that in the signal handler you get direct access to
5713 information about what streams are about to get torn down (in the
5714 GstRTSPClientState).
5715 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
5717 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
5719 * gst/rtsp-server/rtsp-client.c:
5720 * gst/rtsp-server/rtsp-client.h:
5721 client: expose connection
5722 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
5724 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
5727 Automatic update of common submodule
5728 From aed87ae to 3cb3d3c
5730 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5732 * gst/rtsp-server/rtsp-media.c:
5733 * gst/rtsp-server/rtsp-media.h:
5734 * gst/rtsp-server/rtsp-session-media.c:
5735 * gst/rtsp-server/rtsp-session-media.h:
5736 media: add method to get the base_time of the pipeline
5737 Together with a shared clock, this base-time could eventually be sent to
5738 the client so that it can reconstruct the exact running-time of the clock
5741 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5743 * gst/rtsp-server/Makefile.am:
5744 * gst/rtsp-server/rtsp-media.c:
5745 * gst/rtsp-server/rtsp-media.h:
5746 * gst/rtsp-server/rtsp-sdp.c:
5747 media: add GstNetTimeProvider support
5748 Add a property to let the media provide a GstNetTimeProvider for its clock.
5749 Make methods to get the clock and nettimeprovider
5750 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
5751 provider and also the current time of the clock. This should make it possible
5752 for (GStreamer) clients to slave their clock to the server clock.
5754 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
5757 Automatic update of common submodule
5758 From 04c7a1e to aed87ae
5760 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5762 * gst/rtsp-server/rtsp-media.c:
5763 media: wait for buffering to complete
5764 Wait for buffering to complete before changing the state to the target state.
5766 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5768 * gst/rtsp-server/rtsp-media.c:
5769 media: small cleanup
5771 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
5773 * tests/check/gst/rtspserver.c:
5774 tests: remove extra unref in test_setup_non_existing_stream
5775 The unref is not needed anymore, teardown runs without it.
5776 https://bugzilla.gnome.org/show_bug.cgi?id=696542
5778 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
5780 * tests/check/gst/rtspserver.c:
5781 tests: GSocketService cleanup in test_bind_already_in_use
5782 Use g_socket_service_stop so the rtspserver test stops listening for
5783 incoming connections in test_bind_already_in_use.
5784 https://bugzilla.gnome.org/show_bug.cgi?id=696541
5786 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
5788 * gst/rtsp-server/rtsp-media-factory.c:
5789 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
5790 Instead use a GWeakRef which is safe to use
5791 This is a known GLib bug, see:
5792 https://bugzilla.gnome.org/show_bug.cgi?id=667145
5794 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
5796 * gst/rtsp-server/rtsp-client.c:
5797 * gst/rtsp-server/rtsp-media.c:
5798 * gst/rtsp-server/rtsp-media.h:
5799 * gst/rtsp-server/rtsp-sdp.c:
5800 * tests/check/gst/media.c:
5801 * tests/check/gst/rtspserver.c:
5802 rtsp-media/client: Reply to PLAY request with same type of Range
5803 Remember the type of Range from the PLAY request and use the same type for
5806 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
5808 * gst/rtsp-server/rtsp-client.c:
5809 * gst/rtsp-server/rtsp-client.h:
5810 * tests/check/gst/client.c:
5811 rtsp-client: expose uri
5813 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
5815 * tests/check/gst/mediafactory.c:
5816 tests: Hold ref while creating second media
5817 To test if the media aren't shared, make sure we keep the first one while creating a second
5818 otherwise the same memory address may be reused.
5820 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
5823 configure: remove out-of-date comment
5825 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
5828 .gitignore: ignore more build files
5830 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
5832 * tests/check/Makefile.am:
5833 tests: use right _LIBS variable for gst-plugins-base libs
5835 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5837 * tests/check/Makefile.am:
5838 check: add librtp to libs
5840 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
5842 * tests/check/gst/rtspserver.c:
5843 tests: Add test to check selecting a port the server will send from
5845 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
5847 * tests/check/gst/rtspserver.c:
5848 tests: Make sure packets are actually received
5850 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
5852 * gst/rtsp-server/rtsp-stream.c:
5853 stream: Select unicast address from pool if appropriate
5855 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
5857 * gst/rtsp-server/rtsp-stream.c:
5858 stream: Properties are always there in Gst 1.0
5860 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
5862 * tests/check/gst/addresspool.c:
5863 tests: Add tests for unicast addresses in pool
5865 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
5867 * gst/rtsp-server/rtsp-address-pool.c:
5868 * tests/check/gst/addresspool.c:
5869 address-pool: Verify that multicast addresses are used for multicast and vice-versa
5871 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
5873 * docs/libs/gst-rtsp-server-sections.txt:
5874 * gst/rtsp-server/rtsp-address-pool.c:
5875 * gst/rtsp-server/rtsp-address-pool.h:
5876 * gst/rtsp-server/rtsp-stream.c:
5877 * tests/check/gst/addresspool.c:
5878 address-pool: Add unicast addresses
5880 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
5883 * gst/rtsp-server/rtsp-server.c:
5884 * tests/check/gst/rtspserver.c:
5885 rtsp-server: Limit the number of threads per server instance
5886 If we exceed the maximum, just round robin the clients over the existing
5889 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
5891 * gst/rtsp-server/rtsp-server.c:
5892 rtsp-server: No need to store the GMainContext in the client context
5894 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
5896 * tests/check/gst/rtspserver.c:
5897 tests: Add test for client disconnection
5899 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
5901 * tests/check/gst/rtspserver.c:
5902 tests: Test client and session timeouts with multiple threads
5904 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
5906 * gst/rtsp-server/rtsp-address-pool.c:
5907 * gst/rtsp-server/rtsp-auth.c:
5908 * gst/rtsp-server/rtsp-client.c:
5909 * gst/rtsp-server/rtsp-media-factory-uri.c:
5910 * gst/rtsp-server/rtsp-media-factory.c:
5911 * gst/rtsp-server/rtsp-media.c:
5912 * gst/rtsp-server/rtsp-mount-points.c:
5913 * gst/rtsp-server/rtsp-server.c:
5914 * gst/rtsp-server/rtsp-session-media.c:
5915 * gst/rtsp-server/rtsp-session-pool.c:
5916 * gst/rtsp-server/rtsp-session.c:
5917 Document locking and its order
5919 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
5921 * tests/check/gst/rtspserver.c:
5922 tests: Test that slow DESCRIBE don't block other clients
5924 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
5926 * tests/check/gst/client.c:
5927 tests: Add tests for client-requested multicast address
5929 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
5931 * docs/libs/gst-rtsp-server-sections.txt:
5932 docs: Put the various functions in the right sections
5934 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
5936 * docs/libs/gst-rtsp-server-docs.sgml:
5937 * docs/libs/gst-rtsp-server-sections.txt:
5938 * gst/rtsp-server/rtsp-address-pool.c:
5939 * gst/rtsp-server/rtsp-address-pool.h:
5940 docs: Generate docs for GstRTSPAddressPool
5942 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
5944 * gst/rtsp-server/rtsp-client.c:
5945 * gst/rtsp-server/rtsp-stream.c:
5946 * gst/rtsp-server/rtsp-stream.h:
5947 client: Check client provided addresses against the address pool
5949 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
5951 * gst/rtsp-server/rtsp-address-pool.c:
5952 * gst/rtsp-server/rtsp-address-pool.h:
5953 * tests/check/gst/addresspool.c:
5954 address-pool: Add API to request a specific address from the pool
5955 Also add relevant unit tests.
5957 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
5959 * tests/check/gst/mediafactory.c:
5960 tests: Check the passing around of a RTSPAddressPool
5961 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
5962 way down to the stream.
5964 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
5966 * tests/check/gst/addresspool.c:
5967 tests: Add more tests for the address pool
5969 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
5971 * gst/rtsp-server/rtsp-address-pool.c:
5972 address-pool: Fix off by one error
5973 When splitting a port range, the port after a skip is not part of range.
5975 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
5978 Automatic update of common submodule
5979 From 2de221c to 04c7a1e
5981 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
5984 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
5985 AM_CONFIG_HEADER was removed in automake 1.13
5986 https://bugzilla.gnome.org/show_bug.cgi?id=693368
5988 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
5991 Automatic update of common submodule
5992 From a942293 to 2de221c
5994 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5996 * gst/rtsp-server/rtsp-client.c:
5997 client: make sure the watch exists while sending data
5998 Protect the send_func with a lock. This allows us to wait for sending
5999 to complete before changing the send_func and user_data. We add an
6000 extra ref to the watch to make sure that it remains valid during
6002 When closing the connection, set the send_func to NULL
6003 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
6005 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6007 * tests/check/Makefile.am:
6008 tests: use GST_*_1_0 environment variables everywhere
6009 The _1_0 suffixed environment variables override the
6010 non-suffixed ones, so if we're in an environment that
6011 sets the _1_0 suffixed ones, such as jhbuild, we need
6012 to set those to make sure ours actually always get
6015 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6018 Automatic update of common submodule
6019 From acb04d9 to a942293
6021 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6023 * gst/rtsp-server/rtsp-client.c:
6024 rtsp-client: set the client backlog
6025 Set the client backlog to a reasonable default
6027 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
6029 * gst/rtsp-server/rtsp-media.c:
6030 rtsp-media: Make the element a constructor parameter
6031 https://bugzilla.gnome.org/show_bug.cgi?id=689594
6033 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
6035 * docs/libs/Makefile.am:
6036 docs: Link with gcov library when gcov is enabled
6037 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
6039 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6041 * gst/rtsp-server/rtsp-media.c:
6042 media: match prepare with unprepare
6043 Really unprepare when there were an equal amount of prepare calls.
6045 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6047 * gst/rtsp-server/rtsp-media.c:
6048 media: media has to be unprepared in finalize
6049 Because unprepare takes away the last ref on the media.
6051 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6053 * gst/rtsp-server/rtsp-client.c:
6054 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
6055 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
6056 We can't use the refcount to trigger unprepare because it is the unprepare call
6057 that removes the last refcount after all messages are consumed. What we should
6058 probably do is make a prepared refcount and only unprepare when the refcount
6061 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6063 * gst/rtsp-server/rtsp-media.c:
6064 media: let the source unref the last media ref
6065 the last ref to the media is held by the source so we don't need to add more ref
6066 and unrefs, we simply destroy the media when the source is gone.
6068 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6070 * gst/rtsp-server/rtsp-media.c:
6071 media: improve debug
6073 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6075 * gst/rtsp-server/rtsp-media.c:
6077 Make sure we are in the right state when collecting the position and duration.
6078 Only make ourselves PREPARED when we were previously PREPARING.
6080 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6082 * gst/rtsp-server/rtsp-media.c:
6083 media: use g_object_ref/unref for GObjects
6085 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
6087 * gst/rtsp-server/rtsp-client.c:
6088 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
6089 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
6090 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
6091 isn't being used anymore.
6093 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
6095 * gst/rtsp-server/rtsp-media.c:
6096 Fix compiler warning
6098 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
6100 * gst/rtsp-server/rtsp-media-factory-uri.c:
6101 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
6103 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6105 * gst/rtsp-server/rtsp-session-media.h:
6108 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6110 * gst/rtsp-server/rtsp-media.c:
6111 * tests/check/gst/media.c:
6112 media: avoid element leak
6114 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6116 * gst/rtsp-server/rtsp-media.c:
6117 media: require an element in media constructor
6119 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6121 * gst/rtsp-server/rtsp-client.c:
6122 Revert "client: TEARDOWN brings that state to Init again"
6123 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
6124 The object is already disposed, there is no point in setting the state.
6126 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6128 * gst/rtsp-server/rtsp-client.c:
6129 client: TEARDOWN brings that state to Init again
6131 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6133 * docs/libs/gst-rtsp-server-sections.txt:
6134 * examples/test-auth.c:
6135 * gst/rtsp-server/rtsp-auth.c:
6136 * gst/rtsp-server/rtsp-auth.h:
6137 * gst/rtsp-server/rtsp-client.c:
6138 * gst/rtsp-server/rtsp-client.h:
6139 * gst/rtsp-server/rtsp-media-factory-uri.c:
6140 * gst/rtsp-server/rtsp-media-factory-uri.h:
6141 * gst/rtsp-server/rtsp-media-factory.c:
6142 * gst/rtsp-server/rtsp-media-factory.h:
6143 * gst/rtsp-server/rtsp-media.c:
6144 * gst/rtsp-server/rtsp-media.h:
6145 * gst/rtsp-server/rtsp-mount-points.c:
6146 * gst/rtsp-server/rtsp-mount-points.h:
6147 * gst/rtsp-server/rtsp-sdp.c:
6148 * gst/rtsp-server/rtsp-server.c:
6149 * gst/rtsp-server/rtsp-server.h:
6150 * gst/rtsp-server/rtsp-session-media.c:
6151 * gst/rtsp-server/rtsp-session-media.h:
6152 * gst/rtsp-server/rtsp-session-pool.c:
6153 * gst/rtsp-server/rtsp-session-pool.h:
6154 * gst/rtsp-server/rtsp-session.c:
6155 * gst/rtsp-server/rtsp-session.h:
6156 * gst/rtsp-server/rtsp-stream-transport.c:
6157 * gst/rtsp-server/rtsp-stream-transport.h:
6158 * gst/rtsp-server/rtsp-stream.c:
6159 * gst/rtsp-server/rtsp-stream.h:
6160 * tests/check/gst/media.c:
6161 rtsp: make object details private
6162 Make all object details private
6163 Add methods to access private bits
6165 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6167 * tests/check/Makefile.am:
6168 * tests/check/gst/media.c:
6169 tests: add media tests
6171 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6173 * gst/rtsp-server/rtsp-media.c:
6174 media: check if prepared for some methods
6175 Check that the media object is prepared before doing seek and getting the
6176 current position etc.
6177 Add some g_return checks.
6179 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6181 * tests/check/Makefile.am:
6182 * tests/check/gst/mediafactory.c:
6183 tests: add mediafactory test
6185 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6187 * gst/rtsp-server/rtsp-stream.c:
6188 stream: improve debug
6190 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6192 * gst/rtsp-server/rtsp-media.c:
6193 * gst/rtsp-server/rtsp-media.h:
6194 media: unref pipeline in finalize to avoid leaking it
6196 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6198 * gst/rtsp-server/rtsp-media-factory-uri.c:
6199 * gst/rtsp-server/rtsp-media.c:
6200 rtsp: use gst_object_unref on GstObjects
6202 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6204 * gst/rtsp-server/rtsp-media-factory.c:
6205 media-factory: require an url
6207 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6209 * examples/test-uri.c:
6210 examples: fix include
6212 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6214 * gst/rtsp-server/rtsp-server.h:
6215 server: remove unused include
6217 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6219 * tests/check/Makefile.am:
6220 * tests/check/gst/mountpoints.c:
6221 tests: add test for mountpoints
6223 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6225 * gst/rtsp-server/rtsp-client.c:
6226 client: fix factory leak
6227 Keep the factory in the state object only for authorization checks and make
6228 sure we unref it on failure. Also don't keep invalid objects in the state
6231 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6233 * gst/rtsp-server/rtsp-mount-points.c:
6234 mounts: add g_return_if guards
6236 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6238 * tests/check/gst/client.c:
6239 tests: add more tests
6241 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6243 * gst/rtsp-server/rtsp-client.c:
6244 client: improve debug
6246 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6248 * gst/rtsp-server/rtsp-client.c:
6249 client: improve debug and fix leaks
6250 Cleanup the uri and session when there is a bad request.
6252 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6257 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6259 * tests/check/gst/client.c:
6260 test: add test for session in options request
6262 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6264 * gst/rtsp-server/rtsp-client.c:
6265 client: use 454 when session can't be found
6266 We should use 454 when a session can't be found because there was no session
6267 pool configured in the server. This is not a server configuration problem
6268 because the server on which the request is done might not be the same one that
6269 will keep the sessions for us and so it does not need to support sessions.
6271 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6273 * gst/rtsp-server/rtsp-client.c:
6274 client: only free connection when there is one
6275 It's possible that the client doesn't have a connection when we try to free it.
6277 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6279 * tests/check/Makefile.am:
6280 * tests/check/gst/client.c:
6281 tests: add unit test for the client object
6283 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6285 * gst/rtsp-server/rtsp-client.c:
6286 client: small cleanup
6288 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6290 * gst/rtsp-server/rtsp-client.h:
6291 client: remove unused include
6293 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6295 * gst/rtsp-server/rtsp-client.c:
6296 client: fix compilation
6298 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6300 * gst/rtsp-server/rtsp-client.c:
6301 client: call destroy without the lock
6303 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6305 * gst/rtsp-server/rtsp-client.c:
6306 * gst/rtsp-server/rtsp-client.h:
6307 client: make the client usable without a socket
6308 Make a method to let the client handle a message and a callback when the client
6309 wants us to send a response message back. This makes it possible to also use the
6310 client object without the sockets, which should make it easier to test.
6312 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6314 * gst/rtsp-server/rtsp-client.c:
6315 * gst/rtsp-server/rtsp-client.h:
6316 client: small cleanup
6318 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6320 * docs/libs/gst-rtsp-server-sections.txt:
6321 * gst/rtsp-server/rtsp-client.c:
6322 * gst/rtsp-server/rtsp-client.h:
6323 * gst/rtsp-server/rtsp-server.c:
6324 client: remove reference to server
6325 We don't need to keep a ref to the server
6327 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6329 * gst/rtsp-server/rtsp-client.c:
6330 * gst/rtsp-server/rtsp-client.h:
6332 Also add some g_return_if()
6334 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6336 * gst/rtsp-server/rtsp-client.c:
6337 client: log more errors
6339 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6341 * gst/rtsp-server/rtsp-client.c:
6342 client: fix compilation
6344 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6346 * gst/rtsp-server/rtsp-client.c:
6347 * gst/rtsp-server/rtsp-client.h:
6348 client: add generic close-after-send support
6349 Add a property to send_response() to close the connection after the response has
6350 been sent to the client.
6352 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6355 * docs/libs/gst-rtsp-server-docs.sgml:
6356 * docs/libs/gst-rtsp-server-sections.txt:
6357 * docs/libs/gst-rtsp-server.types:
6358 * examples/test-auth.c:
6359 * examples/test-launch.c:
6360 * examples/test-mp4.c:
6361 * examples/test-multicast.c:
6362 * examples/test-multicast2.c:
6363 * examples/test-ogg.c:
6364 * examples/test-readme.c:
6365 * examples/test-sdp.c:
6366 * examples/test-uri.c:
6367 * examples/test-video.c:
6368 * gst/rtsp-server/Makefile.am:
6369 * gst/rtsp-server/rtsp-auth.h:
6370 * gst/rtsp-server/rtsp-client.c:
6371 * gst/rtsp-server/rtsp-client.h:
6372 * gst/rtsp-server/rtsp-media-mapping.c:
6373 * gst/rtsp-server/rtsp-media-mapping.h:
6374 * gst/rtsp-server/rtsp-mount-points.c:
6375 * gst/rtsp-server/rtsp-mount-points.h:
6376 * gst/rtsp-server/rtsp-server.c:
6377 * gst/rtsp-server/rtsp-server.h:
6378 * gst/rtsp-server/rtsp-session-media.c:
6379 * gst/rtsp-server/rtsp-session-pool.c:
6380 * gst/rtsp-server/rtsp-session-pool.h:
6381 * tests/check/gst/rtspserver.c:
6382 MediaMapping -> MountPoints
6383 Describes better what the object manages.
6385 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6388 configure: bump required version of -base
6390 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6392 * gst/rtsp-server/rtsp-media.c:
6395 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6397 * gst/rtsp-server/rtsp-media.c:
6398 * gst/rtsp-server/rtsp-media.h:
6399 media: support more Range formats
6400 Use the new -base methods to convert the Range string into a seek start and stop
6403 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6405 * examples/test-launch.c:
6406 examples: fix whitespace
6408 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6410 * examples/test-auth.c:
6411 test-auth: add example of how to remove sessions
6412 Add an example of the session filter api.
6414 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6416 * examples/test-uri.c:
6417 test-uri: remove mapping example
6419 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6421 * examples/test-uri.c:
6422 test-uri: fix callback signature
6424 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6426 * gst/rtsp-server/rtsp-media-factory.c:
6427 factory: keep ref to factory while media active
6428 While the media from a factory is alive, keep a ref to the factory.
6429 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
6431 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6433 * gst/rtsp-server/rtsp-media-factory-uri.c:
6434 factory-uri: add some debug
6436 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6438 * gst/rtsp-server/rtsp-stream.c:
6439 stream: set udp sources to PLAYING
6440 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
6441 so that it doesn't cause our pipeline to produce ASYNC-DONE.
6443 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6445 * gst/rtsp-server/rtsp-media-factory-uri.c:
6446 factory-uri: take ref to factory
6447 Take a ref to the factory that we place in our list.
6449 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6451 * tests/Makefile.am:
6452 * tests/test-reuse.c:
6453 test: add test for server reuse
6454 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
6456 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
6458 * gst/rtsp-server/rtsp-server.c:
6459 server: start and stop multiple times
6460 Stop listening on the RTSP port when the GSource is removed, so clients
6461 can't connect and the server can be started again.
6462 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
6464 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6466 * gst/rtsp-server/rtsp-server.c:
6467 server: fix small leak
6469 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6471 * gst/rtsp-server/rtsp-media.c:
6472 media: unref source in finish_unprepare
6473 The source is created in prepare, unref it in finish_unprepare.
6474 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
6476 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
6478 * gst/rtsp-server/rtsp-client.c:
6479 * gst/rtsp-server/rtsp-media.c:
6480 rtsp-media: remove bus watch before finalizing
6481 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
6482 * An extra media ref is added for the bus watch. This extra ref is unreffed by
6483 the GDestroyNotify function.
6484 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
6485 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
6486 gst_rtsp_media_unprepare before unreffing the media.
6487 This way, the bus watch will be removed before the media is finalized.
6488 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
6490 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
6492 * gst/rtsp-server/rtsp-client.c:
6493 * gst/rtsp-server/rtsp-client.h:
6494 client: wait until the TEARDOWN response is sent to close the connection
6495 Responses can be sent async so we need to wait until the TEARDOWN response has
6496 been written before we close the connection to the client. This avoids the risk
6497 of writing/polling closed sockets.
6498 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
6500 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
6502 * gst/rtsp-server/rtsp-stream.c:
6503 rtsp-stream: plug socket leak
6504 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
6506 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
6509 Automatic update of common submodule
6510 From 6bb6951 to a72faea
6512 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
6514 * gst/rtsp-server/rtsp-media-factory-uri.c:
6515 rtsp-server: don't use deprecated API
6517 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
6519 * gst/rtsp-server/rtsp-client.c:
6520 rtsp-client: fix unused-but-set-variable compiler warning
6521 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
6523 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6526 * docs/libs/gst-rtsp-server-sections.txt:
6527 * gst/rtsp-server/rtsp-client.c:
6530 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6532 * examples/Makefile.am:
6533 * examples/test-multicast2.c:
6534 examples: add another multicast example
6535 Add an example for how to configure separate multicast ranges for each media
6538 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6540 * examples/test-multicast.c:
6543 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6545 * gst/rtsp-server/rtsp-client.c:
6546 * gst/rtsp-server/rtsp-media.c:
6547 * gst/rtsp-server/rtsp-session-media.c:
6548 * gst/rtsp-server/rtsp-session-media.h:
6549 * gst/rtsp-server/rtsp-stream-transport.c:
6550 * gst/rtsp-server/rtsp-stream-transport.h:
6551 stream: use the address managed by the stream
6552 Use the address managed by the stream for multicast. This allows us to have 1
6553 multicast address for each stream.
6554 Because the address is now managed by the stream we don't have to pass it around
6556 Set the address pool on the streams.
6558 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6560 * gst/rtsp-server/rtsp-client.c:
6561 * gst/rtsp-server/rtsp-media.c:
6562 * gst/rtsp-server/rtsp-stream.c:
6565 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6567 * gst/rtsp-server/rtsp-media.c:
6568 * gst/rtsp-server/rtsp-media.h:
6569 media: add signal for new streams
6570 This allows applications to listen for new streams and configure properties on
6571 them, like the address pool.
6573 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6575 * gst/rtsp-server/rtsp-media.c:
6576 media: configure address pool in new streams
6578 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6580 * gst/rtsp-server/rtsp-stream.c:
6581 * gst/rtsp-server/rtsp-stream.h:
6582 stream: add methods to deal with address pool
6583 Add methods to get and set the address pool for the stream
6584 Add method to allocate and get the multicast addresses for this stream.
6586 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6588 * docs/libs/gst-rtsp-server-sections.txt:
6589 * gst/rtsp-server/rtsp-media.c:
6590 * gst/rtsp-server/rtsp-media.h:
6591 media: remove MTU property
6592 It is a stream property
6594 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6596 * gst/rtsp-server/rtsp-client.c:
6597 client: set blocksize only on stream
6598 Set the blocksize only on the current stream.
6600 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6602 * gst/rtsp-server/rtsp-stream.c:
6603 stream: share src and sink sockets
6604 the allocated socket is in the used-socket property, not socket.
6606 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6608 * gst/rtsp-server/rtsp-address-pool.c:
6609 * gst/rtsp-server/rtsp-address-pool.h:
6610 * gst/rtsp-server/rtsp-client.c:
6611 * gst/rtsp-server/rtsp-session-media.c:
6612 * gst/rtsp-server/rtsp-session-media.h:
6613 * gst/rtsp-server/rtsp-stream-transport.c:
6614 * gst/rtsp-server/rtsp-stream-transport.h:
6615 * tests/check/gst/addresspool.c:
6616 rtsp: make address-pool return an address object
6617 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
6618 store more info in the structure and allows us to more easily return the address
6619 to the right pool when no longer needed.
6620 Pass the address to the StreamTransport so that we can return it to the pool
6621 when the stream transport is freed or changed.
6623 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6625 * examples/Makefile.am:
6626 * examples/test-multicast.c:
6627 examples: add multicast example
6628 Show how to set up the multicast address pool so that media can be
6629 server with multicast.
6631 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6633 * gst/rtsp-server/rtsp-client.c:
6634 * gst/rtsp-server/rtsp-media-factory.c:
6635 * gst/rtsp-server/rtsp-media-factory.h:
6636 * gst/rtsp-server/rtsp-media.c:
6637 * gst/rtsp-server/rtsp-media.h:
6638 rtsp: use AddressPool
6639 Remove the multicast_group property.
6640 Use the configured addresspool to allocate multicast addresses.
6642 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6644 * gst/rtsp-server/rtsp-address-pool.c:
6645 * gst/rtsp-server/rtsp-address-pool.h:
6646 address-pool: add clear method
6648 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6650 * gst/rtsp-server/rtsp-address-pool.c:
6651 address-pool: small cleanups
6653 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6655 * tests/check/Makefile.am:
6656 * tests/check/gst/addresspool.c:
6657 tests: add addresspool unit test
6659 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6661 * gst/rtsp-server/Makefile.am:
6662 * gst/rtsp-server/rtsp-address-pool.c:
6663 * gst/rtsp-server/rtsp-address-pool.h:
6664 address-pool: add object to manage multicast addresses
6665 Make an object that can manage a rage of multicast addresses and ports.
6667 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6669 * gst/rtsp-server/rtsp-server.c:
6670 server: set default max-threads property
6672 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6674 * gst/rtsp-server/rtsp-media.c:
6675 media: wait for concurrent _prepare
6676 If a prepare is busy, wait for the result.
6678 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6680 * gst/rtsp-server/rtsp-media.c:
6681 media: add lock around message handler
6682 We don't want to dispatch messages while we are still processing the result of
6685 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6687 * gst/rtsp-server/rtsp-media.c:
6688 * gst/rtsp-server/rtsp-media.h:
6689 media: add lock to protect state changes
6691 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6693 * gst/rtsp-server/rtsp-stream.c:
6694 * gst/rtsp-server/rtsp-stream.h:
6697 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6699 * gst/rtsp-server/rtsp-stream-transport.c:
6700 * gst/rtsp-server/rtsp-stream-transport.h:
6701 * gst/rtsp-server/rtsp-stream.c:
6702 stream-transport: add keep-alive method
6704 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6706 * gst/rtsp-server/rtsp-stream-transport.c:
6707 * gst/rtsp-server/rtsp-stream-transport.h:
6708 * gst/rtsp-server/rtsp-stream.c:
6709 stream-transport: add method to handle RTP/RTCP
6710 Call new methods instead of poking into the structures directly.
6712 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6714 * gst/rtsp-server/rtsp-session-media.c:
6715 * gst/rtsp-server/rtsp-session-media.h:
6716 session-media: add locking
6718 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6720 * gst/rtsp-server/rtsp-session.c:
6721 * gst/rtsp-server/rtsp-session.h:
6722 session: add locking
6724 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6726 * gst/rtsp-server/rtsp-server.c:
6727 server: free old socket
6729 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6731 * gst/rtsp-server/rtsp-media-mapping.c:
6732 * gst/rtsp-server/rtsp-media-mapping.h:
6733 mapping: add locking
6735 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6737 * gst/rtsp-server/rtsp-media-factory.c:
6738 media-factory: add locking
6740 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6742 * gst/rtsp-server/rtsp-auth.c:
6743 * gst/rtsp-server/rtsp-auth.h:
6746 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6748 * gst/rtsp-server/rtsp-server.c:
6749 * gst/rtsp-server/rtsp-server.h:
6750 server: add max-thread property
6752 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6754 * gst/rtsp-server/rtsp-server.c:
6755 * gst/rtsp-server/rtsp-server.h:
6756 server: use a threadpool for the mainloops
6758 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6760 * gst/rtsp-server/rtsp-client.c:
6761 * gst/rtsp-server/rtsp-client.h:
6762 client: rename method
6763 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
6764 don't really create the client from the socket, we use the socket for the
6767 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6769 * gst/rtsp-server/rtsp-client.c:
6770 * gst/rtsp-server/rtsp-client.h:
6771 * gst/rtsp-server/rtsp-server.c:
6772 server: rework maincontext handling in clients
6773 Make a separate method to attach a client to a MainContext.
6774 Let the server decide in what GMainContext the client will operate and give this
6775 context to the client in attach. Then the server can later decide to use a
6776 separate thread for each client or just use the mainthread.
6778 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6780 * gst/rtsp-server/rtsp-client.c:
6781 * gst/rtsp-server/rtsp-session.c:
6782 * gst/rtsp-server/rtsp-session.h:
6783 session: move session header code in session object
6785 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
6789 * examples/test-auth.c:
6790 * examples/test-launch.c:
6791 * examples/test-mp4.c:
6792 * examples/test-ogg.c:
6793 * examples/test-readme.c:
6794 * examples/test-sdp.c:
6795 * examples/test-uri.c:
6796 * examples/test-video.c:
6797 * gst/rtsp-server/rtsp-auth.c:
6798 * gst/rtsp-server/rtsp-auth.h:
6799 * gst/rtsp-server/rtsp-client.c:
6800 * gst/rtsp-server/rtsp-client.h:
6801 * gst/rtsp-server/rtsp-media-factory-uri.c:
6802 * gst/rtsp-server/rtsp-media-factory-uri.h:
6803 * gst/rtsp-server/rtsp-media-factory.c:
6804 * gst/rtsp-server/rtsp-media-factory.h:
6805 * gst/rtsp-server/rtsp-media-mapping.c:
6806 * gst/rtsp-server/rtsp-media-mapping.h:
6807 * gst/rtsp-server/rtsp-media.c:
6808 * gst/rtsp-server/rtsp-media.h:
6809 * gst/rtsp-server/rtsp-params.c:
6810 * gst/rtsp-server/rtsp-params.h:
6811 * gst/rtsp-server/rtsp-sdp.c:
6812 * gst/rtsp-server/rtsp-sdp.h:
6813 * gst/rtsp-server/rtsp-server.c:
6814 * gst/rtsp-server/rtsp-server.h:
6815 * gst/rtsp-server/rtsp-session-media.c:
6816 * gst/rtsp-server/rtsp-session-media.h:
6817 * gst/rtsp-server/rtsp-session-pool.c:
6818 * gst/rtsp-server/rtsp-session-pool.h:
6819 * gst/rtsp-server/rtsp-session.c:
6820 * gst/rtsp-server/rtsp-session.h:
6821 * gst/rtsp-server/rtsp-stream-transport.c:
6822 * gst/rtsp-server/rtsp-stream-transport.h:
6823 * gst/rtsp-server/rtsp-stream.c:
6824 * gst/rtsp-server/rtsp-stream.h:
6825 * tests/check/gst/rtspserver.c:
6826 * tests/test-cleanup.c:
6829 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6831 * gst/rtsp-server/rtsp-media.c:
6832 * gst/rtsp-server/rtsp-session-media.c:
6833 * gst/rtsp-server/rtsp-session.c:
6834 rtsp-server: added annotations to indicate type of ownership transfer of return values
6835 https://bugzilla.gnome.org/show_bug.cgi?id=680777
6837 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
6840 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
6842 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
6845 * bindings/Makefile.am:
6846 * bindings/vala/Makefile.am:
6847 * bindings/vala/gst-rtsp-server-0.10.deps:
6848 * bindings/vala/gst-rtsp-server-0.10.vapi:
6849 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
6850 * bindings/vala/packages/gst-rtsp-server-0.10.files:
6851 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6852 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6853 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
6855 bindings: remove vala bindings
6856 They'll be reunited with the other GStreamer bindings
6857 https://bugzilla.gnome.org/show_bug.cgi?id=680777
6859 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6861 * gst/rtsp-server/rtsp-client.c:
6862 * gst/rtsp-server/rtsp-session-media.c:
6863 * gst/rtsp-server/rtsp-session-media.h:
6864 * gst/rtsp-server/rtsp-stream-transport.c:
6865 * gst/rtsp-server/rtsp-stream-transport.h:
6866 rtsp: only create transport when needed
6867 Only create the StreamTransport when configured.
6869 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6871 * gst/rtsp-server/rtsp-client.c:
6872 client: small cleanup
6874 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6876 * gst/rtsp-server/rtsp-client.c:
6877 * gst/rtsp-server/rtsp-client.h:
6878 * gst/rtsp-server/rtsp-stream-transport.c:
6879 * gst/rtsp-server/rtsp-stream-transport.h:
6880 rtsp: refactor configuration of transport
6881 Move the configuration of the transport to a place where it makes
6884 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6886 * gst/rtsp-server/rtsp-client.c:
6887 client: refactor transport parsing
6889 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6891 * gst/rtsp-server/rtsp-client.c:
6892 client: refuse to change the MTU on shared media
6893 If we change the MTU of chared media, it changes for all clients.
6894 We don't want to set the MTU to something large for clients that
6897 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6899 * examples/test-mp4.c:
6900 * gst/rtsp-server/rtsp-media.c:
6901 small fixes to docs and debug
6903 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6905 * gst/rtsp-server/rtsp-stream.c:
6906 stream: transports must already have been removed
6908 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6910 * gst/rtsp-server/rtsp-media.c:
6911 * gst/rtsp-server/rtsp-stream.c:
6912 * gst/rtsp-server/rtsp-stream.h:
6913 stream: improve join and leave of the pipeline
6915 Do the cleanup properly
6918 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6920 * gst/rtsp-server/rtsp-media.c:
6921 media: move unprepare below default implementation
6922 Makes it easier to find the default implementation
6924 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6926 * gst/rtsp-server/rtsp-media.c:
6927 media: signal unprepared when we actually finish
6929 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6931 * gst/rtsp-server/rtsp-media.c:
6932 media: no need to unlock, unprepare does that when needed
6934 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6936 * docs/libs/gst-rtsp-server-sections.txt:
6937 * gst/rtsp-server/rtsp-media-factory.h:
6938 * gst/rtsp-server/rtsp-media-mapping.c:
6939 * gst/rtsp-server/rtsp-media.h:
6940 * gst/rtsp-server/rtsp-params.c:
6941 * gst/rtsp-server/rtsp-server.c:
6942 * gst/rtsp-server/rtsp-session-pool.h:
6943 * gst/rtsp-server/rtsp-session.c:
6944 * gst/rtsp-server/rtsp-session.h:
6945 * gst/rtsp-server/rtsp-stream-transport.h:
6946 * gst/rtsp-server/rtsp-stream.h:
6949 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6951 * gst/rtsp-server/rtsp-client.c:
6952 * gst/rtsp-server/rtsp-media-mapping.h:
6953 * gst/rtsp-server/rtsp-media.c:
6954 * gst/rtsp-server/rtsp-media.h:
6955 * gst/rtsp-server/rtsp-server.h:
6956 * gst/rtsp-server/rtsp-stream.c:
6957 * gst/rtsp-server/rtsp-stream.h:
6958 rtsp: fix MTU setting
6959 Fix setting of the MTU. There is no need for a vmethod.
6961 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6966 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6969 configure: bump version number after refactoring
6971 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6973 * gst/rtsp-server/Makefile.am:
6974 * gst/rtsp-server/rtsp-client.c:
6975 * gst/rtsp-server/rtsp-client.h:
6976 * gst/rtsp-server/rtsp-media-factory-uri.c:
6977 * gst/rtsp-server/rtsp-media-factory.c:
6978 * gst/rtsp-server/rtsp-media-factory.h:
6979 * gst/rtsp-server/rtsp-media.c:
6980 * gst/rtsp-server/rtsp-media.h:
6981 * gst/rtsp-server/rtsp-sdp.c:
6982 * gst/rtsp-server/rtsp-session-media.c:
6983 * gst/rtsp-server/rtsp-session-media.h:
6984 * gst/rtsp-server/rtsp-session.c:
6985 * gst/rtsp-server/rtsp-session.h:
6986 * gst/rtsp-server/rtsp-stream-transport.c:
6987 * gst/rtsp-server/rtsp-stream-transport.h:
6988 * gst/rtsp-server/rtsp-stream.c:
6989 * gst/rtsp-server/rtsp-stream.h:
6990 rtsp: massive refactoring
6991 Make GObjects from the remaining simple structures.
6992 Remove GstRTSPSessionStream, it's not needed.
6993 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
6994 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
6995 a GstRTSPStream should be transported to a client.
6996 Rename GstRTSPMediaFactory::get_element -> create_element because that
6997 more accurately describes what it does.
6998 Make nice methods instead of poking in the structures.
6999 Move some methods inside the relevant object source code.
7000 Use GPtrArray to store objects instead of plain arrays, it is more
7001 natural and allows us to more easily clean up.
7002 Move the allocation of udp ports to the Stream object. The Stream object
7003 contains the elements needed to stream the media to a client.
7004 Improve the prepare and unprepare methods. Unprepare should now undo
7005 everything prepare did. Improve also async unprepare when doing EOS on
7006 shutdown. Make sure we always unprepare correctly.
7008 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
7010 * gst/rtsp-server/rtsp-client.c:
7011 rtsp-client: Unref server address clients connected to
7012 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
7014 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
7016 * gst/rtsp-server/rtsp-server.c:
7017 rtsp-server: don't ref server socket if it is NULL
7018 Fixes test_bind_already_in_use unit test again after commit 6a497440.
7019 https://bugzilla.gnome.org/show_bug.cgi?id=686644
7021 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
7023 * tests/check/Makefile.am:
7024 tests: Add libgio link dependency
7025 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
7027 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7029 * gst/rtsp-server/rtsp-media-mapping.c:
7030 * gst/rtsp-server/rtsp-media-mapping.h:
7031 rtsp-media-mapping: rename find_media vfunc to find_factory
7032 The virtual method and class method should have the same name
7033 so it is correctly represented in GIR file
7034 https://bugzilla.gnome.org/show_bug.cgi?id=680777
7036 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7038 * gst/rtsp-server/rtsp-auth.c:
7039 * gst/rtsp-server/rtsp-client.c:
7040 * gst/rtsp-server/rtsp-media-factory-uri.c:
7041 * gst/rtsp-server/rtsp-media-factory.c:
7042 * gst/rtsp-server/rtsp-media-mapping.c:
7043 * gst/rtsp-server/rtsp-media.c:
7044 * gst/rtsp-server/rtsp-server.c:
7045 * gst/rtsp-server/rtsp-session-pool.c:
7046 * gst/rtsp-server/rtsp-session.c:
7047 rtsp-server: fixed comments and GIR annotations
7048 https://bugzilla.gnome.org/show_bug.cgi?id=680777
7050 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
7052 * gst/rtsp-server/rtsp-media-mapping.c:
7053 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
7055 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
7057 * gst/rtsp-server/rtsp-server.c:
7058 rtsp-server: allow binding on port 0 (binds on a random port)
7060 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
7062 * gst/rtsp-server/rtsp-server.c:
7063 * gst/rtsp-server/rtsp-server.h:
7064 rtsp-server: add bound-port property
7065 bound-port can be used to retrieve the port number when the server is bound on
7066 port 0, which binds on a random port.
7068 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
7070 * gst/rtsp-server/rtsp-media-factory.c:
7071 * gst/rtsp-server/rtsp-media-factory.h:
7072 rtsp-media-factory: make ::get_element overridable by GI bindings
7073 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
7074 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
7075 as the invoker for ::get_element(), making it overridable by GI generated
7078 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
7080 * gst/rtsp-server/rtsp-media-factory-uri.c:
7081 rtsp-media-factory-uri: don't autoplug parsers in a loop
7082 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
7085 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
7087 * gst/rtsp-server/Makefile.am:
7088 Explicitly link against gio. Fix link error on mac.
7090 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
7092 * gst/rtsp-server/rtsp-session.c:
7093 session: add ttl to the transport header in SETUP
7094 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
7096 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
7098 * gst/rtsp-server/rtsp-client.c:
7099 * gst/rtsp-server/rtsp-client.h:
7100 * gst/rtsp-server/rtsp-media.c:
7101 client: Use client transport settings for multicast if allowed.
7102 This patch makes it possible for the client to send transport settings for
7103 multicast (destination && ttl). Client settings must be explicitly allowed or
7104 the server will use its own settings.
7105 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
7107 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
7110 Automatic update of common submodule
7111 From 6c0b52c to 6bb6951
7113 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
7115 * gst/rtsp-server/rtsp-client.c:
7116 rtsp-client: do not destroy the rtsp watch
7117 Don't destroy the client watch while dispatching. The rtsp watch is
7118 automatically destroyed after the rtsp watch function closed() has
7120 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
7122 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
7125 Automatic update of common submodule
7126 From 4f962f7 to 6c0b52c
7128 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
7130 * gst/rtsp-server/rtsp-media.c:
7131 media: fix check for seekability
7133 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7135 * gst/rtsp-server/rtsp-client.c:
7136 client: use more GIO
7137 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
7139 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7141 * gst/rtsp-server/rtsp-server.c:
7142 server: remove obsolete includes
7144 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
7146 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
7147 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
7148 be available in "on_new_ssrc". The transports are added in
7149 gst_rtsp_media_set_state when going to PLAYING state. However,
7150 "on_new_ssrc" might be called before this happens.
7151 https://bugzilla.gnome.org/show_bug.cgi?id=683304
7153 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
7155 * gst/rtsp-server/rtsp-client.c:
7156 * gst/rtsp-server/rtsp-client.h:
7157 rtsp-client: add signals for rtsp requests (fixes #683287)
7159 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
7161 * gst/rtsp-server/rtsp-client.c:
7162 * gst/rtsp-server/rtsp-client.h:
7163 add new-session signal to rtsp-client (fixes #683058)
7165 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
7168 Automatic update of common submodule
7169 From 668acee to 4f962f7
7171 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
7173 * gst/rtsp-server/rtsp-server.c:
7174 * tests/check/gst/rtspserver.c:
7175 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
7176 Do not assume that *error is set in g_socket_address_enumerator_next.
7177 Added test_bind_already_in_use unit-test.
7178 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
7180 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
7183 Automatic update of common submodule
7184 From 94ccf4c to 668acee
7186 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
7188 * gst/rtsp-server/rtsp-client.c:
7189 * gst/rtsp-server/rtsp-client.h:
7190 rtsp-client: make create_sdp virtual method
7191 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
7193 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7196 Automatic update of common submodule
7197 From 98e386f to 94ccf4c
7199 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7201 * gst/rtsp-server/rtsp-client.c:
7204 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
7206 * gst/rtsp-server/rtsp-client.c:
7207 * gst/rtsp-server/rtsp-client.h:
7208 * gst/rtsp-server/rtsp-server.c:
7209 * gst/rtsp-server/rtsp-server.h:
7210 rtsp-server: use an existing socket to establish HTTP tunnel
7211 Make it possible to transfer a socket from an HTTP server to be used as
7212 an RTSP over HTTP tunnel.
7214 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
7216 * gst/rtsp-server/rtsp-client.c:
7217 * gst/rtsp-server/rtsp-media.c:
7218 * gst/rtsp-server/rtsp-media.h:
7219 rtsp: Handle the blocksize parameter
7220 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
7222 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
7224 * tests/check/Makefile.am:
7225 * tests/check/gst/rtspserver.c:
7226 Have unit test get header from source dir, not installed dir
7227 This makes compilation of unit tests work in a build directory other
7228 than the source directory.
7229 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
7231 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
7233 * gst/rtsp-server/rtsp-media.c:
7234 rtsp-media: update for gst_element_make_from_uri() changes
7236 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
7239 * tests/Makefile.am:
7240 * tests/check/Makefile.am:
7241 * tests/check/gst/rtspserver.c:
7243 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
7245 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
7247 * gst/rtsp-server/rtsp-media.c:
7248 rtsp-media: don't collect media stats when going to NULL
7249 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
7251 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7253 * gst/rtsp-server/rtsp-client.c:
7254 client: don't leak transports
7256 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
7258 * gst/rtsp-server/rtsp-client.c:
7259 rtsp-client: free transport on no_stream in SETUP handler
7261 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
7263 * gst/rtsp-server/rtsp-client.c:
7264 rtsp-client: changed session media iteration
7265 In client_unlink_session: now don't iterate in session->medias
7266 list where items are removed by gst_rtsp_session_release_media.
7267 Instead, repeatedly remove the first item.
7269 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
7271 * gst/rtsp-server/rtsp-client.c:
7272 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
7273 GstRTSPSessionMedia is not a GObject type. When the
7274 GstRTSPSession is freed, it will free the media.
7276 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
7278 * gst/rtsp-server/rtsp-media-factory.c:
7279 factory: plug pad leak in collect_streams
7280 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
7281 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
7282 will take one reference, and the other reference will otherwise
7285 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
7288 configure: suppress some warnings when debug is disabled
7289 Warnings about unused variables should be suppressed if core has the
7290 debug system disabled.
7291 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
7293 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7295 * docs/libs/Makefile.am:
7296 docs: fix build in uninstalled setup
7297 Include gst-plugins-base libs properly.
7299 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
7301 * docs/libs/gst-rtsp-server.types:
7302 docs: include headers defining rtsp-server object types
7303 Fixes compiler warnings during docs build.
7304 https://bugzilla.gnome.org/show_bug.cgi?id=676824
7306 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
7309 configure: Add warning flags for compiler when configuring
7310 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
7312 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7315 Automatic update of common submodule
7316 From 03a0e57 to 98e386f
7318 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7321 Automatic update of common submodule
7322 From 1fab359 to 03a0e57
7324 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
7326 * gst/rtsp-server/rtsp-client.c:
7327 client: fix GSocketAddress leak in gst_rtsp_client_accept
7328 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
7330 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7333 Automatic update of common submodule
7334 From f1b5a96 to 1fab359
7336 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7339 Automatic update of common submodule
7340 From 92b7266 to f1b5a96
7342 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7345 Automatic update of common submodule
7346 From ec1c4a8 to 92b7266
7348 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7351 Automatic update of common submodule
7352 From 3429ba6 to ec1c4a8
7354 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
7356 * gst/rtsp-server/rtsp-auth.c:
7357 * gst/rtsp-server/rtsp-client.c:
7358 * gst/rtsp-server/rtsp-media-factory-uri.c:
7359 * gst/rtsp-server/rtsp-server.c:
7360 rtsp: fix compiler warnings
7361 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
7363 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7366 Automatic update of common submodule
7367 From dc70203 to 3429ba6
7369 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7371 * gst/rtsp-server/rtsp-client.c:
7372 * gst/rtsp-server/rtsp-media-factory.c:
7373 * gst/rtsp-server/rtsp-media-factory.h:
7374 * gst/rtsp-server/rtsp-media.c:
7375 * gst/rtsp-server/rtsp-media.h:
7376 * gst/rtsp-server/rtsp-server.c:
7377 * gst/rtsp-server/rtsp-server.h:
7378 * gst/rtsp-server/rtsp-session-pool.c:
7379 * gst/rtsp-server/rtsp-session-pool.h:
7380 rtsp-server: port to new thread API
7382 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7385 Automatic update of common submodule
7386 From 6db25be to dc70203
7388 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7390 * gst/rtsp-server/rtsp-auth.c:
7391 * gst/rtsp-server/rtsp-auth.h:
7392 * gst/rtsp-server/rtsp-client.c:
7393 rtsp-server: Fix compilation and compiler warnings
7395 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7399 * gst/rtsp-server/Makefile.am:
7400 configure: Modernize autotools setup a bit
7401 Also we now only create tar.bz2 and tar.xz tarballs.
7403 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7406 Automatic update of common submodule
7407 From 464fe15 to 6db25be
7409 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7412 Automatic update of common submodule
7413 From 7fda524 to 464fe15
7415 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7418 * docs/libs/Makefile.am:
7419 * docs/version.entities.in:
7421 * gst/rtsp-server/Makefile.am:
7422 * pkgconfig/Makefile.am:
7423 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
7424 * pkgconfig/gstreamer-rtsp-server.pc.in:
7425 * tests/Makefile.am:
7426 rtsp-server: Update versioning
7428 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7430 Merge remote-tracking branch 'origin/0.10'
7432 gst/rtsp-server/rtsp-session-pool.c
7434 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7436 * gst/rtsp-server/rtsp-session-pool.c:
7437 rtsp-server: Don't use deprecated GLib API
7439 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7441 Replace master with 0.11
7443 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7445 Merge branch 'master' into 0.11
7447 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7449 Merge branch 'master' into 0.11
7451 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
7454 A couple minor typo fixes
7456 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7458 * gst/rtsp-server/rtsp-media.c:
7459 media: fix state of the appqueue
7461 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7463 * gst/rtsp-server/rtsp-media-factory-uri.c:
7464 factory: use videoconvert
7466 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7468 * gst/rtsp-server/rtsp-media-factory-uri.c:
7469 factory: change to new style caps
7471 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7473 * gst/rtsp-server/rtsp-client.c:
7474 * gst/rtsp-server/rtsp-client.h:
7475 * gst/rtsp-server/rtsp-media-factory-uri.c:
7476 * gst/rtsp-server/rtsp-media.c:
7477 * gst/rtsp-server/rtsp-server.c:
7478 * gst/rtsp-server/rtsp-server.h:
7479 * gst/rtsp-server/rtsp-session-pool.c:
7480 rtsp-server: port to GIO
7483 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7486 configure: fix build
7488 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7491 docs: fix for gst_rtsp_server_set_port() -> _set_service()
7492 https://bugzilla.gnome.org/show_bug.cgi?id=666548
7494 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7497 * examples/Makefile.am:
7498 First rule of gst-rtsp-server club: don't talk about gst-phonon
7500 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7503 * pkgconfig/Makefile.am:
7504 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
7505 * pkgconfig/gstreamer-rtsp-server.pc.in:
7506 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
7507 For consistency with all other modules.
7509 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7511 * gst/rtsp-server/rtsp-client.c:
7512 rtsp-client: update for new map API
7514 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7517 * bindings/Makefile.am:
7518 * bindings/python/Makefile.am:
7519 * bindings/python/arg-types.py:
7520 * bindings/python/codegen/Makefile.am:
7521 * bindings/python/codegen/__init__.py:
7522 * bindings/python/codegen/argtypes.py:
7523 * bindings/python/codegen/code-coverage.py:
7524 * bindings/python/codegen/codegen.py:
7525 * bindings/python/codegen/definitions.py:
7526 * bindings/python/codegen/defsparser.py:
7527 * bindings/python/codegen/docextract.py:
7528 * bindings/python/codegen/docgen.py:
7529 * bindings/python/codegen/fileprefix.override:
7530 * bindings/python/codegen/fileprefixmodule.c:
7531 * bindings/python/codegen/h2def.py:
7532 * bindings/python/codegen/mergedefs.py:
7533 * bindings/python/codegen/mkskel.py:
7534 * bindings/python/codegen/override.py:
7535 * bindings/python/codegen/reversewrapper.py:
7536 * bindings/python/codegen/scmexpr.py:
7537 * bindings/python/rtspserver-types.defs:
7538 * bindings/python/rtspserver.defs:
7539 * bindings/python/rtspserver.override:
7540 * bindings/python/rtspservermodule.c:
7541 * bindings/python/test.py:
7543 python: remove pygst-based python bindings
7544 pygi is the future, apparently.
7546 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
7549 Automatic update of common submodule
7550 From c463bc0 to 7fda524
7552 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7555 Automatic update of common submodule
7556 From 2a59016 to c463bc0
7558 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7561 Automatic update of common submodule
7562 From 0807187 to 2a59016
7564 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7567 Automatic update of common submodule
7568 From 11f0cd5 to 0807187
7570 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7572 * examples/test-auth.c:
7573 example: update for new caps
7575 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7577 * examples/test-video.c:
7578 * gst/rtsp-server/rtsp-client.c:
7579 * gst/rtsp-server/rtsp-media-factory-uri.c:
7580 * gst/rtsp-server/rtsp-media.c:
7581 * gst/rtsp-server/rtsp-media.h:
7582 * gst/rtsp-server/rtsp-session.c:
7583 * gst/rtsp-server/rtsp-session.h:
7584 rtsp-server: port some more to 0.11
7586 Remove bufferlist stuff
7588 Add queue before appsink now that preroll-queue-len is gone.
7589 Update for request pad changes.
7591 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7593 Merge branch 'master' into 0.11
7595 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
7597 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7598 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
7599 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
7601 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
7603 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7604 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
7605 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
7607 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7609 Merge branch 'master' into 0.11
7611 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7613 * gst/rtsp-server/rtsp-media.c:
7614 * gst/rtsp-server/rtsp-media.h:
7615 media: add a seekable boolean
7616 Maintain the seekable state with a new variable instead of reusing the
7619 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
7621 * gst/rtsp-server/rtsp-media.c:
7622 Disallow seek in live media
7624 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7626 Merge branch 'master' into 0.11
7628 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
7630 * gst/rtsp-server/rtsp-server.c:
7631 #ifdef statements for windows socket creation were missing
7633 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
7636 Automatic update of common submodule
7637 From a39eb83 to 11f0cd5
7639 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
7642 Automatic update of common submodule
7643 From 605cd9a to a39eb83
7645 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7647 Merge branch 'master' into 0.11
7649 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7651 * gst/rtsp-server/rtsp-client.c:
7652 client: use method to access property
7654 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7656 * gst/rtsp-server/rtsp-media-factory.c:
7657 * gst/rtsp-server/rtsp-media-factory.h:
7658 media-factory: add protocols property
7659 Add a property to configure the allowed protocols in the media created from the
7662 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7664 * gst/rtsp-server/rtsp-media-factory.c:
7665 * gst/rtsp-server/rtsp-media-factory.h:
7666 media-factory: add media-configure signal
7667 Add signal to allow the application to configure the media after it was created
7670 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7672 * gst/rtsp-server/rtsp-client.c:
7673 client: use method to access property
7675 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7677 * gst/rtsp-server/rtsp-media-factory.c:
7678 * gst/rtsp-server/rtsp-media-factory.h:
7679 media-factory: add protocols property
7680 Add a property to configure the allowed protocols in the media created from the
7683 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7685 * gst/rtsp-server/rtsp-media-factory.c:
7686 * gst/rtsp-server/rtsp-media-factory.h:
7687 media-factory: add media-configure signal
7688 Add signal to allow the application to configure the media after it was created
7691 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7693 Merge branch 'master' into 0.11
7695 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7697 * gst/rtsp-server/rtsp-client.c:
7698 client: use media multicast group
7700 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7702 * gst/rtsp-server/rtsp-media-factory.h:
7703 * gst/rtsp-server/rtsp-server.h:
7704 * gst/rtsp-server/rtsp-session-pool.h:
7705 * gst/rtsp-server/rtsp-session.h:
7708 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
7710 * gst/rtsp-server/rtsp-client.c:
7711 * gst/rtsp-server/rtsp-sdp.h:
7712 sdp: copy and free the server ip address
7713 Copy and free the server ip address to make memory management easier later.
7715 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7717 * gst/rtsp-server/rtsp-media-factory.c:
7718 media-factory: configure multicast in media
7720 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7722 * gst/rtsp-server/rtsp-media.c:
7723 * gst/rtsp-server/rtsp-media.h:
7724 media: add property for multicast group
7725 Add a property to configure the multicast group in the media.
7726 Based on patches from Marc Leeman and Robert Krakora.
7728 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7730 * gst/rtsp-server/rtsp-media-factory.c:
7731 * gst/rtsp-server/rtsp-media-factory.h:
7732 media-factory: add property for multicast group
7733 Add a property to configure the multicast group in the media factory.
7734 Based on patches from Marc Leeman and Robert Krakora.
7736 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7738 * gst/rtsp-server/rtsp-client.c:
7739 client: do configuration of transport in one place
7740 Move the configuration of the transport destination address to where we also
7741 configure the other bits.
7743 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7745 * gst/rtsp-server/rtsp-client.c:
7746 client: use media multicast group
7748 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7750 * gst/rtsp-server/rtsp-media-factory.h:
7751 * gst/rtsp-server/rtsp-server.h:
7752 * gst/rtsp-server/rtsp-session-pool.h:
7753 * gst/rtsp-server/rtsp-session.h:
7756 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
7758 * gst/rtsp-server/rtsp-client.c:
7759 * gst/rtsp-server/rtsp-sdp.h:
7760 sdp: copy and free the server ip address
7761 Copy and free the server ip address to make memory management easier later.
7763 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7765 * gst/rtsp-server/rtsp-media-factory.c:
7766 media-factory: configure multicast in media
7768 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7770 * gst/rtsp-server/rtsp-media.c:
7771 * gst/rtsp-server/rtsp-media.h:
7772 media: add property for multicast group
7773 Add a property to configure the multicast group in the media.
7774 Based on patches from Marc Leeman and Robert Krakora.
7776 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7778 * gst/rtsp-server/rtsp-media-factory.c:
7779 * gst/rtsp-server/rtsp-media-factory.h:
7780 media-factory: add property for multicast group
7781 Add a property to configure the multicast group in the media factory.
7782 Based on patches from Marc Leeman and Robert Krakora.
7784 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7786 * gst/rtsp-server/rtsp-client.c:
7787 client: do configuration of transport in one place
7788 Move the configuration of the transport destination address to where we also
7789 configure the other bits.
7791 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7793 Merge branch 'master' into 0.11
7795 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
7797 * gst/rtsp-server/rtsp-client.c:
7798 client: destroy pipeline on client disconnect with no prior TEARDOWN.
7799 The problem occurs when the client abruptly closes the connection without
7800 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
7801 server is where the pipeline gets torn down. Since this handler is not called,
7802 the pipeline remains and is up and running. Subsequent clients get their own
7803 pipelines and if the do not issue TEARDOWNs then those pipelines will also
7804 remain up and running. This is a resource leak.
7806 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7808 Merge branch 'master' into 0.11
7810 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
7812 * gst/rtsp-server/rtsp-media-factory.c:
7813 * gst/rtsp-server/rtsp-media-factory.h:
7814 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
7815 For example, it can be used to retrieve source elements like appsrc, in a more
7816 convenient way than subclassing get_element.
7818 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7820 Merge branch 'master' into 0.11
7822 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
7824 * gst/rtsp-server/rtsp-server.c:
7825 rtsp-server: hold on to reference while using object
7827 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7829 * gst/rtsp-server/rtsp-media.c:
7832 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7835 configure: use unstable api
7837 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
7839 * gst/rtsp-server/rtsp-client.c:
7840 client: fix reference counting
7842 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
7844 * gst/rtsp-server/rtsp-client.c:
7845 * gst/rtsp-server/rtsp-media.c:
7846 fix compiler warnings about unused variables
7848 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
7850 * examples/test-launch.c:
7851 * examples/test-readme.c:
7852 * examples/test-uri.c:
7853 * examples/test-video.c:
7854 examples: tell rtsp uri when ready
7856 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
7859 Automatic update of common submodule
7860 From 69b981f to 605cd9a
7862 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7864 * gst/rtsp-server/rtsp-client.c:
7865 client: update for buffer API change
7867 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7869 * gst/rtsp-server/Makefile.am:
7870 Makefile.am: 0.10 => @GST_MAJORMINOR@
7872 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7874 * gst/rtsp-server/rtsp-media-factory-uri.c:
7875 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
7877 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7879 * gst/rtsp-server/.gitignore:
7880 .gitignore: 0.10 => 0.11
7882 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7884 * gst/rtsp-server/Makefile.am:
7885 Makefile.am: 0.10 => @GST_MAJORMINOR@
7887 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7889 Merge branch 'master' into 0.11
7891 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
7894 Automatic update of common submodule
7895 From 9e5bbd5 to 69b981f
7897 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
7900 Automatic update of common submodule
7901 From fd35073 to 9e5bbd5
7903 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
7906 Automatic update of common submodule
7907 From 46dfcea to fd35073
7909 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7911 * gst/rtsp-server/rtsp-media-factory-uri.c:
7912 * gst/rtsp-server/rtsp-media.c:
7913 media: port to new caps API
7915 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7917 Merge branch 'master' into 0.11
7919 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
7921 * bindings/vala/gst-rtsp-server-0.10.vapi:
7922 Updated Vala bindings.
7923 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
7925 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
7927 * gst/rtsp-server/rtsp-server.c:
7928 * gst/rtsp-server/rtsp-server.h:
7929 Add a signal for newly connected clients.
7930 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
7932 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
7934 * bindings/python/rtspserver.override:
7935 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
7937 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7939 * gst/rtsp-server/Makefile.am:
7940 * gst/rtsp-server/rtsp-client.c:
7941 * gst/rtsp-server/rtsp-funnel.c:
7942 * gst/rtsp-server/rtsp-funnel.h:
7943 * gst/rtsp-server/rtsp-media.c:
7944 rtsp-server: port to 0.11
7946 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7951 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7953 Merge branch 'master' into 0.11
7958 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7961 Automatic update of common submodule
7962 From c3cafe1 to 46dfcea
7964 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
7966 * bindings/python/Makefile.am:
7967 * bindings/python/rtspserver.defs:
7968 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
7970 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
7972 * bindings/python/arg-types.py:
7973 python bindings: add GstRTSPUrlParam
7974 Needed to implement MediaFactory virtual proxies
7976 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
7978 * bindings/python/arg-types.py:
7979 python bindings: fix returning GstRTSPUrl types
7981 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
7983 * bindings/python/arg-types.py:
7984 python bindings: add arg type for GstRTSPUrl
7986 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
7988 * bindings/python/rtspserver.defs:
7989 python bindings: fix the definition of MediaFactory.collect_stream
7991 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
7994 Automatic update of common submodule
7995 From 1ccbe09 to c3cafe1
7997 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8000 Automatic update of common submodule
8001 From 193b717 to 1ccbe09
8003 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
8006 Automatic update of common submodule
8007 From b77e2bf to 193b717
8009 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8012 build: Include lcov.mak to allow test coverage report generation
8014 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8017 Automatic update of common submodule
8018 From d8814b6 to b77e2bf
8020 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8023 Automatic update of common submodule
8024 From 6aaa286 to d8814b6
8026 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
8029 Automatic update of common submodule
8030 From 6aec6b9 to 6aaa286
8032 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
8035 autogen: wingo signed comment
8037 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
8039 * gst/rtsp-server/rtsp-session-pool.c:
8040 session: use full charset for RTSP session ID
8041 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
8042 session ID more difficult.
8043 https://bugzilla.gnome.org/show_bug.cgi?id=643812
8045 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8047 * gst/rtsp-server/Makefile.am:
8048 rtsp-server: Don't install the funnel header
8050 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
8053 Automatic update of common submodule
8054 From 1de7f6a to 6aec6b9
8056 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8059 configure: require core/base 0.10.31
8060 Needed at least for gst_plugin_feature_rank_compare_func().
8062 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
8065 Automatic update of common submodule
8066 From f94d739 to 1de7f6a
8068 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8070 * gst/rtsp-server/rtsp-media.c:
8071 media: remove more unused code
8073 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8075 * gst/rtsp-server/rtsp-media.c:
8076 * gst/rtsp-server/rtsp-media.h:
8077 media: remove duplicate filtering
8078 Remove the duplicate filtering code now that we have a released -good version.
8079 Give a warning instead.
8081 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8083 * gst/rtsp-server/rtsp-media-factory.c:
8084 * gst/rtsp-server/rtsp-media.c:
8085 media: fix default buffer size
8087 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8089 * gst/rtsp-server/rtsp-media-factory.c:
8090 * gst/rtsp-server/rtsp-media-factory.h:
8091 media-factory: add property to configure the buffer-size
8092 Add a property to configure the kernel UDP buffer size.
8094 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8096 * gst/rtsp-server/rtsp-media.c:
8097 * gst/rtsp-server/rtsp-media.h:
8098 media: add property to configure kernel buffer sizes
8099 Add a property to configure the kernel UDP buffer size.
8101 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8104 configure: set PYGOBJECT_REQ before using it
8105 https://bugzilla.gnome.org/show_bug.cgi?id=640641
8107 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8110 docs: recursive into sub-directories on 'make upload'
8112 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8114 * docs/libs/gst-rtsp-server-docs.sgml:
8115 * docs/version.entities.in:
8116 docs: mention full version these docs are for, not just major-minor
8118 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8123 === release 0.10.8 ===
8125 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8130 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8132 * gst/rtsp-server/rtsp-server.c:
8133 rtsp-server: clarify docs a little
8135 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8137 * gst/rtsp-server/rtsp-media.c:
8138 media: init debug category before starting thread
8140 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8142 * gst/rtsp-server/rtsp-auth.c:
8143 auth: add realm to make it more spec compliant
8145 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8147 * gst/rtsp-server/rtsp-server.c:
8148 * gst/rtsp-server/rtsp-server.h:
8151 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8153 * examples/test-video.c:
8154 example: improve example docs a little
8156 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8158 * gst/rtsp-server/rtsp-server.c:
8159 server: ensure the watch has a ref to the server
8161 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8163 * gst/rtsp-server/rtsp-server.c:
8164 server: simpify channel function
8166 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8168 * gst/rtsp-server/rtsp-server.c:
8169 * gst/rtsp-server/rtsp-server.h:
8170 server: simplify management of channel and source
8171 We don't need to keep around the channel and source objects. Let the mainloop
8172 and the source manage the source and channel respectively.
8174 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8180 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8183 * tests/Makefile.am:
8184 * tests/test-cleanup.c:
8185 tests: add tests directory and cleanup test
8187 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8189 * gst/rtsp-server/rtsp-media-factory-uri.c:
8190 * gst/rtsp-server/rtsp-media-factory.c:
8191 * gst/rtsp-server/rtsp-media-mapping.c:
8192 * gst/rtsp-server/rtsp-media.c:
8193 * gst/rtsp-server/rtsp-session-pool.c:
8194 * gst/rtsp-server/rtsp-session.c:
8195 server: improve debugging in various objects
8197 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8199 * gst/rtsp-server/rtsp-server.c:
8200 server: chain up to the parent finalize
8202 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
8204 * bindings/python/rtspserver-types.defs:
8205 * bindings/python/rtspserver.defs:
8206 * bindings/python/rtspserver.override:
8207 * bindings/python/test.py:
8208 gst-rtsp-server: update python bindings
8210 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8212 * gst/rtsp-server/rtsp-client.c:
8213 client: use the response from the clientstate
8214 Create the response object only once and store in the client state.
8215 Make all methods use the state response,
8217 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8219 * gst/rtsp-server/rtsp-server.c:
8220 server: use signal to keep track of clients
8221 Keep track of all the clients that the server creates and remove them when they
8222 fire the 'closed' signal.
8224 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8226 * gst/rtsp-server/rtsp-client.c:
8227 * gst/rtsp-server/rtsp-client.h:
8228 client: emit signal when closing
8230 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8232 * examples/.gitignore:
8233 * examples/Makefile.am:
8234 * examples/test-auth.c:
8235 * examples/test-video.c:
8236 * gst/rtsp-server/rtsp-auth.c:
8237 * gst/rtsp-server/rtsp-auth.h:
8238 * gst/rtsp-server/rtsp-client.c:
8239 * gst/rtsp-server/rtsp-media-factory.c:
8240 * gst/rtsp-server/rtsp-media.c:
8241 * gst/rtsp-server/rtsp-media.h:
8242 * gst/rtsp-server/rtsp-session-pool.h:
8243 * gst/rtsp-server/rtsp-session.h:
8244 media: enable per factory authorisations
8245 Allow for adding a GstRTSPAuth on the factory and media level and check
8246 permissions when accessing the factory.
8247 Add hints to the auth methods for future more fine grained authorisation.
8248 Add example application for per factory authentication.
8250 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8252 * gst/rtsp-server/rtsp-auth.c:
8253 * gst/rtsp-server/rtsp-auth.h:
8254 * gst/rtsp-server/rtsp-client.c:
8255 * gst/rtsp-server/rtsp-client.h:
8256 * gst/rtsp-server/rtsp-params.c:
8257 * gst/rtsp-server/rtsp-params.h:
8258 rtsp-server: Pass ClientState structure arround
8259 Pass the collected information for the ongoing request in a GstRTSPClientState
8260 structure that we can then pass around to simplify the method arguments. This
8261 will also be handy when we implement logging functionality.
8263 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8265 * gst/rtsp-server/rtsp-media-factory.c:
8266 * gst/rtsp-server/rtsp-media-factory.h:
8267 media-factory: add methods to configure authorisation
8269 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8271 * gst/rtsp-server/rtsp-client.c:
8272 client: unref auth in finalize
8274 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8276 * gst/rtsp-server/rtsp-server.c:
8277 server: unref auth in finalize
8279 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8281 * docs/libs/gst-rtsp-server-docs.sgml:
8282 * docs/libs/gst-rtsp-server-sections.txt:
8283 * docs/libs/gst-rtsp-server.types:
8286 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8288 * gst/rtsp-server/rtsp-server.c:
8289 * gst/rtsp-server/rtsp-server.h:
8290 server: separate create and accept
8291 Create separate create and accept methods so that subclasses can create custom
8293 Configure the server in the client object and prepare for keeping track of
8296 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8298 * gst/rtsp-server/rtsp-client.c:
8299 * gst/rtsp-server/rtsp-client.h:
8300 client: add support for setting the server.
8301 Add support for keeping a ref to the server that started this client
8304 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8306 * gst/rtsp-server/rtsp-auth.c:
8307 auth: fix memleak and add some docs
8308 Fix a memleak of the basic auth token.
8309 Add docs for the helper function
8311 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8313 * gst/rtsp-server/rtsp-auth.c:
8314 * gst/rtsp-server/rtsp-auth.h:
8315 * gst/rtsp-server/rtsp-client.c:
8316 client: delegate setup of auth to the manager
8317 Delegate the configuration of the authentication tokens to the manager object
8320 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8322 * examples/test-video.c:
8323 * gst/rtsp-server/Makefile.am:
8324 * gst/rtsp-server/rtsp-auth.c:
8325 * gst/rtsp-server/rtsp-auth.h:
8326 * gst/rtsp-server/rtsp-client.c:
8327 * gst/rtsp-server/rtsp-client.h:
8328 * gst/rtsp-server/rtsp-server.c:
8329 * gst/rtsp-server/rtsp-server.h:
8330 auth: add authentication object
8331 Add an object that can check the authorization of requests.
8332 Implement basic authentication.
8333 Add example authentication to test-video
8335 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8337 * gst/rtsp-server/rtsp-server.c:
8338 * gst/rtsp-server/rtsp-server.h:
8339 server: move includes back
8340 the includes are needed for sockaddr_in.
8342 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8344 * gst/rtsp-server/rtsp-client.c:
8345 * gst/rtsp-server/rtsp-client.h:
8346 * gst/rtsp-server/rtsp-server.c:
8347 * gst/rtsp-server/rtsp-server.h:
8348 rtsp: move network includes where they are needed
8350 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
8352 * gst/rtsp-server/rtsp-media.h:
8353 rtsp-media.h: Minor corrections in comments.
8356 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
8359 Automatic update of common submodule
8360 From e572c87 to f94d739
8362 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8366 * docs/libs/.gitignore:
8367 * examples/.gitignore:
8368 * gst/rtsp-server/.gitignore:
8371 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8373 * docs/libs/Makefile.am:
8374 docs: We don't build ps/pdf for API reference docs
8376 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8379 Automatic update of common submodule
8380 From ccbaa85 to e572c87
8382 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8385 Automatic update of common submodule
8386 From 46445ad to ccbaa85
8388 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8390 * gst/rtsp-server/Makefile.am:
8391 * gst/rtsp-server/rtsp-funnel.c:
8392 * gst/rtsp-server/rtsp-funnel.h:
8393 * gst/rtsp-server/rtsp-media.c:
8394 funnel: rename fsfunnel to rtspfunnel
8395 Rename the funnel to avoid conflicts with the farsight one.
8397 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8399 * gst/rtsp-server/Makefile.am:
8400 * gst/rtsp-server/fs-funnel.c:
8401 * gst/rtsp-server/fs-funnel.h:
8402 * gst/rtsp-server/rtsp-media.c:
8403 rtsp-media: add and use fsfunnel
8404 Add a copy of fsfunnel to the build because input-selector removed the (broken)
8405 select-all property that we need.
8407 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8409 * gst/rtsp-server/Makefile.am:
8410 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
8411 Use PKG_CONFIG_PATH specified at configure time (if any) as well
8412 for the g-ir-compiler, rather than just assuming the env var has
8415 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8422 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
8424 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8427 * gst/rtsp-server/Makefile.am:
8428 gobject-introspection: fix g-i build for uninstalled setup
8429 Requires gst-plugins-base git (> 0.10.31.2).
8431 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8433 * examples/test-uri.c:
8434 examples: add some more options and comments
8436 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8438 * gst/rtsp-server/rtsp-media-factory-uri.c:
8439 factory-uri: use right property type
8441 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8443 * gst/rtsp-server/rtsp-media-factory-uri.c:
8444 factory-uri: attempt to configure buffer-lists
8445 Attempt to configure buffer lists in the payloader for improved performance.
8447 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8449 * gst/rtsp-server/rtsp-media.c:
8450 media: attempt to configure bigger UDP buffers
8451 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
8452 send buffers with high bitrate streams.
8454 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
8456 * gst/rtsp-server/rtsp-client.c:
8457 client: use the socket length from getsockname
8458 Use the length returned by getsockname to perform the getnameinfo call because
8459 the size can depend on the socket type and platform.
8462 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8464 * docs/libs/gst-rtsp-server-docs.sgml:
8465 * docs/libs/gst-rtsp-server-sections.txt:
8466 docs: add uri factory to the docs
8468 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8470 * gst/rtsp-server/rtsp-client.c:
8471 * gst/rtsp-server/rtsp-media.h:
8474 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8476 * gst/rtsp-server/rtsp-client.c:
8477 * gst/rtsp-server/rtsp-media.c:
8478 * gst/rtsp-server/rtsp-media.h:
8479 * gst/rtsp-server/rtsp-session.c:
8480 * gst/rtsp-server/rtsp-session.h:
8481 rtsp-server: add support for buffer lists
8482 Add support for sending bufferlists received from appsink.
8485 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8487 * gst/rtsp-server/rtsp-client.c:
8488 * gst/rtsp-server/rtsp-media.c:
8489 * gst/rtsp-server/rtsp-media.h:
8490 * gst/rtsp-server/rtsp-sdp.c:
8491 media: make method to retrieve the play range
8492 Make a method to retrieve the playback range so that we can conditionally create
8493 a different range for the SDP and the PLAY requests.
8495 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8497 * gst/rtsp-server/rtsp-media.c:
8498 * gst/rtsp-server/rtsp-media.h:
8499 media: add signal to notify of state changes
8501 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8503 * gst/rtsp-server/rtsp-client.h:
8504 client: cleanup headers
8506 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8508 * gst/rtsp-server/rtsp-client.c:
8511 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8513 * gst/rtsp-server/rtsp-media-factory-uri.c:
8514 * gst/rtsp-server/rtsp-media-factory-uri.h:
8515 factory-uri: add support for gstpay
8516 Add an option to prefer gstpay over decoder + raw payloader.
8518 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8520 * gst/rtsp-server/rtsp-media-factory-uri.c:
8521 * gst/rtsp-server/rtsp-media-factory-uri.h:
8522 factory-uri: rework the autoplugger.
8523 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
8526 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8528 * gst/rtsp-server/rtsp-media-factory-uri.c:
8529 factory-uri: use better factory filter
8530 Make better payloader filter based on autoplug rank and RTP use case.
8532 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8535 Automatic update of common submodule
8536 From 169462a to 46445ad
8538 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8540 * gst/rtsp-server/rtsp-server.c:
8541 server: set SO_REUSEADDR before bind
8542 Set the SO_REUSEADDR _before_ bind() to make it actually work.
8544 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8546 * gst/rtsp-server/rtsp-media.c:
8547 * gst/rtsp-server/rtsp-media.h:
8548 media: emit prepared signal when prepared
8549 Make a 'prepared' signal and emit it when we successfully prepared the element.
8550 This signal can be used to configure the media object after it has been prepared
8553 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
8556 Automatic update of common submodule
8557 From 011bcc8 to 169462a
8559 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
8561 python an optional dependency
8562 * configure.ac: Move up valgrind and g-i checks. Make the python
8563 dependency optional, as it was before.
8565 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8567 Merge branch 'master' into 0.11
8572 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8574 * gst/rtsp-server/rtsp-media.c:
8575 media: update range when active clients changed
8576 When we changed the number of active clients, update the current range
8577 information because we want the second client connecting to a shared resource
8578 continue from where the stream currently.
8580 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8582 * gst/rtsp-server/rtsp-media-factory-uri.c:
8583 * gst/rtsp-server/rtsp-media-factory-uri.h:
8584 factory-uri: add colorspace and fix pt
8585 Rework the way we pass data to the autoplugger.
8586 When we have raw caps, plug a converter element to make pluggin to raw
8587 payloaders more successful.
8588 Make sure all dynamically plugged payloaders have a unique payload types.
8590 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8592 * examples/Makefile.am:
8593 * examples/test-uri.c:
8594 example: add example of the uri factory
8596 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8598 * gst/rtsp-server/Makefile.am:
8599 * gst/rtsp-server/rtsp-media-factory-uri.c:
8600 * gst/rtsp-server/rtsp-media-factory-uri.h:
8601 * gst/rtsp-server/rtsp-server.h:
8602 factory-uri: add a factory to stream any URI
8603 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
8606 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8608 * gst/rtsp-server/rtsp-media.c:
8609 * gst/rtsp-server/rtsp-media.h:
8610 media: ignore spurious ASYNC_DONE messages
8611 When we are dynamically adding pads, the addition of the udpsrc elements will
8612 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
8613 the real ASYNC_DONE when everything is prerolled.
8615 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8617 * gst/rtsp-server/rtsp-media-factory.c:
8618 * gst/rtsp-server/rtsp-media-factory.h:
8619 media-factory: make lock macro
8621 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
8623 * gst/rtsp-server/rtsp-client.c:
8624 rtsp-server: Remove unused variable and dead assignment
8626 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
8628 * examples/test-launch.c:
8629 * examples/test-mp4.c:
8630 * examples/test-ogg.c:
8631 * examples/test-readme.c:
8632 * examples/test-sdp.c:
8633 * examples/test-video.c:
8634 examples: Run gst-indent
8636 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
8638 * gst/rtsp-server/rtsp-client.c:
8639 * gst/rtsp-server/rtsp-media-factory.c:
8640 * gst/rtsp-server/rtsp-media-mapping.c:
8641 * gst/rtsp-server/rtsp-media.c:
8642 * gst/rtsp-server/rtsp-params.c:
8643 * gst/rtsp-server/rtsp-sdp.c:
8644 * gst/rtsp-server/rtsp-server.c:
8645 * gst/rtsp-server/rtsp-session-pool.c:
8646 * gst/rtsp-server/rtsp-session.c:
8647 rtsp-server: Run gst-indent
8648 Since it wasn't using the upstream common previously, there was no
8649 indentation check before commiting.
8651 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
8653 * gst/rtsp-server/rtsp-media-mapping.h:
8654 * gst/rtsp-server/rtsp-media.c:
8655 * gst/rtsp-server/rtsp-media.h:
8656 * gst/rtsp-server/rtsp-sdp.c:
8657 * gst/rtsp-server/rtsp-session-pool.h:
8658 * gst/rtsp-server/rtsp-session.c:
8659 * gst/rtsp-server/rtsp-session.h:
8660 rtsp-server: Some more doc fixups
8662 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8665 Makefile: Add cruft-cleaning support
8667 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8672 * docs/libs/Makefile.am:
8673 * docs/libs/gst-rtsp-server-docs.sgml:
8674 * docs/libs/gst-rtsp-server-sections.txt:
8675 * docs/libs/gst-rtsp-server.types:
8676 * docs/version.entities.in:
8677 docs: Add gtk-doc build system
8679 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8681 * gst/rtsp-server/Makefile.am:
8682 Makefile.am: Use standard GIR make behaviour
8684 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8688 autogen/configure: Bring more in sync to standard gst module behaviour
8690 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8692 * gst/rtsp-server/rtsp-media.c:
8693 media: warn and fail when gstrtpbin is not found
8695 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8698 configure: open 0.11 branch
8700 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
8704 Add common submodule
8706 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
8709 * common/Makefile.am:
8710 * common/c-to-xml.py:
8712 * common/coverage/coverage-report-entry.pl:
8713 * common/coverage/coverage-report.pl:
8714 * common/coverage/coverage-report.xsl:
8715 * common/coverage/lcov.mak:
8716 * common/gettext.patch:
8717 * common/glib-gen.mak:
8718 * common/gst-autogen.sh:
8719 * common/gst-xmlinspect.py:
8721 * common/gstdoc-scangobj:
8722 * common/gtk-doc-plugins.mak:
8723 * common/gtk-doc.mak:
8724 * common/m4/.gitignore:
8725 * common/m4/Makefile.am:
8727 * common/m4/as-ac-expand.m4:
8728 * common/m4/as-auto-alt.m4:
8729 * common/m4/as-compiler-flag.m4:
8730 * common/m4/as-compiler.m4:
8731 * common/m4/as-docbook.m4:
8732 * common/m4/as-libtool-tags.m4:
8733 * common/m4/as-libtool.m4:
8734 * common/m4/as-python.m4:
8735 * common/m4/as-scrub-include.m4:
8736 * common/m4/as-version.m4:
8737 * common/m4/ax_create_stdint_h.m4:
8738 * common/m4/check.m4:
8739 * common/m4/glib-gettext.m4:
8740 * common/m4/gst-arch.m4:
8741 * common/m4/gst-args.m4:
8742 * common/m4/gst-check.m4:
8743 * common/m4/gst-debuginfo.m4:
8744 * common/m4/gst-default.m4:
8745 * common/m4/gst-doc.m4:
8746 * common/m4/gst-error.m4:
8747 * common/m4/gst-feature.m4:
8748 * common/m4/gst-function.m4:
8749 * common/m4/gst-gettext.m4:
8750 * common/m4/gst-glib2.m4:
8751 * common/m4/gst-libxml2.m4:
8752 * common/m4/gst-plugindir.m4:
8753 * common/m4/gst-valgrind.m4:
8754 * common/m4/gtk-doc.m4:
8755 * common/m4/introspection.m4:
8757 * common/mangle-tmpl.py:
8758 * common/plugins.xsl:
8760 * common/release.mak:
8761 * common/scangobj-merge.py:
8762 * common/upload.mak:
8763 common: Remove static version
8765 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
8767 * common/m4/introspection.m4:
8768 Update introspection.m4 to match usage
8770 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8774 Remove old stuff from the README
8776 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8781 === release 0.10.7 ===
8783 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8788 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8790 * examples/test-ogg.c:
8791 test-ogg: remove parsers
8792 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
8793 buffers with timestamps. Using the parsers also seems to break things.
8795 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8797 * bindings/vala/gst-rtsp-server-0.10.vapi:
8798 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8799 Updated Vala bindings
8801 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8803 * common/m4/introspection.m4:
8805 * gst/rtsp-server/Makefile.am:
8806 Added initial gobject-introspection support
8808 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8810 * gst/rtsp-server/rtsp-media-factory.c:
8811 media-factory: don't use host for shared hash key
8812 When we generate the key to share made between connections, don't include the
8813 host used to connect so that we can share media even if between clients that
8814 connected with localhost and ones with the ip address.
8816 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8818 * bindings/vala/Makefile.am:
8819 build: fix distcheck
8821 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8823 * bindings/vala/gst-rtsp-server-0.10.vapi:
8824 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8825 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8826 Update Vala bindings
8828 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8830 * bindings/vala/Makefile.am:
8832 Fix configure checks and installation location for Vala bindings
8835 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8840 === release 0.10.6 ===
8842 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8845 configure: release 0.10.6
8847 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8849 * gst/rtsp-server/rtsp-media.c:
8850 media: help the compiler a little
8852 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8854 * gst/rtsp-server/rtsp-media.c:
8855 * gst/rtsp-server/rtsp-media.h:
8856 * gst/rtsp-server/rtsp-session.c:
8857 media: cleanup media transport before freeing
8858 Cleanup the media transport data before freeing. In particular, remove the qdata
8859 from the rtpsource object.
8861 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8863 * gst/rtsp-server/rtsp-media-factory.c:
8864 * gst/rtsp-server/rtsp-media-factory.h:
8865 * gst/rtsp-server/rtsp-media.c:
8866 * gst/rtsp-server/rtsp-media.h:
8867 media-factory: add eos-shutdown property
8868 Add an eos-shutdown property that will send an EOS to the pipeline before
8869 shutting it down. This allows for nice cleanup in case of a muxer.
8872 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8874 * gst/rtsp-server/rtsp-media.c:
8875 * gst/rtsp-server/rtsp-media.h:
8876 media: use multiudpsink send-duplicates when we can
8877 If we have a new enough multiudpsink with the send-duplicates property, use this
8878 instead of doing our own filtering. Our custom filtering code should eventually
8879 be removed when we can depend on a released -good.
8881 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8883 * gst/rtsp-server/rtsp-media.c:
8884 media: don't leak destinations
8885 Refactor and cleanup the destinations array when the stream is destroyed.
8887 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8889 * gst/rtsp-server/rtsp-media.c:
8890 * gst/rtsp-server/rtsp-media.h:
8891 media: don't add udp addresses multiple times
8892 Keep track of the udp addresses we added to udpsink and never add the same udp
8893 destination twice. This avoids duplicate packets when using multicast.
8895 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8897 * gst/rtsp-server/rtsp-server.c:
8898 server: disable use of SO_LINGER
8899 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
8900 server close()s the connection.
8902 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8904 * gst/rtsp-server/rtsp-server.c:
8905 server: use 5 second linger period in SO_LINGER
8906 Wait 5 seconds before clearing the send buffers and reseting the connection with
8907 the client when we do a close. This should be enough time to get the message to
8911 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
8913 * gst/rtsp-server/rtsp-server.c:
8914 server: use SO_LINGER
8915 SO_LINGER on the socket will make sure that any pending data on the socket is
8916 flushed ASAP and that the socket connection is reset. This makes sure that the
8917 socket can be reused immediately.
8920 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8923 README: add blurb about shared media factories
8925 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
8927 * gst/rtsp-server/rtsp-media.c:
8928 Add stdlib.h for atoi()
8930 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8932 * bindings/python/Makefile.am:
8933 * bindings/vala/Makefile.am:
8934 build: distcheck fixes
8935 Fix 'make distcheck', somewhat (it still fails because it tries to
8936 install files into /usr/share/vala/vapi/ irrespective of the
8939 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8942 configure: bump core/base requirements to released version
8943 Makes things less confusing for people.
8945 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8948 configure: fail if GStreamer core/base requirements are not met
8950 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8952 * gst/rtsp-server/rtsp-client.c:
8953 client: improve client cleanups
8954 Make sure the session does not timeout when using TCP. We need to do this
8955 because quicktime player does not send RTCP for some reason in tunneled
8957 Refactor some cleanup code.
8960 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8962 * gst/rtsp-server/rtsp-session.c:
8963 * gst/rtsp-server/rtsp-session.h:
8964 session: add support for prevent session timeouts
8965 Add an atomix counter to prevent session timeouts when we are, for example,
8968 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8970 * gst/rtsp-server/rtsp-client.c:
8971 client: fix unlink on session timeouts
8972 When our session times out, make sure we unlink all streams in this
8974 Remove the tunnelid when closing the connection.
8976 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8978 * gst/rtsp-server/rtsp-session.c:
8979 session: small cleanups
8981 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8983 * gst/rtsp-server/rtsp-client.c:
8984 client: handle lost_tunnel callbacks
8985 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
8986 hashtable so that we can reuse it for when the client reopens the POST
8988 Close the connection after a TEARDOWN.
8989 Make sure or watchid is cleared when the watch is removed.
8992 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8994 * gst/rtsp-server/rtsp-client.c:
8995 * gst/rtsp-server/rtsp-media.c:
8996 * gst/rtsp-server/rtsp-sdp.c:
8997 rtsp-server: add more support for multicast
8999 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9002 * gst/rtsp-server/rtsp-media.c:
9003 * gst/rtsp-server/rtsp-media.h:
9004 media: allow configuration of allowed lower transport
9006 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9008 * gst/rtsp-server/rtsp-client.h:
9009 * gst/rtsp-server/rtsp-media.c:
9010 * gst/rtsp-server/rtsp-media.h:
9011 * gst/rtsp-server/rtsp-sdp.c:
9012 * gst/rtsp-server/rtsp-sdp.h:
9013 * gst/rtsp-server/rtsp-server.c:
9014 rtsp: keep track of server ip and ipv6
9015 Keep track of how the client connected to the server and setup the udp ports
9016 with the same protocol.
9017 Copy the server ip address in the SDP so that clients can send RTCP back to
9020 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9022 * gst/rtsp-server/rtsp-session.c:
9025 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9027 * gst/rtsp-server/rtsp-client.c:
9028 client: use right size for malloc
9030 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9032 * gst/rtsp-server/rtsp-server.c:
9033 server: comment ipv6 server listening address
9035 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9037 * gst/rtsp-server/rtsp-media.c:
9038 media: allow for ipv6 sockets
9040 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9042 * gst/rtsp-server/rtsp-server.c:
9043 * gst/rtsp-server/rtsp-server.h:
9044 server: rework server part
9045 Allow setting a bind address, make sure we can deal with ipv6.
9046 Remove the port property and change with the service property.
9048 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9050 * gst/rtsp-server/rtsp-media.h:
9051 media: update comments a little
9053 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9055 * gst/rtsp-server/rtsp-client.c:
9056 client: make content-base better
9057 Use the URI formatting functions to make a content-base. Also make sure that
9058 there is a trailing / at the end.
9060 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9062 * gst/rtsp-server/rtsp-client.c:
9063 client: guard against invalid paths
9065 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9067 * examples/test-video.c:
9068 test: catch server bind errors
9070 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
9072 * gst/rtsp-server/rtsp-media.c:
9073 rtspmedia: emit "unprepared" if _prepare fails.
9074 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
9075 media object is removed from its factory's cache.
9077 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9079 * gst/rtsp-server/rtsp-media.c:
9080 media: collect media position when seek completes
9082 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
9084 * gst/rtsp-server/rtsp-client.c:
9085 client: call unlink_streams in client finalize
9088 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9090 * gst/rtsp-server/rtsp-media.c:
9091 media: limit the time to wait to something huge
9092 Avoid waiting forever but limit the timeout to 20 seconds.
9094 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9096 * gst/rtsp-server/rtsp-sdp.c:
9097 sdp: reindent and check for prepared status
9099 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9101 * gst/rtsp-server/rtsp-media.c:
9102 * gst/rtsp-server/rtsp-media.h:
9103 * gst/rtsp-server/rtsp-session.c:
9104 media: avoid doing _get_state() for state changes
9105 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
9106 until the media is prerolled or in error. This avoids doing a blocking call of
9107 gst_element_get_state() that can cause lockups when there is an error.
9110 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9112 * gst/rtsp-server/rtsp-media.c:
9115 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9117 * gst/rtsp-server/rtsp-media-factory.c:
9118 media-factory: better error handling
9119 Improve the error handling a bit.
9121 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9123 * gst/rtsp-server/rtsp-client.c:
9124 client: rework transport parsing
9125 Rework the transport parsing code so that we can ignore transports we don't
9126 support instead of just picking the first one we can parse.
9127 Configure a (for now hardcoded) destination for multicast transports.
9129 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9131 * gst/rtsp-server/rtsp-media.c:
9132 media: set multicast sink parameters
9133 Disable loop and automatic multicast join on the udpsink elements.
9134 Add some more debug info.
9135 Reset some state variables in the right place.
9136 Use the right port numbers for multicast.
9138 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9140 * gst/rtsp-server/rtsp-session.c:
9141 session: handle transport setup correctly
9142 Handle UDP, MCAST and TCP transport negotiation more correctly.
9143 Store the server session SSRC in the transport.
9145 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9147 * gst/rtsp-server/rtsp-client.c:
9148 rtsp-client: implement error_full
9149 Implement error_full to avoid some segfaults when the rtspconnection calls it.
9152 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9155 * gst/rtsp-server/rtsp-client.c:
9156 * gst/rtsp-server/rtsp-server.c:
9157 docs: update docs and comments
9159 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
9161 * gst/rtsp-server/rtsp-sdp.c:
9162 sdp: make server work better when behind a proxy
9164 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9166 * gst/rtsp-server/rtsp-client.c:
9167 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
9169 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9171 * gst/rtsp-server/rtsp-client.c:
9172 * gst/rtsp-server/rtsp-media-factory.c:
9173 * gst/rtsp-server/rtsp-media-mapping.c:
9174 * gst/rtsp-server/rtsp-media.c:
9175 * gst/rtsp-server/rtsp-server.c:
9176 * gst/rtsp-server/rtsp-session-pool.c:
9177 * gst/rtsp-server/rtsp-session.c:
9178 Use GStreamer's debugging subsystem
9180 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9182 * gst/rtsp-server/rtsp-media-factory.c:
9183 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
9185 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9190 === release 0.10.5 ===
9192 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9197 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9200 configure: bump required versions
9202 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
9204 * gst/rtsp-server/rtsp-client.c:
9205 client: call weak-unref on client->sessions from finalize
9208 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9210 * gst/rtsp-server/rtsp-media.c:
9211 media: Fixed crasher where caps got unref'ed too often
9213 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9216 * pkgconfig/.gitignore:
9217 * pkgconfig/Makefile.am:
9218 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
9219 Added pkg-config file to use gst-rtsp-server uninstalled
9221 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9223 * gst/rtsp-server/rtsp-media.c:
9224 media: add some docs
9226 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
9228 * gst/rtsp-server/rtsp-client.c:
9229 rtsp: Use gst_rtsp_watch_send_message().
9230 Use gst_rtsp_watch_send_message() since the old API which used
9231 gst_rtsp_watch_queue_message() has been deprecated.
9233 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9238 === release 0.10.4 ===
9240 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9245 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9247 * gst/rtsp-server/rtsp-client.c:
9248 * gst/rtsp-server/rtsp-session.c:
9249 * gst/rtsp-server/rtsp-session.h:
9250 rtsp: allocate channels in TCP mode
9251 When the client does not provide us with channels in TCP mode, allocate channels
9254 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9256 * gst/rtsp-server/rtsp-client.c:
9257 client: don't crash when tunnelid is missing
9258 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
9259 don't crash but return an error response to the client.
9262 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9264 * bindings/vala/gst-rtsp-server-0.10.vapi:
9265 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9266 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9267 bindings: update vala bindings with new method
9269 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9271 * gst/rtsp-server/rtsp-session-pool.c:
9272 * gst/rtsp-server/rtsp-session-pool.h:
9273 sessionpool: add function to filter sessions
9274 Add generic function to retrieve/remove sessions.
9276 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9279 configure: bump core/base requirements to release
9281 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9283 * gst/rtsp-server/rtsp-media.c:
9284 media: fix indentation
9286 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9288 * gst/rtsp-server/rtsp-media.c:
9289 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
9291 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9293 * gst/rtsp-server/rtsp-media.c:
9294 set state and remove elements of media in for loop
9296 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
9298 * bindings/vala/gst-rtsp-server-0.10.vapi:
9299 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9300 Added gst_rtsp_media_remove_elements function to Vala bindings
9302 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
9304 * gst/rtsp-server/rtsp-media.c:
9305 * gst/rtsp-server/rtsp-media.h:
9306 Added gst_rtsp_media_remove_elements function
9308 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
9310 * gst/rtsp-server/rtsp-media.c:
9311 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
9313 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9315 * bindings/vala/gst-rtsp-server-0.10.vapi:
9316 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9317 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9318 Updated Vala bindings
9320 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9322 * gst/rtsp-server/rtsp-media.c:
9323 * gst/rtsp-server/rtsp-media.h:
9324 Added vmethod unprepare to GstRTSPMedia
9325 The default implementation sets the state of the pipeline to GST_STATE_NULL
9327 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9329 * gst/rtsp-server/rtsp-media-factory.c:
9330 * gst/rtsp-server/rtsp-media-factory.h:
9331 Made collect_streams function public
9333 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9335 * gst/rtsp-server/rtsp-media-factory.c:
9336 * gst/rtsp-server/rtsp-media-factory.h:
9337 * gst/rtsp-server/rtsp-media.c:
9338 Added vmethod create_pipeline to GstRTSPMediaFactory
9339 The pipeline is created in this method and the GstRTSPMedia's element is added to it
9341 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9343 * gst/rtsp-server/rtsp-client.c:
9344 client: use g_source_destroy()
9345 We need to use g_source_destroy() because we might have added the source to a
9346 different main context than the default one.
9348 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9350 * gst/rtsp-server/Makefile.am:
9351 * gst/rtsp-server/rtsp-client.c:
9352 * gst/rtsp-server/rtsp-params.c:
9353 * gst/rtsp-server/rtsp-params.h:
9354 rtsp: prepare for handling GET/SET_PARAMETER
9355 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
9357 Fix return codes of handlers.
9359 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9361 * gst/rtsp-server/rtsp-media.c:
9362 media: don't leak session pads
9364 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9366 * gst/rtsp-server/rtsp-media.c:
9367 media: clean up the messages a bit
9369 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9371 * gst/rtsp-server/rtsp-sdp.c:
9372 sdp: warn and skip streams without media
9374 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9376 * bindings/vala/gst-rtsp-server-0.10.vapi:
9377 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9378 vala: Fixed typo in header file of RTSPMediaStream
9380 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9382 * gst/rtsp-server/rtsp-media.c:
9385 Make dumping RTCP stats configurable
9387 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9389 * gst/rtsp-server/rtsp-media.c:
9390 media: be less verbose and leak less
9392 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9394 * gst/rtsp-server/rtsp-media.c:
9395 media: don't leak the destination address
9397 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9399 * gst/rtsp-server/rtsp-client.c:
9400 * gst/rtsp-server/rtsp-media.c:
9401 * gst/rtsp-server/rtsp-media.h:
9402 * gst/rtsp-server/rtsp-session.c:
9403 * gst/rtsp-server/rtsp-session.h:
9404 rtsp: use RTCP to keep the session alive
9405 Use the RTCP rtcp-from stats field to find the associated session and use this
9406 to keep the session alive.
9408 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9410 * gst/rtsp-server/rtsp-session.c:
9411 session: add 5sec to the real session timeout
9412 Allow the session to live 5sec longer before really timing out. This should give
9413 clients some extra time to keep the session active.
9415 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9417 * gst/rtsp-server/rtsp-client.c:
9418 client: replay OK to GET/SET_PARAMETER
9419 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
9420 so that we return OK for those requests.
9422 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9424 * gst/rtsp-server/rtsp-media.c:
9425 * gst/rtsp-server/rtsp-media.h:
9426 media: keep track of active transports
9427 Keep track of which transport is active to avoid closing the connection too
9429 Remove the destination transport also when going to NULL.
9430 Print some stats about the SDES and other RTCP messages we receive from the
9433 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9435 * examples/.gitignore:
9436 * examples/Makefile.am:
9437 * examples/test-sdp.c:
9438 example: add SDP relay example
9440 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9442 * gst/rtsp-server/rtsp-media.c:
9443 media: also count active TCP connections
9445 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9447 * gst/rtsp-server/rtsp-media-factory.c:
9448 * gst/rtsp-server/rtsp-media.c:
9449 * gst/rtsp-server/rtsp-media.h:
9450 rtsp: add support for dynamic elements
9451 Add support for dynamic elements.
9452 Don't set live pipelines back to paused.
9454 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9456 * gst/rtsp-server/rtsp-sdp.c:
9457 sdp: don't add encoding name when absent in caps
9459 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9461 * gst/rtsp-server/rtsp-client.c:
9462 client: warn when we can't do RTP-Info
9464 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9466 * gst/rtsp-server/rtsp-media-factory.c:
9467 factory: factor out the stream construction
9469 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9471 * gst/rtsp-server/rtsp-client.c:
9472 client: only add RTP-Info when we have the info
9473 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
9476 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9481 === release 0.10.3 ===
9483 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9487 - Fixes a bug where it put the wrong verion in pkgconfig
9488 - Link RTP and RTCP sources
9490 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9492 * gst/rtsp-server/rtsp-media.c:
9493 * gst/rtsp-server/rtsp-media.h:
9494 media: link the RTP udpsrc to the session manager
9495 Link the RTP udpsrc and the appsrc to the session manager so that they don't
9496 shut down when the client sends a packet to open firewalls.
9498 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9500 * pkgconfig/gst-rtsp-server.pc.in:
9501 Don't use hard-coded version number in pkg-config file
9503 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9508 === release 0.10.2 ===
9510 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9515 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9518 * common/m4/.gitignore:
9519 * examples/.gitignore:
9520 * pkgconfig/.gitignore:
9521 add some .gitignore files
9523 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9525 * gst/rtsp-server/rtsp-media.c:
9526 media: seek to key frames
9528 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9530 * gst/rtsp-server/rtsp-media.c:
9531 media: emit the unprepared signal by id
9532 Emit the unprepared signal by id instead of name and set the media as
9535 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9537 * gst/rtsp-server/rtsp-media.c:
9538 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
9540 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9542 * gst/rtsp-server/rtsp-server.c:
9543 Added finalize function to GstRTPSPServer to unref session pool and media mapping
9545 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9547 * bindings/vala/gst-rtsp-server-0.10.vapi:
9548 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9549 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9550 Updated vala bindings
9552 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9554 * gst/rtsp-server/Makefile.am:
9555 * gst/rtsp-server/rtsp-client.c:
9556 * gst/rtsp-server/rtsp-media.c:
9557 server: use appsink and appsrc with the API
9558 Use the appsink/appsrc API instead of the signals for higher
9561 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9563 * examples/test-ogg.c:
9564 tests: set the payload type correctly
9566 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9568 * gst/rtsp-server/rtsp-media-factory.c:
9569 factory: connect to the unprepare signal
9570 Connect to the unprepare signal for non-reusable media so that we can remove
9571 them from the cache.
9573 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9575 * gst/rtsp-server/rtsp-media.c:
9576 * gst/rtsp-server/rtsp-media.h:
9577 media: add signal to notify of unprepare
9579 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9581 * gst/rtsp-server/rtsp-media.c:
9582 * gst/rtsp-server/rtsp-media.h:
9583 media: more work on making the media shared
9584 Add a reusable flag to medias, indicating that they can be reused after a state
9588 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9590 * examples/test-readme.c:
9591 examples: mark the example as shared for testing
9593 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9595 * gst/rtsp-server/rtsp-media.c:
9596 * gst/rtsp-server/rtsp-media.h:
9597 client: support shared media
9598 Always perform the state actions even if the target state of the pipeline is
9599 already correct, we still want to add/remove the transports when we are dealing
9601 Keep a counter of the number of active transports for a media so that we can use
9602 this to perform a state change when needed.
9603 Perform a state change of the pipeline only when the first transport was added
9604 or when there are no active transports.
9606 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9608 * gst/rtsp-server/rtsp-client.c:
9609 client: fix refcounting crasher
9610 Don't need to remove the weak refs in the finalize methods, they are already
9611 removed in the dispose.
9612 Don't register the callback with a DestroyNofity.
9614 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9616 * gst/rtsp-server/rtsp-client.c:
9617 Fix rtsp client refcount management in TCP mode.
9618 Don't unref a client ref we never had. Fixes an unref
9619 of an already-free client object after a client
9620 teardown request for me.
9622 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9624 * gst/rtsp-server/rtsp-session.c:
9625 docs: fix typo in API docs
9627 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9629 * gst/rtsp-server/rtsp-media.c:
9631 Keep the udp sources in playing even if we go to paused. unlock the sources when
9633 Add some more debug info.
9634 Only seek when we need to.
9635 Keep track of the position when we go to paused.
9637 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9639 * gst/rtsp-server/rtsp-client.c:
9640 * gst/rtsp-server/rtsp-media.c:
9641 * gst/rtsp-server/rtsp-media.h:
9642 Add beginnings of seeking.
9643 Parse the Range header and perform a seek on the pipeline for the requested
9644 position. It's disabled currently until I figure out what's going wrong.
9646 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9648 * gst/rtsp-server/rtsp-client.c:
9649 allow pause requests for now.
9652 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9654 * gst/rtsp-server/rtsp-client.c:
9655 Remove weak ref on the session in teardown
9656 We need to remove our weakref from the session when we do a teardown because
9657 else we close the TCP connection prematurely.
9659 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9661 * gst/rtsp-server/rtsp-client.c:
9662 * gst/rtsp-server/rtsp-client.h:
9663 * gst/rtsp-server/rtsp-session-pool.c:
9664 Do some more session cleanup
9665 Make session timeout kill the TCP connection that currently watches the
9667 Remove the client timeout property.
9669 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9671 * gst/rtsp-server/rtsp-client.c:
9672 * gst/rtsp-server/rtsp-client.h:
9673 * gst/rtsp-server/rtsp-media.c:
9674 * gst/rtsp-server/rtsp-media.h:
9675 * gst/rtsp-server/rtsp-server.c:
9676 * gst/rtsp-server/rtsp-session.c:
9677 * gst/rtsp-server/rtsp-session.h:
9679 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
9682 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9684 * examples/Makefile.am:
9685 * examples/test-launch.c:
9686 Add example server that takes launch lines
9687 Add an example server that streams any -launch line.
9689 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9691 * examples/test-readme.c:
9692 * gst/rtsp-server/rtsp-client.c:
9693 * gst/rtsp-server/rtsp-media.c:
9694 * gst/rtsp-server/rtsp-media.h:
9695 Add support for live streams
9696 Add support for live streams and ranges
9697 Start on handling TCP data transfer.
9699 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9701 * gst/rtsp-server/rtsp-media.c:
9702 Free the pipeline before other things
9705 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9707 * gst/rtsp-server/rtsp-client.c:
9708 Only free the pending tunnel if there is one
9711 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9713 * gst/rtsp-server/rtsp-client.c:
9714 * gst/rtsp-server/rtsp-client.h:
9715 * gst/rtsp-server/rtsp-media.c:
9716 rtsp-server: Add support for tunneling
9717 Add support for tunneling over HTTP.
9718 Use new connection methods to retrieve the url.
9719 Dispatch messages based on the message type instead of blindly
9720 assuming it's always a request.
9721 Keep track of the watch id so that we can remove it later.
9722 Set the media pipeline to NULL before unreffing the pipeline.
9724 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9726 * gst/rtsp-server/rtsp-client.c:
9727 * gst/rtsp-server/rtsp-client.h:
9728 Fix for channel -> watch rename in gstreamer
9729 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
9731 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9733 * gst/rtsp-server/rtsp-client.c:
9734 * gst/rtsp-server/rtsp-client.h:
9736 Use the async RTSP channels instead of spawning a new thread for each client.
9737 If a sessionid is specified in a request, fail if we don't have the session.
9739 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9741 * gst/rtsp-server/rtsp-media.c:
9742 Add better debug info
9743 Add some better debug info.
9745 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9747 * examples/test-video.c:
9749 Add support for session timeouts in the example.
9751 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9753 * gst/rtsp-server/rtsp-session-pool.c:
9754 * gst/rtsp-server/rtsp-session-pool.h:
9755 Pass GTimeVal around for performance reasons
9756 Get the current time only once and pass it around so that sessions don't have to
9757 get the current time anymore.
9758 Add experimental support for a GSource that dispatches when the session needs to
9761 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9763 * gst/rtsp-server/rtsp-session.c:
9764 * gst/rtsp-server/rtsp-session.h:
9765 Add better support for session timeouts
9766 Add a method to request the number of milliseconds when a session will timeout.
9768 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9770 * gst/rtsp-server/rtsp-media.c:
9771 * gst/rtsp-server/rtsp-media.h:
9772 Add suport for RTP manager monitoring
9773 Add the first stage in monitoring the rtp manager.
9774 Make sure we don't update the state to something we don't want.
9776 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9778 * gst/rtsp-server/rtsp-client.c:
9779 Add support for session keepalive
9780 Get and update the session timeout for all requests. get the session as early as
9783 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9785 * gst/rtsp-server/rtsp-media-factory.h:
9786 * gst/rtsp-server/rtsp-media.c:
9787 * gst/rtsp-server/rtsp-media.h:
9788 Handle media bus messages
9789 Handle media bus messages in a custom mainloop and dispatch them to the
9790 RTSPMedia objects. Let the default implementation handle some common messages.
9792 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9794 * gst/rtsp-server/rtsp-client.c:
9795 * gst/rtsp-server/rtsp-session-pool.c:
9796 * gst/rtsp-server/rtsp-session.c:
9797 Some more session timeout handling
9798 Move the session header setting code to a central place so that we always add
9799 the timeout parameter too.
9800 Handle timeouts by running the session cleanup code.
9801 Stop media before cleaning up.
9803 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9805 * gst/rtsp-server/rtsp-client.c:
9806 * gst/rtsp-server/rtsp-client.h:
9807 Add timeout property
9808 Add a timeout property ot the client and make the other properties into GObject
9811 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9813 * gst/rtsp-server/rtsp-session-pool.c:
9814 Use getters and setters in property code
9815 Use the getters and setters for the timeout property instead of locking
9818 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9820 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
9822 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9824 * gst/rtsp-server/rtsp-session-pool.c:
9825 * gst/rtsp-server/rtsp-session-pool.h:
9826 * gst/rtsp-server/rtsp-session.c:
9827 * gst/rtsp-server/rtsp-session.h:
9828 Add more timeout stuff
9829 Add method to check if a session is expired.
9830 Add method to perform cleanup on a session pool.
9832 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9834 * gst/rtsp-server/rtsp-client.c:
9835 * gst/rtsp-server/rtsp-session-pool.c:
9836 * gst/rtsp-server/rtsp-session-pool.h:
9837 * gst/rtsp-server/rtsp-session.c:
9838 * gst/rtsp-server/rtsp-session.h:
9839 Add beginnings of session timeouts and limits
9840 Add the timeout value to the Session header for unusual timeout values.
9841 Allow us to configure a limit to the amount of active sessions in a pool. Set a
9842 limit on the amount of retry we do after a sessionid collision.
9843 Add properties to the sessionid and the timeout of a session. Keep track of
9844 creation time and last access time for sessions.
9846 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9848 * gst/rtsp-server/rtsp-client.c:
9849 * gst/rtsp-server/rtsp-media.c:
9850 * gst/rtsp-server/rtsp-media.h:
9851 * gst/rtsp-server/rtsp-sdp.c:
9852 * gst/rtsp-server/rtsp-session-pool.c:
9853 * gst/rtsp-server/rtsp-session.c:
9854 * gst/rtsp-server/rtsp-session.h:
9855 Cleanup of sessions and more
9856 Fix the refcounting of media and sessions in the client. Properly clean up the
9857 session data when the client performs a teardown.
9858 Add Server header to responses.
9859 Allow for multiple uri setups in one session.
9860 Add Range header to the PLAY response and add the range attribute to the SDP
9862 Fix the session pool remove method, it used the wrong key in the hashtable. Also
9863 give the ownership of the sessionid to the session object.
9865 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9867 * gst/rtsp-server/rtsp-server.c:
9868 * gst/rtsp-server/rtsp-server.h:
9870 Rename the 'server_port' variable to simply 'port'.
9872 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9875 * gst/rtsp-server/rtsp-client.c:
9876 * gst/rtsp-server/rtsp-media.c:
9877 * gst/rtsp-server/rtsp-media.h:
9878 * gst/rtsp-server/rtsp-session.c:
9879 * gst/rtsp-server/rtsp-session.h:
9880 Rework the way we handle transports for streams
9881 Make the media accept an array of transports for the streams that we have
9882 configured for the play/pause requests.
9883 Implement server states for a client and its media.
9884 Require 0.10.22.1 (git HEAD) of gstreamer.
9886 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9888 * gst/rtsp-server/rtsp-client.c:
9889 * gst/rtsp-server/rtsp-media-factory.c:
9890 Drop const from functions dealing with urls
9891 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
9892 have the right const in them.
9894 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9896 * gst/rtsp-server/rtsp-client.c:
9897 * gst/rtsp-server/rtsp-media.c:
9898 * gst/rtsp-server/rtsp-sdp.c:
9902 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9904 * gst/rtsp-server/rtsp-client.c:
9905 * gst/rtsp-server/rtsp-media-factory.c:
9906 * gst/rtsp-server/rtsp-media.c:
9907 * gst/rtsp-server/rtsp-media.h:
9909 Don't keep a reference to the GstRTSPMedia in the stream.
9910 Free more things when freeing the GstRTSPMedia.
9912 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9915 * gst/rtsp-server/rtsp-media-factory.c:
9916 * gst/rtsp-server/rtsp-media-factory.h:
9917 * gst/rtsp-server/rtsp-media.c:
9918 * gst/rtsp-server/rtsp-media.h:
9919 * gst/rtsp-server/rtsp-server.c:
9920 * gst/rtsp-server/rtsp-server.h:
9921 More docs and small cleanups
9922 Add some more docs and update the README
9923 Cleanup some method names.
9924 Remove an unneeded idx field in the GstRTSPMediaStream
9926 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9929 * examples/Makefile.am:
9930 * examples/test-readme.c:
9931 Add a README and more example code
9932 Add a README file that contains a small introduction on how to use the server
9933 along with the example code explained in the readme.
9935 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9937 * gst/rtsp-server/rtsp-media.c:
9938 * gst/rtsp-server/rtsp-server.c:
9939 Fix some leaks and change default port
9940 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
9941 we finished the initial preroll. If we keep them locked, setting the pipeline to
9942 NULL will not stop and clean up the sources correctly.
9943 Change the default RTSP port to 8554 aka the official alternative RTSP port.
9945 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9947 * gst/rtsp-server/rtsp-session.c:
9948 * gst/rtsp-server/rtsp-session.h:
9949 Cleanups to the session object
9950 Remove some unneeded variables in the session state of a stream such as the
9951 owner media and the server transport.
9952 Get the configuration of a media stream in a session based on the media_stream
9953 in the original object instead of our cached index.
9954 Free more data in the finalize method.
9956 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9958 * gst/rtsp-server/rtsp-client.c:
9959 * gst/rtsp-server/rtsp-client.h:
9960 Cleanups and reuse media from DESCRIBE
9961 Handle thread create errors.
9962 Rename some internal methods to better match what they actually do.
9963 Handle misconfiguration of session_pool and media_mapping gracefully.
9964 Cache the DESCRIBE media and uri in the client connection and reuse them when
9965 we receive a SETUP request in the same connection for the same uri.
9966 Cleanup the client connection object.
9968 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
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 Add shared properties to media and factory
9975 Add the shared property to media.
9976 Implement some simple caching in the factory depending on if the media is shared
9979 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9981 * gst/rtsp-server/rtsp-client.c:
9982 Add a little comment
9983 Add some comment about the content-base header.
9985 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9987 * examples/Makefile.am:
9988 * examples/test-mp4.c:
9989 * examples/test-ogg.c:
9990 * examples/test-video.c:
9991 * gst/rtsp-server/Makefile.am:
9992 * gst/rtsp-server/rtsp-client.c:
9993 * gst/rtsp-server/rtsp-client.h:
9994 * gst/rtsp-server/rtsp-media-factory.c:
9995 * gst/rtsp-server/rtsp-media-factory.h:
9996 * gst/rtsp-server/rtsp-media.c:
9997 * gst/rtsp-server/rtsp-media.h:
9998 * gst/rtsp-server/rtsp-sdp.c:
9999 * gst/rtsp-server/rtsp-sdp.h:
10000 * gst/rtsp-server/rtsp-server.c:
10001 * gst/rtsp-server/rtsp-server.h:
10002 * gst/rtsp-server/rtsp-session.c:
10003 * gst/rtsp-server/rtsp-session.h:
10004 Reorganize things, prepare for media sharing
10005 Added various other test server examples
10006 Move the SDP message generation to a separate helper.
10007 Refactor common code for finding the session.
10008 Add content-base for realplayer compatibility
10009 Clean up request uris before processing for better vlc compatibility.
10010 Move prerolling and pipeline construction to the RTSPMedia object.
10011 Use multiudpsink for future pipeline reuse.
10013 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10016 Back to development
10019 === release 0.10.1 ===
10021 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10024 Make 0.10.1 release
10027 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10029 * bindings/vala/Makefile.am:
10031 Add more directories and files to the dist.
10033 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10035 * bindings/python/Makefile.am:
10036 * bindings/python/rtspserver.override:
10037 Fixed compile error of python bindings
10039 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10041 * bindings/vala/gst-rtsp-server-0.10.vapi:
10042 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
10043 Marked values as nullable accordingly
10045 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10047 * bindings/vala/gst-rtsp-server-0.10.vapi:
10048 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
10049 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
10050 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
10051 Updated Vala bindings
10053 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10055 * gst/rtsp-server/rtsp-client.c:
10056 * gst/rtsp-server/rtsp-media-mapping.c:
10057 * gst/rtsp-server/rtsp-media-mapping.h:
10058 * gst/rtsp-server/rtsp-media.h:
10059 * gst/rtsp-server/rtsp-session-pool.h:
10060 Cleanups and doc updates
10061 Add some more documentation and do some minor cleanups here and there.
10063 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10065 * gst/rtsp-server/rtsp-client.c:
10066 * gst/rtsp-server/rtsp-media-factory.c:
10067 * gst/rtsp-server/rtsp-media-factory.h:
10068 * gst/rtsp-server/rtsp-media.c:
10069 * gst/rtsp-server/rtsp-media.h:
10070 * gst/rtsp-server/rtsp-session.c:
10071 * gst/rtsp-server/rtsp-session.h:
10073 Rename GstRTSPMediaBin to GstRTSPMedia
10074 Parse the request url into a GstRTSPUri object and pass this object to the
10075 various handlers and methods that require the uri.
10077 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10081 Add some more docs and remove some old code from the example.
10083 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10085 * gst/rtsp-server/rtsp-client.c:
10086 Handle state change failures better
10087 Handle state change failures better when changing the state of the pipeline to
10090 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10092 * gst/rtsp-server/rtsp-media-factory.c:
10093 * gst/rtsp-server/rtsp-media-factory.h:
10094 Make element creation more extendible
10095 Add get_element vmethod to the default MediaFactory so that subclasses can just
10096 override that method and still use the default logic for making a MediaBin from
10099 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10102 * gst/rtsp-server/Makefile.am:
10103 * gst/rtsp-server/rtsp-client.c:
10104 * gst/rtsp-server/rtsp-client.h:
10105 * gst/rtsp-server/rtsp-media-factory.c:
10106 * gst/rtsp-server/rtsp-media-factory.h:
10107 * gst/rtsp-server/rtsp-media-mapping.c:
10108 * gst/rtsp-server/rtsp-media-mapping.h:
10109 * gst/rtsp-server/rtsp-media.c:
10110 * gst/rtsp-server/rtsp-media.h:
10111 * gst/rtsp-server/rtsp-server.c:
10112 * gst/rtsp-server/rtsp-server.h:
10113 * gst/rtsp-server/rtsp-session.c:
10114 * gst/rtsp-server/rtsp-session.h:
10115 Make the server handle arbitrary pipelines
10116 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
10117 The GstMediaBin object has a handle to a bin with elements and to a list of
10118 GstMediaStream objects that this bin produces.
10119 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
10120 with methods to register and remove those mappings.
10121 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
10122 used by the server instance.
10123 Modify the example application so that it shows how to create custom pipelines
10124 attached to a specific mount point.
10125 Various misc cleanps.
10127 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10129 * gst/rtsp-server/rtsp-server.c:
10130 * gst/rtsp-server/rtsp-server.h:
10131 Allow setting a custom media factory for a server
10133 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10135 * gst/rtsp-server/rtsp-client.c:
10136 * gst/rtsp-server/rtsp-client.h:
10137 Allow setting a custom media factory for a client.
10139 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10141 * gst/rtsp-server/Makefile.am:
10142 Add Makefile entry for the media factory
10144 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10146 * gst/rtsp-server/rtsp-media-factory.c:
10147 * gst/rtsp-server/rtsp-media-factory.h:
10148 Add media factory to map urls to media pipeline objects.
10150 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10152 * gst/rtsp-server/rtsp-media.c:
10153 * gst/rtsp-server/rtsp-media.h:
10154 Add comments. Remove unused field
10156 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10158 * gst/rtsp-server/rtsp-session-pool.c:
10159 * gst/rtsp-server/rtsp-session-pool.h:
10160 Allow custom session pools to override the session id allocation algorithms Add some comments.
10162 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10164 * gst/rtsp-server/rtsp-session.h:
10167 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10169 * gst/rtsp-server/rtsp-client.c:
10170 * gst/rtsp-server/rtsp-client.h:
10171 Move the connection code in one place Add some comments
10173 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10175 * gst/rtsp-server/rtsp-server.c:
10176 * gst/rtsp-server/rtsp-server.h:
10177 Make vmethod to create and accept new clients. Add some docs.
10179 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10181 * gst/rtsp-server/rtsp-server.c:
10182 * gst/rtsp-server/rtsp-server.h:
10183 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
10185 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10187 * gst/rtsp-server/rtsp-client.c:
10188 * gst/rtsp-server/rtsp-client.h:
10189 Name the parameters more appropriately.
10191 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10193 * gst/rtsp-server/rtsp-session-pool.c:
10194 Do some more cleanup of the session pool.
10196 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10198 * gst/rtsp-server/Makefile.am:
10199 * gst/rtsp-server/rtsp-client.c:
10200 Check if return value of gst_rtsp_session_get_media is not NULL
10202 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10204 * gst/rtsp-server/Makefile.am:
10205 Install rtsp-session and rtsp-session-pool headers
10207 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10212 * bindings/python/Makefile.am:
10213 * bindings/python/arg-types.py:
10214 * bindings/python/codegen/Makefile.am:
10215 * bindings/python/codegen/__init__.py:
10216 * bindings/python/codegen/argtypes.py:
10217 * bindings/python/codegen/code-coverage.py:
10218 * bindings/python/codegen/codegen.py:
10219 * bindings/python/codegen/definitions.py:
10220 * bindings/python/codegen/defsparser.py:
10221 * bindings/python/codegen/docextract.py:
10222 * bindings/python/codegen/docgen.py:
10223 * bindings/python/codegen/fileprefix.override:
10224 * bindings/python/codegen/fileprefixmodule.c:
10225 * bindings/python/codegen/h2def.py:
10226 * bindings/python/codegen/mergedefs.py:
10227 * bindings/python/codegen/mkskel.py:
10228 * bindings/python/codegen/override.py:
10229 * bindings/python/codegen/reversewrapper.py:
10230 * bindings/python/codegen/scmexpr.py:
10231 * bindings/python/rtspserver-types.defs:
10232 * bindings/python/rtspserver.defs:
10233 * bindings/python/rtspserver.override:
10234 * bindings/python/rtspservermodule.c:
10236 Add python bindings.
10238 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10240 * bindings/Makefile.am:
10242 Don't go into python dir when requirements for python bindings are missing
10244 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10246 * bindings/Makefile.am:
10247 * bindings/vala/Makefile.am:
10249 Install Vala bindings if vala is available
10251 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10253 * bindings/vala/gst-rtsp-server-0.10.deps:
10254 * bindings/vala/gst-rtsp-server-0.10.vapi:
10255 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
10256 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
10257 * bindings/vala/packages/gst-rtsp-server-0.10.files:
10258 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
10259 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
10260 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
10261 Regenerated Vala bindings
10263 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10265 * bindings/vala/gst-rtsp-server.vapi:
10266 * bindings/vala/packages/gst-rtsp-server.metadata:
10267 Fixed typo in included headers for vala bindings
10269 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10273 * pkgconfig/Makefile.am:
10274 * pkgconfig/gst-rtsp-server.pc.in:
10275 Added pkgconfig file
10277 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
10279 * bindings/vala/gst-rtsp-server.vapi:
10280 * bindings/vala/packages/gst-rtsp-server.excludes:
10281 * bindings/vala/packages/gst-rtsp-server.gi:
10282 * bindings/vala/packages/gst-rtsp-server.metadata:
10283 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
10285 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
10287 * bindings/vala/gst-rtsp-server.vapi:
10288 * bindings/vala/packages/gst-rtsp-server.deps:
10289 * bindings/vala/packages/gst-rtsp-server.files:
10290 * bindings/vala/packages/gst-rtsp-server.gi:
10291 * bindings/vala/packages/gst-rtsp-server.metadata:
10292 * bindings/vala/packages/gst-rtsp-server.namespace:
10293 Added Vala bindings
10295 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
10297 * gst/rtsp-server/rtsp-session.c:
10298 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
10300 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
10302 * examples/Makefile.am:
10303 * gst/rtsp-server/Makefile.am:
10304 Put GStreamer version in library name
10306 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10308 * examples/Makefile.am:
10309 * gst/rtsp-server/Makefile.am:
10310 Fix some issues to pass distcheck
10312 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10314 * gst/rtsp-server/rtsp-server.c:
10315 Added port property to GstRTSPServer class.
10317 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10322 * examples/Makefile.am:
10325 * gst/rtsp-server/Makefile.am:
10326 * gst/rtsp-server/rtsp-client.c:
10327 * gst/rtsp-server/rtsp-client.h:
10328 * gst/rtsp-server/rtsp-media.c:
10329 * gst/rtsp-server/rtsp-media.h:
10330 * gst/rtsp-server/rtsp-server.c:
10331 * gst/rtsp-server/rtsp-server.h:
10332 * gst/rtsp-server/rtsp-session-pool.c:
10333 * gst/rtsp-server/rtsp-session-pool.h:
10334 * gst/rtsp-server/rtsp-session.c:
10335 * gst/rtsp-server/rtsp-session.h:
10337 Split in library and example program
10339 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
10341 * src/rtsp-client.h:
10342 Removed obsolete variable
10344 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
10346 * src/rtsp-client.c:
10347 * src/rtsp-client.h:
10348 Removed pipeline variable GstRTSPClient, because it's only used in one function
10350 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10352 * src/rtsp-media.c:
10353 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
10355 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
10357 * src/rtsp-session.c:
10358 Initialize some more vars.
10360 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
10362 * src/rtsp-session.c:
10363 Initialize variable to avoid compiler warning.
10365 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
10368 Add a reasonable generic .gitignore