1 === release 1.11.91 ===
3 2017-04-27 Sebastian Dröge <slomo@coaxion.net>
8 2017-04-24 20:30:37 +0100 Tim-Philipp Müller <tim@centricular.com>
11 Automatic update of common submodule
12 From 60aeef6 to 48a5d85
14 2017-04-13 14:20:10 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
16 * gst/rtsp-server/rtsp-media-factory.c:
17 * gst/rtsp-server/rtsp-media.c:
18 * gst/rtsp-server/rtsp-session.c:
19 * gst/rtsp-server/rtsp-stream.c:
20 gi: Fix some annotations and docstrings
22 2017-04-13 13:52:26 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
24 * gst/rtsp-server/meson.build:
29 2017-04-10 23:51:12 +0100 Tim-Philipp Müller <tim@centricular.com>
33 Automatic update of common submodule
34 From 39ac2f5 to 60aeef6
36 === release 1.11.90 ===
38 2017-04-07 16:35:03 +0300 Sebastian Dröge <sebastian@centricular.com>
44 * gst-rtsp-server.doap:
48 2017-03-27 18:19:33 +0100 Tim-Philipp Müller <tim@centricular.com>
50 * examples/test-launch.c:
51 examples: make test-launch pipeline shared by default as well
53 2017-02-27 19:10:44 +0200 Sebastian Dröge <sebastian@centricular.com>
55 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
56 gstreamer-rtsp-server: Add both srcdir and builddir to the include path
57 Just the build dir is not going to work for srcdir!=builddir.
59 2017-02-24 15:59:54 +0200 Sebastian Dröge <sebastian@centricular.com>
64 2017-02-24 15:37:49 +0200 Sebastian Dröge <sebastian@centricular.com>
69 === release 1.11.2 ===
71 2017-02-24 15:10:07 +0200 Sebastian Dröge <sebastian@centricular.com>
77 * gst-rtsp-server.doap:
80 2017-02-14 20:40:26 +0000 Tim-Philipp Müller <tim@centricular.com>
83 meson: dist meson build files
84 Ship meson build files in tarballs, so people who use tarballs
85 in their builds can start playing with meson already.
87 2017-02-07 23:39:37 +1100 Jan Schmidt <jan@centricular.com>
89 * examples/test-record.c:
90 examples/test-record: Add extra line to initial printout
91 Add an example line of how to deliver a stream to the
94 2017-01-19 14:57:19 +0200 Sebastian Dröge <sebastian@centricular.com>
96 * gst/rtsp-server/rtsp-client.c:
97 rtsp-client: Also handle the (S|G)ET_PARAMETER case of size==0 || !data as keep-alive
98 If there is no Content-Length header, no body would be allocated and the
99 '\0' would also not be appended to the body.
101 2017-01-19 14:24:07 +0200 Sebastian Dröge <sebastian@centricular.com>
103 * gst/rtsp-server/rtsp-client.c:
104 rtsp-client: Fix handling of keep-alive GET_PARAMETER/SET_PARAMETER
105 While they logically have 0 bytes length, GstRTSPConnection is appending
106 a '\0' to everything making the size be 1 instead.
108 2017-01-13 12:39:36 +0000 Tim-Philipp Müller <tim@centricular.com>
113 2017-01-12 19:04:23 +0200 Sebastian Dröge <sebastian@centricular.com>
115 * gst/rtsp-server/rtsp-session.c:
116 rtsp-session: Only remove deprecated API if requested to do so, not just when disabling
117 gst_rtsp_session_is_expired() and gst_rtsp_session_next_timeout() were
120 2017-01-12 16:32:59 +0200 Sebastian Dröge <sebastian@centricular.com>
125 === release 1.11.1 ===
127 2017-01-12 16:14:46 +0200 Sebastian Dröge <sebastian@centricular.com>
133 * gst-rtsp-server.doap:
134 * win32/common/libgstrtspserver.def:
137 2017-01-10 08:34:50 +0100 Patricia Muscalu <patricia@axis.com>
139 * gst/rtsp-server/rtsp-stream.c:
140 rtsp-stream: corrected if-statement in _get_server_port()
141 This bug was accidentally introduced while fixing a segfault
142 in _get_server_port() function.
143 https://bugzilla.gnome.org/show_bug.cgi?id=776345
145 2017-01-09 14:12:05 +0100 Patricia Muscalu <patricia@axis.com>
147 * gst/rtsp-server/rtsp-stream.c:
148 * tests/check/gst/stream.c:
149 rtsp-stream: fixed segmenation fault in _get_server_port()
150 Calling function gst_rtsp_stream_get_server_port() results in
151 segmenation fault in the RTP/RTSP/TCP case.
152 Port that the server will use to receive RTCP makes only
153 sense in the UDP case, however the function should handle
154 the TCP case in a nicer way.
155 https://bugzilla.gnome.org/show_bug.cgi?id=776345
157 2017-01-09 12:22:40 +0300 Aleksandr Slobodeniuk <alenuke@yandex.ru>
159 * gst/rtsp-server/rtsp-media-factory.c:
160 dosc: Fix a little typo
161 https://bugzilla.gnome.org/show_bug.cgi?id=777037
163 2017-01-04 16:20:54 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
165 * pkgconfig/Makefile.am:
166 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
167 * pkgconfig/meson.build:
168 meson: generate pkg-config -uninstalled pc files
169 Generating those files is useful for users building the GStreamer stack
170 using meson and having to link it to another project which is still
172 https://bugzilla.gnome.org/show_bug.cgi?id=776810
174 2017-01-04 16:11:08 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
176 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
177 pkgconfig: fix -uninstalled pc file
178 pcfiledir was never defined so the paths were wrong.
179 https://bugzilla.gnome.org/show_bug.cgi?id=776867
181 2016-12-21 13:41:50 +0100 Patricia Muscalu <patricia@axis.com>
183 * gst/rtsp-server/rtsp-stream.c:
184 * tests/check/gst/rtspserver.c:
185 rtsp-stream: Fixed TCP transport case
186 Make sure that the appsink element is actually added to
187 the bin before trying to link it with the elements in it.
188 https://bugzilla.gnome.org/show_bug.cgi?id=776343
190 2016-12-16 17:26:04 +0000 Tim-Philipp Müller <tim@centricular.com>
196 Remove generated .spec file
197 Likely extremely bitrotten, and we should not ship this anyway.
199 2016-12-03 08:21:02 +0100 Edward Hervey <bilboed@bilboed.com>
202 Automatic update of common submodule
203 From f980fd9 to 39ac2f5
205 2016-12-02 15:40:09 +0100 Edward Hervey <edward@centricular.com>
207 * gst/rtsp-server/rtsp-media.c:
208 media: Fix pt map caps
209 Since decryption is handled within rtpbin, all outcoming stream
210 caps will be application/x-rtp (i.e. regular rtp)
211 Fixes RECORD with SRTP streams
213 2016-12-02 15:38:04 +0100 Edward Hervey <edward@centricular.com>
215 * gst/rtsp-server/rtsp-media-factory.c:
216 media-factory: Create media objects with the proper transport mode
217 The function called immediately afterwards (collect_streams()) will
218 need it to work properly
220 2016-12-02 14:36:50 +0200 Sebastian Dröge <sebastian@centricular.com>
222 * gst/rtsp-server/rtsp-auth.c:
223 rtsp-auth: Don't remove digest-auth nonces that already/still have a client connected
225 2016-12-01 18:04:34 +0200 Sebastian Dröge <sebastian@centricular.com>
227 * gst/rtsp-server/rtsp-media-factory.c:
228 rtsp-media-factory: Don't create a pipeline for the media pipeline string
229 We're going to put a pipeline into a pipeline otherwise, which is not
232 2016-10-25 15:41:28 +0300 Kseniia Vasilchuk <vasilchukkseniia@gmail.com>
234 * gst/rtsp-server/rtsp-media.c:
235 media: Fix race condition around finish_unprepare() if called multiple time
236 https://bugzilla.gnome.org/show_bug.cgi?id=755329
238 2016-11-30 14:06:36 +1100 Jan Schmidt <jan@centricular.com>
240 * gst/rtsp-sink/gstrtspclientsink.c:
241 rtspclientsink: Don't leave stale pointer after unref
242 Fix a warning on shutdown - don't keep a pointer to an
243 alread-unreffed object.
245 2016-11-26 11:24:50 +0000 Tim-Philipp Müller <tim@centricular.com>
248 common: use https protocol for common submodule
249 https://bugzilla.gnome.org/show_bug.cgi?id=775110
251 2016-11-21 23:29:56 +1100 Matthew Waters <matthew@centricular.com>
253 * gst/rtsp-server/rtsp-stream.c:
254 stream: block the output of rtpbin instead of the source pipeline
255 85c52e194bcb81928b96614be0ae47d59eccb1ce introduced a more correct
256 detection of the srtp rollover counter to add to the SDP.
257 Unfortunately, it was incomplete for live pipelines where the logic
258 blocks the source bin before creating the SDP and thus would never have
259 the necessary informaiton to create a correct SDP with srtp encryption.
260 Move the pad blocks to rtpbin's output pads instead so that the
261 necessary information can be created before we need the information for
263 https://bugzilla.gnome.org/show_bug.cgi?id=770239
265 2016-11-21 16:02:39 +0100 Dag Gullberg <dagg@axis.com>
267 * gst/rtsp-server/rtsp-client.c:
268 rtsp-client: add IDLE timeout, before session exists
269 The RTSP server will not timeout an idle RTSP connection
270 (note this is different from doing timeout on a RTSP
272 At least for Apache this is a problem when running RTSP over
273 HTTPS since it uses one of the threads (there is a rather
274 limited number) that are available for handling requests.
275 https://bugzilla.gnome.org/show_bug.cgi?id=771830
277 2016-11-23 09:45:08 +0000 Tim-Philipp Müller <tim@centricular.com>
282 2016-11-21 13:05:50 +0100 Göran Jönsson <goranjn@axis.com>
284 * gst/rtsp-server/rtsp-stream.c:
285 rtsp-stream: Set close-socket FALSE on UDP src:es
286 With this RTSP server can use the sockets independent on the udpsrc
288 When the udp src is finalized it will unref socket and when g_socket
289 is finalized the socket will be closed.
290 https://bugzilla.gnome.org/show_bug.cgi?id=765673
292 2016-11-18 17:47:13 +0200 Sebastian Dröge <sebastian@centricular.com>
294 * gst/rtsp-sink/gstrtspclientsink.c:
295 rtspclientsink: Move to new helper function to parse authentication responses
296 https://bugzilla.gnome.org/show_bug.cgi?id=774416
298 2016-11-16 08:42:24 +0200 Sebastian Dröge <sebastian@centricular.com>
300 * examples/Makefile.am:
301 * examples/test-auth-digest.c:
302 * gst/rtsp-server/rtsp-auth.c:
303 * gst/rtsp-server/rtsp-auth.h:
304 * win32/common/libgstrtspserver.def:
305 rtsp-auth: Add support for Digest authentication
306 https://bugzilla.gnome.org/show_bug.cgi?id=774416
308 2016-11-17 09:41:53 -0800 Scott D Phillips <scott.d.phillips@intel.com>
311 * gst/rtsp-server/meson.build:
313 * tests/check/meson.build:
315 * win32/common/libgstrtspserver.def:
316 Enable building with MSVC
317 https://bugzilla.gnome.org/show_bug.cgi?id=774640
319 2016-11-18 20:23:14 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
322 meson: gstreamer gst_check_dep does not exist on windows
324 2016-11-17 09:43:37 -0800 Scott D Phillips <scott.d.phillips@intel.com>
326 * gst/rtsp-server/rtsp-client.c:
327 client: update do_send_message to match type GstRTSPClientSendFunc
328 This type mismatch fails building with MSVC
329 https://bugzilla.gnome.org/show_bug.cgi?id=774640
331 2016-11-11 14:42:08 +0200 Sebastian Dröge <sebastian@centricular.com>
333 * gst/rtsp-server/rtsp-sdp.c:
334 rtsp-sdp: Fix indentation
336 2016-11-10 05:16:00 +0000 Neha Arora <arora.neha@samsung.com>
338 * gst/rtsp-server/rtsp-media.c:
339 rtsp-media: Only signal "new-state" if the state has actually changed
340 https://bugzilla.gnome.org/show_bug.cgi?id=774173
342 2016-08-24 11:39:13 +0200 Branko Subasic <branko@axis.com>
344 * gst/rtsp-server/rtsp-client.c:
345 * gst/rtsp-server/rtsp-client.h:
346 client: emit signal in the beginning of each rtsp request
347 These signals let the application validate the requests, configure the
348 media/stream in a certain way and also generate error status code in
349 case of error or bad request.
350 https://bugzilla.gnome.org/show_bug.cgi?id=758062
352 2016-11-01 18:10:35 +0000 Tim-Philipp Müller <tim@centricular.com>
355 meson: update version
357 === release 1.11.0 ===
359 2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com>
364 === release 1.10.0 ===
366 2016-11-01 18:06:46 +0200 Sebastian Dröge <sebastian@centricular.com>
372 * gst-rtsp-server.doap:
375 2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com>
377 * tests/check/gst/rtspserver.c:
378 * tests/check/gst/stream.c:
379 tests: try to avoid using the same ports in different tests
380 Causes problems with client multicast tests otherwise if
381 tests are run in parallel.
382 https://bugzilla.gnome.org/show_bug.cgi?id=773640
384 2016-10-28 17:50:59 +0100 Tim-Philipp Müller <tim@centricular.com>
386 * tests/check/gst/client.c:
387 tests: client: use fail_unless_equals_foo() for better failure reporting
389 2016-09-26 11:16:04 +0200 Göran Jönsson <goranjn@axis.com>
391 * gst/rtsp-server/rtsp-client.c:
392 rtsp-client: Session filter in unwatch session
393 Call session filter with filter_session_media as paramer in
394 client_unwatch_session if using drop_backlog = FALSE.
395 In client_unwatch_session its allowed to grow the watchs backlog.
396 If using drop_backlog = FALSE and the backlog is full it will cause
397 a deadlock when setting session media state to NULL
398 if the backlog is not allowed to grow.
399 https://bugzilla.gnome.org/show_bug.cgi?id=771983
401 2016-10-20 21:40:18 +0100 Tim-Philipp Müller <tim@centricular.com>
404 meson: add fallbacks for gst modules
407 2016-09-14 17:48:39 +0300 Nikita Bobkov <NikitaDBobkov@gmail.com>
409 * gst/rtsp-server/rtsp-client.c:
410 rtsp-client: Fix factory leaking in find_media() in error cases
411 https://bugzilla.gnome.org/show_bug.cgi?id=771488
413 2016-10-06 11:47:50 -0400 Xavier Claessens <xavier.claessens@collabora.com>
415 * gst/rtsp-server/rtsp-stream.c:
416 stream: Fix randomly missing streams from SDP with dynamic elements
417 When using dynamic elements, gst_rtsp_stream_join_bin() is called from
418 "pad-added" signal. In that case priv->srcpad could already have its caps,
419 and they'll be sent to priv->send_src[0] pad. That means that when it
420 connects "notify::caps" signal, that pad could already have received its
421 caps and the signal won't be emitted anymore.
422 In that case priv->caps stay to NULL and when building the SDP that stream
423 gets ignored. Leading to missing video or audio when playing in client side.
424 https://bugzilla.gnome.org/show_bug.cgi?id=772478
426 2016-09-30 11:42:08 +0100 Tim-Philipp Müller <tim@centricular.com>
429 meson: update version
431 === release 1.9.90 ===
433 2016-09-30 13:04:12 +0300 Sebastian Dröge <sebastian@centricular.com>
439 * gst-rtsp-server.doap:
442 2016-09-17 13:17:19 +0100 Ian Jamison <ian.dev@arkver.com>
444 * gst/rtsp-server/rtsp-media-factory.c:
445 * gst/rtsp-server/rtsp-media.c:
446 * gst/rtsp-server/rtsp-stream.c:
447 rtsp-server: Hint that set_multicast_iface expects the name of the interface
448 To prevent any possibly confusion with IPs or anything else.
449 https://bugzilla.gnome.org/show_bug.cgi?id=771530
451 2016-09-18 09:58:55 -0400 Sebastian Dröge <sebastian@centricular.com>
453 * gst/rtsp-server/rtsp-media-factory.c:
454 * gst/rtsp-server/rtsp-media.c:
455 rtsp-media: Call g_free() instead of g_object_unref() on multicast-iface strings
456 https://bugzilla.gnome.org/show_bug.cgi?id=763000#c5
458 2016-09-14 11:31:15 +0200 Sebastian Dröge <sebastian@centricular.com>
461 configure: Depend on gstreamer 1.9.2.1
463 2016-09-10 20:52:31 +1000 Jan Schmidt <jan@centricular.com>
467 Automatic update of common submodule
468 From b18d820 to f980fd9
470 2016-09-10 09:58:31 +1000 Jan Schmidt <jan@centricular.com>
474 Automatic update of common submodule
475 From 6f2d209 to b18d820
477 2016-09-07 18:44:34 +0300 Sebastian Dröge <sebastian@centricular.com>
479 * gst/rtsp-server/rtsp-stream.c:
480 rtsp-stream: Remove unused _locked() variant of a function
481 It was added during refactoring.
483 2016-09-07 10:21:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
485 * gst/rtsp-server/rtsp-stream.c:
486 stream: cosmetic cleanup
487 https://bugzilla.gnome.org/show_bug.cgi?id=766612
489 2016-09-07 10:16:19 -0400 Xavier Claessens <xavier.claessens@collabora.com>
491 * gst/rtsp-server/rtsp-stream.c:
492 stream: Compare IP addresses case insensitive in more places
493 https://bugzilla.gnome.org/show_bug.cgi?id=766612
495 2016-09-07 10:12:18 -0400 Xavier Claessens <xavier.claessens@collabora.com>
498 * gst/rtsp-server/rtsp-stream.c:
499 stream: Fix leaked joined_bin
500 There is no need to keep a strong ref on it, and _leave_bin() was
501 setting it to NULL before calling g_clear_object() so it was leaked.
502 https://bugzilla.gnome.org/show_bug.cgi?id=766612
504 2016-09-06 19:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
506 * gst/rtsp-server/rtsp-stream.c:
507 rtsp-stream: Compare IP address strings case insensitive
508 Otherwise IPv6 addresses might fail this comparision.
510 2016-09-06 19:10:21 +0300 Sebastian Dröge <sebastian@centricular.com>
512 * gst/rtsp-server/rtsp-stream.c:
513 rtsp-stream: Bind multicast sockets to ANY as before
514 https://bugzilla.gnome.org/show_bug.cgi?id=766612#c48
516 2016-09-05 18:31:36 +0300 Kseniia <vasilchukkseniia@gmail.com>
518 * gst/rtsp-server/rtsp-session.c:
519 rtsp-session: Fix segfault when doing keep-alive after removing the session
520 If keep-alive happens after removing the session but before finalizing the
521 stream transport, we would segfault.
522 https://bugzilla.gnome.org/show_bug.cgi?id=750544
524 2016-09-05 18:04:50 +0300 Sebastian Dröge <sebastian@centricular.com>
526 * gst/rtsp-server/rtsp-stream.c:
527 rtsp-stream: Always create multicast UDP elements if the protocol flag is set
528 Adding them later will cause deadlocks due to
529 1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
530 2) adding the multicast sink
531 3) waiting for it to get data to preroll again
532 3) never happens because the queues after the tee are full.
534 2016-09-05 16:32:57 +0300 Sebastian Dröge <sebastian@centricular.com>
536 * gst/rtsp-server/rtsp-stream.c:
537 rtsp-stream: Fix up various multicast related issues
539 2016-09-05 13:40:59 +0300 Sebastian Dröge <sebastian@centricular.com>
541 * tests/check/gst/stream.c:
542 tests: Fix compilation
544 2016-07-28 15:33:05 -0400 Xavier Claessens <xavier.claessens@collabora.com>
546 * gst/rtsp-server/rtsp-client.c:
547 * gst/rtsp-server/rtsp-stream.c:
548 * tests/check/gst/stream.c:
549 stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast
550 This is basically reverting changes introduced in commit f62a9a7,
551 because it was introducing various regressions:
552 - It introduces a leak of udpsrc elements that got wrongly fixed by adding
553 an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
554 ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
555 - If a mcast client connects, it creates a new socket in SETUP to try to respect
556 the destination/port given by the client in the transport, and overrides the
557 socket already set on the udpsink element. That means that if we already had a
558 client connected, the source address on the udp packets it receives suddenly
560 - If a 2nd mcast client connects, the destination/port in its transport is
561 ignored but its transport wasn't updated.
562 What this patch does:
563 - Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
564 - Always have a tee+queue when udp is enabled. This could be optimized
565 again in a later patch, but is more complicated. If no unicast clients
566 connects then those elements are useless, this could be also optimized
568 - When mcast transport is added, it creates a new set of udpsrc/udpsink,
569 seperated from those for unicast clients. Since we already support only
570 one mcast address, we also create only one set of elements.
571 https://bugzilla.gnome.org/show_bug.cgi?id=766612
573 2016-07-28 15:20:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
575 * gst/rtsp-server/rtsp-stream.c:
576 stream: factor our plug_src function
577 https://bugzilla.gnome.org/show_bug.cgi?id=766612
579 2016-07-21 21:46:16 -0400 Xavier Claessens <xavier.claessens@collabora.com>
581 * gst/rtsp-server/rtsp-stream.c:
582 stream: factor out plug_sink function
583 https://bugzilla.gnome.org/show_bug.cgi?id=766612
585 2016-07-20 23:05:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
587 * gst/rtsp-server/rtsp-stream.c:
588 stream: small documentation clarification
589 https://bugzilla.gnome.org/show_bug.cgi?id=766612
591 2016-07-20 15:35:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
593 * gst/rtsp-server/rtsp-stream.c:
594 stream: rename addr_v4/6 to mcast_addr_v4/6 for clarity
595 https://bugzilla.gnome.org/show_bug.cgi?id=766612
597 2016-07-14 11:10:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
599 * gst/rtsp-server/rtsp-stream.c:
600 stream: Keep a ref on joined bin
601 https://bugzilla.gnome.org/show_bug.cgi?id=766612
603 2016-07-20 15:11:32 -0400 Xavier Claessens <xavier.claessens@collabora.com>
605 * gst/rtsp-server/rtsp-stream.c:
607 https://bugzilla.gnome.org/show_bug.cgi?id=766612
609 2016-07-20 23:18:23 -0400 Xavier Claessens <xavier.claessens@collabora.com>
611 * gst/rtsp-server/rtsp-stream.c:
612 stream: small fix in error code path
613 https://bugzilla.gnome.org/show_bug.cgi?id=766612
615 2016-07-20 20:09:57 -0400 Xavier Claessens <xavier.claessens@collabora.com>
617 * gst/rtsp-server/rtsp-stream.c:
618 Revert "rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc"
619 This partly reverts commit cba045e1b19fad6e689e10206f57903e15f1229a,
620 but keeps unit tests.
621 https://bugzilla.gnome.org/show_bug.cgi?id=766612
623 2016-09-01 12:33:00 +0300 Sebastian Dröge <sebastian@centricular.com>
628 === release 1.9.2 ===
630 2016-09-01 12:32:51 +0300 Sebastian Dröge <sebastian@centricular.com>
636 * gst-rtsp-server.doap:
639 2016-01-27 01:03:52 +0000 Tim-Philipp Müller <tim@centricular.com>
642 * examples/meson.build:
644 * gst/rtsp-server/meson.build:
645 * gst/rtsp-sink/meson.build:
647 * pkgconfig/meson.build:
648 * tests/check/meson.build:
650 Add support for Meson as alternative/parallel build system
651 https://github.com/mesonbuild/meson
653 2016-08-26 21:56:13 +0200 Josep Torra <n770galaxy@gmail.com>
656 * tests/check/Makefile.am:
657 build: silence error about pthread for 'make check' in osx
658 Fixes "clang: error: argument unused during compilation: '-pthread'"
660 2015-09-25 15:04:00 +0000 Nikita Bobkov <NikitaDBobkov@gmail.com>
662 * gst/rtsp-server/rtsp-client.c:
663 rtsp-client: Fix leaking of media in error cases
664 With additional fixes by Kseniya Vasilchuk <vasilchukkseniia@gmail.com>
665 and myself to make the media refcounting a bit easier to follow.
666 https://bugzilla.gnome.org/show_bug.cgi?id=755632
668 2016-08-02 15:08:22 +0300 Sebastian Dröge <sebastian@centricular.com>
670 * gst/rtsp-server/rtsp-client.c:
671 rtsp-client: Fix leaking of session in error cases
672 https://bugzilla.gnome.org/show_bug.cgi?id=755632
674 2016-07-11 21:16:04 +0200 Stefan Sauer <ensonic@users.sf.net>
677 Automatic update of common submodule
678 From f363b32 to f49c55e
680 2016-07-06 13:51:15 +0300 Sebastian Dröge <sebastian@centricular.com>
685 === release 1.9.1 ===
687 2016-07-06 13:28:12 +0300 Sebastian Dröge <sebastian@centricular.com>
693 * gst-rtsp-server.doap:
696 2016-06-24 02:02:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
699 configure: Need to add -DGST_STATIC_COMPILATION when building only statically
700 https://bugzilla.gnome.org/show_bug.cgi?id=767463
702 2016-06-21 11:49:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
705 Automatic update of common submodule
706 From ac2f647 to f363b32
708 2016-04-14 22:56:11 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
710 * gst/rtsp-server/rtsp-sdp.c:
711 * gst/rtsp-server/rtsp-sdp.h:
712 * gst/rtsp-server/rtsp-stream.c:
713 * gst/rtsp-server/rtsp-stream.h:
714 sdp: add rollover counters for all sender SSRC
715 We add different crypto sessions in MIKEY, one for each sender
716 SSRC. Currently, all of them will have the same security policy, 0.
717 The rollover counters are obtained from the srtpenc element using the
719 https://bugzilla.gnome.org/show_bug.cgi?id=730539
721 2016-06-07 20:44:42 +0100 Tim-Philipp Müller <tim@centricular.com>
723 * gst/rtsp-server/rtsp-media-factory.h:
724 * gst/rtsp-server/rtsp-server.h:
727 2016-05-25 10:28:43 +0100 Tim-Philipp Müller <tim@centricular.com>
729 * gst/rtsp-server/Makefile.am:
730 g-i: pass compiler env to g-ir-scanner
731 It's what introspection.mak does as well. Should
732 fix spurious build failures on gnome-continuous
733 (caused by g-ir-scanner getting compiler details
734 via python which is broken in some environments
735 so passing the compiler details bypasses that).
737 2016-05-18 16:48:44 +0100 Ian <ian.arkver.dev@gmail.com>
739 * gst/rtsp-server/rtsp-session.c:
740 rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header
741 This works with rtspsrc and live555, but fails with e.g. ffmpeg.
742 https://bugzilla.gnome.org/show_bug.cgi?id=766619
744 2016-03-07 14:48:38 +0100 Edward Hervey <bilboed@bilboed.com>
746 * gst/rtsp-sink/gstrtspclientsink.c:
747 rtspclientsink: Check return value of sscanf
748 And just make sure we always have 0/0 if we have an error
751 2016-04-25 08:55:25 -0400 Jake Foytik <jake.foytik@ipconfigure.com>
753 * gst/rtsp-server/rtsp-stream.c:
754 * tests/check/gst/rtspserver.c:
755 * tests/check/gst/stream.c:
756 rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
757 - Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
758 - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
759 - Create unit test for shared media.
760 https://bugzilla.gnome.org/show_bug.cgi?id=764744
762 2016-04-11 10:55:23 +0300 Sebastian Dröge <sebastian@centricular.com>
764 * gst/rtsp-server/rtsp-stream.c:
765 rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
766 For IPv6 addresses, binding to a multicast group does not work on Linux
767 either. Always bind to ANY and then later join the multicast group.
768 https://bugzilla.gnome.org/show_bug.cgi?id=764679
770 2016-04-14 10:05:02 +0100 Julien Isorce <j.isorce@samsung.com>
773 Automatic update of common submodule
774 From 6f2d209 to ac2f647
776 2016-04-06 10:09:46 +0200 Patricia Muscalu <patricia@axis.com>
778 * gst/rtsp-server/rtsp-thread-pool.c:
779 rtsp-thread-pool: explained why GSource is a part of ThreadImpl
780 Clarified why it is necessary to add source information to
781 GstRTSPThreadImpl. See the reported bug in GLib:
782 https://bugzilla.gnome.org/show_bug.cgi?id=720186
783 for more information.
784 https://bugzilla.gnome.org/show_bug.cgi?id=761702
786 2016-04-04 12:58:38 +0300 Sebastian Dröge <sebastian@centricular.com>
788 * examples/Makefile.am:
789 examples: Clean up CFLAGS/LDADD even more
790 The internal .la should come first and is part of LDADD, as is
793 2016-04-04 12:39:39 +0300 Sebastian Dröge <sebastian@centricular.com>
795 * examples/Makefile.am:
796 examples: Clean up CFLAGS/LDADD to link with the correct versions of all libraries
798 2016-04-03 12:06:29 +0300 Sebastian Dröge <sebastian@centricular.com>
800 * gst/rtsp-server/Makefile.am:
801 rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS)
803 2015-12-30 18:39:05 +0200 Sebastian Dröge <sebastian@centricular.com>
805 * gst/rtsp-server/rtsp-client.c:
806 * gst/rtsp-server/rtsp-media-factory.c:
807 * gst/rtsp-server/rtsp-media-factory.h:
808 * gst/rtsp-server/rtsp-media.c:
809 * gst/rtsp-server/rtsp-media.h:
810 * gst/rtsp-server/rtsp-sdp.c:
811 * gst/rtsp-server/rtsp-stream.c:
812 * gst/rtsp-server/rtsp-stream.h:
813 rtsp-server: Implement clock signalling according to RFC7273
814 For NTP and PTP clocks we signal the actual clock that is used and signal
815 the direct media clock offset.
816 For all other clocks we at least signal that it's the local sender clock.
817 This allows receivers to know which clock was used to generate the media and
818 its RTP timestamps. Receivers can then implement network synchronization,
819 either absolute or at least relative by getting the sender clock rate directly
820 via NTP/PTP instead of estimating it from RTP timestamps and packet receive
822 https://bugzilla.gnome.org/show_bug.cgi?id=760005
824 2016-03-02 19:42:58 +0200 Sebastian Dröge <sebastian@centricular.com>
826 * gst/rtsp-sink/gstrtspclientsink.c:
827 rtspclientsink: Add support for setting the multicast interface
828 https://bugzilla.gnome.org/show_bug.cgi?id=763000
830 2016-03-02 19:42:13 +0200 Sebastian Dröge <sebastian@centricular.com>
832 * gst/rtsp-server/rtsp-media-factory.c:
833 * gst/rtsp-server/rtsp-media-factory.h:
834 * gst/rtsp-server/rtsp-media.c:
835 * gst/rtsp-server/rtsp-media.h:
836 * gst/rtsp-server/rtsp-stream.c:
837 * gst/rtsp-server/rtsp-stream.h:
838 rtsp-media: Add support for setting the multicast interface
839 https://bugzilla.gnome.org/show_bug.cgi?id=763000
841 2016-03-07 08:50:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
843 * gst/rtsp-sink/gstrtspclientsink.c:
844 rtspclientsink: use new gst_element_class_add_static_pad_template()
845 https://bugzilla.gnome.org/show_bug.cgi?id=763196
847 2016-03-24 13:33:43 +0200 Sebastian Dröge <sebastian@centricular.com>
852 === release 1.8.0 ===
854 2016-03-24 13:00:35 +0200 Sebastian Dröge <sebastian@centricular.com>
860 * gst-rtsp-server.doap:
863 2016-03-16 23:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
865 * gst/rtsp-server/rtsp-stream.c:
866 rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
867 This would get us NO_PREROLL in the bin again and break seeking.
868 Thanks to Carlos Rafael Giani for helping to debug this!
869 https://bugzilla.gnome.org/show_bug.cgi?id=740509
871 === release 1.7.91 ===
873 2016-03-15 12:26:13 +0200 Sebastian Dröge <sebastian@centricular.com>
879 * gst-rtsp-server.doap:
882 2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
884 * gst/rtsp-server/rtsp-stream.c:
885 rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
886 Without this, RECORD pipelines are broken because
887 a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
888 added later. Previously it was there earlier and due to NO_PREROLL caused the
889 pipeline to preroll immediately
890 b) the udpsrc for the pipeline is added later and never set to PLAYING state,
891 as the corresponding code previously was only for PLAY pipelines.
892 https://bugzilla.gnome.org/show_bug.cgi?id=763281
894 2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
896 * gst/rtsp-server/rtsp-stream.c:
897 rtsp-stream: Fix typo in the docstring
898 gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
900 2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
902 * gst/rtsp-server/rtsp-stream.c:
903 rtsp-stream: Disable multicast loopback for all our sockets
904 On Windows this is a receiver-side setting, on Linux a sender-side setting. As
905 we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
906 loopback setting on the socket... while udpsink does which unfortunately has
907 no effect here on Windows but on Linux.
908 https://bugzilla.gnome.org/show_bug.cgi?id=757488
910 2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
912 * tests/check/gst/stream.c:
913 stream tests: added new tests
914 Test a case when the address pool only contains multicast addresses
915 and the client is requesting unicast udp.
916 Added tests for multicast ports allocation.
917 https://bugzilla.gnome.org/show_bug.cgi?id=757488
919 2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
921 * gst/rtsp-server/rtsp-stream.c:
922 rtsp-stream: Only bind multicast sockets to ANY on Windows
923 On Linux it is still needed to bind to the multicast address
924 to filter out random other packets, while on Windows binding
925 to multicast addresses just fails.
927 2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
929 * gst/rtsp-server/rtsp-stream.c:
930 rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
931 Otherwise we fail to allocate UDP ports if the pool only contains multicast
932 addresses, which is something that used to work before. For unicast addresses
933 if the pool contains none, we just allocate them as if there is no pool at
935 https://bugzilla.gnome.org/show_bug.cgi?id=757488
937 2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
939 * gst/rtsp-server/rtsp-client.c:
940 * gst/rtsp-server/rtsp-stream.c:
941 rtsp-server: Fix indentation
943 2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
945 * gst/rtsp-server/rtsp-stream.c:
946 rtsp-stream: Don't bind the sockets to multicast addresses
947 This works on Linux but fails completely on Windows. You're supposed
948 to bind to ANY and then join the multicast group.
949 https://bugzilla.gnome.org/show_bug.cgi?id=757488
951 === release 1.7.90 ===
953 2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
959 * gst-rtsp-server.doap:
962 2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
965 Automatic update of common submodule
966 From b64f03f to 6f2d209
968 2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
970 * gst/rtsp-sink/gstrtspclientsink.c:
971 * tests/check/gst/rtspclientsink.c:
972 rtspsink: Fix some leaks in rtspclientsink and the unit test.
973 https://bugzilla.gnome.org/show_bug.cgi?id=762525
975 2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
977 * tests/check/gst/media.c:
978 * tests/check/gst/rtspclientsink.c:
979 * tests/check/gst/rtspserver.c:
980 * tests/check/gst/stream.c:
981 tests: unit test fixes
982 Removed port allocation test from the media suite.
983 The port allocation failure is now in the stream suite.
985 Make sure that the media is suspended after the DESCRIBE request
986 before reconfiguring the UDP sinks.
988 In the RECORD case we have to set async property to false
989 for the appsink element in the test in order to make sure
990 that the media pipeline doesn't hang in start_preroll().
991 https://bugzilla.gnome.org/show_bug.cgi?id=757488
993 2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
995 * gst/rtsp-server/rtsp-client.c:
996 * gst/rtsp-server/rtsp-stream.c:
997 * gst/rtsp-server/rtsp-stream.h:
998 rtsp-stream: postpone UDP socket allocation until SETUP
999 Postpone the allocation of the UDP sockets until we know
1000 what transport has been chosen by the client.
1001 Both unicast and multicast UDP sources are created in one
1003 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1005 2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
1007 * gst/rtsp-server/rtsp-stream.c:
1008 rtsp-stream: postpone the creation of the UDP sources
1009 Code refactoring: allocate the UDP ports after the sender and
1010 the reciver parts have been created.
1011 We postpone the creation of the UDP sources until the UDP
1012 ports have been allocated.
1013 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1015 2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
1017 * gst/rtsp-server/rtsp-stream.c:
1018 rtsp-stream: added function for setting UDP sources to PLAYING state
1019 Code refactoring: Introduced a function for setting UDP sources
1021 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1023 2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
1025 * gst/rtsp-server/rtsp-stream.c:
1026 rtsp-stream: added function for creating and configuring UDP sources
1027 Code refactoring: create and configure UDP sources in a separate function.
1028 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1030 2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
1032 * gst/rtsp-server/rtsp-stream.c:
1033 rtsp-stream: added function for RTP/RTCP socket configuration
1034 Code refactoring: configure RTP and RTCP sockets for UDP sinks
1035 in a separate function.
1036 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1038 2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
1040 * gst/rtsp-server/rtsp-stream.c:
1041 rtsp-stream: added function for creating and configuring UDP sinks
1042 Code refactoring: create and configure UDP sinks in a separate function.
1043 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1045 2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
1047 * gst/rtsp-server/rtsp-stream.c:
1048 rtsp-stream: added helper function for creating the sender/receiver parts
1049 Code refactoring: introduced helper function for creating
1050 the receiver and the sender parts of the streaming pipeline.
1051 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1053 2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
1058 === release 1.7.2 ===
1060 2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
1066 * gst-rtsp-server.doap:
1069 2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
1071 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
1072 uninstalled.pc: add support for non libtool build systems
1073 Currently the .la path is provided which requires to use libtool as
1074 mentioned in the GStreamer manual section-helloworld-compilerun.html.
1075 It is fine as long as the application is built using libtool.
1076 So currently it is not possible to compile a GStreamer application
1077 within gst-uninstalled with CMake or other build system different
1079 This patch allows to do the following in gst-uninstalled env:
1080 gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
1081 gstreamer-rtsp-server-1.0)
1082 Previously it required to prepend libtool --mode=link
1083 https://bugzilla.gnome.org/show_bug.cgi?id=720778
1085 2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
1087 * gst/rtsp-sink/gstrtspclientsink.c:
1088 rtspclientsink: remove check for impossible condition
1089 Goto error label checks stream to see if it needs to be unreferenced before
1090 returning, but this goto jumps happens before the stream is ever set, so it
1091 will always be NULL in this error label.
1094 2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
1096 * gst/rtsp-sink/gstrtspclientsink.c:
1097 rtspclientsink: clean switch statements
1098 Coverity demands for fallthrough statements to be clearly commented,
1099 to distinguish from accidental fall throughs. And it also needs all
1100 cases to finish with a break, even if the break is never going to be
1101 executed like in the case of a continue jump.
1105 2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
1107 * tests/check/Makefile.am:
1108 tests: extend the AM_TESTS_ENVIRONMENT from check.mak
1109 To get the CK_DEFAULT_TIMEOUT defined for all tests
1110 Also removes a 120 seconds timeout that was set as default
1111 explicitly in this module
1112 https://bugzilla.gnome.org/show_bug.cgi?id=761472
1114 2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
1118 Automatic update of common submodule
1119 From 86e4663 to b64f03f
1121 2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
1123 * gst/rtsp-server/rtsp-media.c:
1124 rtsp-media: fix state_lock not locked again when preroll fails
1125 https://bugzilla.gnome.org/show_bug.cgi?id=761399
1127 2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
1130 configure: Move plugin specific flags below all the others
1131 They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
1132 -no-undefined. And -no-undefined is required on Windows to build DLLs.
1134 2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
1136 * gst/rtsp-sink/gstrtspclientsink.c:
1137 rtspclientsink: Simplify slightly using new -base API
1138 Use the new Mikey and SDP API in the base plugins libs
1139 to simplify some code.
1140 https://bugzilla.gnome.org/show_bug.cgi?id=758180
1142 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1147 * gst/rtsp-sink/Makefile.am:
1148 * gst/rtsp-sink/gstrtspclientsink.c:
1149 * gst/rtsp-sink/gstrtspclientsink.h:
1150 * gst/rtsp-sink/plugin.c:
1151 * tests/check/Makefile.am:
1152 * tests/check/gst/rtspclientsink.c:
1153 rtspsink: Add rtspclientsink element
1154 Add an rtspclientsink element that accepts streams for which
1155 there is a registered payloader and sends them to
1156 an RTSP server using RECORD.
1157 Sending is synchronised to the pipeline clock. Payload-types
1158 are automatically selected. The 'new-payloader' signal is fired
1159 for custom configuration of payloaders when they are created.
1160 Can now stream a movie like this:
1162 ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
1163 decodebin name=depay1 ! audioconvert ! autoaudiosink )"
1165 gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
1166 queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
1167 https://bugzilla.gnome.org/show_bug.cgi?id=758180
1169 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1171 * gst/rtsp-server/rtsp-stream.c:
1172 * gst/rtsp-server/rtsp-stream.h:
1173 rtsp-stream: Add functions for using rtsp-stream from the client
1174 Add a boolean to indicate that the rtsp-stream is running on the
1175 'client' side of an RTSP connection, for sending streams via
1176 RECORD. In that case, the roles of the client/server ports
1177 in transport setup are swapped.
1178 https://bugzilla.gnome.org/show_bug.cgi?id=758180
1180 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1182 * gst/rtsp-server/rtsp-sdp.c:
1183 * gst/rtsp-server/rtsp-sdp.h:
1184 rtsp-sdp: Add gst_rtsp_sdp_from_stream()
1185 A new function that adds info from a GstRTSPStream into an SDP message.
1186 https://bugzilla.gnome.org/show_bug.cgi?id=758180
1188 2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
1190 * gst/rtsp-server/rtsp-media.c:
1191 rtsp-media: Fix mutex beeing unlocked while they should be locked
1192 https://bugzilla.gnome.org/show_bug.cgi?id=761226
1194 2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
1196 * gst/rtsp-server/rtsp-media-factory.c:
1197 rtsp-media-factory: add missing break in "clock" property setter
1200 2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
1202 * gst/rtsp-server/rtsp-stream.c:
1203 rtsp-stream: fixed assert during update transport
1204 When RTSP server trying update transport during multicast, it throws an
1205 assert. The assert is thrown because it is trying to get the parent of
1206 an non-existing funnel element.
1207 https://bugzilla.gnome.org/show_bug.cgi?id=760150
1209 2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
1211 * gst/rtsp-server/rtsp-permissions.h:
1212 * gst/rtsp-server/rtsp-thread-pool.h:
1213 * gst/rtsp-server/rtsp-token.h:
1214 docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
1215 gtk-doc can handle static inline functions just fine these days,
1216 there's no need for this stuff any more.
1218 2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1220 * gst/rtsp-server/rtsp-media.c:
1221 * gst/rtsp-server/rtsp-sdp.c:
1222 sdp: replace duplicated codes to call new base sdp apis
1223 https://bugzilla.gnome.org/show_bug.cgi?id=745880
1225 2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
1227 * examples/test-netclock.c:
1228 test-netclock: Use the new API to configure a clock directly
1230 2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
1232 * gst/rtsp-server/rtsp-media-factory.c:
1233 * gst/rtsp-server/rtsp-media-factory.h:
1234 * gst/rtsp-server/rtsp-media.c:
1235 * gst/rtsp-server/rtsp-media.h:
1236 rtsp-media: Add API to directly configure a clock on the media pipelines
1238 2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
1240 * gst/rtsp-server/rtsp-media.c:
1241 rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
1243 2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
1245 * gst/rtsp-server/rtsp-media-factory.c:
1246 rtsp-media-factory: Add FIXME for 2.0
1248 2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
1250 * gst/rtsp-server/rtsp-stream.c:
1251 rtsp-stream: Fix indentation
1253 2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1255 * gst/rtsp-server/rtsp-media.c:
1256 rtsp-media: Do not prepare media after media times out
1257 Deferred calls to start_prepare() can be deferred past the point until
1258 which wait_preroll() and by proxy gst_rtsp_media_get_status() is
1259 prepared to wait. Previously there was no lock and no check for this
1260 situation. This meant that a media could be prepared and unprepared
1261 simultaneously by two different threads. Now a lock is in place and a
1262 suitable check is done.
1263 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
1265 2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
1267 * gst/rtsp-server/rtsp-client.c:
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 property to decide if sending media should be stopped when a client disconnects without TEARDOWN
1273 Without TEARDOWN it might be desireable to keep the media running and continue
1274 sending data to the client, even if the RTSP connection itself is
1276 Only do this for session medias that have only UDP transports. If there's at
1277 least on TCP transport, it will stop working and cause problems when the
1278 connection is disconnected.
1279 https://bugzilla.gnome.org/show_bug.cgi?id=758999
1281 2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
1286 === release 1.7.1 ===
1288 2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
1294 * gst-rtsp-server.doap:
1297 2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
1300 configure: Make -Bsymbolic check work with clang.
1301 Update the -Bsymbolic check with the version glib has. This version
1303 https://bugzilla.gnome.org/show_bug.cgi?id=759713
1305 2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
1307 * gst/rtsp-server/rtsp-session-pool.c:
1308 rtsp-session-pool: Avoid dollar sign ($) in session ids
1309 Live555 in VLC strips off dollar signs and then gets very confused,
1310 we don't loose too much entropy by just skipping it.
1312 2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
1314 * gst/rtsp-server/rtsp-address-pool.h:
1315 * gst/rtsp-server/rtsp-auth.h:
1316 * gst/rtsp-server/rtsp-client.h:
1317 * gst/rtsp-server/rtsp-media-factory-uri.h:
1318 * gst/rtsp-server/rtsp-media-factory.h:
1319 * gst/rtsp-server/rtsp-media.h:
1320 * gst/rtsp-server/rtsp-mount-points.h:
1321 * gst/rtsp-server/rtsp-permissions.h:
1322 * gst/rtsp-server/rtsp-server.h:
1323 * gst/rtsp-server/rtsp-session-media.h:
1324 * gst/rtsp-server/rtsp-session-pool.h:
1325 * gst/rtsp-server/rtsp-session.h:
1326 * gst/rtsp-server/rtsp-stream-transport.h:
1327 * gst/rtsp-server/rtsp-stream.h:
1328 * gst/rtsp-server/rtsp-thread-pool.h:
1329 * gst/rtsp-server/rtsp-token.h:
1330 rtsp-server: Add g_autoptr() support to all types
1331 https://bugzilla.gnome.org/show_bug.cgi?id=754464
1333 2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
1335 * gst/rtsp-server/rtsp-stream.c:
1336 rtsp-stream: fixed valgrind error
1337 Fixed the valgrind error in unit test. The UDP source created during
1338 gst_rtsp_stream_join_bin() was not released while destroying the rtp
1340 https://bugzilla.gnome.org/show_bug.cgi?id=759010
1342 2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1346 Automatic update of common submodule
1347 From b319909 to 86e4663
1349 2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
1351 * gst/rtsp-server/rtsp-client.c:
1352 rtsp-client: suspend media during setup request
1353 SETUP request from clients needs to suspend the media to clear the
1354 prerolled buffers. Otherwise it will not affect the prerolled buffer
1355 and the prerolled buffers will be incorrect (for example block-size
1356 from setup request will not affect the prerolled buffer unless the
1357 media is suspended).
1358 https://bugzilla.gnome.org/show_bug.cgi?id=758268
1360 2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
1362 * gst/rtsp-server/rtsp-stream.c:
1363 rtsp-stream: create stream pipeline based on transport
1364 Based on the protocol, create the rtsp stream pipeline. If only TCP or
1365 only UDP is set as the transport protocol, it will not add the extra tee
1366 or queue element to the pipeline. Both these elements will be added, if
1367 it supports both TCP and UDP protocols. This improves the pipeline
1368 performance when one protocol is present.
1369 https://bugzilla.gnome.org/show_bug.cgi?id=758179
1371 2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
1373 * gst/rtsp-server/rtsp-stream.c:
1374 rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
1375 Adding them when not needed will start some logic inside rtpbin that might be
1376 problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
1377 would start up a rtpjitterbuffer and behave in weird ways.
1378 We still set up the UDP sources for RTP receiving for a sender media to be
1379 able to receive any packets sent by the client for NAT traversal. They will
1380 all go to a fakesink though.
1381 Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
1382 NO_PREROLL, which will cause deadlocks when seeking the media as it will never
1383 receive ASYNC_DONE after a seek.
1384 https://bugzilla.gnome.org/show_bug.cgi?id=758319
1386 2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
1388 * gst/rtsp-server/rtsp-stream.c:
1389 rtsp-stream: Disable multicast loopback for the multicast udp sources too
1390 On POSIX this setting is for sender sockets, on Windows for receiver sockets.
1391 Previously we were only setting this for sender sockets, which caused looped
1392 back packets to be received on Windows if a multicast transport was used.
1394 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1396 * examples/test-record-auth.c:
1397 * examples/test-record.c:
1398 examples: Actually use the provided port in the record examples
1400 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1402 * examples/test-record-auth.c:
1403 test-record-auth: Add the option to build in TLS support
1405 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1407 * examples/test-auth.c:
1408 test-auth: Use an 'anonymous' user for unauthenticated default
1409 There's a comment on one of the resources that 'user' and 'admin'
1410 shouldn't even be able to see it, but they can if the default
1411 token is 'admin2', since that gives them access anyway.
1413 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1415 * examples/.gitignore:
1416 * examples/Makefile.am:
1417 * examples/test-record-auth.c:
1418 Add test-record-auth example
1420 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1422 * gst/rtsp-server/rtsp-client.c:
1423 * tests/check/gst/client.c:
1424 rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
1426 2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
1428 * gst/rtsp-server/rtsp-server.c:
1429 rtsp-server: Change the logic so we don't pop a NULL context
1430 When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
1431 will sometimes fail. This call is made before any context is pushed
1432 resulting in an attempt to pop a NULL context.
1433 https://bugzilla.gnome.org/show_bug.cgi?id=757949
1435 2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
1437 * tests/check/gst/rtspserver.c:
1438 rtspserver: Add udp-mcast transport SETUP test
1439 Refactor utility functions in the test file so they can handle
1440 more than UDP and TCP as lower transport.
1441 https://bugzilla.gnome.org/show_bug.cgi?id=756969
1443 2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
1445 * gst/rtsp-server/rtsp-stream.c:
1446 rtsp-stream: Always unref return value of gst_object_get_parent()
1447 Fixes a leak of a GstBin in the udp-mcast case.
1448 https://bugzilla.gnome.org/show_bug.cgi?id=756968
1450 2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
1453 Automatic update of common submodule
1454 From b99800a to b319909
1456 2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
1459 Use new GST_ENABLE_EXTRA_CHECKS #define
1460 https://bugzilla.gnome.org/show_bug.cgi?id=756870
1462 2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
1465 Automatic update of common submodule
1466 From 6babecd to b99800a
1468 2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
1471 Update GLib dependency to 2.40.0
1473 2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1475 * examples/test-mp4.c:
1476 * gst/rtsp-server/rtsp-stream.c:
1477 stream: listen to sender ssrc signals
1478 https://bugzilla.gnome.org/show_bug.cgi?id=746747
1480 2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
1483 common: update for new suppression
1484 Makes check-valgrind pass with glib 2.46
1486 2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1488 * gst/rtsp-server/rtsp-media.c:
1489 rtsp-media: Take reference to media that will be prepared
1490 default_prepare() takes a transfer-none reference GstRTSPMedia object.
1491 Later on a g_idle_source_new() is created and a pointer to the media
1492 object is passed as user data. If the media is freed before the idle
1493 source is dispatched the media object pointer is invalid, but the idle
1494 source callback expects it to still be valid. To fix this a reference to
1495 the media object is taken when registering the source callback function
1496 and a corresponding release of the reference is done when the souce is
1498 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
1500 2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
1502 * examples/test-launch.c:
1503 * examples/test-mp4.c:
1504 * examples/test-ogg.c:
1505 * examples/test-record.c:
1506 * examples/test-uri.c:
1507 rtsp-server: Fix memory leaks when context parse fails
1508 When g_option_context_parse fails, context and error variables are not getting free'd
1509 which results in memory leaks. Free'ing the same.
1510 And replacing g_error_free with g_clear_error, which checks if the error being passed
1511 is not NULL and sets the variable to NULL on free'ing.
1512 https://bugzilla.gnome.org/show_bug.cgi?id=753863
1514 2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
1519 === release 1.6.0 ===
1521 2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
1527 * gst-rtsp-server.doap:
1530 === release 1.5.91 ===
1532 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
1538 * gst-rtsp-server.doap:
1541 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
1543 * docs/libs/gst-rtsp-server-sections.txt:
1544 * gst/rtsp-server/rtsp-stream.c:
1545 stream: fix docs for recently-added get/set_buffer_size API
1546 https://bugzilla.gnome.org/show_bug.cgi?id=749095
1548 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
1550 * gst/rtsp-server/rtsp-media.c:
1551 rtsp-media: Don't crash on encrypted RTX SDP
1552 In parse_keymgmt(), don't mutate the input string that's been passed
1553 as const, especially since we might need the original value again if
1554 the same key info applies to multiple streams (RTX, for example).
1555 https://bugzilla.gnome.org/show_bug.cgi?id=754753
1557 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
1559 * examples/test-mp4.c:
1560 test-mp4: Support filenames with spaces in them. Error out on too few arguments
1562 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
1564 * examples/test-record.c:
1565 test-record: Check parameter count and print out help
1566 If no launch pipeline was supplied, print out some help
1568 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
1570 * gst/rtsp-server/rtsp-media.c:
1571 * gst/rtsp-server/rtsp-stream.c:
1572 * gst/rtsp-server/rtsp-stream.h:
1573 rtsp-stream: Implement UDP buffer size setting.
1574 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
1576 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
1577 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
1579 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
1581 * gst/rtsp-server/rtsp-media.h:
1582 rtsp-media: Fix small typo causing gtk-doc to complain
1584 === release 1.5.90 ===
1586 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
1592 * gst-rtsp-server.doap:
1595 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1597 * gst/rtsp-server/rtsp-media-factory.c:
1598 media-factory: get port number through gst_rtsp_url_get_port
1599 https://bugzilla.gnome.org/show_bug.cgi?id=753473
1601 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
1603 * tests/check/gst/media.c:
1604 media-test: Removing unnecessary assertion
1605 https://bugzilla.gnome.org/show_bug.cgi?id=753385
1607 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1609 * gst/rtsp-server/rtsp-server.c:
1610 Document that source keeps a ref on server until it's destroyed
1611 https://bugzilla.gnome.org/show_bug.cgi?id=749227
1613 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1615 * tests/check/gst/media.c:
1616 media-test: Test for multiple dynamic payload
1617 https://bugzilla.gnome.org/show_bug.cgi?id=753385
1619 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1621 * gst/rtsp-server/rtsp-media.c:
1622 media: Only add fakesink once per pipeline
1623 The intention is to prevent going PLAYING state before pads are created.
1624 If there was mutilple dynamic payload, it would leak few fakesink and
1625 actually prevent from ever reaching playing state.
1626 https://bugzilla.gnome.org/show_bug.cgi?id=753385
1628 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1630 * gst/rtsp-server/rtsp-media.c:
1631 Revert "rtsp-media: Only add 1 fakesink per pipeline"
1632 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
1634 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1636 * gst/rtsp-server/rtsp-media.c:
1637 rtsp-media: Only add 1 fakesink per pipeline
1638 There should be only one fakesink per pipeline, not per dynpay. This
1639 would lead to element naming clash.
1641 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
1643 * gst/rtsp-server/rtsp-media.c:
1644 rtsp-media: assertion error due to wrong condition check
1645 In media to caps function, reserved_keys array is being used for variable i,
1646 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
1647 changed it to variable j
1648 https://bugzilla.gnome.org/show_bug.cgi?id=753009
1650 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
1652 * gst/rtsp-server/rtsp-media.c:
1653 rtsp-media: Strip keys from the fmtp that we use internally in our caps
1654 Skip keys from the fmtp, which we already use ourselves for the
1655 caps. Some software is adding random things like clock-rate into
1656 the fmtp, and we would otherwise here set a string-typed clock-rate
1657 in the caps... and thus fail to create valid RTP caps
1658 https://bugzilla.gnome.org/show_bug.cgi?id=753009
1660 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1662 * gst/rtsp-server/rtsp-thread-pool.c:
1663 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
1664 https://bugzilla.gnome.org/show_bug.cgi?id=752640
1666 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
1669 Automatic update of common submodule
1670 From f74b2df to 9aed1d7
1672 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
1677 === release 1.5.2 ===
1679 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
1685 * gst-rtsp-server.doap:
1688 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
1690 * gst/rtsp-server/rtsp-client.c:
1691 * gst/rtsp-server/rtsp-client.h:
1692 * tests/check/gst/client.c:
1693 rtsp-client: allow application to decide what requirements are supported
1694 Add "check-requirements" signal and vfunc to allow application
1695 (and subclasses) to check the requirements.
1696 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
1697 https://bugzilla.gnome.org/show_bug.cgi?id=749417
1699 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1702 Automatic update of common submodule
1703 From 6015d26 to f74b2df
1705 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
1707 * gst/rtsp-server/rtsp-media.c:
1708 rtsp-media: Always use real payloader when creating streams
1709 A bin that contains the real payloader might be used as payloader. In this
1710 case we have to get the real payloader for the various properties it provides.
1711 Example use cases for this are bins that payload some media and then have
1712 additional elements that add metadata or RTP extension headers to the stream.
1713 https://bugzilla.gnome.org/show_bug.cgi?id=750800
1715 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
1717 * examples/test-netclock-client.c:
1718 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
1720 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
1722 * examples/test-netclock-client.c:
1723 * examples/test-netclock.c:
1724 test-netclock: Use new ntp-time-source property on rtpbin
1725 Select the clock time to be used as NTP time source. This allows proper
1726 synchronization between receivers, independent of sharing base times, and just
1727 requires them to use the same clock.
1729 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
1731 * examples/test-netclock-client.c:
1732 * examples/test-netclock.c:
1733 test-netclock: Setting the same base time on sender and receiver is not necessary
1734 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
1736 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1738 * gst/rtsp-server/rtsp-stream.c:
1739 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
1740 https://bugzilla.gnome.org/show_bug.cgi?id=750764
1742 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1744 * docs/libs/gst-rtsp-server.types:
1745 docs: add missing types
1746 https://bugzilla.gnome.org/show_bug.cgi?id=750764
1748 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1750 * docs/libs/gst-rtsp-server-sections.txt:
1751 docs: add missing apis
1752 https://bugzilla.gnome.org/show_bug.cgi?id=750764
1754 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
1756 * examples/test-netclock-client.c:
1757 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
1759 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1761 * docs/libs/gst-rtsp-server-sections.txt:
1762 * gst/rtsp-server/rtsp-auth.c:
1763 * gst/rtsp-server/rtsp-auth.h:
1764 GstRTSPAuth: Add client certificate authentication support
1765 https://bugzilla.gnome.org/show_bug.cgi?id=750471
1767 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
1769 * examples/test-netclock-client.c:
1770 test-netclock-client: Use new GstClock API to wait for clock synchronization
1772 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
1774 * examples/test-netclock-client.c:
1775 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
1776 A mainloop is needed to get glimagesink to display something on OSX, and
1777 the source-setup signal just makes things a little bit easier.
1779 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
1782 Automatic update of common submodule
1783 From d9a3353 to 6015d26
1785 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
1788 Automatic update of common submodule
1789 From d37af32 to d9a3353
1791 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
1794 Automatic update of common submodule
1795 From 21ba2e5 to d37af32
1797 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
1800 Automatic update of common submodule
1801 From c408583 to 21ba2e5
1803 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
1805 * docs/libs/Makefile.am:
1806 docs: remove variables that we define in the snippet from common
1807 This is syncing our Makefile.am with upstream gtkdoc.
1809 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
1812 Automatic update of common submodule
1813 From 44a3517 to c408583
1815 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
1820 === release 1.5.1 ===
1822 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
1828 * gst-rtsp-server.doap:
1831 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
1833 * gst/rtsp-server/rtsp-client.c:
1834 rtsp-client: No flush during Teardown.
1835 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
1836 backlog is empty it can happen that just a part of a message will be
1837 sent and rest is in backlog queue. If then flush during teardown
1838 just a part of message will be sent.This can lead to client miss
1839 teardown response since it expect to get the last part of message.
1840 The flushing during teardown was introduced to fix a deadlock that now
1841 is fixed more generally in handle_request by temporary setting backlog
1843 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
1845 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
1847 * tests/check/Makefile.am:
1848 tests: Use AM_TESTS_ENVIRONMENT
1849 Needed by the new automake test runner and the
1850 current version of the common submodule.
1852 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
1854 * gst/rtsp-server/rtsp-media.h:
1855 * gst/rtsp-server/rtsp-stream.h:
1856 rtsp-server: Use single-include rtsp header to make sure we get all definitions
1858 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
1860 * gst/rtsp-server/rtsp-media.c:
1861 rtsp-media: Mark some more functions static
1863 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
1865 * gst/rtsp-server/rtsp-media.c:
1866 rtsp-media: Only unblock the media in suspend() when actually changing the state
1867 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
1869 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
1871 * examples/test-video-rtx.c:
1872 examples: Use AVPF profile for the RTX example
1874 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
1876 * gst/rtsp-server/rtsp-sdp.c:
1877 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
1879 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1881 * gst/rtsp-server/rtsp-stream.c:
1882 rtsp-stream: get valid clock-rate from last-sample
1883 clock-rate in last-sample's caps is integer, not unsigned.
1884 To get this value properly, variable needs to be type-casted to int.
1885 https://bugzilla.gnome.org/show_bug.cgi?id=747614
1887 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
1891 autogen.sh: only run autopoint if gettext requested in configure.ac
1892 Not just because there happens to be a po directory.
1893 https://bugzilla.gnome.org/show_bug.cgi?id=748058
1895 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
1898 Revert "configure.ac: uncomment gettext version setup"
1899 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
1900 We don't need a gettext setup here and there's no po
1901 directory either, so no reason why autopoint would be
1902 run in the first place.
1903 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
1905 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
1907 * examples/test-multicast.c:
1908 * examples/test-multicast2.c:
1909 * examples/test-sdp.c:
1910 * examples/test-video-rtx.c:
1911 * examples/test-video.c:
1912 * tests/test-cleanup.c:
1913 * tests/test-reuse.c:
1914 Fix timeout function signatures across tests and examples
1916 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
1918 * tests/check/Makefile.am:
1919 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
1920 Make sure the test environment is set up.
1921 https://bugzilla.gnome.org//show_bug.cgi?id=747624
1923 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
1926 configure: bump automake requirement to 1.14 and autoconf to 2.69
1927 This is only required for builds from git, people can still
1928 build tarballs if they only have older autotools.
1929 https://bugzilla.gnome.org//show_bug.cgi?id=747624
1931 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
1934 configure.ac: uncomment gettext version setup
1935 Fixes autogen.sh. It would run autopoint, which would complain
1936 that it could not find the gettext version in configure.ac.
1937 https://bugzilla.gnome.org/show_bug.cgi?id=748058
1939 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1941 * examples/test-video-rtx.c:
1942 test-video-rtx: set exact payload type to PCMA payloader
1943 Setting wrong payload type causes failure to do retransmission through audio stream
1944 https://bugzilla.gnome.org/show_bug.cgi?id=747839
1946 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1948 * gst/rtsp-server/rtsp-media.c:
1949 * gst/rtsp-server/rtsp-stream.c:
1950 * gst/rtsp-server/rtsp-stream.h:
1951 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
1952 Because of duplicated g_signal_connect for request-aux-sender signal,
1953 wrong stream pointer is passed to the signal handler.
1954 Instead of passing each stream, pass stream array and get the relevant stream.
1955 https://bugzilla.gnome.org/show_bug.cgi?id=747839
1957 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
1961 Update autogen.sh to latest version from common
1962 Fixes build after aclocal_check etc. helpers have been removed.
1964 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
1967 Automatic update of common submodule
1968 From bc76a8b to c8fb372
1970 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
1972 * gst/rtsp-server/rtsp-stream.c:
1973 rtsp-stream: Limit the queues to 1 buffer
1974 We only need them to be able to pre-roll, queueing up more data here
1975 is only going to harm latency and memory usage.
1977 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
1979 * gst/rtsp-server/rtsp-stream.c:
1980 rtsp-stream: Update comment and ASCII art to the latest code
1981 We have a queue in front of the udpsink too to prevent the pipeline from
1984 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
1986 * gst/rtsp-server/rtsp-stream.c:
1987 rtsp-media: Properly return first rtptime
1988 Instead we where returning first GstBuffer timestamp. This would result
1989 in clock skew and unwanted behaviour in RTSP playback.
1990 https://bugzilla.gnome.org/show_bug.cgi?id=746479
1992 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
1994 * gst/rtsp-server/rtsp-stream.c:
1995 rtsp-stream: Don't leave buffer mapped
1996 If the seq is NULL, the RTP buffer was left mapped. We should always
1999 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
2004 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
2006 * gst/rtsp-server/rtsp-media-factory.c:
2007 * tests/check/gst/client.c:
2008 Fix double semicolons
2010 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
2012 * gst/rtsp-server/rtsp-stream.c:
2013 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
2014 This gives more accurate values than asking the payloader. There might be
2015 queueing happening between the payloader and the sink.
2016 https://bugzilla.gnome.org/show_bug.cgi?id=745704
2018 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
2020 * gst/rtsp-server/rtsp-media.c:
2021 rtsp-media: Don't seek for PLAY if the position will not change
2022 https://bugzilla.gnome.org/show_bug.cgi?id=745704
2024 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
2026 * gst/rtsp-server/rtsp-media.c:
2027 rtsp-media: Don't include payload type in the caps for framesize
2028 When the sdp media attribute framesize are converted to caps
2029 the <payload> should not be included.
2030 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
2031 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
2033 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
2035 * gst/rtsp-server/rtsp-sdp.c:
2036 rtsp-sdp: add payload type to the sdp framesize attribute
2037 The sdp framesize attribute is desribed in RFC6064. It is specified
2038 for payloading of H263 and has the following form
2039 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
2040 should be added to the caps in a payloader and the <payload type> should
2041 be added by the rtsp-server.
2042 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
2044 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
2046 * examples/test-uri.c:
2047 examples: test-uri: fix tainted variable
2048 Insignificant but this keeps Coverity happy.
2051 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
2053 * examples/.gitignore:
2054 * examples/Makefile.am:
2055 * examples/test-netclock-client.c:
2056 * examples/test-netclock.c:
2057 examples: Add a simple example of network synch for live streams.
2058 An example server and client that works for synchronising live streams
2059 only - as it can't support pause/play.
2061 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
2063 * gst/rtsp-server/rtsp-media-factory.c:
2064 * gst/rtsp-server/rtsp-media-factory.h:
2065 rtsp-media-factory: Add functions to set/get the media gtype
2066 Allow specifying the GType of a GstRtspMedia subclass to create
2067 as a simpler way to get the factory to create a custom
2068 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
2070 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
2072 * gst/rtsp-server/rtsp-media.c:
2073 rtsp-media: fix double unlock in _get_buffer_size()
2074 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
2075 because of double g_mutex_unlock () usage.
2076 https://bugzilla.gnome.org/show_bug.cgi?id=745434
2078 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
2080 * gst/rtsp-server/rtsp-session-pool.c:
2081 * gst/rtsp-server/rtsp-session.c:
2082 * gst/rtsp-server/rtsp-session.h:
2083 rtsp-session: Use monotonic time for RTSP session timeout
2084 Changed RTSP session timeout handling to monotonic time
2085 and deprecating the API for current system time.
2086 This fixes timeouts when the system time changes.
2087 https://bugzilla.gnome.org/show_bug.cgi?id=743346
2089 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
2091 * gst/rtsp-server/rtsp-client.c:
2092 * gst/rtsp-server/rtsp-media.c:
2093 rtsp-client: Only error out in PLAY if seeking actually failed
2094 If the media was just not seekable, we continue from whatever position we are
2095 and let the client decide if that is what is wanted or not.
2096 Only if the actual seek failed, we can't really recover and should error out.
2098 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
2100 * gst/rtsp-server/rtsp-stream.c:
2101 rtsp-stream: Add necessary queues between tee and multiudpsink
2102 https://bugzilla.gnome.org/show_bug.cgi?id=744379
2104 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
2106 * gst/rtsp-server/rtsp-client.c:
2107 * gst/rtsp-server/rtsp-media.c:
2108 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
2109 Instead error out properly the same way as if the SEEKING query already
2112 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
2114 * gst/rtsp-server/rtsp-stream.h:
2115 rtsp-stream: minor code formatting fix
2117 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
2119 * gst/rtsp-server/rtsp-media.c:
2120 rtsp-media: fix logic for collect_streams
2121 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
2122 all streams it knows if it got any, and can check if the transport mode is OK.
2125 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
2127 * gst/rtsp-server/rtsp-media.c:
2128 rtsp-media: Don't set the transport mode based on what elements we find
2129 Just print a warning if the one that was set before disagrees with what
2130 elements we found. It must already be set to something before as this
2131 function is called after we received the SDP from ANNOUNCE in RECORD mode,
2132 and we would reject ANNOUNCE if the RECORD flag was not set.
2134 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
2136 * tests/check/gst/rtspserver.c:
2137 tests: rtspserver: rename shadowed variable
2138 We have two different 'sink' variables here,
2139 rename one of them for clarity.
2141 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
2143 * gst/rtsp-server/rtsp-client.c:
2144 rtsp-client: fix awkward if clause
2146 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
2148 * examples/test-uri.c:
2149 examples: test-uri: improve uri argument handling and accept file names
2150 Print an error if the argument passed is not a URI and can't
2151 be converted into one, or no arguments have been provided.
2153 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
2155 * examples/test-uri.c:
2156 examples: test-uri: don't remove mount point after 10 seconds
2157 It's very irritating when trying to test stuff repeatedly
2158 and serves no real purpose other than showing that it can
2161 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
2163 * examples/.gitignore:
2164 examples: add new test-record to .gitignore
2166 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
2168 * examples/test-record.c:
2169 * gst/rtsp-server/rtsp-client.c:
2170 * gst/rtsp-server/rtsp-media-factory.c:
2171 * gst/rtsp-server/rtsp-media-factory.h:
2172 * gst/rtsp-server/rtsp-media.c:
2173 * gst/rtsp-server/rtsp-media.h:
2174 * tests/check/gst/rtspserver.c:
2175 rtsp-media: Use flags to distinguish between PLAY and RECORD media
2177 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
2179 * examples/test-record.c:
2180 test-record: Set latency for playback-style example to 2s instead of 200ms
2182 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
2184 * tests/check/gst/rtspserver.c:
2185 tests: add some unit tests for ANNOUNCE and RECORD
2186 https://bugzilla.gnome.org/show_bug.cgi?id=743175
2188 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
2190 * gst/rtsp-server/rtsp-client.c:
2191 rtsp-client: fix a couple of leaks in handle_announce
2193 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
2195 * gst/rtsp-server/rtsp-media-factory.c:
2196 * gst/rtsp-server/rtsp-media-factory.h:
2197 * gst/rtsp-server/rtsp-media.c:
2198 * gst/rtsp-server/rtsp-media.h:
2199 rtsp-media: Expose latency setting for setting the rtpbin latency
2201 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
2203 * examples/test-record.c:
2204 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
2206 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
2208 * gst/rtsp-server/rtsp-stream.c:
2209 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
2211 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
2213 * examples/Makefile.am:
2214 * examples/test-record.c:
2215 * gst/rtsp-server/rtsp-client.c:
2216 * gst/rtsp-server/rtsp-client.h:
2217 * gst/rtsp-server/rtsp-media-factory.c:
2218 * gst/rtsp-server/rtsp-media-factory.h:
2219 * gst/rtsp-server/rtsp-media.c:
2220 * gst/rtsp-server/rtsp-media.h:
2221 * gst/rtsp-server/rtsp-session-media.c:
2222 * gst/rtsp-server/rtsp-stream.c:
2223 * gst/rtsp-server/rtsp-stream.h:
2224 Add initial support for RECORD
2225 We currently only support media that is RECORD or PLAY only, not both at once.
2226 https://bugzilla.gnome.org/show_bug.cgi?id=743175
2228 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
2230 * gst/rtsp-server/rtsp-stream.c:
2231 rtsp-stream: RTCP and RTP transport cache cookies seperated
2232 RTCP packets were not sent because the same tr_cache_cookie was used for
2233 both RTP and RTCP. So only one of the tr_cache lists were populated
2234 depending on which one was sent first. If the tr_cache list is not
2235 populated then no packets can be sent. Most often this happened to be
2236 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
2237 resulted in both the tr_cache_lists to be populated regardless of which
2239 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
2241 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
2243 * gst/rtsp-server/rtsp-stream.c:
2244 rtsp-stream: fix false compiler warning
2245 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
2247 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
2249 * gst/rtsp-server/rtsp-client.c:
2250 rtsp-client: log interleaved data received
2252 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
2254 * gst/rtsp-server/rtsp-client.c:
2255 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
2257 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
2259 * gst/rtsp-server/rtsp-client.c:
2260 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
2262 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
2264 * gst/rtsp-server/rtsp-client.c:
2265 rtsp-client: Use a random session ID in the SDP
2266 RFC4566 Section 5.2 says that it should make the username, session id,
2267 nettype, addrtype and unicast address tuple globally unique. Always using
2268 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
2269 Instead let's create a 64 bit random number, which at least brings us
2270 closer to the goal of global uniqueness.
2271 https://tools.ietf.org/html/rfc4566#section-5.2
2273 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
2275 * examples/test-launch.c:
2276 * examples/test-mp4.c:
2277 * examples/test-ogg.c:
2278 * examples/test-uri.c:
2279 examples: Don't call gst_init() and gst_get_option_group()
2280 The latter calls the former at the appropriate time.
2282 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
2284 * gst/rtsp-server/rtsp-client.c:
2285 rtsp-client: Drop trailing \0 of RTSP DATA messages
2286 We add a trailing \0 in GstRTSPConnection to make parsing of
2287 string message bodies easier (e.g. the SDP from DESCRIBE) but
2288 for actual data this means we have to drop it or otherwise
2289 create invalid data.
2291 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
2293 * gst/rtsp-server/rtsp-stream.c:
2294 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
2295 Fixes crash when two threads access handle_new_sample() at the same
2296 time, one for RTP, one for RTCP.
2297 Otherwise, when iterating over the transports cache, it might be modified by
2298 another thread at the same time if the transports cookie has changed.
2299 https://bugzilla.gnome.org/show_bug.cgi?id=742954
2301 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
2303 * gst/rtsp-server/rtsp-stream.c:
2304 rtsp-stream: Set format=TIME on our app sources for TCP
2306 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
2308 * gst/rtsp-server/rtsp-session-pool.c:
2309 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
2310 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
2311 RFC 2326 states that session IDs may consist of alphanumeric as well as
2312 the safe characters $-_.+ -- N.B. the percent character is not allowed.
2313 Previously the session ID was URI-escaped, this meant that any character
2314 which was not alphanumeric or any of the characters +-._~ would be
2315 percent encoded. While the RFC (surprisingly) mentions that linear white
2316 space in session IDs should be URI-escaped, it does not say anything
2317 about other characters. Moreover no white space is allowed in the
2318 session ID. Finally the percent character which is the result of
2319 URI-escaping is not allowed in a session ID.
2320 So there is no reason to do any URI-escaping, and now it is removed.
2321 https://bugzilla.gnome.org/show_bug.cgi?id=742869
2323 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
2326 Automatic update of common submodule
2327 From f2c6b95 to bc76a8b
2329 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
2332 Fix 'make check' from top-level directory
2334 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2336 * examples/test-launch.c:
2337 * examples/test-mp4.c:
2338 * examples/test-ogg.c:
2339 * examples/test-uri.c:
2340 examples: Add command-line parsing and take a 'port' argument
2341 This allows users to run multiple servers on different ports for testing.
2342 Only done for examples that actually take arguments and hence are capable of
2343 outputting different streams for each instance on each port.
2344 https://bugzilla.gnome.org/show_bug.cgi?id=742115
2346 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
2348 * gst/rtsp-server/rtsp-client.c:
2349 * gst/rtsp-server/rtsp-client.h:
2350 rtsp-client: Add a send_message default signal handler
2351 This allows subclasses to easily hook into the response sending
2352 mechanism without doing everything from a signal, which seems
2353 awkward from subclasses.
2355 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
2358 Automatic update of common submodule
2359 From ef1ffdc to f2c6b95
2361 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2365 configure: add --disable-examples switch
2366 https://bugzilla.gnome.org/show_bug.cgi?id=741678
2368 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
2370 * examples/.gitignore:
2371 * examples/Makefile.am:
2372 * examples/test-video-rtx.c:
2373 examples: add a retransmisison example implementing RFC4588
2374 Currently only SSRC-multiplexed rtx streams are supported
2376 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
2378 * gst/rtsp-server/rtsp-stream.c:
2379 rtsp-stream: Fix some minor memory leaks
2381 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
2383 * gst/rtsp-server/rtsp-media.c:
2384 rtsp-media: Some minor cleanup
2386 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
2388 * gst/rtsp-server/rtsp-stream.c:
2389 rtsp-stream: Fix compiler warnings
2390 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
2391 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
2393 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
2394 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
2397 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
2399 * docs/libs/gst-rtsp-server-sections.txt:
2400 * gst/rtsp-server/rtsp-media-factory.c:
2401 * gst/rtsp-server/rtsp-media-factory.h:
2402 * gst/rtsp-server/rtsp-media.c:
2403 * gst/rtsp-server/rtsp-media.h:
2404 * gst/rtsp-server/rtsp-sdp.c:
2405 * gst/rtsp-server/rtsp-stream.c:
2406 * gst/rtsp-server/rtsp-stream.h:
2407 media: implement ssrc-multiplexed retransmission support
2408 based off RFC 4588 and the server-rtpaux example in -good
2410 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
2412 * gst/rtsp-server/rtsp-client.c:
2413 * gst/rtsp-server/rtsp-stream-transport.c:
2414 * gst/rtsp-server/rtsp-stream.c:
2415 rtsp: Ref transports in hash table.
2416 Also ref streams for transports.
2417 This solves a crash when reciving a rtcp after teardown but before
2419 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
2421 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
2424 Automatic update of common submodule
2425 From 7bb2bce to ef1ffdc
2427 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
2429 * gst/rtsp-server/rtsp-client.c:
2430 client: refactor cleanup of cached media
2432 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
2434 * tests/check/gst/client.c:
2436 The session leak is now fixed, lets remove those FIXME comments.
2438 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
2440 * tests/check/gst/rtspserver.c:
2441 tests: Test to setup two sessions on one connection
2442 https://bugzilla.gnome.org/show_bug.cgi?id=739112
2444 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
2446 * tests/check/gst/rtspserver.c:
2447 tests: Test setup with tcp transport
2448 https://bugzilla.gnome.org/show_bug.cgi?id=739112
2450 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
2452 * gst/rtsp-server/rtsp-client.c:
2453 client: Configure transport after creating session media
2454 The default implementation of configure_client_transport() in
2455 rtsp-client uses the session media when it chooses channels for
2456 interleaved traffic.
2457 https://bugzilla.gnome.org/show_bug.cgi?id=739112
2459 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
2461 * gst/rtsp-server/rtsp-client.c:
2462 * gst/rtsp-server/rtsp-session-media.c:
2463 client: Stop caching media in client when doing setup
2464 If the media has been managed by a session media, it should not be
2465 cached in the client any longer. The GstRTSPSessionMedia object is now
2466 responsible for unpreparing the GstRTSPMedia object using
2467 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
2469 https://bugzilla.gnome.org/show_bug.cgi?id=739112
2471 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2473 * gst/rtsp-server/rtsp-stream.c:
2474 rtsp-stream: unref srtp decoder when leaving bin
2475 https://bugzilla.gnome.org/show_bug.cgi?id=739481
2477 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2479 * gst/rtsp-server/rtsp-client.c:
2480 rtsp-client: mikey memory leaks
2481 https://bugzilla.gnome.org/show_bug.cgi?id=739383
2483 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
2486 Automatic update of common submodule
2487 From 84d06cd to 7bb2bce
2489 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
2492 Parallelise 'make check-valgrind'
2494 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
2497 Automatic update of common submodule
2498 From a8c8939 to 84d06cd
2500 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
2503 Automatic update of common submodule
2504 From 36388a1 to a8c8939
2506 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
2508 * gst/rtsp-server/rtsp-media.c:
2509 rtsp-media: deactivate media when shutting down from paused
2510 This was only done when going directly from playing.
2511 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
2513 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2515 * gst/rtsp-server/rtsp-client.c:
2516 * gst/rtsp-server/rtsp-context.h:
2517 rtsp-client: add stream transport to context
2518 We add the stream transport to the context so we can get the configured
2519 client stream transport in the setup request signal.
2520 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
2522 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2524 * gst/rtsp-server/rtsp-stream.c:
2525 stream: release lock even not all transports have been removed
2526 We don't want to keep the lock even we return FALSE because not all the
2527 transports have been removed. This could lead into a deadlock.
2528 https://bugzilla.gnome.org/show_bug.cgi?id=737797
2530 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
2532 * gst/rtsp-server/rtsp-sdp.c:
2533 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
2534 These were renamed in GstRTPBasePayload in 1.0
2536 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2538 * gst/rtsp-server/rtsp-client.c:
2539 client: set session media to NULL without the lock
2540 We need to set session medias to NULL without the client lock otherwise
2541 we can end up in a deadlock if another thread is waiting for the lock
2542 and media unprepare is also waiting for that thread to end.
2543 https://bugzilla.gnome.org/show_bug.cgi?id=737690
2545 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
2547 * gst/rtsp-server/rtsp-media.c:
2548 rtsp-media: Set state to UNPREPARING in all cases
2550 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
2552 * gst/rtsp-server/rtsp-media.c:
2553 media: set state to unpreparing when unprepare is initiated
2554 https://bugzilla.gnome.org/show_bug.cgi?id=737675
2556 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
2558 * gst/rtsp-server/rtsp-client.c:
2559 rtsp-client: Remove backlog limit while processings requests
2560 If the backlog limit is kept two cases of deadlocks may be
2561 encountered when streaming over TCP. Without the backlog
2562 limit this deadlocks can not happen, at the expence of
2564 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
2566 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
2568 * gst/rtsp-server/rtsp-client.c:
2569 rtsp-client: do not free main context before rtsp watch
2570 https://bugzilla.gnome.org/show_bug.cgi?id=737110
2572 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
2574 * tests/check/gst/rtspserver.c:
2575 tests: Extend unit test timeout to accomodate for valgrind
2576 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
2578 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
2580 * gst/rtsp-server/rtsp-client.c:
2581 * gst/rtsp-server/rtsp-session.c:
2582 * gst/rtsp-server/rtsp-stream-transport.c:
2583 rtsp-*: Treat sending packets to clients as keepalive
2584 As long as gst-rtsp-server can successfully send RTP/RTCP data to
2585 clients then the client must be reading. This change makes the server
2586 timeout the connection if the client stops reading.
2587 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
2589 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
2591 * gst/rtsp-server/rtsp-client.c:
2592 rtsp-client: Allow backlog to grow while expiring session
2593 Allow the send backlog in the RTSP watch to grow to unlimited size while
2594 attempting to bring the media pipeline to NULL due to a session
2595 expiring. Without this change the appsink element cannot change state
2596 because it is blocked while rendering data in the new_sample callback.
2597 This callback will block until it has successfully put the data into the
2598 send backlog. There is a chance that the send backlog is full at this
2599 point which means that the callback may block for a long time, possibly
2600 forever. Therefore the media pipeline may also be prevented from
2601 changing state for a long time.
2602 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
2604 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
2606 * gst/rtsp-server/rtsp-client.c:
2607 rtsp-client: Make old compilers happy
2608 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
2609 Just in case that guint8 doesn't fit in a pointer. Just in case ...
2611 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
2613 * gst/rtsp-server/rtsp-client.c:
2614 client: raise the backlog limits before pausing
2615 We need to raise the backlog limits before pausing the pipeline or else
2616 the appsink might be blocking in the render method in wait_backlog() and
2617 we would deadlock waiting for paused.
2618 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
2620 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
2622 * gst/rtsp-server/rtsp-client.c:
2623 client: make define for the WATCH_BACKLOG
2624 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
2626 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
2628 * gst/rtsp-server/rtsp-client.c:
2629 client: simplify session transport handling
2630 link/unlink of the transport in a session was done to keep track of all
2631 TCP transports and to send RTP/RTCP data to the streams. We can simplify
2632 that by putting all the TCP transports in a hashtable indexed with the
2634 We also don't need to link/unlink the transports when we pause/resume
2635 the streams. The same effect is already achieved when we pause/play the
2636 media. Indeed, when we pause the media, the transport is removed from
2637 the media and the callbacks will not be called anymore.
2638 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
2640 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
2642 * gst/rtsp-server/rtsp-stream-transport.c:
2643 * gst/rtsp-server/rtsp-stream-transport.h:
2644 stream-transport: make method to handle received data
2645 Make a method to handle the data received on a channel. It sends the
2646 data to the stream of the transport on the RTP or RTCP pads based on
2649 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
2651 * examples/test-mp4.c:
2652 test: add example of dumping RTCP reports
2654 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
2656 * gst/rtsp-server/rtsp-media.c:
2657 * gst/rtsp-server/rtsp-stream.c:
2658 * gst/rtsp-server/rtsp-stream.h:
2659 rtsp-media: Make sure that sequence numbers are monotonic after pause
2660 The sequence number is not monotonic for RTP packets after pause. The
2661 reason is basepayloader generates a randon sequence number when the
2662 pipeline goes from ready to pause. With this fix generation of sequence
2663 number will be monotonic when going from pause to play request.
2664 https://bugzilla.gnome.org/show_bug.cgi?id=736017
2666 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
2668 * gst/rtsp-server/rtsp-client.c:
2669 rtsp-client: Protect saved clients watch with a mutex
2670 Fixes a crash when close() is called while merging clients
2671 in handle_tunnel(). In that case close() would destroy the
2672 watch while it is still being used in handle_tunnel().
2673 https://bugzilla.gnome.org/show_bug.cgi?id=735570
2675 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
2677 * gst/rtsp-server/rtsp-stream.c:
2678 rtsp-stream: Remove the multicast group udp sources when removing from the bin
2680 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
2682 * gst/rtsp-server/rtsp-media.c:
2683 * gst/rtsp-server/rtsp-stream.c:
2684 * gst/rtsp-server/rtsp-stream.h:
2685 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
2686 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
2687 seeking and will always continue counting the time. This leads to
2688 the NPT after a backwards seek to be something completely different
2689 to the actual seek position.
2690 https://bugzilla.gnome.org/show_bug.cgi?id=732644
2692 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
2694 * examples/test-appsrc.c:
2695 examples: fix another reference leak
2696 gst_rtsp_media_get_element() returns a new ref.
2698 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2700 * examples/test-appsrc.c:
2701 examples: unref element after usage
2702 gst_bin_get_by_name_recurse_up() returns an element
2703 reference that must be unreffed after usage.
2704 https://bugzilla.gnome.org/show_bug.cgi?id=734546
2706 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
2708 * gst/rtsp-server/rtsp-media.c:
2709 signals: Fix copy-pasto in target-state signal offset
2711 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
2715 Makefile: Add usage of build-checks step
2716 Allows building checks without running them
2718 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
2720 * gst/rtsp-server/rtsp-stream.c:
2721 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
2722 When a UDP multicast transport is used it is expected that the server listens
2723 for RTP and RTCP packets on the multicast group with the corresponding port.
2724 Without this we will never get RTCP packets from clients in multicast mode.
2725 https://bugzilla.gnome.org/show_bug.cgi?id=732238
2727 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
2732 === release 1.4.0 ===
2734 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
2740 * gst-rtsp-server.doap:
2743 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
2745 * gst/rtsp-server/rtsp-media.h:
2746 media: correct misspelled words in description
2747 https://bugzilla.gnome.org/show_bug.cgi?id=733244
2749 === release 1.3.91 ===
2751 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
2757 * gst-rtsp-server.doap:
2760 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
2762 * docs/libs/gst-rtsp-server-sections.txt:
2765 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
2767 * gst/rtsp-server/rtsp-server.c:
2768 server: implement client REMOVE filter
2770 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
2772 * gst/rtsp-server/rtsp-client.c:
2773 * gst/rtsp-server/rtsp-client.h:
2774 client: expose _close() method
2775 Expose a previously internal close method to close the client
2778 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
2780 * gst/rtsp-server/rtsp-session-pool.c:
2781 session-pool: signal session-removed outside of the lock
2782 Release the lock before emiting the session-removed signal.
2784 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
2786 * gst/rtsp-server/rtsp-client.c:
2787 * gst/rtsp-server/rtsp-server.c:
2788 * gst/rtsp-server/rtsp-session-pool.c:
2789 * gst/rtsp-server/rtsp-session.c:
2790 * gst/rtsp-server/rtsp-stream.c:
2791 filter: Release lock in filter functions
2792 Release the object lock before calling the filter functions. We need to
2793 keep a cookie to detect when the list changed during the filter
2794 callback. We also keep a hashtable to make sure we only call the filter
2795 function once for each object in case of concurrent modification.
2796 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
2798 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
2800 * gst/rtsp-server/rtsp-client.c:
2801 client: check if watch is set in handle_teardown()
2802 The unit tests run without a watch
2804 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
2806 * tests/check/gst/client.c:
2807 client tests: send teardown to cleanup session
2809 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
2811 * tests/check/gst/rtspserver.c:
2812 server tests: send teardown to cleanup session
2814 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
2816 * gst/rtsp-server/rtsp-client.c:
2817 client: keep ref to client for the session removed handler
2818 This extra ref will be dropped when all client sessions have been
2819 removed. A session is removed when a client sends teardown, closes its
2820 endpoint of the TCP connection or the sessions expires.
2821 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
2823 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
2825 * gst/rtsp-server/rtsp-client.c:
2826 * gst/rtsp-server/rtsp-session.c:
2827 * tests/check/gst/client.c:
2828 client: manage media in session as a last step
2829 Once we manage a media in a session, we can't unmanage it anymore
2830 without destroying it. Therefore, first check everything before we
2831 manage the media, otherwise if something is wrong we have no way to
2833 If we created a new session and something went wrong, remove the session
2834 again. Fixes a leak in the unit test.
2836 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
2838 * examples/test-mp4.c:
2839 * examples/test-ogg.c:
2840 examples: print 'stream ready at url' for mp4 and ogg example
2842 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
2844 * gst/rtsp-server/rtsp-client.c:
2845 * gst/rtsp-server/rtsp-sdp.c:
2846 rtsp: fix for MIKEY api change
2848 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
2850 * gst/rtsp-server/rtsp-client.c:
2851 client: free watch context only once
2852 The watch context is freed when the source is destroyed. Avoids
2853 a CRITICAL when we try to unref the context twice.
2855 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
2857 * gst/rtsp-server/rtsp-client.c:
2860 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
2862 * gst/rtsp-server/rtsp-client.c:
2863 client: protect sessions with lock
2864 Protect the list of sessions with the lock.
2865 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
2867 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
2869 * gst/rtsp-server/rtsp-client.c:
2870 Client: keep a ref to the session
2871 Don't just keep a weak ref to the session objects but use a hard ref. We
2872 will be notified when a session is removed from the pool (expired) with
2873 the new session-removed signal.
2874 Don't automatically close the RTSP connection when all the sessions of
2875 a client are removed, a client can continue to operate and it can create
2876 a new session if it wants. If you want to remove the client from the
2877 server, you have to use gst_rtsp_server_client_filter() now.
2878 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
2879 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
2881 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
2883 * gst/rtsp-server/rtsp-session-pool.c:
2884 * gst/rtsp-server/rtsp-session-pool.h:
2885 session-pool: add session-removed signal
2886 Add a signal to be notified when a session is removed from the pool.
2888 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
2890 * gst/rtsp-server/Makefile.am:
2891 * gst/rtsp-server/rtsp-server.h:
2892 Make rtsp-server.h a single-include header, use it for G-I
2893 https://bugzilla.gnome.org/show_bug.cgi?id=732411
2895 === release 1.3.90 ===
2897 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
2903 * gst-rtsp-server.doap:
2906 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
2908 * gst/rtsp-server/rtsp-stream.c:
2909 stream: crypto can be NULL
2911 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
2913 * gst/rtsp-server/rtsp-client.c:
2914 * gst/rtsp-server/rtsp-media.c:
2915 * gst/rtsp-server/rtsp-mount-points.c:
2916 introspection: add missing allow-none annotations
2917 https://bugzilla.gnome.org/show_bug.cgi?id=730952
2919 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
2921 * gst/rtsp-server/rtsp-address-pool.c:
2922 * gst/rtsp-server/rtsp-media.c:
2923 * gst/rtsp-server/rtsp-session-media.c:
2924 * gst/rtsp-server/rtsp-session-pool.c:
2925 * gst/rtsp-server/rtsp-stream-transport.c:
2926 * gst/rtsp-server/rtsp-stream.c:
2927 * gst/rtsp-server/rtsp-token.c:
2928 introspection: add (nullable) annotations to return values
2929 https://bugzilla.gnome.org/show_bug.cgi?id=730952
2931 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
2933 * gst/rtsp-server/rtsp-client.c:
2934 * gst/rtsp-server/rtsp-stream.c:
2935 gi: improve annotations
2936 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
2938 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
2940 * gst/rtsp-server/rtsp-client.c:
2941 * gst/rtsp-server/rtsp-media-factory.c:
2942 * gst/rtsp-server/rtsp-media.c:
2943 * gst/rtsp-server/rtsp-server.c:
2944 signals: use generic marshal function
2945 Use the generic C marshal function.
2946 Use more explicit type instead of G_TYPE_POINTER
2948 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
2950 * gst/rtsp-server/rtsp-context.h:
2951 context: add type macro
2953 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
2955 * gst/rtsp-server/rtsp-client.c:
2956 * gst/rtsp-server/rtsp-sdp.c:
2957 * gst/rtsp-server/rtsp-sdp.h:
2958 sdp: hide key length defines
2959 They don't have a namespace.
2961 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
2966 === release 1.3.3 ===
2968 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
2974 * gst-rtsp-server.doap:
2977 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2979 * gst/rtsp-server/rtsp-client.c:
2980 * gst/rtsp-server/rtsp-sdp.c:
2981 * gst/rtsp-server/rtsp-sdp.h:
2982 mikey: add different key length parameters
2983 Add encryption and authentication key length parameters to MIKEY. For
2984 the encoders, the key lengths are obtained from the cipher and auth
2985 algorithms set in the caps. For the decoders, they are obtained while
2986 parsing the key management from the client.
2987 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
2989 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
2991 * tests/check/gst/stream.c:
2992 stream tests: Make sure we get right multicast address from stream
2993 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
2995 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
2997 * gst/rtsp-server/rtsp-client.c:
2998 client: ref the context until rtsp watch is alive
2999 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
3001 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
3003 * gst/rtsp-server/rtsp-client.c:
3004 client: Destroy the rtsp watch after connection close
3006 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
3008 * gst/rtsp-server/rtsp-media.c:
3009 media: fix confusing comment
3011 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
3013 * gst/rtsp-server/rtsp-session.c:
3014 rtsp-session: Timeout in header.
3015 Adding the possbilty to always have timout in header.
3016 This is configurabe with setting "timeout-always-visible".
3017 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
3019 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
3024 === release 1.3.2 ===
3026 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
3033 * gst-rtsp-server.doap:
3036 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
3039 Automatic update of common submodule
3040 From 211fa5f to 1f5d3c3
3042 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
3044 * gst/rtsp-server/rtsp-client.c:
3045 client: store TCP ports in transport
3046 Store the TCP ports in the transport when we are doing RTSP over TCP.
3047 This way, we can easily get to the ports from the transport.
3048 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
3050 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
3052 * gst/rtsp-server/rtsp-stream.c:
3053 stream: add signals for new RTP/RTCP encoders
3054 New signals to allow the user to configure the dynamically created
3056 https://bugzilla.gnome.org/show_bug.cgi?id=730228
3058 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
3060 * gst/rtsp-server/rtsp-media.c:
3061 * gst/rtsp-server/rtsp-media.h:
3062 media: Make suspend()/unsuspend() virtual
3063 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
3065 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
3067 * gst/rtsp-server/rtsp-client.c:
3068 client: fix send-message signal marshaller
3069 Use generic marshalling for the send-message signal. It has
3070 two POINTER arguments, not just one.
3071 https://bugzilla.gnome.org/show_bug.cgi?id=729900
3073 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
3075 * tests/check/gst/media.c:
3076 tests: add and remove pads only once
3077 In this test we simulate a dynamic pad by watching the caps event.
3078 Because of renegotiation in the base payloader now, this caps is sent
3079 multiple times but we can only deal with 1 invocation, use a variable to
3080 only 'add and remove' the pad once.
3082 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
3084 * tests/check/gst/rtspserver.c:
3085 tests: add unit test for correct handling of Require headers
3086 https://bugzilla.gnome.org/show_bug.cgi?id=729426
3088 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
3090 * gst/rtsp-server/rtsp-client.c:
3091 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
3092 Servers must handle Require headers and must report a failure
3093 if they don't handle any of the Required options, see RFC 2326,
3094 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
3095 https://bugzilla.gnome.org/show_bug.cgi?id=729426
3097 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
3102 === release 1.3.1 ===
3104 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
3110 * gst-rtsp-server.doap:
3113 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
3116 Automatic update of common submodule
3117 From bcb1518 to 211fa5f
3119 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
3124 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
3126 * tests/check/gst/sessionmedia.c:
3127 tests: fix memory leak in sessionmedia unit test
3129 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
3131 * gst/rtsp-server/rtsp-client.c:
3132 client: emit a signal before sending a message
3133 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
3135 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
3137 * gst/rtsp-server/rtsp-client.c:
3138 client: pass context to send_message
3139 Pass the current context to send_message, we will need it later.
3141 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
3143 * gst/rtsp-server/rtsp-client.c:
3144 client: fix typo in comment
3146 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
3148 * gst/rtsp-server/rtsp-media.c:
3149 media: Do not stop thread twice if default_prepare() fails
3151 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
3153 * gst/rtsp-server/rtsp-client.c:
3154 client: set the watch to flushing before going to NULL
3155 First set the watch to flushing so that we unblock any current and
3156 future attempt to send data on the watch, Then set the pipeline to
3158 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
3160 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
3162 * gst/rtsp-server/rtsp-session-pool.c:
3163 * tests/check/gst/sessionpool.c:
3164 rtsp-session-pool: Fixes annotation
3165 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
3166 in the sessionpool test.
3167 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
3169 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
3171 * gst/rtsp-server/rtsp-media.c:
3172 * gst/rtsp-server/rtsp-media.h:
3173 media: make media_prepare virtual
3174 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
3176 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
3178 * gst/rtsp-server/rtsp-media.c:
3179 * tests/check/gst/media.c:
3180 media: stop the thread in more error cases
3182 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
3184 * gst/rtsp-server/rtsp-media.c:
3185 * tests/check/gst/media.c:
3186 media: allow NULL as the thread
3187 Use the default context whan passing a NULL thread.
3189 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
3191 * gst/rtsp-server/rtsp-client.c:
3192 rtsp-client: indent cleanup
3193 Coverity was moaning about unreachable code, and I think it was just
3194 confused by { being before the label. We'll see if it pops up again.
3197 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
3199 * gst/rtsp-server/rtsp-client.c:
3200 * gst/rtsp-server/rtsp-media.c:
3201 client: Add drop-backlog property
3202 When we have too many messages queued for a client (currently hardcoded
3203 to 100) we overflow and drop the messages. Add a drop-backlog property
3204 to control this behaviour. Setting this property to FALSE will retry
3205 to send the messages to the client by waiting for more room in the
3207 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
3209 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
3211 * gst/rtsp-server/rtsp-client.c:
3212 client: support for POST before GET when setting up a tunnel
3214 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
3216 * gst/rtsp-server/rtsp-client.c:
3217 client: remove watch of the second client after http tunnel setup
3218 The second client will be freed after the HTTP tunnel has been set up.
3219 Make sure it's RTSP watch is never dispatched again.
3220 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
3222 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
3224 * gst/rtsp-server/rtsp-media.c:
3225 * tests/check/gst/media.c:
3226 media: Make media_prepare() fail if port allocation fails
3227 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
3229 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
3231 * tests/check/gst/media.c:
3232 media test: cleanup the thread pool in tests
3234 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
3236 * gst/rtsp-server/rtsp-media.c:
3237 * tests/check/gst/media.c:
3238 rtsp-media: Unblock blocked streams in unprepare
3239 The streams will be blocked when a live media is prepared.
3240 The streams should be unblocked in gst_rtsp_media_unprepare.
3241 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
3243 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
3245 * gst/rtsp-server/rtsp-media.c:
3246 media: release the state lock when going to NULL
3247 Set our state to UNPREPARING and release the state-lock before
3248 setting the pipeline to the NULL state. This way, any pad-added
3249 callback will be able to take the state-lock and check that we are now
3250 unpreparing instead of deadlocking.
3251 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
3253 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
3255 * gst/rtsp-server/rtsp-media.c:
3256 media: protect status with lock
3257 Make sure we only update the status with the lock.
3259 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
3261 * gst/rtsp-server/rtsp-client.c:
3262 * gst/rtsp-server/rtsp-sdp.c:
3263 rtsp: update for MIKEY API changes
3265 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
3267 * gst/rtsp-server/rtsp-client.c:
3268 client: parse the mikey response from the client
3269 Parse the mikey response from the client and update the policy for
3272 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
3274 * gst/rtsp-server/rtsp-stream.c:
3275 * gst/rtsp-server/rtsp-stream.h:
3276 stream: add method to set crypto info
3277 Make a method to configure the crypto information of a stream.
3278 Set udpsrc in READY instead of PAUSED so that we can configure caps
3281 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
3283 * gst/rtsp-server/rtsp-client.c:
3284 client: cleanup error paths
3286 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
3288 * gst/rtsp-server/rtsp-media.c:
3291 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
3293 * examples/test-video.c:
3294 test: enable SRTP only on RTSPS
3295 We only want to enable SRTP when doing rtsp over TLS so that we can
3296 exchange the keys in a secure way.
3298 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
3300 * examples/test-video.c:
3301 test: print an error on failure
3303 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
3306 * examples/test-video.c:
3307 * gst/rtsp-server/rtsp-sdp.c:
3308 * gst/rtsp-server/rtsp-stream.c:
3309 * tests/check/Makefile.am:
3310 stream: add SRTP support
3311 Install srtp encoder and decoder elements in rtpbin
3314 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3316 * tests/check/Makefile.am:
3317 * tests/check/gst/sessionpool.c:
3318 tests: Add unit tests for sessionpool
3319 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
3321 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3323 * tests/check/gst/threadpool.c:
3324 tests: Improve code coverage of rtsp-threadpool tests
3325 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
3327 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3329 * tests/check/gst/sessionmedia.c:
3330 tests: Improve code coverage for rtsp-session-media
3331 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
3333 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3335 gobject-introspection: Add annotations to support language bindings
3336 In addition a few cosmetic changes:
3337 * Adjust the order of arguments
3338 * Fix typo: occured -> occurred
3339 * Fix indentation after Return:-clauses
3340 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
3342 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3344 * gst/rtsp-server/rtsp-stream.c:
3345 rtsp-stream: Don't mix IPv4 and IPv6 addresses
3346 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
3348 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
3350 * gst/rtsp-server/rtsp-stream.c:
3351 stream: take caps after the session manager
3352 Take the caps for the SDP after they leave the rtpbin so that we can
3353 also get the properties added by rtpbin elements.
3355 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
3357 * gst/rtsp-server/rtsp-stream.c:
3358 stream: release lock while pushing out packets
3359 Keep a cache of the transports and use this to iterate the transport
3360 while pushing packets. This allows us to release the lock early.
3361 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
3363 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
3365 * gst/rtsp-server/rtsp-client.c:
3366 * gst/rtsp-server/rtsp-client.h:
3367 rtsp-client: vmethod for modifying tunnel GET response
3368 Add a vmethod tunnel_http_response where the response to the HTTP GET
3369 for tunneled connections can be modified.
3370 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
3372 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
3374 * gst/rtsp-server/rtsp-sdp.c:
3375 sdp: make 1 media line per profile
3376 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
3377 line in the SDP for each profile. The client is then supposed to pick
3378 one of the profiles in the SETUP request. Because the m= lines have the
3379 same pt, the client also knows that only 1 option is possible.
3381 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
3383 * gst/rtsp-server/rtsp-media-factory.c:
3384 * gst/rtsp-server/rtsp-media-factory.h:
3385 * gst/rtsp-server/rtsp-media.c:
3386 factory: add profile property and pass to media and streams
3388 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
3390 * examples/test-multicast.c:
3391 * gst/rtsp-server/rtsp-sdp.c:
3392 sdp: pass multicast connection for multicast-only stream
3393 Pass the multicast address of the stream in the connection info in the
3394 SDP so that clients try a multicast connection first.
3395 Only allow multicast connections in the test-multicast example. Also
3396 increase the TTL a little.
3398 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3401 .gitignore: Ignore gcov intermediate files
3402 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
3404 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
3406 * gst/rtsp-server/rtsp-stream.c:
3407 stream: release some locks in error cases
3409 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3411 docs: Enable and fix gtk-doc warnings
3412 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
3413 * addresspool/mediafactory: Add missing annotation colon
3414 * stream: Annotate return value
3415 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
3417 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
3420 Automatic update of common submodule
3421 From fe1672e to bcb1518
3423 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
3426 Automatic update of common submodule
3427 From 1a07da9 to fe1672e
3429 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
3431 * examples/Makefile.am:
3432 examples: use LDADD for libs instead of LDFLAGS
3434 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
3437 configure: make sure releases are in .doap file
3439 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
3441 * examples/test-cgroups.c:
3442 examples: test-cgroups: don't put code with side effects into g_assert()
3443 The g_assert() might get compiled out with the right
3444 compiler/preprocessor flags.
3446 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
3448 * examples/.gitignore:
3449 examples: add cgroup test binary to .gitignore
3451 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
3453 * examples/test-cgroups.c:
3454 examples: fix cgroup test build
3455 Fixes build failure caused by compiler warning:
3456 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
3458 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
3461 .gitignore: ignore temp files created in the course of 'make check'
3463 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
3465 * gst/rtsp-server/rtsp-media.c:
3466 rtsp-media: don't loose frames handling new PLAY request
3467 If client supplied a range check if the range specifies the start point.
3468 If not, then do an accurate seek to the current position. If a start
3469 point was specified do do a key unit seek to make sure the streaming
3470 starts with decodeable frames.
3471 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
3473 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
3475 * gst/rtsp-server/rtsp-media.c:
3476 Revert "media: only flush when setting a new start position"
3477 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
3478 We need to do the flush in all cases, demuxer block currently for
3481 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
3483 * gst/rtsp-server/rtsp-media.c:
3484 media: only flush when setting a new start position
3485 Only flush the pipeline when we change the start position with
3487 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
3489 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
3491 * gst/rtsp-server/rtsp-stream.c:
3492 stream: set ttl-mc before adding the socket
3493 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
3494 never be set on socket.
3495 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
3497 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
3499 * gst/rtsp-server/rtsp-media.c:
3500 media: stop thread if media is already prepared
3501 in gst_rtsp_media_prepare() the thread is not used if media is already
3502 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
3504 https://bugzilla.gnome.org/show_bug.cgi?id=724182
3506 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
3509 build: Ship gst-rtsp-server.doap file
3511 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
3513 * tests/check/gst/rtspserver.c:
3514 tests: Fix another compiler warning with gcc
3516 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
3518 * gst/rtsp-server/rtsp-client.c:
3519 * gst/rtsp-server/rtsp-mount-points.c:
3520 * gst/rtsp-server/rtsp-stream.c:
3521 * tests/check/gst/client.c:
3522 rtsp-server: Fix lots of compiler warnings with clang
3524 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
3527 * gst-rtsp-server.doap:
3528 * tests/Makefile.am:
3529 configure: Synchronise with the configure scripts of the other modules
3531 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
3534 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
3536 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
3538 * gst/rtsp-server/rtsp-media.c:
3539 * gst/rtsp-server/rtsp-stream.c:
3540 Revert "rtsp-server: support build against last stable release"
3541 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
3542 Let us require 1.2.3 now, which is going to be released in a few
3545 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
3547 * gst/rtsp-server/rtsp-session-media.c:
3548 * gst/rtsp-server/rtsp-stream-transport.c:
3549 session: improve RTP-Info
3550 Ignore streams that can't generate RTP-Info instead of failing.
3551 Don't return the empty string when all streams are unconfigured but
3552 return NULL so that we don't generate and empty RTP-Info header.
3553 Improve docs a little.
3555 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
3557 * gst/rtsp-server/rtsp-session-media.c:
3558 Don't free rtpinfo GString when it is NULL
3559 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
3561 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
3563 * gst/rtsp-server/rtsp-media.c:
3564 media: only set keyframe flag when modifying start
3565 Only set the keyframe flag when we modify the start position. The
3566 keyframe flag should probably be ignored when no change is requested but
3567 until we can claim this is all documented properly and all demuxer
3568 implement this, avoid setting the flag.
3569 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
3571 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
3573 * gst/rtsp-server/rtsp-thread-pool.c:
3574 thread-pool: Unref source after mainloop has quit to avoid races in GLib
3575 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
3577 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
3579 * gst/rtsp-server/rtsp-stream.c:
3580 stream: handle NULL seqnum and rtptime arguments
3582 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
3584 * gst/rtsp-server/rtsp-thread-pool.c:
3585 * tests/check/gst/threadpool.c:
3586 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
3587 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
3589 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
3591 * gst/rtsp-server/rtsp-stream.c:
3592 stream: add fallback for missing stats property
3593 Use a fallback when the payloader does not have a stats property
3594 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
3596 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
3599 Automatic update of common submodule
3600 From f7bc1c3 to 1a07da9
3602 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
3604 * gst/rtsp-server/rtsp-stream.c:
3605 stream: don't leak stats structure
3606 Don't leak the stats structure and deal with NULL stats.
3608 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
3610 * gst/rtsp-server/rtsp-stream.c:
3611 stream: Get rtpinfo properties atomically from payloader
3612 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
3614 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
3616 * gst/rtsp-server/rtsp-media.c:
3617 media: refactor state change functions and signals
3618 Make functions to set the target state and the pipeline state and emit
3619 the signals from those functions.
3621 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
3623 * gst/rtsp-server/rtsp-media.c:
3624 * gst/rtsp-server/rtsp-media.h:
3625 media: add signal to notify of pending state changes
3627 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
3629 * gst/rtsp-server/rtsp-media.c:
3630 * gst/rtsp-server/rtsp-stream.c:
3631 rtsp-server: support build against last stable release
3632 Until 1.2.3 is out with the new get_type function and we
3635 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
3637 * gst/rtsp-server/rtsp-stream.c:
3638 stream: fix compilation
3640 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
3642 * gst/rtsp-server/rtsp-media.c:
3643 * gst/rtsp-server/rtsp-media.h:
3644 * gst/rtsp-server/rtsp-stream.c:
3645 * gst/rtsp-server/rtsp-stream.h:
3646 stream: add property to configure profiles
3648 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
3650 * gst/rtsp-server/rtsp-client.c:
3651 client: let stream check supported transport
3652 Delegate the check if a transport is allowed to the stream.
3653 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
3655 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
3657 * gst/rtsp-server/rtsp-stream.c:
3658 * gst/rtsp-server/rtsp-stream.h:
3659 stream: add method to check supported transport
3660 Add a method to check if a transport is supported
3662 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
3665 configure.ac: Only check for gstreamer-check, not check
3666 We include check in gstreamer-check since quite some time now.
3668 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
3670 * gst/rtsp-server/rtsp-session-media.c:
3671 * gst/rtsp-server/rtsp-stream-transport.c:
3672 * gst/rtsp-server/rtsp-stream.c:
3673 * gst/rtsp-server/rtsp-stream.h:
3674 stream: return clock-rate from get_rtpinfo
3675 And use it to correct the rtptime to the requested start-time.
3676 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
3678 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
3680 * gst/rtsp-server/rtsp-session-media.c:
3681 * gst/rtsp-server/rtsp-stream-transport.c:
3682 * gst/rtsp-server/rtsp-stream-transport.h:
3683 session-media: calculate start-time
3685 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
3687 * gst/rtsp-server/rtsp-stream-transport.c:
3688 * gst/rtsp-server/rtsp-stream.c:
3689 * gst/rtsp-server/rtsp-stream.h:
3690 stream: also return the running-time
3691 Return the running-time in the rtpinfo as well.
3693 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
3695 * gst/rtsp-server/rtsp-client.c:
3696 * gst/rtsp-server/rtsp-session-media.c:
3697 * gst/rtsp-server/rtsp-session-media.h:
3698 * gst/rtsp-server/rtsp-stream-transport.c:
3699 * gst/rtsp-server/rtsp-stream-transport.h:
3700 session-media: let the session-media make the RTPInfo
3701 Add method to create the RTPInfo for a stream-transport.
3702 Add method to create the RTPInfo for all stream-transports in a
3704 Use the session-media RTPInfo code in client. This allows us to refactor
3705 another method to link the TCP callbacks.
3707 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
3709 mount-points: sort sequence before g_sequence_lookup
3710 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
3711 sort sequence if dirty, otherwise lookup will fail.
3712 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
3714 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
3717 configure: rename package from gst-rtsp to gst-rtsp-server
3718 To match git module name and avoid confusion with the
3719 rtsp lib in gst-plugins-base and rtsp plugin in -good.
3721 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
3724 configure: bump core/base/good requirement to 1.2.0
3725 Bump to released stable version and make implicit
3726 requirements explicit.
3728 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
3733 Fix broken gettext setup which is not used anyway
3735 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
3738 Automatic update of common submodule
3739 From dbedaa0 to d48bed3
3741 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
3743 * gst/rtsp-server/rtsp-client.c:
3744 * gst/rtsp-server/rtsp-media.c:
3745 * gst/rtsp-server/rtsp-media.h:
3746 media: add setup_sdp vmethod
3747 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
3748 gst_rtsp_media_setup_sdp.
3749 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
3751 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
3753 * gst/rtsp-server/rtsp-stream.c:
3754 rtsp-stream: Check return value of sscanf
3755 streamid is only valid if sscanf matched something.
3757 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
3759 * gst/rtsp-server/rtsp-client.c:
3760 rtsp-client: Fix iteration
3761 Wouldn't even enter the code block otherwise (i++ was used as the check
3762 and not the postfix).
3764 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
3766 * gst/rtsp-server/rtsp-client.c:
3767 * gst/rtsp-server/rtsp-client.h:
3768 client: add vmethod to configure media and streams
3769 Implement a vmethod that can be used to configure the media and the
3770 streams based on the current context. Handle the blocksize handling in
3771 the default handler.
3772 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
3774 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
3777 Make git ignore more unit test binaries
3779 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
3781 * gst/rtsp-server/rtsp-address-pool.h:
3782 * gst/rtsp-server/rtsp-auth.h:
3783 * gst/rtsp-server/rtsp-client.h:
3784 * gst/rtsp-server/rtsp-context.h:
3785 * gst/rtsp-server/rtsp-media-factory-uri.h:
3786 * gst/rtsp-server/rtsp-media-factory.h:
3787 * gst/rtsp-server/rtsp-media.h:
3788 * gst/rtsp-server/rtsp-mount-points.h:
3789 * gst/rtsp-server/rtsp-server.h:
3790 * gst/rtsp-server/rtsp-session-media.h:
3791 * gst/rtsp-server/rtsp-session-pool.h:
3792 * gst/rtsp-server/rtsp-session.h:
3793 * gst/rtsp-server/rtsp-stream-transport.h:
3794 * gst/rtsp-server/rtsp-stream.h:
3795 * gst/rtsp-server/rtsp-thread-pool.h:
3796 * gst/rtsp-server/rtsp-token.h:
3797 rtsp-server: add padding to many public structures
3798 Not mini objects though, since they are not subclassable
3799 anyway, nor kept on the stack or inlined in a structure.
3801 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
3803 media: add new create_rtpbin vmethod
3804 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
3805 https://bugzilla.gnome.org/show_bug.cgi?id=719734
3807 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
3809 * tests/check/gst/media.c:
3810 tests: fix memory leak, free test's thread pool
3811 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
3813 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
3815 * gst/rtsp-server/rtsp-stream-transport.c:
3816 stream-transport: free url in finalize
3818 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
3820 * gst/rtsp-server/rtsp-media.c:
3821 media: also do state change in suspended state
3823 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
3825 * gst/rtsp-server/rtsp-client.c:
3826 * gst/rtsp-server/rtsp-media.c:
3827 media: also handle prepare and range in suspended state
3828 When we are suspended, we are already prepared.
3829 We can get the range in the suspended state.
3831 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
3833 * tests/check/Makefile.am:
3834 * tests/check/gst/sessionmedia.c:
3835 check: add test for uri in setup
3836 Added unit tests for the new functionality in GstRTSPStreamTransport.
3837 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
3839 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
3841 * gst/rtsp-server/rtsp-client.c:
3842 client: store setup uri and use in PLAY response
3843 Store the uri used when doing the setup and use that in the PLAY
3845 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
3847 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
3849 * gst/rtsp-server/rtsp-stream-transport.c:
3850 * gst/rtsp-server/rtsp-stream-transport.h:
3851 stream-transport: add method to get/set url
3853 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
3855 * gst/rtsp-server/rtsp-client.c:
3856 client: suspend after SDP and unsuspend before PLAYING
3857 Based on patches by Ognyan Tonchev <ognyan@axis.com>
3858 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
3860 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
3862 * gst/rtsp-server/rtsp-media-factory.c:
3863 * gst/rtsp-server/rtsp-media-factory.h:
3864 * gst/rtsp-server/rtsp-media.c:
3865 * gst/rtsp-server/rtsp-media.h:
3866 * gst/rtsp-server/rtsp-session-media.c:
3867 * gst/rtsp-server/rtsp-session.c:
3868 * tests/check/gst/media.c:
3869 * tests/check/gst/mediafactory.c:
3870 media: add suspend modes
3871 Add support for different suspend modes. The stream is suspended right after
3872 producing the SDP and after PAUSE. Different suspend modes are available that
3873 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
3874 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
3875 state and RESET will bring the pipeline to the NULL state.
3876 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
3877 this means that the pipeline needs to be prerolled again.
3878 Base on patches by Ognyan Tonchev <ognyan@axis.com>
3879 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3881 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
3883 * gst/rtsp-server/rtsp-media.c:
3884 media: start live streams in blocked state
3885 Start live streams in the blocked state and make them preroll using the
3886 messages. This ensure that no data is played by the sink until we explicitly
3887 unblock the stream right before going to PLAYING.
3888 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3890 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
3892 * gst/rtsp-server/rtsp-media.c:
3893 media: refactor starting and waiting for preroll
3894 Based on patches from Ognyan Tonchev <ognyan@axis.com>
3895 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3897 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
3899 * gst/rtsp-server/rtsp-stream.c:
3900 * gst/rtsp-server/rtsp-stream.h:
3901 stream: add API to block streams
3902 Add an API to block on the streams and make it post a message.
3903 Based on patch by Ognyan Tonchev <ognyan@axis.com>
3904 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3906 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
3908 * docs/libs/Makefile.am:
3909 docs: Specify the override file
3910 Even if it's empty (for now) it avoids make distcheck complaining
3912 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
3914 * gst/rtsp-server/rtsp-media.c:
3915 media: move default implementations to where they are used
3917 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
3919 * gst/rtsp-server/rtsp-media.c:
3920 media: take the right lock in gst_rtsp_media_set_pipeline_state()
3921 We need to take the state_lock when calling this method.
3923 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
3925 * gst/rtsp-server/rtsp-media.c:
3926 media: handle add-added on non-bins too
3927 Handle dynamic payloaders that are not bins, as used in the unit-test.
3929 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3931 * gst/rtsp-server/rtsp-media-factory.c:
3932 * gst/rtsp-server/rtsp-media-factory.h:
3933 * gst/rtsp-server/rtsp-media.c:
3934 rtsp-media/-factory: Fix request pad name comments
3935 These must be escaped for gtk-doc to parse the comments without warnings.
3937 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
3939 rtsp-media: remove transports if media is in error status
3940 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
3941 trying to change to GST_STATE_NULL and media is in error status, we
3942 remove all transports.
3943 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
3945 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
3947 * gst/rtsp-server/rtsp-media.c:
3948 rtsp-media: use element metadata to find payloader
3949 Use the element metadata to find the payloader instead of checking
3951 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
3953 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
3955 rtsp-stream: add getter for payload type
3956 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
3957 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
3958 element and create the stream with this one instead of the dynpay%d
3960 https://bugzilla.gnome.org/show_bug.cgi?id=712396
3962 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3964 * gst/rtsp-server/rtsp-client.c:
3965 * gst/rtsp-server/rtsp-context.h:
3966 * gst/rtsp-server/rtsp-media.c:
3967 * gst/rtsp-server/rtsp-mount-points.c:
3968 * gst/rtsp-server/rtsp-server.c:
3969 * gst/rtsp-server/rtsp-token.c:
3970 rtsp-*: Refer to NULL as a constant in comments
3972 https://bugzilla.gnome.org/show_bug.cgi?id=714988
3974 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3976 rtsp-*: Fix type name typos in comments
3977 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
3978 * rtsp-auth: Refer to part of constant name as text
3979 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
3980 * rtsp-session-media: Fix GstRTSPSessionMedia typo
3981 * rtsp-stream: Fix typo when refering to GstBin
3982 https://bugzilla.gnome.org/show_bug.cgi?id=714988
3984 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3987 * docs/libs/gst-rtsp-server-docs.sgml:
3988 * docs/libs/gst-rtsp-server-sections.txt:
3989 docs: Improve documentation
3990 * Include annotation-glossary to quiet gtk-doc
3991 * Rename remaining ClientState -> Context
3992 * Rename object hierarchy file
3993 * Remove stale chapter references
3994 * Add missing function and object references
3995 * Include missing GstRTSPAddressPoolResult
3996 https://bugzilla.gnome.org/show_bug.cgi?id=714988
3998 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
4000 * gst/rtsp-server/rtsp-client.c:
4001 * gst/rtsp-server/rtsp-server.c:
4002 * gst/rtsp-server/rtsp-session-pool.c:
4003 * gst/rtsp-server/rtsp-session.c:
4004 * gst/rtsp-server/rtsp-stream.c:
4005 rtsp-server: sprinkle some allow-none annotations for g-i
4007 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
4009 * gst/rtsp-server/rtsp-stream.c:
4010 * gst/rtsp-server/rtsp-stream.h:
4011 stream: add method to filter transports
4012 Add a method to safely iterate and collect the stream transports
4013 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
4015 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
4017 * gst/rtsp-server/rtsp-client.c:
4018 * gst/rtsp-server/rtsp-server.c:
4019 * gst/rtsp-server/rtsp-session-pool.c:
4020 * gst/rtsp-server/rtsp-session.c:
4021 rtsp: allow NULL func in filters
4022 Passing a null function make the filters return a list of
4025 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
4027 * gst/rtsp-server/rtsp-address-pool.c:
4028 * tests/check/gst/addresspool.c:
4029 address-pool: fix address increment
4030 Use a guint instead of guint8 to increment the address. It's still not
4031 completely correct because a guint might not be able to hold the complete
4032 address range, but that's an enhacement for later.
4033 Add unit test to test improved behaviour.
4034 https://bugzilla.gnome.org/show_bug.cgi?id=708237
4036 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
4038 * gst/rtsp-server/rtsp-client.c:
4039 * tests/check/gst/client.c:
4040 client: allow absolute path in requests
4041 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
4043 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
4045 * gst/rtsp-server/rtsp-client.c:
4046 * gst/rtsp-server/rtsp-client.h:
4047 client: make make_path_from_uri a vmethod
4049 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
4051 * docs/libs/gst-rtsp-server-sections.txt:
4052 * gst/rtsp-server/rtsp-stream.c:
4053 * gst/rtsp-server/rtsp-stream.h:
4054 * tests/check/Makefile.am:
4055 * tests/check/gst/stream.c:
4056 stream: Add functions to get rtp and rtcp sockets
4057 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
4059 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
4061 * gst/rtsp-server/rtsp-context.c:
4062 * gst/rtsp-server/rtsp-context.h:
4063 context: defing a GType for the context
4064 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
4066 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
4068 * gst/rtsp-server/Makefile.am:
4069 * gst/rtsp-server/rtsp-auth.c:
4070 * gst/rtsp-server/rtsp-context.c:
4071 * gst/rtsp-server/rtsp-media.c:
4072 * gst/rtsp-server/rtsp-mount-points.c:
4073 * gst/rtsp-server/rtsp-server.h:
4074 * gst/rtsp-server/rtsp-session-media.c:
4075 * gst/rtsp-server/rtsp-session.c:
4076 * gst/rtsp-server/rtsp-stream.c:
4077 Fixed several GIR warnings
4079 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
4081 * gst/rtsp-server/rtsp-auth.c:
4084 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
4086 * tests/check/Makefile.am:
4087 * tests/check/gst/token.c:
4088 tests: Add unit tests for token
4089 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
4091 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
4093 * gst/rtsp-server/rtsp-token.c:
4094 token: Validate args for gst_rtsp_token_is_allowed
4095 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
4097 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
4099 * gst/rtsp-server/rtsp-token.c:
4100 token: Fix bug when creating empty token
4101 We always want to have a valid GstStructure in the token.
4102 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
4104 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
4106 * gst/rtsp-server/rtsp-thread-pool.c:
4107 thread-pool: avoid race in shutdown
4108 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
4109 don't actually stop the mainloop ever. Solve this race by adding an idle source
4110 to the mainloop that calls the _quit. This way we immediately exit the mainloop
4111 if quit was called before we started it.
4113 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
4115 * tests/check/Makefile.am:
4116 * tests/check/gst/permissions.c:
4117 tests: Add unit tests for permissions
4118 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
4120 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
4122 * tests/check/gst/mediafactory.c:
4123 tests: Test mediafactory permissions
4124 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
4126 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
4128 * gst/rtsp-server/rtsp-permissions.c:
4129 permissions: Fix refcounting when adding/removing roles
4130 Previously a role that was removed was unreffed twice, and when
4131 replacing an existing role the replaced role was freed while still being
4132 referenced. Both bugs are now fixed.
4133 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
4135 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
4137 * tests/check/gst/media.c:
4138 * tests/check/gst/mediafactory.c:
4139 * tests/check/gst/rtspserver.c:
4140 tests: Check gst_rtsp_url_parse return value
4141 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
4143 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
4146 Automatic update of common submodule
4147 From 865aa20 to dbedaa0
4149 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
4151 * gst/rtsp-server/rtsp-server.c:
4152 rtsp-server: Fix socket leak
4153 https://bugzilla.gnome.org/show_bug.cgi?id=710088
4155 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
4157 * gst/rtsp-server/rtsp-session-pool.c:
4158 rtsp-session-pool: Make sure session IDs are properly URI-escaped
4159 https://bugzilla.gnome.org/show_bug.cgi?id=643812
4161 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4163 * examples/.gitignore:
4164 * examples/test-video.c:
4165 examples: fix compilation when WITH_AUTH is defined
4166 https://bugzilla.gnome.org/show_bug.cgi?id=710228
4168 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
4171 gitignore: Add new test binary
4173 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
4175 * tests/check/Makefile.am:
4176 * tests/check/gst/threadpool.c:
4177 thread-pool: Add unit test for the thread pools
4178 https://bugzilla.gnome.org/show_bug.cgi?id=710228
4180 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
4182 * gst/rtsp-server/rtsp-thread-pool.c:
4183 thread-pool: Fix thread leak when reusing threads
4184 https://bugzilla.gnome.org/show_bug.cgi?id=709730
4186 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
4188 * gst/rtsp-server/rtsp-server.c:
4189 * tests/check/gst/rtspserver.c:
4190 tests: fixed racy behavior in rtspserver tests
4191 https://bugzilla.gnome.org/show_bug.cgi?id=710078
4193 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
4195 * tests/check/gst/addresspool.c:
4196 tests: Improve address pool unit tests
4197 Add a range with mixed IPV4 and IPV6 addresses to pool.
4198 Get an IPV4 address from an IPV6-only pool.
4199 Get an IPV6 address from an IPV4-only pool.
4200 Reserve a IPV6 address from an IPV4-only pool.
4201 Check for unicast addresses in multicast-only pool.
4202 Check for unicast addresses in uni-/multicast-mixed pool.
4203 https://bugzilla.gnome.org/show_bug.cgi?id=710128
4205 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4207 * gst/rtsp-server/rtsp-client.c:
4208 client: append query string in PAUSE/PLAY/TEARDOWN as well
4210 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
4212 * gst/rtsp-server/rtsp-client.c:
4213 client: Add query to control path
4214 If the SETUP url contains a query it must be appended to the control
4215 path so that it matches any already created stream in the media. The
4216 query will also be appended to the session media path.
4218 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4220 * gst/rtsp-server/rtsp-media.c:
4221 rtsp-media: remove old line
4223 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
4225 * gst/rtsp-server/rtsp-stream.c:
4226 stream: Correct control comparison
4227 https://bugzilla.gnome.org/show_bug.cgi?id=709176
4229 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4231 * gst/rtsp-server/rtsp-media.c:
4232 media: Check dynamically if the pipeline supports seeking
4233 We should not depend on whether or not the pipeline state change
4234 returned NO_PREROLL or not. A media could dynamically change its
4235 element and switch from seekable to non seekable so it's best to test
4236 the seekable nature of the pipeline dynamically when we try to do a seek.
4238 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4240 * gst/rtsp-server/rtsp-media.c:
4241 media: Return FALSE if seeking is not supported
4243 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4245 * gst/rtsp-server/rtsp-media.c:
4246 rtsp-media: don't seek accurate by default
4247 Accurate seeking is perhaps a little overkill in the most common situation and
4248 causes some formats (mp3) over slow media to seek extremely slowly.
4250 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
4252 * tests/check/gst/rtspserver.c:
4253 tests: fix unit test
4254 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
4256 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
4258 * gst/rtsp-server/rtsp-client.c:
4259 client: Reply 400 if media cannot be constructed
4260 Reply 400 Bad Request instead of 503 Service Unavailable if media
4261 cannot be constructed in SETUP.
4262 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
4264 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
4266 * gst/rtsp-server/rtsp-client.c:
4267 client: Send setup reply once only
4268 If find_media() failed in handle_setup_request() two replies was sent.
4269 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
4271 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
4274 Automatic update of common submodule
4275 From 6b03ba7 to 865aa20
4277 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
4279 * gst/rtsp-server/rtsp-server.c:
4280 server: Emit client-connected signal earlier
4281 Emit client-connected before the client ref is given to a GSource,
4282 otherwise client-connected can be emitted after the client object has
4285 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
4287 * gst/rtsp-server/rtsp-address-pool.c:
4288 * gst/rtsp-server/rtsp-address-pool.h:
4289 * gst/rtsp-server/rtsp-stream.c:
4290 * tests/check/gst/addresspool.c:
4291 addresspool: return reason of failure
4292 Let gst_rtsp_address_pool_reserve_address() return the reason why
4293 the address could not be reserved.
4294 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
4296 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
4299 autogen.sh: Sync behaviour with other GStreamer modules
4300 Allows building from outside of tree amongst other things
4302 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
4305 Automatic update of common submodule
4306 From b613661 to 6b03ba7
4308 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
4311 Automatic update of common submodule
4312 From 74a6857 to b613661
4314 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
4317 Automatic update of common submodule
4318 From 01a7a46 to 74a6857
4320 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
4322 * gst/rtsp-server/rtsp-client.c:
4323 client: Do not read beyond end of path string
4324 If the setup was done without a control url, make sure we don't try to read the
4325 non-existing control string and crash.
4327 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4329 * gst/rtsp-server/rtsp-client.c:
4330 client: Fix RTPInfo header
4331 Refactor the method to make the content_base.
4332 Use the content-base and the control url to construct the RTPInfo
4335 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4337 * gst/rtsp-server/rtsp-client.c:
4338 client: map url to path only in describe
4339 Only map the request url to a path in the DESCRIBE method. The SDP then
4340 contains the base and control urls that should be used to SETUP/PAUSE/
4341 PLAY/TEARDOWN the media.
4343 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4345 * gst/rtsp-server/rtsp-client.c:
4346 Revert "client: map URL to path in requests"
4347 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
4348 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
4349 contains the base and control urls which are used in the SETUP, PLAY,
4350 PAUSE and TEARDOWN requests.
4352 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4354 * gst/rtsp-server/rtsp-client.c:
4355 client: map URL to path in requests
4357 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4359 * gst/rtsp-server/rtsp-client.c:
4360 * gst/rtsp-server/rtsp-mount-points.c:
4361 * gst/rtsp-server/rtsp-mount-points.h:
4362 mount-points: make vmethod to make path from uri
4363 Make a vmethod to transform an url into a path. The path is then used to lookup
4364 the factory. This makes it possible to also use other bits of the url, such as
4365 the query parameters, to locate the factory.
4367 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
4369 * gst/rtsp-server/rtsp-thread-pool.c:
4370 * gst/rtsp-server/rtsp-thread-pool.h:
4371 thread-pool: Add cleanup to wait for the threadpool to finish
4372 Also fix race condition if two threads are asking for the first
4373 thread from the thread pool at once. This would case two internal
4374 GThreadPools to be created.
4375 https://bugzilla.gnome.org/show_bug.cgi?id=707753
4377 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
4379 * gst/rtsp-server/rtsp-client.c:
4380 * tests/check/gst/client.c:
4381 client: free threadpool
4382 https://bugzilla.gnome.org/show_bug.cgi?id=707638
4384 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
4386 * tests/check/gst/mountpoints.c:
4387 mountpoints tests: unref matched factories
4388 https://bugzilla.gnome.org/show_bug.cgi?id=707638
4390 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
4392 * tests/check/gst/media.c:
4393 media tests: unref thread pool and caps
4394 https://bugzilla.gnome.org/show_bug.cgi?id=707638
4396 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
4398 * gst/rtsp-server/rtsp-auth.c:
4399 * gst/rtsp-server/rtsp-media-factory.c:
4400 * gst/rtsp-server/rtsp-media.c:
4401 auth, media, media-factory: unref permissions
4402 https://bugzilla.gnome.org/show_bug.cgi?id=707638
4404 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4406 * examples/Makefile.am:
4407 Makefile: add rule for appsrc example
4409 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4411 * examples/test-appsrc.c:
4412 tests: add appsrc example
4413 Add an example on how to use appsrc to feed the server pipeline with data.
4415 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
4417 * gst/rtsp-server/rtsp-client.c:
4418 rtsp-client: remove query part from content-base string
4419 Make sure that after the control url has been resolved, it's
4420 not a part of the query-string.
4421 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
4423 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4425 * gst/rtsp-server/rtsp-client.c:
4426 client: don't check url in response
4427 There is no url or method in the response to check
4429 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4431 * gst/rtsp-server/rtsp-client.c:
4432 * gst/rtsp-server/rtsp-client.h:
4433 Add handle-response signal for when we receive a GET_PARAMETER response
4435 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4437 * gst/rtsp-server/rtsp-server.c:
4438 Fix gst_rtsp_server_client_filter, using wrong variable type
4440 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
4442 * gst/rtsp-server/rtsp-media-factory-uri.c:
4443 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
4444 For AAC we need to check for framed=true instead of parsed=true.
4445 https://bugzilla.gnome.org/show_bug.cgi?id=701384
4447 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4449 * gst/rtsp-server/rtsp-stream.c:
4450 stream: optimize pipeline for protocols
4451 When TCP is not an allowed protocol for the stream, avoid creating the
4452 appsrc/appsink/queue and tee elements.
4454 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4456 * gst/rtsp-server/rtsp-media.c:
4457 media: set protocols on streams
4459 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4461 * gst/rtsp-server/rtsp-client.c:
4462 client: use protocols supported by stream
4464 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4466 * gst/rtsp-server/rtsp-media-factory.c:
4467 * gst/rtsp-server/rtsp-media.c:
4468 * gst/rtsp-server/rtsp-stream.c:
4469 media-factory: allow all protocols
4471 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4473 * gst/rtsp-server/rtsp-media.c:
4474 media: configure protocols in new streams
4476 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4478 * gst/rtsp-server/rtsp-stream.c:
4479 * gst/rtsp-server/rtsp-stream.h:
4480 stream: add protocols property
4482 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4484 * gst/rtsp-server/rtsp-media.c:
4485 rtsp-media: send state in "new-state" signal
4486 https://bugzilla.gnome.org/show_bug.cgi?id=705110
4488 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
4491 build: add subdir-objects to AM_INIT_AUTOMAKE
4492 Fixes warnings with automake 1.14
4493 https://bugzilla.gnome.org/show_bug.cgi?id=705350
4495 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4497 * docs/libs/gst-rtsp-server-sections.txt:
4498 * gst/rtsp-server/rtsp-client.c:
4499 * gst/rtsp-server/rtsp-server.c:
4500 * gst/rtsp-server/rtsp-server.h:
4501 server: add method to iterate clients of server
4503 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4505 * gst/rtsp-server/rtsp-media.c:
4506 * gst/rtsp-server/rtsp-media.h:
4507 Add vmethod for rtsp-media subclass to access rtpbin
4509 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4511 * gst/rtsp-server/rtsp-client.h:
4512 small documentation fix
4514 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4516 * gst/rtsp-server/rtsp-client.c:
4517 Do not take range header if range is invalid
4519 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4521 * docs/libs/gst-rtsp-server-sections.txt:
4522 * gst/rtsp-server/rtsp-media.c:
4523 media: add docs for new method
4525 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4527 * gst/rtsp-server/rtsp-media.c:
4528 * gst/rtsp-server/rtsp-media.h:
4529 Add API to rtsp-media set the pipeline's state
4531 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4533 * gst/rtsp-server/rtsp-media.c:
4534 Update current position/duration when gst_rtsp_media_get_range_string is called
4536 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4538 * examples/test-cgroups.c:
4539 tests: add some more docs
4541 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4543 * examples/test-cgroups.c:
4544 * gst/rtsp-server/Makefile.am:
4545 * gst/rtsp-server/rtsp-auth.c:
4546 * gst/rtsp-server/rtsp-auth.h:
4547 * gst/rtsp-server/rtsp-client.c:
4548 * gst/rtsp-server/rtsp-client.h:
4549 * gst/rtsp-server/rtsp-context.c:
4550 * gst/rtsp-server/rtsp-context.h:
4551 * gst/rtsp-server/rtsp-params.c:
4552 * gst/rtsp-server/rtsp-params.h:
4553 * gst/rtsp-server/rtsp-server.c:
4554 * gst/rtsp-server/rtsp-thread-pool.c:
4555 * gst/rtsp-server/rtsp-thread-pool.h:
4556 * tests/check/gst/client.c:
4557 ClientState -> Context
4558 Rename the clientstate to context and put the code in a separate file.
4560 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4562 * examples/test-auth.c:
4563 * gst/rtsp-server/rtsp-auth.c:
4564 * gst/rtsp-server/rtsp-auth.h:
4565 auth: add support for default token
4566 The default token is used when the user is not authenticated and can be used to
4567 give minimal permissions.
4569 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4571 * examples/test-auth.c:
4572 * gst/rtsp-server/rtsp-auth.c:
4573 auth: use defines when possible
4575 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4577 * gst/rtsp-server/rtsp-address-pool.c:
4578 address-pool: improve docs
4580 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4582 * gst/rtsp-server/rtsp-permissions.c:
4583 permissions: add the role to the copy
4585 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
4587 * gst/rtsp-server/rtsp-permissions.c:
4588 permissions: Also copy the roles
4590 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
4592 * gst/rtsp-server/rtsp-permissions.c:
4593 permissions: Make it build
4595 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4597 * gst/rtsp-server/rtsp-address-pool.h:
4600 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4602 * docs/libs/gst-rtsp-server-sections.txt:
4603 * gst/rtsp-server/rtsp-auth.c:
4604 * gst/rtsp-server/rtsp-auth.h:
4605 * gst/rtsp-server/rtsp-media.c:
4606 * gst/rtsp-server/rtsp-session-media.c:
4607 * gst/rtsp-server/rtsp-stream-transport.c:
4608 * gst/rtsp-server/rtsp-stream-transport.h:
4609 * gst/rtsp-server/rtsp-stream.c:
4610 * tests/check/gst/client.c:
4613 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4615 * docs/libs/gst-rtsp-server-sections.txt:
4616 * gst/rtsp-server/rtsp-address-pool.c:
4617 * gst/rtsp-server/rtsp-address-pool.h:
4618 * tests/check/gst/addresspool.c:
4619 * tests/check/gst/rtspserver.c:
4620 address-pool: cleanups
4621 Remove redundant method, improve docs.
4623 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4625 * docs/libs/gst-rtsp-server-sections.txt:
4626 * gst/rtsp-server/rtsp-auth.h:
4627 * gst/rtsp-server/rtsp-permissions.c:
4628 * gst/rtsp-server/rtsp-permissions.h:
4629 * gst/rtsp-server/rtsp-token.c:
4632 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4634 * gst/rtsp-server/rtsp-permissions.c:
4635 permissions: implement _remove_role
4637 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4639 * gst/rtsp-server/rtsp-permissions.c:
4640 permissions: update docs
4642 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4644 * tests/check/gst/client.c:
4645 tests: simplify tests
4646 Client settings are now disabled by default so we don't need an auth
4647 module to disable them.
4649 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4651 * gst/rtsp-server/rtsp-auth.c:
4652 auth: add default authorizations
4653 When no auth module is specified, use our table of defaults to look up the
4654 default value of the check instead of always allowing everything. This was
4655 we can disallow client settings by default.
4657 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4660 README: update readme
4662 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4664 * gst/rtsp-server/rtsp-thread-pool.c:
4665 * gst/rtsp-server/rtsp-thread-pool.h:
4666 thread-pool: add more docs
4668 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4670 * gst/rtsp-server/rtsp-thread-pool.c:
4671 * gst/rtsp-server/rtsp-thread-pool.h:
4672 thread-pool: fix race in thread reuse
4673 If we try to reuse a thread right after we made it stop, we end up using a
4674 stopped thread. Catch this case and only reuse threads that are not stopping.
4676 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4678 * gst/rtsp-server/rtsp-server.c:
4679 server: add small debug
4681 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4683 * tests/check/gst/client.c:
4685 Add some permissions to media so we can use the auth and enable
4688 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4690 * gst/rtsp-server/rtsp-client.c:
4691 client: support pushed context in handle_request
4692 If we already have a pushed state, reuse it and add our own things. This makes
4693 it easier to write tests.
4695 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4697 * gst/rtsp-server/rtsp-auth.c:
4698 auth: don't auth on methods
4699 Don't authorize on methods anymore but on the resources that we
4700 try to access, this is more flexible.
4701 Move the authorization checks to where they are needed and let the
4702 check return the response on error.
4704 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4706 * gst/rtsp-server/rtsp-mount-points.c:
4707 mount-points: add some debug
4709 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4711 * tests/check/gst/client.c:
4712 tests: almost fix test
4714 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4716 * gst/rtsp-server/rtsp-auth.c:
4717 * gst/rtsp-server/rtsp-auth.h:
4718 * gst/rtsp-server/rtsp-client.c:
4719 * gst/rtsp-server/rtsp-client.h:
4720 * gst/rtsp-server/rtsp-server.c:
4721 * gst/rtsp-server/rtsp-server.h:
4722 auth: let the auth module check client_settings
4723 Let the auth module decide if client settings are allowed for the
4726 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4728 * gst/rtsp-server/rtsp-token.c:
4729 * gst/rtsp-server/rtsp-token.h:
4730 token: add method to check boolean permission
4732 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4734 * examples/test-auth.c:
4735 * examples/test-cgroups.c:
4736 * gst/rtsp-server/rtsp-token.c:
4737 * gst/rtsp-server/rtsp-token.h:
4738 token: simplify token constructor
4739 Use variable arguments to make easier API.
4741 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4743 * examples/test-auth.c:
4744 * examples/test-cgroups.c:
4745 * gst/rtsp-server/rtsp-media-factory.c:
4746 * gst/rtsp-server/rtsp-media-factory.h:
4747 media-factory: add convenience API for factory
4749 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4751 * examples/test-auth.c:
4752 * examples/test-cgroups.c:
4753 * gst/rtsp-server/rtsp-permissions.c:
4754 * gst/rtsp-server/rtsp-permissions.h:
4755 permissions: simplify API a little
4756 Avoid passing GstStructure in the add_role method, use varargs instead
4757 to construct the structure behind the scenes. We can then also use the
4758 structure name as the role and simplify some more logic.
4760 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4762 * gst/rtsp-server/rtsp-auth.c:
4765 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4767 * gst/rtsp-server/rtsp-auth.c:
4768 * gst/rtsp-server/rtsp-auth.h:
4769 * gst/rtsp-server/rtsp-client.c:
4770 auth: handle unauthorized response
4771 Move handling of the unauthorized response to the auth module, it can add
4772 the appropriate headers to request authorization for the required method
4773 much better than the client.
4775 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4777 * gst/rtsp-server/rtsp-client.c:
4778 * gst/rtsp-server/rtsp-client.h:
4779 client: allow for sending any message, not only requests
4780 Change the _send_request() method to _send_message() so that we
4781 can both send requests and replies.
4783 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4785 * docs/libs/gst-rtsp-server-sections.txt:
4786 * gst/rtsp-server/rtsp-server.h:
4789 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4791 * examples/test-video.c:
4792 * gst/rtsp-server/rtsp-auth.c:
4793 * gst/rtsp-server/rtsp-auth.h:
4794 * gst/rtsp-server/rtsp-server.c:
4795 * gst/rtsp-server/rtsp-server.h:
4796 auth: move TLS handling to auth module
4797 Remove the TLS settings on the server and move it to the auth module because
4798 that is where security related bits go.
4800 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4802 * gst/rtsp-server/rtsp-client.c:
4803 * gst/rtsp-server/rtsp-client.h:
4804 client: add state push/pop
4806 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4808 * gst/rtsp-server/rtsp-client.c:
4809 * gst/rtsp-server/rtsp-client.h:
4810 client: add connection to state
4812 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4814 * gst/rtsp-server/rtsp-mount-points.c:
4815 mount-points: fix debug
4817 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4819 * tests/check/gst/media.c:
4820 tests: fix media test
4822 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4824 * gst/rtsp-server/rtsp-thread-pool.c:
4825 thread-pool: we don't require a state
4827 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4829 * gst/rtsp-server/rtsp-server.c:
4830 server: let context ref the server
4831 So that we don't risk losing the server object early anc crash.
4833 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4835 * tests/check/gst/client.c:
4836 tests: fix client test
4838 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4841 * docs/libs/gst-rtsp-server-docs.sgml:
4842 * docs/libs/gst-rtsp-server-sections.txt:
4843 * gst/rtsp-server/rtsp-address-pool.c:
4844 * gst/rtsp-server/rtsp-auth.c:
4845 * gst/rtsp-server/rtsp-client.c:
4846 * gst/rtsp-server/rtsp-client.h:
4847 * gst/rtsp-server/rtsp-media-factory-uri.c:
4848 * gst/rtsp-server/rtsp-media-factory.c:
4849 * gst/rtsp-server/rtsp-media-factory.h:
4850 * gst/rtsp-server/rtsp-media.c:
4851 * gst/rtsp-server/rtsp-mount-points.c:
4852 * gst/rtsp-server/rtsp-params.c:
4853 * gst/rtsp-server/rtsp-permissions.c:
4854 * gst/rtsp-server/rtsp-sdp.c:
4855 * gst/rtsp-server/rtsp-server.c:
4856 * gst/rtsp-server/rtsp-server.h:
4857 * gst/rtsp-server/rtsp-session-media.c:
4858 * gst/rtsp-server/rtsp-session-pool.c:
4859 * gst/rtsp-server/rtsp-session.c:
4860 * gst/rtsp-server/rtsp-stream-transport.c:
4861 * gst/rtsp-server/rtsp-stream.c:
4862 * gst/rtsp-server/rtsp-thread-pool.c:
4863 * gst/rtsp-server/rtsp-token.c:
4866 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4868 * gst/rtsp-server/rtsp-session-pool.c:
4869 * gst/rtsp-server/rtsp-session-pool.h:
4870 session-pool: make vmethod to create a session
4871 Make a vmethod to create a sessions so that subclasses can create
4872 custom session objects
4874 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4876 * gst/rtsp-server/rtsp-auth.c:
4877 * gst/rtsp-server/rtsp-media-factory.h:
4878 * gst/rtsp-server/rtsp-media.h:
4879 * gst/rtsp-server/rtsp-mount-points.h:
4880 * gst/rtsp-server/rtsp-session-pool.h:
4881 * gst/rtsp-server/rtsp-stream.h:
4884 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4886 * docs/libs/gst-rtsp-server-docs.sgml:
4887 * docs/libs/gst-rtsp-server-sections.txt:
4888 * gst/rtsp-server/rtsp-address-pool.c:
4889 * gst/rtsp-server/rtsp-address-pool.h:
4890 * gst/rtsp-server/rtsp-auth.c:
4891 * gst/rtsp-server/rtsp-client.h:
4892 * gst/rtsp-server/rtsp-media-factory.h:
4893 * gst/rtsp-server/rtsp-media.c:
4894 * gst/rtsp-server/rtsp-media.h:
4895 * gst/rtsp-server/rtsp-permissions.c:
4896 * gst/rtsp-server/rtsp-permissions.h:
4897 * gst/rtsp-server/rtsp-server.h:
4898 * gst/rtsp-server/rtsp-session-media.c:
4899 * gst/rtsp-server/rtsp-session-media.h:
4900 * gst/rtsp-server/rtsp-session-pool.h:
4901 * gst/rtsp-server/rtsp-session.h:
4902 * gst/rtsp-server/rtsp-stream-transport.h:
4903 * gst/rtsp-server/rtsp-stream.c:
4904 * gst/rtsp-server/rtsp-thread-pool.h:
4907 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4910 * examples/Makefile.am:
4911 configure: compile cgroup example conditionally
4912 Only compile the cgroup example when we have libcgroup
4914 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4917 * examples/Makefile.am:
4918 * examples/test-cgroups.c:
4919 examples: add cgroups example
4921 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4923 * tests/check/gst/rtspserver.c:
4924 tests: fix compilation
4926 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4928 * gst/rtsp-server/rtsp-thread-pool.c:
4929 thread-pool: fix vmethod invocation
4931 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4933 * gst/rtsp-server/rtsp-thread-pool.c:
4934 * gst/rtsp-server/rtsp-thread-pool.h:
4935 thread-pool: store thread type in thread
4937 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4939 * gst/rtsp-server/rtsp-client.c:
4940 client: pass thread from pool to media _prepare
4941 Get a thread from the configured threadpool and pass it to the prepare method of
4944 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4946 * gst/rtsp-server/rtsp-media.c:
4947 * gst/rtsp-server/rtsp-media.h:
4948 media: Accept a thread in _prepare
4949 Remove out own threadpool handling and use the provided thread and
4950 maincontext for the bus messages and the state changes.
4952 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4954 * gst/rtsp-server/rtsp-server.c:
4955 server: configure client thread pool
4957 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4959 * gst/rtsp-server/rtsp-client.c:
4960 * gst/rtsp-server/rtsp-client.h:
4961 client: add method to configure thread pool
4963 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4965 * gst/rtsp-server/rtsp-client.h:
4966 * gst/rtsp-server/rtsp-server.c:
4967 * gst/rtsp-server/rtsp-server.h:
4968 server: use thread pool
4969 Use the thread pool instead of doing our own thing.
4971 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4973 * gst/rtsp-server/Makefile.am:
4974 * gst/rtsp-server/rtsp-thread-pool.c:
4975 * gst/rtsp-server/rtsp-thread-pool.h:
4976 thread-pool: add object to manage threads
4977 Add an object to manage the client and media threads.
4979 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4981 * gst/rtsp-server/rtsp-auth.c:
4982 auth: debug authorization check
4984 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4986 * gst/rtsp-server/rtsp-media.c:
4987 media: start media pipeline in context
4988 Start the media pipeline in the provided context (or our default one
4989 when NULL). This makes sure that we run the bus thread in this context and that
4990 all media threads are children of this context.
4992 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4994 * gst/rtsp-server/rtsp-media-factory.c:
4995 factory: pass permissions to media by default
4997 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4999 * examples/test-auth.c:
5000 test: add permissions to auth test
5001 Ass some permissions to the media factory in the test.
5003 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5005 * gst/rtsp-server/rtsp-auth.c:
5006 * gst/rtsp-server/rtsp-auth.h:
5007 * gst/rtsp-server/rtsp-client.c:
5008 auth: simplify auth checks
5009 Remove client from methods, it's now in the state
5010 Perform the check specified by the string, use the information from the
5011 thread local context.
5013 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5015 * gst/rtsp-server/rtsp-client.c:
5016 * gst/rtsp-server/rtsp-client.h:
5017 client: add state to current thread
5018 Add the client to the ClientState object.
5019 Place the ClientState on the current thread.
5021 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5023 * gst/rtsp-server/rtsp-media-factory.c:
5024 * gst/rtsp-server/rtsp-media-factory.h:
5025 * gst/rtsp-server/rtsp-media.c:
5026 * gst/rtsp-server/rtsp-media.h:
5027 media: make it possible to set permissions
5028 Make it possible to set permissions on media and media factory objects
5030 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5032 * gst/rtsp-server/Makefile.am:
5033 * gst/rtsp-server/rtsp-permissions.c:
5034 * gst/rtsp-server/rtsp-permissions.h:
5035 permissions: add permissions object
5036 Add a mini object to store permissions based on a role.
5038 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5040 * examples/test-auth.c:
5041 * gst/rtsp-server/rtsp-auth.c:
5042 * gst/rtsp-server/rtsp-auth.h:
5043 * gst/rtsp-server/rtsp-client.c:
5044 auth: add auth checks
5045 Add an enum with auth checks and implement the checks in the auth object.
5046 Perform the checks from the client.
5048 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5050 * examples/test-auth.c:
5051 * gst/rtsp-server/rtsp-auth.c:
5052 * gst/rtsp-server/rtsp-auth.h:
5053 * gst/rtsp-server/rtsp-client.h:
5054 auth: use the token after authentication
5055 After we authenticated a user, keep the Token around in the state.
5057 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5059 * gst/rtsp-server/rtsp-client.c:
5060 * gst/rtsp-server/rtsp-media.c:
5061 * gst/rtsp-server/rtsp-media.h:
5062 * tests/check/gst/media.c:
5063 media: add optional context for bus messages
5064 Add an optional mainloop to _prepare that will handle the bus messages instead
5065 of always using the shared mainloop.
5067 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5069 * gst/rtsp-server/Makefile.am:
5070 * gst/rtsp-server/rtsp-token.c:
5071 * gst/rtsp-server/rtsp-token.h:
5072 token: add authorization token
5073 Add a simply miniobject that contains the authorizations. The object contains a
5074 GstStructure that hold all authorization fields. When a user is authenticated,
5075 the auth module will create a Token for the user. The token is then used to
5076 check what operations the user is allowed to do and various other configuration
5079 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5081 * examples/test-auth.c:
5082 * gst/rtsp-server/rtsp-auth.c:
5083 * gst/rtsp-server/rtsp-auth.h:
5084 * gst/rtsp-server/rtsp-client.c:
5085 * gst/rtsp-server/rtsp-client.h:
5086 * gst/rtsp-server/rtsp-media-factory.c:
5087 * gst/rtsp-server/rtsp-media-factory.h:
5088 * gst/rtsp-server/rtsp-media.c:
5089 * gst/rtsp-server/rtsp-media.h:
5090 auth: remove auth from media and factory
5091 Remove the auth object from media and factory. We want to have the RTSPClient
5092 authenticate and authorize resources, there is no need to place another auth
5093 manager on the media/factory.
5095 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5097 * examples/test-auth.c:
5098 * gst/rtsp-server/rtsp-auth.c:
5099 * gst/rtsp-server/rtsp-auth.h:
5100 * gst/rtsp-server/rtsp-client.h:
5101 auth: add support for multiple basic auth tokens
5102 Make it possible to add multiple basic authorisation tokens to one authorization
5103 object. Associate with each token an authorization group that will define what
5104 capabilities are allowed.
5106 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5108 * gst/rtsp-server/rtsp-client.c:
5109 client: error out on non-aggregate control
5110 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
5112 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5114 * gst/rtsp-server/rtsp-client.c:
5115 client: rework setup request a little
5116 Cache the media in DESCRIBE based on the longest matching path with the uri
5117 that we can find in the mount points.
5118 Rework the setup request a little to get the media from the session or from
5119 the longest matching path, this way we can derive the control string as
5120 everything after the path instead of hardcoding it.
5121 Find the stream based on the control string and only open a session when all
5124 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5126 * gst/rtsp-server/rtsp-media.c:
5127 * gst/rtsp-server/rtsp-media.h:
5128 media: add method to find a stream by control url
5130 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5132 * gst/rtsp-server/rtsp-stream.c:
5133 * gst/rtsp-server/rtsp-stream.h:
5134 stream: add method to check control url of stream
5136 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5138 * gst/rtsp-server/rtsp-client.c:
5139 * gst/rtsp-server/rtsp-session-media.c:
5140 * gst/rtsp-server/rtsp-session-media.h:
5141 * gst/rtsp-server/rtsp-session.c:
5142 * gst/rtsp-server/rtsp-session.h:
5143 session: use path matching for session media
5144 Use a path string instead of a uri to lookup session media in the sessions. Also
5145 use path matching to find the largest possible path that matches.
5147 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5149 * gst/rtsp-server/rtsp-client.c:
5150 * gst/rtsp-server/rtsp-mount-points.c:
5151 * gst/rtsp-server/rtsp-mount-points.h:
5152 * tests/check/gst/mountpoints.c:
5153 mount-points: remove useless vmethod
5154 Making lookups in the mount points should not be done with a URL, if there is a
5155 mapping to be done from URL to mount points, we'll need to do it somewhere
5158 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5160 * gst/rtsp-server/rtsp-mount-points.c:
5161 * gst/rtsp-server/rtsp-mount-points.h:
5162 * tests/check/gst/mountpoints.c:
5163 mount-points: improve mount point searching
5164 Use a GSequence to keep track of the mount points.
5165 Match a URL to the longest matching registered mount point. This should be the
5166 URL to perform aggreagate control and the remainder is the stream specific
5168 Add some unit tests for this.
5170 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
5172 * gst/rtsp-server/Makefile.am:
5173 rtsp-server: Allow building of static library
5175 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5177 * tests/check/gst/mediafactory.c:
5178 tests: fix compilation
5180 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5182 * gst/rtsp-server/rtsp-sdp.c:
5183 sdp: get control string from stream
5184 Use the control string as configured in the stream.
5186 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5188 * gst/rtsp-server/rtsp-stream.c:
5189 * gst/rtsp-server/rtsp-stream.h:
5190 stream: add methods and property to set control string
5192 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5194 * gst/rtsp-server/rtsp-client.c:
5196 Rename variables for clarity
5197 Keep media in state when we can
5199 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5201 * gst/rtsp-server/rtsp-client.c:
5202 * gst/rtsp-server/rtsp-stream.c:
5203 * gst/rtsp-server/rtsp-stream.h:
5204 stream: add more support for IPv6
5205 Rename _get_address to _get_multicast_address in GstRTSPStream to
5206 make it clear that this function only deals with multicast.
5207 Make it possible to have both an IPv4 and IPv6 multicast address on
5208 a stream. Give the client an IPv4 or IPv6 address depending on the
5209 address it used to connect to the server.
5210 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
5212 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5214 * gst/rtsp-server/rtsp-client.c:
5217 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5219 * gst/rtsp-server/rtsp-stream.c:
5220 stream: handle failed port allocation
5221 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
5222 can't allocate any family at all. Also keep track of what port families we
5224 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
5226 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5228 * gst/rtsp-server/rtsp-stream.c:
5229 stream: improve docs
5231 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5233 * gst/rtsp-server/rtsp-stream-transport.c:
5234 stream-transport: remove old if 0 block
5236 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
5238 * tests/check/gst/client.c:
5240 gst_rtsp_client_get_uri() has been removed
5241 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
5243 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5245 * gst/rtsp-server/rtsp-client.c:
5246 * gst/rtsp-server/rtsp-client.h:
5247 client: add method to filter managed sessions
5248 Add a method to filter the sessions managed by this client connection.
5249 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
5251 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5253 * gst/rtsp-server/rtsp-client.c:
5254 * gst/rtsp-server/rtsp-client.h:
5255 client: remove _get_uri() method
5256 Remove the get_uri() method on the client. A client has no uri, the uri
5257 property is an internal property to manage the last cached media for
5260 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5262 * gst/rtsp-server/rtsp-media-factory.h:
5263 media-factory: fix typo
5265 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
5267 * gst/rtsp-server/rtsp-media.c:
5268 rtsp-media: Do not leak the query in default_query_stop
5269 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
5271 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5273 * gst/rtsp-server/rtsp-media.c:
5274 media: don't unlock when conversion fails
5275 Don't unlock the state lock when conversion fails because it was not locked.
5277 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5279 * gst/rtsp-server/rtsp-media.c:
5280 * gst/rtsp-server/rtsp-media.h:
5281 Add query_position and query_stop vmethods to rtsp-media
5283 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5285 * gst/rtsp-server/rtsp-media.c:
5286 Fix typo in property install for rtsp-media's time-provider
5288 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5290 * gst/rtsp-server/rtsp-client.c:
5291 * gst/rtsp-server/rtsp-client.h:
5292 client: clean some variables
5293 Clean some variables and add some guards to _send_request()
5295 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5297 * gst/rtsp-server/rtsp-client.c:
5298 * gst/rtsp-server/rtsp-client.h:
5299 Add gst_rtsp_client_send_request API
5300 This makes it possible to send arbitrary messages to a client, such as
5301 SET_PARAMETER or GET_PARAMETER
5303 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5305 * gst/rtsp-server/rtsp-media.c:
5306 * gst/rtsp-server/rtsp-media.h:
5307 media: add _get_element() method
5308 Add method to get the element used when creating the media.
5309 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
5311 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5313 * gst/rtsp-server/rtsp-media.c:
5316 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
5318 * gst/rtsp-server/rtsp-stream.c:
5319 * gst/rtsp-server/rtsp-stream.h:
5320 stream: allow access to the rtp session
5321 https://bugzilla.gnome.org/show_bug.cgi?id=703004
5323 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
5325 * gst/rtsp-server/rtsp-stream.c:
5326 * gst/rtsp-server/rtsp-stream.h:
5327 dscp qos support in gst-rtsp-stream
5328 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
5330 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5332 * tests/check/gst/rtspserver.c:
5334 Actually do what the comment says. Also keep the old code around, not sure what
5335 should happen when you get a 454 from a TEARDOWN, does it close the connection?
5336 it currently doesn't.
5338 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5340 * gst/rtsp-server/rtsp-client.c:
5341 client: also watch newly created session
5342 When we newly created a session, start watching it immediately instead of
5343 on the next request.
5345 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
5347 * tests/check/gst/client.c:
5348 tests: add unit test for new-session
5349 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
5351 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5353 * gst/rtsp-server/rtsp-client.c:
5354 client: emit new-session when new session is created
5355 Only emit new-session when we created a new session for a client, not when a
5356 client picked up a previous session.
5357 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
5359 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
5361 * gst/rtsp-server/rtsp-client.c:
5362 client: handle asterisk as path in requests
5363 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
5365 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5367 * gst/rtsp-server/rtsp-media.c:
5368 media: handle segment query format mismatch
5369 It's possible that the segment query returns with a different format than what
5370 we asked for, handle this case also.
5372 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
5374 * gst/rtsp-server/rtsp-media.c:
5375 media: use segment stop in collect_media_stats
5376 Use segment stop instead of duration as range end point.
5377 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
5379 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
5381 * gst/rtsp-server/rtsp-media.c:
5382 * tests/check/gst/media.c:
5383 rtsp-media: Do not leak the element in take_pipeline
5384 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
5386 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
5388 * gst/rtsp-server/rtsp-client.c:
5389 * gst/rtsp-server/rtsp-client.h:
5390 rtsp-client: Make configure_client_transport virtual
5391 This patch makes configure_client_transport virtual. The functionality is
5392 needed to handle some weird clients sending multicast transport settings as url
5394 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
5396 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
5398 * gst/rtsp-server/rtsp-client.c:
5399 * gst/rtsp-server/rtsp-client.h:
5400 rtsp-client: Make param_set and param_get virtual
5401 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
5403 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
5405 * gst/rtsp-server/rtsp-client.c:
5406 * gst/rtsp-server/rtsp-media.c:
5407 * gst/rtsp-server/rtsp-media.h:
5408 media: convert_range replaces get_range_times
5409 get_range_times worked for handling UTC ranges for seeks, but we also
5410 need to convert back from NPT to the requested unit in
5411 get_range_string. convert_range is now used for both.
5412 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
5414 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5416 * gst/rtsp-server/rtsp-client.c:
5417 * gst/rtsp-server/rtsp-sdp.c:
5418 * gst/rtsp-server/rtsp-sdp.h:
5419 sdp: cleanup sdp info
5420 We don't need to pass the proto, we can more easily check a boolean.
5421 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
5423 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
5425 * gst/rtsp-server/rtsp-sdp.c:
5426 use 0.0.0.0 or :: for c= line instead of server address
5428 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
5430 * gst/rtsp-server/rtsp-client.c:
5431 use local address, not remote, in SDP
5432 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
5434 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5437 Automatic update of common submodule
5438 From 098c0d7 to 01a7a46
5440 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
5442 * gst/rtsp-server/rtsp-media.c:
5443 * gst/rtsp-server/rtsp-media.h:
5444 media: possibility to override range time conversion
5445 Make it possible to override the conversion from GstRTSPTimeRange to
5446 GstClockTimes, that is done before seeking on the media
5447 pipeline. Overriding can be useful for UTC ranges, where the default
5448 conversion gives nanoseconds since 1900.
5449 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
5451 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
5453 * gst/rtsp-server/rtsp-server.c:
5454 * gst/rtsp-server/rtsp-server.h:
5455 rtsp-server: Expose the use_client_settings API
5456 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
5458 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
5460 * gst/rtsp-server/rtsp-client.c:
5461 * gst/rtsp-server/rtsp-stream.c:
5462 * gst/rtsp-server/rtsp-stream.h:
5463 rtspstream: handle both ipv4 and ipv6 clients
5464 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
5466 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5468 * gst/rtsp-server/rtsp-sdp.c:
5469 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
5470 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
5471 We already have a way to place extra attributes in the SDP by using a string
5472 property with prefix x- or a- in the caps.
5474 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5476 * gst/rtsp-server/rtsp-sdp.c:
5477 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
5478 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
5479 We already have a way to place extra attributes in the SDP, just make a string
5480 property in the payloader with a- or x- prefix.
5482 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5484 * gst/rtsp-server/rtsp-sdp.c:
5485 rtsp: place a- and x- properties as attributes
5486 application/x-rtp has properties with a- and x- prefixes that should be
5487 placed as attributes in the SDP for the media instead of being added to the
5490 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5492 * examples/Makefile.am:
5493 * examples/test-video.c:
5494 example: add TLS example
5496 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5498 * gst/rtsp-server/rtsp-server.c:
5499 * gst/rtsp-server/rtsp-server.h:
5500 server: add support for TLS
5501 Add methods to set and get a TLS certificate.
5502 Add vmethod to configure a new connection. By default, configure the TLS
5503 certificate in a new connection if needed.
5505 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5507 * gst/rtsp-server/rtsp-server.c:
5508 * gst/rtsp-server/rtsp-server.h:
5509 server: remove accept_client vmethod
5510 This vmethod is not very useful so remove it.
5512 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5514 * gst/rtsp-server/rtsp-server.c:
5515 server: don't crash on NULL GError
5517 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
5519 * gst/rtsp-server/rtsp-session-pool.c:
5520 rtsp-session-pool: corrected session timeout detection
5521 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
5523 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5525 * gst/rtsp-server/rtsp-client.c:
5526 client: improve debug
5528 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5530 * gst/rtsp-server/rtsp-client.c:
5531 * gst/rtsp-server/rtsp-client.h:
5532 * gst/rtsp-server/rtsp-server.c:
5533 server: refactor connection setup
5534 Let the server accept the socket connection and construct a GstRTSPConnection
5535 from it. Remove the code from the client and let the client only deal with
5536 a fully configure GstRTSPConnection object.
5537 We will need this later when the server will configure the connection for
5540 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5542 * gst/rtsp-server/rtsp-stream.c:
5543 stream: keep the transport object alive
5544 Keep the transport object alive while we have it as qdata on the
5547 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
5549 * gst/rtsp-server/rtsp-client.c:
5550 * gst/rtsp-server/rtsp-server.c:
5551 rtsp-server: Do not crash on nmapping of server
5552 * generate error when gst_rtsp_connection_accept fails
5553 * do not stop accepting incoming connections because
5554 accepting a client fails
5555 https://bugzilla.gnome.org/show_bug.cgi?id=701072
5557 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
5559 * gst/rtsp-server/rtsp-client.c:
5560 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
5561 https://bugzilla.gnome.org/show_bug.cgi?id=700953
5563 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
5565 * gst/rtsp-server/rtsp-sdp.c:
5566 rtsp-sdp: Parse framerate caps field and set SDP attribute
5567 The SDP attribute and its format is described in RFC4566.
5568 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
5570 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
5572 * gst/rtsp-server/rtsp-sdp.c:
5573 rtsp-sdp: Parse width/height from caps and set SDP attribute
5574 The SDP attribute and its format is described in RFC6064.
5575 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
5577 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
5579 * gst/rtsp-server/rtsp-sdp.c:
5580 * tests/check/gst/client.c:
5581 rtsp-sdp: add bandwidth line
5582 https://bugzilla.gnome.org/show_bug.cgi?id=699220
5584 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5587 Automatic update of common submodule
5588 From 5edcd85 to 098c0d7
5590 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
5592 * tests/check/gst/media.c:
5593 tests: add dynamic payloader prepare/unprepare check
5595 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5597 * gst/rtsp-server/rtsp-media.c:
5598 media: release lock when removing fakesink
5600 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5602 * gst/rtsp-server/rtsp-stream.c:
5603 stream: set elements to NULL before removing
5604 When removing a stream, set the elements to NULL first. This avoids
5605 element-is-not-in-NULL-state errors when we dispose the elements.
5607 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
5610 Automatic update of common submodule
5611 From 3cb3d3c to 5edcd85
5613 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5615 * gst/rtsp-server/rtsp-media.c:
5616 * gst/rtsp-server/rtsp-media.h:
5617 media: listen to pad-removed signals
5618 Listen to the pad-removed signal and remove the stream associated with the
5620 Add signal to be notified of the removed pad.
5621 Remove the fakesink in unprepare()
5622 Fix signatures of the signal methods
5624 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5626 * examples/test-sdp.c:
5627 tests: add example of reusable pipelines
5629 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
5631 * gst/rtsp-server/rtsp-stream.c:
5632 * gst/rtsp-server/rtsp-stream.h:
5633 stream: add method to get the srcpad
5635 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
5637 * tests/check/gst/media.c:
5638 check: add media prepare/unprepare test
5639 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
5641 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
5643 * gst/rtsp-server/rtsp-media.c:
5644 media: disconnect from signal handlers in unprepare()
5645 We connected to the pad-added and no-more-pads signals in prepare() so
5646 we need to disconnect from them in unprepare().
5647 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
5649 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
5651 * gst/rtsp-server/rtsp-media.c:
5652 media: don't free streams array
5653 Don't free the streams array in the unprepare() method, they were not
5655 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
5657 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
5659 * gst/rtsp-server/rtsp-media.c:
5660 media: don't unref the pipeline in unprepare
5661 Unprepare() should undo what prepare() does. Because the pipeline is
5662 not created in prepare(), we should not unref it in unprepare()
5664 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
5666 * gst/rtsp-server/rtsp-stream.c:
5667 stream: clear session and caps for reuse
5668 Set the session and caps to NULL after unref otherwise we might unref
5670 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
5672 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
5674 * gst/rtsp-server/rtsp-client.c:
5675 client: send out teardown signal before tearing down
5676 The advantage is that in the signal handler you get direct access to
5677 information about what streams are about to get torn down (in the
5678 GstRTSPClientState).
5679 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
5681 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
5683 * gst/rtsp-server/rtsp-client.c:
5684 * gst/rtsp-server/rtsp-client.h:
5685 client: expose connection
5686 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
5688 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
5691 Automatic update of common submodule
5692 From aed87ae to 3cb3d3c
5694 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5696 * gst/rtsp-server/rtsp-media.c:
5697 * gst/rtsp-server/rtsp-media.h:
5698 * gst/rtsp-server/rtsp-session-media.c:
5699 * gst/rtsp-server/rtsp-session-media.h:
5700 media: add method to get the base_time of the pipeline
5701 Together with a shared clock, this base-time could eventually be sent to
5702 the client so that it can reconstruct the exact running-time of the clock
5705 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5707 * gst/rtsp-server/Makefile.am:
5708 * gst/rtsp-server/rtsp-media.c:
5709 * gst/rtsp-server/rtsp-media.h:
5710 * gst/rtsp-server/rtsp-sdp.c:
5711 media: add GstNetTimeProvider support
5712 Add a property to let the media provide a GstNetTimeProvider for its clock.
5713 Make methods to get the clock and nettimeprovider
5714 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
5715 provider and also the current time of the clock. This should make it possible
5716 for (GStreamer) clients to slave their clock to the server clock.
5718 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
5721 Automatic update of common submodule
5722 From 04c7a1e to aed87ae
5724 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5726 * gst/rtsp-server/rtsp-media.c:
5727 media: wait for buffering to complete
5728 Wait for buffering to complete before changing the state to the target state.
5730 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5732 * gst/rtsp-server/rtsp-media.c:
5733 media: small cleanup
5735 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
5737 * tests/check/gst/rtspserver.c:
5738 tests: remove extra unref in test_setup_non_existing_stream
5739 The unref is not needed anymore, teardown runs without it.
5740 https://bugzilla.gnome.org/show_bug.cgi?id=696542
5742 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
5744 * tests/check/gst/rtspserver.c:
5745 tests: GSocketService cleanup in test_bind_already_in_use
5746 Use g_socket_service_stop so the rtspserver test stops listening for
5747 incoming connections in test_bind_already_in_use.
5748 https://bugzilla.gnome.org/show_bug.cgi?id=696541
5750 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
5752 * gst/rtsp-server/rtsp-media-factory.c:
5753 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
5754 Instead use a GWeakRef which is safe to use
5755 This is a known GLib bug, see:
5756 https://bugzilla.gnome.org/show_bug.cgi?id=667145
5758 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
5760 * gst/rtsp-server/rtsp-client.c:
5761 * gst/rtsp-server/rtsp-media.c:
5762 * gst/rtsp-server/rtsp-media.h:
5763 * gst/rtsp-server/rtsp-sdp.c:
5764 * tests/check/gst/media.c:
5765 * tests/check/gst/rtspserver.c:
5766 rtsp-media/client: Reply to PLAY request with same type of Range
5767 Remember the type of Range from the PLAY request and use the same type for
5770 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
5772 * gst/rtsp-server/rtsp-client.c:
5773 * gst/rtsp-server/rtsp-client.h:
5774 * tests/check/gst/client.c:
5775 rtsp-client: expose uri
5777 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
5779 * tests/check/gst/mediafactory.c:
5780 tests: Hold ref while creating second media
5781 To test if the media aren't shared, make sure we keep the first one while creating a second
5782 otherwise the same memory address may be reused.
5784 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
5787 configure: remove out-of-date comment
5789 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
5792 .gitignore: ignore more build files
5794 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
5796 * tests/check/Makefile.am:
5797 tests: use right _LIBS variable for gst-plugins-base libs
5799 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5801 * tests/check/Makefile.am:
5802 check: add librtp to libs
5804 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
5806 * tests/check/gst/rtspserver.c:
5807 tests: Add test to check selecting a port the server will send from
5809 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
5811 * tests/check/gst/rtspserver.c:
5812 tests: Make sure packets are actually received
5814 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
5816 * gst/rtsp-server/rtsp-stream.c:
5817 stream: Select unicast address from pool if appropriate
5819 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
5821 * gst/rtsp-server/rtsp-stream.c:
5822 stream: Properties are always there in Gst 1.0
5824 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
5826 * tests/check/gst/addresspool.c:
5827 tests: Add tests for unicast addresses in pool
5829 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
5831 * gst/rtsp-server/rtsp-address-pool.c:
5832 * tests/check/gst/addresspool.c:
5833 address-pool: Verify that multicast addresses are used for multicast and vice-versa
5835 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
5837 * docs/libs/gst-rtsp-server-sections.txt:
5838 * gst/rtsp-server/rtsp-address-pool.c:
5839 * gst/rtsp-server/rtsp-address-pool.h:
5840 * gst/rtsp-server/rtsp-stream.c:
5841 * tests/check/gst/addresspool.c:
5842 address-pool: Add unicast addresses
5844 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
5847 * gst/rtsp-server/rtsp-server.c:
5848 * tests/check/gst/rtspserver.c:
5849 rtsp-server: Limit the number of threads per server instance
5850 If we exceed the maximum, just round robin the clients over the existing
5853 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
5855 * gst/rtsp-server/rtsp-server.c:
5856 rtsp-server: No need to store the GMainContext in the client context
5858 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
5860 * tests/check/gst/rtspserver.c:
5861 tests: Add test for client disconnection
5863 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
5865 * tests/check/gst/rtspserver.c:
5866 tests: Test client and session timeouts with multiple threads
5868 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
5870 * gst/rtsp-server/rtsp-address-pool.c:
5871 * gst/rtsp-server/rtsp-auth.c:
5872 * gst/rtsp-server/rtsp-client.c:
5873 * gst/rtsp-server/rtsp-media-factory-uri.c:
5874 * gst/rtsp-server/rtsp-media-factory.c:
5875 * gst/rtsp-server/rtsp-media.c:
5876 * gst/rtsp-server/rtsp-mount-points.c:
5877 * gst/rtsp-server/rtsp-server.c:
5878 * gst/rtsp-server/rtsp-session-media.c:
5879 * gst/rtsp-server/rtsp-session-pool.c:
5880 * gst/rtsp-server/rtsp-session.c:
5881 Document locking and its order
5883 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
5885 * tests/check/gst/rtspserver.c:
5886 tests: Test that slow DESCRIBE don't block other clients
5888 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
5890 * tests/check/gst/client.c:
5891 tests: Add tests for client-requested multicast address
5893 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
5895 * docs/libs/gst-rtsp-server-sections.txt:
5896 docs: Put the various functions in the right sections
5898 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
5900 * docs/libs/gst-rtsp-server-docs.sgml:
5901 * docs/libs/gst-rtsp-server-sections.txt:
5902 * gst/rtsp-server/rtsp-address-pool.c:
5903 * gst/rtsp-server/rtsp-address-pool.h:
5904 docs: Generate docs for GstRTSPAddressPool
5906 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
5908 * gst/rtsp-server/rtsp-client.c:
5909 * gst/rtsp-server/rtsp-stream.c:
5910 * gst/rtsp-server/rtsp-stream.h:
5911 client: Check client provided addresses against the address pool
5913 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
5915 * gst/rtsp-server/rtsp-address-pool.c:
5916 * gst/rtsp-server/rtsp-address-pool.h:
5917 * tests/check/gst/addresspool.c:
5918 address-pool: Add API to request a specific address from the pool
5919 Also add relevant unit tests.
5921 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
5923 * tests/check/gst/mediafactory.c:
5924 tests: Check the passing around of a RTSPAddressPool
5925 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
5926 way down to the stream.
5928 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
5930 * tests/check/gst/addresspool.c:
5931 tests: Add more tests for the address pool
5933 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
5935 * gst/rtsp-server/rtsp-address-pool.c:
5936 address-pool: Fix off by one error
5937 When splitting a port range, the port after a skip is not part of range.
5939 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
5942 Automatic update of common submodule
5943 From 2de221c to 04c7a1e
5945 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
5948 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
5949 AM_CONFIG_HEADER was removed in automake 1.13
5950 https://bugzilla.gnome.org/show_bug.cgi?id=693368
5952 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
5955 Automatic update of common submodule
5956 From a942293 to 2de221c
5958 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5960 * gst/rtsp-server/rtsp-client.c:
5961 client: make sure the watch exists while sending data
5962 Protect the send_func with a lock. This allows us to wait for sending
5963 to complete before changing the send_func and user_data. We add an
5964 extra ref to the watch to make sure that it remains valid during
5966 When closing the connection, set the send_func to NULL
5967 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
5969 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5971 * tests/check/Makefile.am:
5972 tests: use GST_*_1_0 environment variables everywhere
5973 The _1_0 suffixed environment variables override the
5974 non-suffixed ones, so if we're in an environment that
5975 sets the _1_0 suffixed ones, such as jhbuild, we need
5976 to set those to make sure ours actually always get
5979 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5982 Automatic update of common submodule
5983 From acb04d9 to a942293
5985 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5987 * gst/rtsp-server/rtsp-client.c:
5988 rtsp-client: set the client backlog
5989 Set the client backlog to a reasonable default
5991 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
5993 * gst/rtsp-server/rtsp-media.c:
5994 rtsp-media: Make the element a constructor parameter
5995 https://bugzilla.gnome.org/show_bug.cgi?id=689594
5997 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5999 * docs/libs/Makefile.am:
6000 docs: Link with gcov library when gcov is enabled
6001 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
6003 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6005 * gst/rtsp-server/rtsp-media.c:
6006 media: match prepare with unprepare
6007 Really unprepare when there were an equal amount of prepare calls.
6009 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6011 * gst/rtsp-server/rtsp-media.c:
6012 media: media has to be unprepared in finalize
6013 Because unprepare takes away the last ref on the media.
6015 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6017 * gst/rtsp-server/rtsp-client.c:
6018 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
6019 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
6020 We can't use the refcount to trigger unprepare because it is the unprepare call
6021 that removes the last refcount after all messages are consumed. What we should
6022 probably do is make a prepared refcount and only unprepare when the refcount
6025 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6027 * gst/rtsp-server/rtsp-media.c:
6028 media: let the source unref the last media ref
6029 the last ref to the media is held by the source so we don't need to add more ref
6030 and unrefs, we simply destroy the media when the source is gone.
6032 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6034 * gst/rtsp-server/rtsp-media.c:
6035 media: improve debug
6037 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6039 * gst/rtsp-server/rtsp-media.c:
6041 Make sure we are in the right state when collecting the position and duration.
6042 Only make ourselves PREPARED when we were previously PREPARING.
6044 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6046 * gst/rtsp-server/rtsp-media.c:
6047 media: use g_object_ref/unref for GObjects
6049 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
6051 * gst/rtsp-server/rtsp-client.c:
6052 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
6053 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
6054 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
6055 isn't being used anymore.
6057 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
6059 * gst/rtsp-server/rtsp-media.c:
6060 Fix compiler warning
6062 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
6064 * gst/rtsp-server/rtsp-media-factory-uri.c:
6065 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
6067 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6069 * gst/rtsp-server/rtsp-session-media.h:
6072 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6074 * gst/rtsp-server/rtsp-media.c:
6075 * tests/check/gst/media.c:
6076 media: avoid element leak
6078 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6080 * gst/rtsp-server/rtsp-media.c:
6081 media: require an element in media constructor
6083 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6085 * gst/rtsp-server/rtsp-client.c:
6086 Revert "client: TEARDOWN brings that state to Init again"
6087 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
6088 The object is already disposed, there is no point in setting the state.
6090 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6092 * gst/rtsp-server/rtsp-client.c:
6093 client: TEARDOWN brings that state to Init again
6095 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6097 * docs/libs/gst-rtsp-server-sections.txt:
6098 * examples/test-auth.c:
6099 * gst/rtsp-server/rtsp-auth.c:
6100 * gst/rtsp-server/rtsp-auth.h:
6101 * gst/rtsp-server/rtsp-client.c:
6102 * gst/rtsp-server/rtsp-client.h:
6103 * gst/rtsp-server/rtsp-media-factory-uri.c:
6104 * gst/rtsp-server/rtsp-media-factory-uri.h:
6105 * gst/rtsp-server/rtsp-media-factory.c:
6106 * gst/rtsp-server/rtsp-media-factory.h:
6107 * gst/rtsp-server/rtsp-media.c:
6108 * gst/rtsp-server/rtsp-media.h:
6109 * gst/rtsp-server/rtsp-mount-points.c:
6110 * gst/rtsp-server/rtsp-mount-points.h:
6111 * gst/rtsp-server/rtsp-sdp.c:
6112 * gst/rtsp-server/rtsp-server.c:
6113 * gst/rtsp-server/rtsp-server.h:
6114 * gst/rtsp-server/rtsp-session-media.c:
6115 * gst/rtsp-server/rtsp-session-media.h:
6116 * gst/rtsp-server/rtsp-session-pool.c:
6117 * gst/rtsp-server/rtsp-session-pool.h:
6118 * gst/rtsp-server/rtsp-session.c:
6119 * gst/rtsp-server/rtsp-session.h:
6120 * gst/rtsp-server/rtsp-stream-transport.c:
6121 * gst/rtsp-server/rtsp-stream-transport.h:
6122 * gst/rtsp-server/rtsp-stream.c:
6123 * gst/rtsp-server/rtsp-stream.h:
6124 * tests/check/gst/media.c:
6125 rtsp: make object details private
6126 Make all object details private
6127 Add methods to access private bits
6129 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6131 * tests/check/Makefile.am:
6132 * tests/check/gst/media.c:
6133 tests: add media tests
6135 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6137 * gst/rtsp-server/rtsp-media.c:
6138 media: check if prepared for some methods
6139 Check that the media object is prepared before doing seek and getting the
6140 current position etc.
6141 Add some g_return checks.
6143 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6145 * tests/check/Makefile.am:
6146 * tests/check/gst/mediafactory.c:
6147 tests: add mediafactory test
6149 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6151 * gst/rtsp-server/rtsp-stream.c:
6152 stream: improve debug
6154 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6156 * gst/rtsp-server/rtsp-media.c:
6157 * gst/rtsp-server/rtsp-media.h:
6158 media: unref pipeline in finalize to avoid leaking it
6160 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6162 * gst/rtsp-server/rtsp-media-factory-uri.c:
6163 * gst/rtsp-server/rtsp-media.c:
6164 rtsp: use gst_object_unref on GstObjects
6166 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6168 * gst/rtsp-server/rtsp-media-factory.c:
6169 media-factory: require an url
6171 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6173 * examples/test-uri.c:
6174 examples: fix include
6176 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6178 * gst/rtsp-server/rtsp-server.h:
6179 server: remove unused include
6181 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6183 * tests/check/Makefile.am:
6184 * tests/check/gst/mountpoints.c:
6185 tests: add test for mountpoints
6187 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6189 * gst/rtsp-server/rtsp-client.c:
6190 client: fix factory leak
6191 Keep the factory in the state object only for authorization checks and make
6192 sure we unref it on failure. Also don't keep invalid objects in the state
6195 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6197 * gst/rtsp-server/rtsp-mount-points.c:
6198 mounts: add g_return_if guards
6200 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6202 * tests/check/gst/client.c:
6203 tests: add more tests
6205 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6207 * gst/rtsp-server/rtsp-client.c:
6208 client: improve debug
6210 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6212 * gst/rtsp-server/rtsp-client.c:
6213 client: improve debug and fix leaks
6214 Cleanup the uri and session when there is a bad request.
6216 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6221 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6223 * tests/check/gst/client.c:
6224 test: add test for session in options request
6226 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6228 * gst/rtsp-server/rtsp-client.c:
6229 client: use 454 when session can't be found
6230 We should use 454 when a session can't be found because there was no session
6231 pool configured in the server. This is not a server configuration problem
6232 because the server on which the request is done might not be the same one that
6233 will keep the sessions for us and so it does not need to support sessions.
6235 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6237 * gst/rtsp-server/rtsp-client.c:
6238 client: only free connection when there is one
6239 It's possible that the client doesn't have a connection when we try to free it.
6241 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6243 * tests/check/Makefile.am:
6244 * tests/check/gst/client.c:
6245 tests: add unit test for the client object
6247 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6249 * gst/rtsp-server/rtsp-client.c:
6250 client: small cleanup
6252 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6254 * gst/rtsp-server/rtsp-client.h:
6255 client: remove unused include
6257 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6259 * gst/rtsp-server/rtsp-client.c:
6260 client: fix compilation
6262 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6264 * gst/rtsp-server/rtsp-client.c:
6265 client: call destroy without the lock
6267 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6269 * gst/rtsp-server/rtsp-client.c:
6270 * gst/rtsp-server/rtsp-client.h:
6271 client: make the client usable without a socket
6272 Make a method to let the client handle a message and a callback when the client
6273 wants us to send a response message back. This makes it possible to also use the
6274 client object without the sockets, which should make it easier to test.
6276 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6278 * gst/rtsp-server/rtsp-client.c:
6279 * gst/rtsp-server/rtsp-client.h:
6280 client: small cleanup
6282 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6284 * docs/libs/gst-rtsp-server-sections.txt:
6285 * gst/rtsp-server/rtsp-client.c:
6286 * gst/rtsp-server/rtsp-client.h:
6287 * gst/rtsp-server/rtsp-server.c:
6288 client: remove reference to server
6289 We don't need to keep a ref to the server
6291 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6293 * gst/rtsp-server/rtsp-client.c:
6294 * gst/rtsp-server/rtsp-client.h:
6296 Also add some g_return_if()
6298 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6300 * gst/rtsp-server/rtsp-client.c:
6301 client: log more errors
6303 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6305 * gst/rtsp-server/rtsp-client.c:
6306 client: fix compilation
6308 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6310 * gst/rtsp-server/rtsp-client.c:
6311 * gst/rtsp-server/rtsp-client.h:
6312 client: add generic close-after-send support
6313 Add a property to send_response() to close the connection after the response has
6314 been sent to the client.
6316 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6319 * docs/libs/gst-rtsp-server-docs.sgml:
6320 * docs/libs/gst-rtsp-server-sections.txt:
6321 * docs/libs/gst-rtsp-server.types:
6322 * examples/test-auth.c:
6323 * examples/test-launch.c:
6324 * examples/test-mp4.c:
6325 * examples/test-multicast.c:
6326 * examples/test-multicast2.c:
6327 * examples/test-ogg.c:
6328 * examples/test-readme.c:
6329 * examples/test-sdp.c:
6330 * examples/test-uri.c:
6331 * examples/test-video.c:
6332 * gst/rtsp-server/Makefile.am:
6333 * gst/rtsp-server/rtsp-auth.h:
6334 * gst/rtsp-server/rtsp-client.c:
6335 * gst/rtsp-server/rtsp-client.h:
6336 * gst/rtsp-server/rtsp-media-mapping.c:
6337 * gst/rtsp-server/rtsp-media-mapping.h:
6338 * gst/rtsp-server/rtsp-mount-points.c:
6339 * gst/rtsp-server/rtsp-mount-points.h:
6340 * gst/rtsp-server/rtsp-server.c:
6341 * gst/rtsp-server/rtsp-server.h:
6342 * gst/rtsp-server/rtsp-session-media.c:
6343 * gst/rtsp-server/rtsp-session-pool.c:
6344 * gst/rtsp-server/rtsp-session-pool.h:
6345 * tests/check/gst/rtspserver.c:
6346 MediaMapping -> MountPoints
6347 Describes better what the object manages.
6349 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6352 configure: bump required version of -base
6354 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6356 * gst/rtsp-server/rtsp-media.c:
6359 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6361 * gst/rtsp-server/rtsp-media.c:
6362 * gst/rtsp-server/rtsp-media.h:
6363 media: support more Range formats
6364 Use the new -base methods to convert the Range string into a seek start and stop
6367 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6369 * examples/test-launch.c:
6370 examples: fix whitespace
6372 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6374 * examples/test-auth.c:
6375 test-auth: add example of how to remove sessions
6376 Add an example of the session filter api.
6378 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6380 * examples/test-uri.c:
6381 test-uri: remove mapping example
6383 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6385 * examples/test-uri.c:
6386 test-uri: fix callback signature
6388 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6390 * gst/rtsp-server/rtsp-media-factory.c:
6391 factory: keep ref to factory while media active
6392 While the media from a factory is alive, keep a ref to the factory.
6393 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
6395 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6397 * gst/rtsp-server/rtsp-media-factory-uri.c:
6398 factory-uri: add some debug
6400 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6402 * gst/rtsp-server/rtsp-stream.c:
6403 stream: set udp sources to PLAYING
6404 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
6405 so that it doesn't cause our pipeline to produce ASYNC-DONE.
6407 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6409 * gst/rtsp-server/rtsp-media-factory-uri.c:
6410 factory-uri: take ref to factory
6411 Take a ref to the factory that we place in our list.
6413 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6415 * tests/Makefile.am:
6416 * tests/test-reuse.c:
6417 test: add test for server reuse
6418 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
6420 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
6422 * gst/rtsp-server/rtsp-server.c:
6423 server: start and stop multiple times
6424 Stop listening on the RTSP port when the GSource is removed, so clients
6425 can't connect and the server can be started again.
6426 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
6428 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6430 * gst/rtsp-server/rtsp-server.c:
6431 server: fix small leak
6433 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6435 * gst/rtsp-server/rtsp-media.c:
6436 media: unref source in finish_unprepare
6437 The source is created in prepare, unref it in finish_unprepare.
6438 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
6440 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
6442 * gst/rtsp-server/rtsp-client.c:
6443 * gst/rtsp-server/rtsp-media.c:
6444 rtsp-media: remove bus watch before finalizing
6445 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
6446 * An extra media ref is added for the bus watch. This extra ref is unreffed by
6447 the GDestroyNotify function.
6448 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
6449 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
6450 gst_rtsp_media_unprepare before unreffing the media.
6451 This way, the bus watch will be removed before the media is finalized.
6452 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
6454 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
6456 * gst/rtsp-server/rtsp-client.c:
6457 * gst/rtsp-server/rtsp-client.h:
6458 client: wait until the TEARDOWN response is sent to close the connection
6459 Responses can be sent async so we need to wait until the TEARDOWN response has
6460 been written before we close the connection to the client. This avoids the risk
6461 of writing/polling closed sockets.
6462 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
6464 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
6466 * gst/rtsp-server/rtsp-stream.c:
6467 rtsp-stream: plug socket leak
6468 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
6470 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
6473 Automatic update of common submodule
6474 From 6bb6951 to a72faea
6476 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
6478 * gst/rtsp-server/rtsp-media-factory-uri.c:
6479 rtsp-server: don't use deprecated API
6481 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
6483 * gst/rtsp-server/rtsp-client.c:
6484 rtsp-client: fix unused-but-set-variable compiler warning
6485 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
6487 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6490 * docs/libs/gst-rtsp-server-sections.txt:
6491 * gst/rtsp-server/rtsp-client.c:
6494 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6496 * examples/Makefile.am:
6497 * examples/test-multicast2.c:
6498 examples: add another multicast example
6499 Add an example for how to configure separate multicast ranges for each media
6502 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6504 * examples/test-multicast.c:
6507 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6509 * gst/rtsp-server/rtsp-client.c:
6510 * gst/rtsp-server/rtsp-media.c:
6511 * gst/rtsp-server/rtsp-session-media.c:
6512 * gst/rtsp-server/rtsp-session-media.h:
6513 * gst/rtsp-server/rtsp-stream-transport.c:
6514 * gst/rtsp-server/rtsp-stream-transport.h:
6515 stream: use the address managed by the stream
6516 Use the address managed by the stream for multicast. This allows us to have 1
6517 multicast address for each stream.
6518 Because the address is now managed by the stream we don't have to pass it around
6520 Set the address pool on the streams.
6522 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6524 * gst/rtsp-server/rtsp-client.c:
6525 * gst/rtsp-server/rtsp-media.c:
6526 * gst/rtsp-server/rtsp-stream.c:
6529 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6531 * gst/rtsp-server/rtsp-media.c:
6532 * gst/rtsp-server/rtsp-media.h:
6533 media: add signal for new streams
6534 This allows applications to listen for new streams and configure properties on
6535 them, like the address pool.
6537 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6539 * gst/rtsp-server/rtsp-media.c:
6540 media: configure address pool in new streams
6542 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6544 * gst/rtsp-server/rtsp-stream.c:
6545 * gst/rtsp-server/rtsp-stream.h:
6546 stream: add methods to deal with address pool
6547 Add methods to get and set the address pool for the stream
6548 Add method to allocate and get the multicast addresses for this stream.
6550 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6552 * docs/libs/gst-rtsp-server-sections.txt:
6553 * gst/rtsp-server/rtsp-media.c:
6554 * gst/rtsp-server/rtsp-media.h:
6555 media: remove MTU property
6556 It is a stream property
6558 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6560 * gst/rtsp-server/rtsp-client.c:
6561 client: set blocksize only on stream
6562 Set the blocksize only on the current stream.
6564 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6566 * gst/rtsp-server/rtsp-stream.c:
6567 stream: share src and sink sockets
6568 the allocated socket is in the used-socket property, not socket.
6570 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6572 * gst/rtsp-server/rtsp-address-pool.c:
6573 * gst/rtsp-server/rtsp-address-pool.h:
6574 * gst/rtsp-server/rtsp-client.c:
6575 * gst/rtsp-server/rtsp-session-media.c:
6576 * gst/rtsp-server/rtsp-session-media.h:
6577 * gst/rtsp-server/rtsp-stream-transport.c:
6578 * gst/rtsp-server/rtsp-stream-transport.h:
6579 * tests/check/gst/addresspool.c:
6580 rtsp: make address-pool return an address object
6581 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
6582 store more info in the structure and allows us to more easily return the address
6583 to the right pool when no longer needed.
6584 Pass the address to the StreamTransport so that we can return it to the pool
6585 when the stream transport is freed or changed.
6587 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6589 * examples/Makefile.am:
6590 * examples/test-multicast.c:
6591 examples: add multicast example
6592 Show how to set up the multicast address pool so that media can be
6593 server with multicast.
6595 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6597 * gst/rtsp-server/rtsp-client.c:
6598 * gst/rtsp-server/rtsp-media-factory.c:
6599 * gst/rtsp-server/rtsp-media-factory.h:
6600 * gst/rtsp-server/rtsp-media.c:
6601 * gst/rtsp-server/rtsp-media.h:
6602 rtsp: use AddressPool
6603 Remove the multicast_group property.
6604 Use the configured addresspool to allocate multicast addresses.
6606 2012-11-14 16:17:33 +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 address-pool: add clear method
6612 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6614 * gst/rtsp-server/rtsp-address-pool.c:
6615 address-pool: small cleanups
6617 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6619 * tests/check/Makefile.am:
6620 * tests/check/gst/addresspool.c:
6621 tests: add addresspool unit test
6623 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6625 * gst/rtsp-server/Makefile.am:
6626 * gst/rtsp-server/rtsp-address-pool.c:
6627 * gst/rtsp-server/rtsp-address-pool.h:
6628 address-pool: add object to manage multicast addresses
6629 Make an object that can manage a rage of multicast addresses and ports.
6631 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6633 * gst/rtsp-server/rtsp-server.c:
6634 server: set default max-threads property
6636 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6638 * gst/rtsp-server/rtsp-media.c:
6639 media: wait for concurrent _prepare
6640 If a prepare is busy, wait for the result.
6642 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6644 * gst/rtsp-server/rtsp-media.c:
6645 media: add lock around message handler
6646 We don't want to dispatch messages while we are still processing the result of
6649 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6651 * gst/rtsp-server/rtsp-media.c:
6652 * gst/rtsp-server/rtsp-media.h:
6653 media: add lock to protect state changes
6655 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6657 * gst/rtsp-server/rtsp-stream.c:
6658 * gst/rtsp-server/rtsp-stream.h:
6661 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6663 * gst/rtsp-server/rtsp-stream-transport.c:
6664 * gst/rtsp-server/rtsp-stream-transport.h:
6665 * gst/rtsp-server/rtsp-stream.c:
6666 stream-transport: add keep-alive method
6668 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6670 * gst/rtsp-server/rtsp-stream-transport.c:
6671 * gst/rtsp-server/rtsp-stream-transport.h:
6672 * gst/rtsp-server/rtsp-stream.c:
6673 stream-transport: add method to handle RTP/RTCP
6674 Call new methods instead of poking into the structures directly.
6676 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6678 * gst/rtsp-server/rtsp-session-media.c:
6679 * gst/rtsp-server/rtsp-session-media.h:
6680 session-media: add locking
6682 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6684 * gst/rtsp-server/rtsp-session.c:
6685 * gst/rtsp-server/rtsp-session.h:
6686 session: add locking
6688 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6690 * gst/rtsp-server/rtsp-server.c:
6691 server: free old socket
6693 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6695 * gst/rtsp-server/rtsp-media-mapping.c:
6696 * gst/rtsp-server/rtsp-media-mapping.h:
6697 mapping: add locking
6699 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6701 * gst/rtsp-server/rtsp-media-factory.c:
6702 media-factory: add locking
6704 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6706 * gst/rtsp-server/rtsp-auth.c:
6707 * gst/rtsp-server/rtsp-auth.h:
6710 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6712 * gst/rtsp-server/rtsp-server.c:
6713 * gst/rtsp-server/rtsp-server.h:
6714 server: add max-thread property
6716 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6718 * gst/rtsp-server/rtsp-server.c:
6719 * gst/rtsp-server/rtsp-server.h:
6720 server: use a threadpool for the mainloops
6722 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6724 * gst/rtsp-server/rtsp-client.c:
6725 * gst/rtsp-server/rtsp-client.h:
6726 client: rename method
6727 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
6728 don't really create the client from the socket, we use the socket for the
6731 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6733 * gst/rtsp-server/rtsp-client.c:
6734 * gst/rtsp-server/rtsp-client.h:
6735 * gst/rtsp-server/rtsp-server.c:
6736 server: rework maincontext handling in clients
6737 Make a separate method to attach a client to a MainContext.
6738 Let the server decide in what GMainContext the client will operate and give this
6739 context to the client in attach. Then the server can later decide to use a
6740 separate thread for each client or just use the mainthread.
6742 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6744 * gst/rtsp-server/rtsp-client.c:
6745 * gst/rtsp-server/rtsp-session.c:
6746 * gst/rtsp-server/rtsp-session.h:
6747 session: move session header code in session object
6749 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
6753 * examples/test-auth.c:
6754 * examples/test-launch.c:
6755 * examples/test-mp4.c:
6756 * examples/test-ogg.c:
6757 * examples/test-readme.c:
6758 * examples/test-sdp.c:
6759 * examples/test-uri.c:
6760 * examples/test-video.c:
6761 * gst/rtsp-server/rtsp-auth.c:
6762 * gst/rtsp-server/rtsp-auth.h:
6763 * gst/rtsp-server/rtsp-client.c:
6764 * gst/rtsp-server/rtsp-client.h:
6765 * gst/rtsp-server/rtsp-media-factory-uri.c:
6766 * gst/rtsp-server/rtsp-media-factory-uri.h:
6767 * gst/rtsp-server/rtsp-media-factory.c:
6768 * gst/rtsp-server/rtsp-media-factory.h:
6769 * gst/rtsp-server/rtsp-media-mapping.c:
6770 * gst/rtsp-server/rtsp-media-mapping.h:
6771 * gst/rtsp-server/rtsp-media.c:
6772 * gst/rtsp-server/rtsp-media.h:
6773 * gst/rtsp-server/rtsp-params.c:
6774 * gst/rtsp-server/rtsp-params.h:
6775 * gst/rtsp-server/rtsp-sdp.c:
6776 * gst/rtsp-server/rtsp-sdp.h:
6777 * gst/rtsp-server/rtsp-server.c:
6778 * gst/rtsp-server/rtsp-server.h:
6779 * gst/rtsp-server/rtsp-session-media.c:
6780 * gst/rtsp-server/rtsp-session-media.h:
6781 * gst/rtsp-server/rtsp-session-pool.c:
6782 * gst/rtsp-server/rtsp-session-pool.h:
6783 * gst/rtsp-server/rtsp-session.c:
6784 * gst/rtsp-server/rtsp-session.h:
6785 * gst/rtsp-server/rtsp-stream-transport.c:
6786 * gst/rtsp-server/rtsp-stream-transport.h:
6787 * gst/rtsp-server/rtsp-stream.c:
6788 * gst/rtsp-server/rtsp-stream.h:
6789 * tests/check/gst/rtspserver.c:
6790 * tests/test-cleanup.c:
6793 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6795 * gst/rtsp-server/rtsp-media.c:
6796 * gst/rtsp-server/rtsp-session-media.c:
6797 * gst/rtsp-server/rtsp-session.c:
6798 rtsp-server: added annotations to indicate type of ownership transfer of return values
6799 https://bugzilla.gnome.org/show_bug.cgi?id=680777
6801 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
6804 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
6806 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
6809 * bindings/Makefile.am:
6810 * bindings/vala/Makefile.am:
6811 * bindings/vala/gst-rtsp-server-0.10.deps:
6812 * bindings/vala/gst-rtsp-server-0.10.vapi:
6813 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
6814 * bindings/vala/packages/gst-rtsp-server-0.10.files:
6815 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6816 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6817 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
6819 bindings: remove vala bindings
6820 They'll be reunited with the other GStreamer bindings
6821 https://bugzilla.gnome.org/show_bug.cgi?id=680777
6823 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6825 * gst/rtsp-server/rtsp-client.c:
6826 * gst/rtsp-server/rtsp-session-media.c:
6827 * gst/rtsp-server/rtsp-session-media.h:
6828 * gst/rtsp-server/rtsp-stream-transport.c:
6829 * gst/rtsp-server/rtsp-stream-transport.h:
6830 rtsp: only create transport when needed
6831 Only create the StreamTransport when configured.
6833 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6835 * gst/rtsp-server/rtsp-client.c:
6836 client: small cleanup
6838 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6840 * gst/rtsp-server/rtsp-client.c:
6841 * gst/rtsp-server/rtsp-client.h:
6842 * gst/rtsp-server/rtsp-stream-transport.c:
6843 * gst/rtsp-server/rtsp-stream-transport.h:
6844 rtsp: refactor configuration of transport
6845 Move the configuration of the transport to a place where it makes
6848 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6850 * gst/rtsp-server/rtsp-client.c:
6851 client: refactor transport parsing
6853 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6855 * gst/rtsp-server/rtsp-client.c:
6856 client: refuse to change the MTU on shared media
6857 If we change the MTU of chared media, it changes for all clients.
6858 We don't want to set the MTU to something large for clients that
6861 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6863 * examples/test-mp4.c:
6864 * gst/rtsp-server/rtsp-media.c:
6865 small fixes to docs and debug
6867 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6869 * gst/rtsp-server/rtsp-stream.c:
6870 stream: transports must already have been removed
6872 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6874 * gst/rtsp-server/rtsp-media.c:
6875 * gst/rtsp-server/rtsp-stream.c:
6876 * gst/rtsp-server/rtsp-stream.h:
6877 stream: improve join and leave of the pipeline
6879 Do the cleanup properly
6882 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6884 * gst/rtsp-server/rtsp-media.c:
6885 media: move unprepare below default implementation
6886 Makes it easier to find the default implementation
6888 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6890 * gst/rtsp-server/rtsp-media.c:
6891 media: signal unprepared when we actually finish
6893 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6895 * gst/rtsp-server/rtsp-media.c:
6896 media: no need to unlock, unprepare does that when needed
6898 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6900 * docs/libs/gst-rtsp-server-sections.txt:
6901 * gst/rtsp-server/rtsp-media-factory.h:
6902 * gst/rtsp-server/rtsp-media-mapping.c:
6903 * gst/rtsp-server/rtsp-media.h:
6904 * gst/rtsp-server/rtsp-params.c:
6905 * gst/rtsp-server/rtsp-server.c:
6906 * gst/rtsp-server/rtsp-session-pool.h:
6907 * gst/rtsp-server/rtsp-session.c:
6908 * gst/rtsp-server/rtsp-session.h:
6909 * gst/rtsp-server/rtsp-stream-transport.h:
6910 * gst/rtsp-server/rtsp-stream.h:
6913 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6915 * gst/rtsp-server/rtsp-client.c:
6916 * gst/rtsp-server/rtsp-media-mapping.h:
6917 * gst/rtsp-server/rtsp-media.c:
6918 * gst/rtsp-server/rtsp-media.h:
6919 * gst/rtsp-server/rtsp-server.h:
6920 * gst/rtsp-server/rtsp-stream.c:
6921 * gst/rtsp-server/rtsp-stream.h:
6922 rtsp: fix MTU setting
6923 Fix setting of the MTU. There is no need for a vmethod.
6925 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6930 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6933 configure: bump version number after refactoring
6935 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6937 * gst/rtsp-server/Makefile.am:
6938 * gst/rtsp-server/rtsp-client.c:
6939 * gst/rtsp-server/rtsp-client.h:
6940 * gst/rtsp-server/rtsp-media-factory-uri.c:
6941 * gst/rtsp-server/rtsp-media-factory.c:
6942 * gst/rtsp-server/rtsp-media-factory.h:
6943 * gst/rtsp-server/rtsp-media.c:
6944 * gst/rtsp-server/rtsp-media.h:
6945 * gst/rtsp-server/rtsp-sdp.c:
6946 * gst/rtsp-server/rtsp-session-media.c:
6947 * gst/rtsp-server/rtsp-session-media.h:
6948 * gst/rtsp-server/rtsp-session.c:
6949 * gst/rtsp-server/rtsp-session.h:
6950 * gst/rtsp-server/rtsp-stream-transport.c:
6951 * gst/rtsp-server/rtsp-stream-transport.h:
6952 * gst/rtsp-server/rtsp-stream.c:
6953 * gst/rtsp-server/rtsp-stream.h:
6954 rtsp: massive refactoring
6955 Make GObjects from the remaining simple structures.
6956 Remove GstRTSPSessionStream, it's not needed.
6957 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
6958 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
6959 a GstRTSPStream should be transported to a client.
6960 Rename GstRTSPMediaFactory::get_element -> create_element because that
6961 more accurately describes what it does.
6962 Make nice methods instead of poking in the structures.
6963 Move some methods inside the relevant object source code.
6964 Use GPtrArray to store objects instead of plain arrays, it is more
6965 natural and allows us to more easily clean up.
6966 Move the allocation of udp ports to the Stream object. The Stream object
6967 contains the elements needed to stream the media to a client.
6968 Improve the prepare and unprepare methods. Unprepare should now undo
6969 everything prepare did. Improve also async unprepare when doing EOS on
6970 shutdown. Make sure we always unprepare correctly.
6972 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
6974 * gst/rtsp-server/rtsp-client.c:
6975 rtsp-client: Unref server address clients connected to
6976 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
6978 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
6980 * gst/rtsp-server/rtsp-server.c:
6981 rtsp-server: don't ref server socket if it is NULL
6982 Fixes test_bind_already_in_use unit test again after commit 6a497440.
6983 https://bugzilla.gnome.org/show_bug.cgi?id=686644
6985 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
6987 * tests/check/Makefile.am:
6988 tests: Add libgio link dependency
6989 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
6991 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6993 * gst/rtsp-server/rtsp-media-mapping.c:
6994 * gst/rtsp-server/rtsp-media-mapping.h:
6995 rtsp-media-mapping: rename find_media vfunc to find_factory
6996 The virtual method and class method should have the same name
6997 so it is correctly represented in GIR file
6998 https://bugzilla.gnome.org/show_bug.cgi?id=680777
7000 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7002 * gst/rtsp-server/rtsp-auth.c:
7003 * gst/rtsp-server/rtsp-client.c:
7004 * gst/rtsp-server/rtsp-media-factory-uri.c:
7005 * gst/rtsp-server/rtsp-media-factory.c:
7006 * gst/rtsp-server/rtsp-media-mapping.c:
7007 * gst/rtsp-server/rtsp-media.c:
7008 * gst/rtsp-server/rtsp-server.c:
7009 * gst/rtsp-server/rtsp-session-pool.c:
7010 * gst/rtsp-server/rtsp-session.c:
7011 rtsp-server: fixed comments and GIR annotations
7012 https://bugzilla.gnome.org/show_bug.cgi?id=680777
7014 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
7016 * gst/rtsp-server/rtsp-media-mapping.c:
7017 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
7019 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
7021 * gst/rtsp-server/rtsp-server.c:
7022 rtsp-server: allow binding on port 0 (binds on a random port)
7024 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
7026 * gst/rtsp-server/rtsp-server.c:
7027 * gst/rtsp-server/rtsp-server.h:
7028 rtsp-server: add bound-port property
7029 bound-port can be used to retrieve the port number when the server is bound on
7030 port 0, which binds on a random port.
7032 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
7034 * gst/rtsp-server/rtsp-media-factory.c:
7035 * gst/rtsp-server/rtsp-media-factory.h:
7036 rtsp-media-factory: make ::get_element overridable by GI bindings
7037 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
7038 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
7039 as the invoker for ::get_element(), making it overridable by GI generated
7042 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
7044 * gst/rtsp-server/rtsp-media-factory-uri.c:
7045 rtsp-media-factory-uri: don't autoplug parsers in a loop
7046 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
7049 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
7051 * gst/rtsp-server/Makefile.am:
7052 Explicitly link against gio. Fix link error on mac.
7054 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
7056 * gst/rtsp-server/rtsp-session.c:
7057 session: add ttl to the transport header in SETUP
7058 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
7060 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
7062 * gst/rtsp-server/rtsp-client.c:
7063 * gst/rtsp-server/rtsp-client.h:
7064 * gst/rtsp-server/rtsp-media.c:
7065 client: Use client transport settings for multicast if allowed.
7066 This patch makes it possible for the client to send transport settings for
7067 multicast (destination && ttl). Client settings must be explicitly allowed or
7068 the server will use its own settings.
7069 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
7071 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
7074 Automatic update of common submodule
7075 From 6c0b52c to 6bb6951
7077 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
7079 * gst/rtsp-server/rtsp-client.c:
7080 rtsp-client: do not destroy the rtsp watch
7081 Don't destroy the client watch while dispatching. The rtsp watch is
7082 automatically destroyed after the rtsp watch function closed() has
7084 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
7086 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
7089 Automatic update of common submodule
7090 From 4f962f7 to 6c0b52c
7092 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
7094 * gst/rtsp-server/rtsp-media.c:
7095 media: fix check for seekability
7097 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7099 * gst/rtsp-server/rtsp-client.c:
7100 client: use more GIO
7101 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
7103 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7105 * gst/rtsp-server/rtsp-server.c:
7106 server: remove obsolete includes
7108 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
7110 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
7111 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
7112 be available in "on_new_ssrc". The transports are added in
7113 gst_rtsp_media_set_state when going to PLAYING state. However,
7114 "on_new_ssrc" might be called before this happens.
7115 https://bugzilla.gnome.org/show_bug.cgi?id=683304
7117 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
7119 * gst/rtsp-server/rtsp-client.c:
7120 * gst/rtsp-server/rtsp-client.h:
7121 rtsp-client: add signals for rtsp requests (fixes #683287)
7123 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
7125 * gst/rtsp-server/rtsp-client.c:
7126 * gst/rtsp-server/rtsp-client.h:
7127 add new-session signal to rtsp-client (fixes #683058)
7129 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
7132 Automatic update of common submodule
7133 From 668acee to 4f962f7
7135 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
7137 * gst/rtsp-server/rtsp-server.c:
7138 * tests/check/gst/rtspserver.c:
7139 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
7140 Do not assume that *error is set in g_socket_address_enumerator_next.
7141 Added test_bind_already_in_use unit-test.
7142 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
7144 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
7147 Automatic update of common submodule
7148 From 94ccf4c to 668acee
7150 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
7152 * gst/rtsp-server/rtsp-client.c:
7153 * gst/rtsp-server/rtsp-client.h:
7154 rtsp-client: make create_sdp virtual method
7155 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
7157 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7160 Automatic update of common submodule
7161 From 98e386f to 94ccf4c
7163 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7165 * gst/rtsp-server/rtsp-client.c:
7168 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
7170 * gst/rtsp-server/rtsp-client.c:
7171 * gst/rtsp-server/rtsp-client.h:
7172 * gst/rtsp-server/rtsp-server.c:
7173 * gst/rtsp-server/rtsp-server.h:
7174 rtsp-server: use an existing socket to establish HTTP tunnel
7175 Make it possible to transfer a socket from an HTTP server to be used as
7176 an RTSP over HTTP tunnel.
7178 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
7180 * gst/rtsp-server/rtsp-client.c:
7181 * gst/rtsp-server/rtsp-media.c:
7182 * gst/rtsp-server/rtsp-media.h:
7183 rtsp: Handle the blocksize parameter
7184 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
7186 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
7188 * tests/check/Makefile.am:
7189 * tests/check/gst/rtspserver.c:
7190 Have unit test get header from source dir, not installed dir
7191 This makes compilation of unit tests work in a build directory other
7192 than the source directory.
7193 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
7195 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
7197 * gst/rtsp-server/rtsp-media.c:
7198 rtsp-media: update for gst_element_make_from_uri() changes
7200 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
7203 * tests/Makefile.am:
7204 * tests/check/Makefile.am:
7205 * tests/check/gst/rtspserver.c:
7207 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
7209 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
7211 * gst/rtsp-server/rtsp-media.c:
7212 rtsp-media: don't collect media stats when going to NULL
7213 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
7215 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7217 * gst/rtsp-server/rtsp-client.c:
7218 client: don't leak transports
7220 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
7222 * gst/rtsp-server/rtsp-client.c:
7223 rtsp-client: free transport on no_stream in SETUP handler
7225 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
7227 * gst/rtsp-server/rtsp-client.c:
7228 rtsp-client: changed session media iteration
7229 In client_unlink_session: now don't iterate in session->medias
7230 list where items are removed by gst_rtsp_session_release_media.
7231 Instead, repeatedly remove the first item.
7233 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
7235 * gst/rtsp-server/rtsp-client.c:
7236 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
7237 GstRTSPSessionMedia is not a GObject type. When the
7238 GstRTSPSession is freed, it will free the media.
7240 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
7242 * gst/rtsp-server/rtsp-media-factory.c:
7243 factory: plug pad leak in collect_streams
7244 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
7245 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
7246 will take one reference, and the other reference will otherwise
7249 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
7252 configure: suppress some warnings when debug is disabled
7253 Warnings about unused variables should be suppressed if core has the
7254 debug system disabled.
7255 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
7257 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7259 * docs/libs/Makefile.am:
7260 docs: fix build in uninstalled setup
7261 Include gst-plugins-base libs properly.
7263 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
7265 * docs/libs/gst-rtsp-server.types:
7266 docs: include headers defining rtsp-server object types
7267 Fixes compiler warnings during docs build.
7268 https://bugzilla.gnome.org/show_bug.cgi?id=676824
7270 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
7273 configure: Add warning flags for compiler when configuring
7274 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
7276 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7279 Automatic update of common submodule
7280 From 03a0e57 to 98e386f
7282 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7285 Automatic update of common submodule
7286 From 1fab359 to 03a0e57
7288 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
7290 * gst/rtsp-server/rtsp-client.c:
7291 client: fix GSocketAddress leak in gst_rtsp_client_accept
7292 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
7294 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7297 Automatic update of common submodule
7298 From f1b5a96 to 1fab359
7300 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7303 Automatic update of common submodule
7304 From 92b7266 to f1b5a96
7306 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7309 Automatic update of common submodule
7310 From ec1c4a8 to 92b7266
7312 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7315 Automatic update of common submodule
7316 From 3429ba6 to ec1c4a8
7318 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
7320 * gst/rtsp-server/rtsp-auth.c:
7321 * gst/rtsp-server/rtsp-client.c:
7322 * gst/rtsp-server/rtsp-media-factory-uri.c:
7323 * gst/rtsp-server/rtsp-server.c:
7324 rtsp: fix compiler warnings
7325 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
7327 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7330 Automatic update of common submodule
7331 From dc70203 to 3429ba6
7333 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7335 * gst/rtsp-server/rtsp-client.c:
7336 * gst/rtsp-server/rtsp-media-factory.c:
7337 * gst/rtsp-server/rtsp-media-factory.h:
7338 * gst/rtsp-server/rtsp-media.c:
7339 * gst/rtsp-server/rtsp-media.h:
7340 * gst/rtsp-server/rtsp-server.c:
7341 * gst/rtsp-server/rtsp-server.h:
7342 * gst/rtsp-server/rtsp-session-pool.c:
7343 * gst/rtsp-server/rtsp-session-pool.h:
7344 rtsp-server: port to new thread API
7346 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7349 Automatic update of common submodule
7350 From 6db25be to dc70203
7352 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7354 * gst/rtsp-server/rtsp-auth.c:
7355 * gst/rtsp-server/rtsp-auth.h:
7356 * gst/rtsp-server/rtsp-client.c:
7357 rtsp-server: Fix compilation and compiler warnings
7359 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7363 * gst/rtsp-server/Makefile.am:
7364 configure: Modernize autotools setup a bit
7365 Also we now only create tar.bz2 and tar.xz tarballs.
7367 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7370 Automatic update of common submodule
7371 From 464fe15 to 6db25be
7373 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7376 Automatic update of common submodule
7377 From 7fda524 to 464fe15
7379 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7382 * docs/libs/Makefile.am:
7383 * docs/version.entities.in:
7385 * gst/rtsp-server/Makefile.am:
7386 * pkgconfig/Makefile.am:
7387 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
7388 * pkgconfig/gstreamer-rtsp-server.pc.in:
7389 * tests/Makefile.am:
7390 rtsp-server: Update versioning
7392 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7394 Merge remote-tracking branch 'origin/0.10'
7396 gst/rtsp-server/rtsp-session-pool.c
7398 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7400 * gst/rtsp-server/rtsp-session-pool.c:
7401 rtsp-server: Don't use deprecated GLib API
7403 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7405 Replace master with 0.11
7407 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7409 Merge branch 'master' into 0.11
7411 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7413 Merge branch 'master' into 0.11
7415 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
7418 A couple minor typo fixes
7420 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7422 * gst/rtsp-server/rtsp-media.c:
7423 media: fix state of the appqueue
7425 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7427 * gst/rtsp-server/rtsp-media-factory-uri.c:
7428 factory: use videoconvert
7430 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7432 * gst/rtsp-server/rtsp-media-factory-uri.c:
7433 factory: change to new style caps
7435 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7437 * gst/rtsp-server/rtsp-client.c:
7438 * gst/rtsp-server/rtsp-client.h:
7439 * gst/rtsp-server/rtsp-media-factory-uri.c:
7440 * gst/rtsp-server/rtsp-media.c:
7441 * gst/rtsp-server/rtsp-server.c:
7442 * gst/rtsp-server/rtsp-server.h:
7443 * gst/rtsp-server/rtsp-session-pool.c:
7444 rtsp-server: port to GIO
7447 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7450 configure: fix build
7452 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7455 docs: fix for gst_rtsp_server_set_port() -> _set_service()
7456 https://bugzilla.gnome.org/show_bug.cgi?id=666548
7458 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7461 * examples/Makefile.am:
7462 First rule of gst-rtsp-server club: don't talk about gst-phonon
7464 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7467 * pkgconfig/Makefile.am:
7468 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
7469 * pkgconfig/gstreamer-rtsp-server.pc.in:
7470 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
7471 For consistency with all other modules.
7473 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7475 * gst/rtsp-server/rtsp-client.c:
7476 rtsp-client: update for new map API
7478 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7481 * bindings/Makefile.am:
7482 * bindings/python/Makefile.am:
7483 * bindings/python/arg-types.py:
7484 * bindings/python/codegen/Makefile.am:
7485 * bindings/python/codegen/__init__.py:
7486 * bindings/python/codegen/argtypes.py:
7487 * bindings/python/codegen/code-coverage.py:
7488 * bindings/python/codegen/codegen.py:
7489 * bindings/python/codegen/definitions.py:
7490 * bindings/python/codegen/defsparser.py:
7491 * bindings/python/codegen/docextract.py:
7492 * bindings/python/codegen/docgen.py:
7493 * bindings/python/codegen/fileprefix.override:
7494 * bindings/python/codegen/fileprefixmodule.c:
7495 * bindings/python/codegen/h2def.py:
7496 * bindings/python/codegen/mergedefs.py:
7497 * bindings/python/codegen/mkskel.py:
7498 * bindings/python/codegen/override.py:
7499 * bindings/python/codegen/reversewrapper.py:
7500 * bindings/python/codegen/scmexpr.py:
7501 * bindings/python/rtspserver-types.defs:
7502 * bindings/python/rtspserver.defs:
7503 * bindings/python/rtspserver.override:
7504 * bindings/python/rtspservermodule.c:
7505 * bindings/python/test.py:
7507 python: remove pygst-based python bindings
7508 pygi is the future, apparently.
7510 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
7513 Automatic update of common submodule
7514 From c463bc0 to 7fda524
7516 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7519 Automatic update of common submodule
7520 From 2a59016 to c463bc0
7522 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7525 Automatic update of common submodule
7526 From 0807187 to 2a59016
7528 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7531 Automatic update of common submodule
7532 From 11f0cd5 to 0807187
7534 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7536 * examples/test-auth.c:
7537 example: update for new caps
7539 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7541 * examples/test-video.c:
7542 * gst/rtsp-server/rtsp-client.c:
7543 * gst/rtsp-server/rtsp-media-factory-uri.c:
7544 * gst/rtsp-server/rtsp-media.c:
7545 * gst/rtsp-server/rtsp-media.h:
7546 * gst/rtsp-server/rtsp-session.c:
7547 * gst/rtsp-server/rtsp-session.h:
7548 rtsp-server: port some more to 0.11
7550 Remove bufferlist stuff
7552 Add queue before appsink now that preroll-queue-len is gone.
7553 Update for request pad changes.
7555 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7557 Merge branch 'master' into 0.11
7559 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
7561 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7562 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
7563 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
7565 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
7567 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7568 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
7569 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
7571 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7573 Merge branch 'master' into 0.11
7575 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7577 * gst/rtsp-server/rtsp-media.c:
7578 * gst/rtsp-server/rtsp-media.h:
7579 media: add a seekable boolean
7580 Maintain the seekable state with a new variable instead of reusing the
7583 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
7585 * gst/rtsp-server/rtsp-media.c:
7586 Disallow seek in live media
7588 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7590 Merge branch 'master' into 0.11
7592 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
7594 * gst/rtsp-server/rtsp-server.c:
7595 #ifdef statements for windows socket creation were missing
7597 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
7600 Automatic update of common submodule
7601 From a39eb83 to 11f0cd5
7603 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
7606 Automatic update of common submodule
7607 From 605cd9a to a39eb83
7609 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7611 Merge branch 'master' into 0.11
7613 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7615 * gst/rtsp-server/rtsp-client.c:
7616 client: use method to access property
7618 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7620 * gst/rtsp-server/rtsp-media-factory.c:
7621 * gst/rtsp-server/rtsp-media-factory.h:
7622 media-factory: add protocols property
7623 Add a property to configure the allowed protocols in the media created from the
7626 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7628 * gst/rtsp-server/rtsp-media-factory.c:
7629 * gst/rtsp-server/rtsp-media-factory.h:
7630 media-factory: add media-configure signal
7631 Add signal to allow the application to configure the media after it was created
7634 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7636 * gst/rtsp-server/rtsp-client.c:
7637 client: use method to access property
7639 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7641 * gst/rtsp-server/rtsp-media-factory.c:
7642 * gst/rtsp-server/rtsp-media-factory.h:
7643 media-factory: add protocols property
7644 Add a property to configure the allowed protocols in the media created from the
7647 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7649 * gst/rtsp-server/rtsp-media-factory.c:
7650 * gst/rtsp-server/rtsp-media-factory.h:
7651 media-factory: add media-configure signal
7652 Add signal to allow the application to configure the media after it was created
7655 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7657 Merge branch 'master' into 0.11
7659 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7661 * gst/rtsp-server/rtsp-client.c:
7662 client: use media multicast group
7664 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7666 * gst/rtsp-server/rtsp-media-factory.h:
7667 * gst/rtsp-server/rtsp-server.h:
7668 * gst/rtsp-server/rtsp-session-pool.h:
7669 * gst/rtsp-server/rtsp-session.h:
7672 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
7674 * gst/rtsp-server/rtsp-client.c:
7675 * gst/rtsp-server/rtsp-sdp.h:
7676 sdp: copy and free the server ip address
7677 Copy and free the server ip address to make memory management easier later.
7679 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7681 * gst/rtsp-server/rtsp-media-factory.c:
7682 media-factory: configure multicast in media
7684 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7686 * gst/rtsp-server/rtsp-media.c:
7687 * gst/rtsp-server/rtsp-media.h:
7688 media: add property for multicast group
7689 Add a property to configure the multicast group in the media.
7690 Based on patches from Marc Leeman and Robert Krakora.
7692 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7694 * gst/rtsp-server/rtsp-media-factory.c:
7695 * gst/rtsp-server/rtsp-media-factory.h:
7696 media-factory: add property for multicast group
7697 Add a property to configure the multicast group in the media factory.
7698 Based on patches from Marc Leeman and Robert Krakora.
7700 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7702 * gst/rtsp-server/rtsp-client.c:
7703 client: do configuration of transport in one place
7704 Move the configuration of the transport destination address to where we also
7705 configure the other bits.
7707 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7709 * gst/rtsp-server/rtsp-client.c:
7710 client: use media multicast group
7712 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7714 * gst/rtsp-server/rtsp-media-factory.h:
7715 * gst/rtsp-server/rtsp-server.h:
7716 * gst/rtsp-server/rtsp-session-pool.h:
7717 * gst/rtsp-server/rtsp-session.h:
7720 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
7722 * gst/rtsp-server/rtsp-client.c:
7723 * gst/rtsp-server/rtsp-sdp.h:
7724 sdp: copy and free the server ip address
7725 Copy and free the server ip address to make memory management easier later.
7727 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7729 * gst/rtsp-server/rtsp-media-factory.c:
7730 media-factory: configure multicast in media
7732 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7734 * gst/rtsp-server/rtsp-media.c:
7735 * gst/rtsp-server/rtsp-media.h:
7736 media: add property for multicast group
7737 Add a property to configure the multicast group in the media.
7738 Based on patches from Marc Leeman and Robert Krakora.
7740 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7742 * gst/rtsp-server/rtsp-media-factory.c:
7743 * gst/rtsp-server/rtsp-media-factory.h:
7744 media-factory: add property for multicast group
7745 Add a property to configure the multicast group in the media factory.
7746 Based on patches from Marc Leeman and Robert Krakora.
7748 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7750 * gst/rtsp-server/rtsp-client.c:
7751 client: do configuration of transport in one place
7752 Move the configuration of the transport destination address to where we also
7753 configure the other bits.
7755 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7757 Merge branch 'master' into 0.11
7759 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
7761 * gst/rtsp-server/rtsp-client.c:
7762 client: destroy pipeline on client disconnect with no prior TEARDOWN.
7763 The problem occurs when the client abruptly closes the connection without
7764 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
7765 server is where the pipeline gets torn down. Since this handler is not called,
7766 the pipeline remains and is up and running. Subsequent clients get their own
7767 pipelines and if the do not issue TEARDOWNs then those pipelines will also
7768 remain up and running. This is a resource leak.
7770 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7772 Merge branch 'master' into 0.11
7774 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
7776 * gst/rtsp-server/rtsp-media-factory.c:
7777 * gst/rtsp-server/rtsp-media-factory.h:
7778 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
7779 For example, it can be used to retrieve source elements like appsrc, in a more
7780 convenient way than subclassing get_element.
7782 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7784 Merge branch 'master' into 0.11
7786 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
7788 * gst/rtsp-server/rtsp-server.c:
7789 rtsp-server: hold on to reference while using object
7791 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7793 * gst/rtsp-server/rtsp-media.c:
7796 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7799 configure: use unstable api
7801 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
7803 * gst/rtsp-server/rtsp-client.c:
7804 client: fix reference counting
7806 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
7808 * gst/rtsp-server/rtsp-client.c:
7809 * gst/rtsp-server/rtsp-media.c:
7810 fix compiler warnings about unused variables
7812 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
7814 * examples/test-launch.c:
7815 * examples/test-readme.c:
7816 * examples/test-uri.c:
7817 * examples/test-video.c:
7818 examples: tell rtsp uri when ready
7820 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
7823 Automatic update of common submodule
7824 From 69b981f to 605cd9a
7826 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7828 * gst/rtsp-server/rtsp-client.c:
7829 client: update for buffer API change
7831 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7833 * gst/rtsp-server/Makefile.am:
7834 Makefile.am: 0.10 => @GST_MAJORMINOR@
7836 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7838 * gst/rtsp-server/rtsp-media-factory-uri.c:
7839 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
7841 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7843 * gst/rtsp-server/.gitignore:
7844 .gitignore: 0.10 => 0.11
7846 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7848 * gst/rtsp-server/Makefile.am:
7849 Makefile.am: 0.10 => @GST_MAJORMINOR@
7851 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7853 Merge branch 'master' into 0.11
7855 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
7858 Automatic update of common submodule
7859 From 9e5bbd5 to 69b981f
7861 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
7864 Automatic update of common submodule
7865 From fd35073 to 9e5bbd5
7867 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
7870 Automatic update of common submodule
7871 From 46dfcea to fd35073
7873 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7875 * gst/rtsp-server/rtsp-media-factory-uri.c:
7876 * gst/rtsp-server/rtsp-media.c:
7877 media: port to new caps API
7879 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7881 Merge branch 'master' into 0.11
7883 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
7885 * bindings/vala/gst-rtsp-server-0.10.vapi:
7886 Updated Vala bindings.
7887 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
7889 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
7891 * gst/rtsp-server/rtsp-server.c:
7892 * gst/rtsp-server/rtsp-server.h:
7893 Add a signal for newly connected clients.
7894 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
7896 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
7898 * bindings/python/rtspserver.override:
7899 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
7901 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7903 * gst/rtsp-server/Makefile.am:
7904 * gst/rtsp-server/rtsp-client.c:
7905 * gst/rtsp-server/rtsp-funnel.c:
7906 * gst/rtsp-server/rtsp-funnel.h:
7907 * gst/rtsp-server/rtsp-media.c:
7908 rtsp-server: port to 0.11
7910 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7915 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7917 Merge branch 'master' into 0.11
7922 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7925 Automatic update of common submodule
7926 From c3cafe1 to 46dfcea
7928 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
7930 * bindings/python/Makefile.am:
7931 * bindings/python/rtspserver.defs:
7932 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
7934 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
7936 * bindings/python/arg-types.py:
7937 python bindings: add GstRTSPUrlParam
7938 Needed to implement MediaFactory virtual proxies
7940 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
7942 * bindings/python/arg-types.py:
7943 python bindings: fix returning GstRTSPUrl types
7945 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
7947 * bindings/python/arg-types.py:
7948 python bindings: add arg type for GstRTSPUrl
7950 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
7952 * bindings/python/rtspserver.defs:
7953 python bindings: fix the definition of MediaFactory.collect_stream
7955 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
7958 Automatic update of common submodule
7959 From 1ccbe09 to c3cafe1
7961 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7964 Automatic update of common submodule
7965 From 193b717 to 1ccbe09
7967 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
7970 Automatic update of common submodule
7971 From b77e2bf to 193b717
7973 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7976 build: Include lcov.mak to allow test coverage report generation
7978 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7981 Automatic update of common submodule
7982 From d8814b6 to b77e2bf
7984 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7987 Automatic update of common submodule
7988 From 6aaa286 to d8814b6
7990 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
7993 Automatic update of common submodule
7994 From 6aec6b9 to 6aaa286
7996 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
7999 autogen: wingo signed comment
8001 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
8003 * gst/rtsp-server/rtsp-session-pool.c:
8004 session: use full charset for RTSP session ID
8005 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
8006 session ID more difficult.
8007 https://bugzilla.gnome.org/show_bug.cgi?id=643812
8009 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8011 * gst/rtsp-server/Makefile.am:
8012 rtsp-server: Don't install the funnel header
8014 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
8017 Automatic update of common submodule
8018 From 1de7f6a to 6aec6b9
8020 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8023 configure: require core/base 0.10.31
8024 Needed at least for gst_plugin_feature_rank_compare_func().
8026 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
8029 Automatic update of common submodule
8030 From f94d739 to 1de7f6a
8032 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8034 * gst/rtsp-server/rtsp-media.c:
8035 media: remove more unused code
8037 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8039 * gst/rtsp-server/rtsp-media.c:
8040 * gst/rtsp-server/rtsp-media.h:
8041 media: remove duplicate filtering
8042 Remove the duplicate filtering code now that we have a released -good version.
8043 Give a warning instead.
8045 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8047 * gst/rtsp-server/rtsp-media-factory.c:
8048 * gst/rtsp-server/rtsp-media.c:
8049 media: fix default buffer size
8051 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8053 * gst/rtsp-server/rtsp-media-factory.c:
8054 * gst/rtsp-server/rtsp-media-factory.h:
8055 media-factory: add property to configure the buffer-size
8056 Add a property to configure the kernel UDP buffer size.
8058 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8060 * gst/rtsp-server/rtsp-media.c:
8061 * gst/rtsp-server/rtsp-media.h:
8062 media: add property to configure kernel buffer sizes
8063 Add a property to configure the kernel UDP buffer size.
8065 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8068 configure: set PYGOBJECT_REQ before using it
8069 https://bugzilla.gnome.org/show_bug.cgi?id=640641
8071 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8074 docs: recursive into sub-directories on 'make upload'
8076 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8078 * docs/libs/gst-rtsp-server-docs.sgml:
8079 * docs/version.entities.in:
8080 docs: mention full version these docs are for, not just major-minor
8082 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8087 === release 0.10.8 ===
8089 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8094 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8096 * gst/rtsp-server/rtsp-server.c:
8097 rtsp-server: clarify docs a little
8099 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8101 * gst/rtsp-server/rtsp-media.c:
8102 media: init debug category before starting thread
8104 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8106 * gst/rtsp-server/rtsp-auth.c:
8107 auth: add realm to make it more spec compliant
8109 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8111 * gst/rtsp-server/rtsp-server.c:
8112 * gst/rtsp-server/rtsp-server.h:
8115 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8117 * examples/test-video.c:
8118 example: improve example docs a little
8120 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8122 * gst/rtsp-server/rtsp-server.c:
8123 server: ensure the watch has a ref to the server
8125 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8127 * gst/rtsp-server/rtsp-server.c:
8128 server: simpify channel function
8130 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8132 * gst/rtsp-server/rtsp-server.c:
8133 * gst/rtsp-server/rtsp-server.h:
8134 server: simplify management of channel and source
8135 We don't need to keep around the channel and source objects. Let the mainloop
8136 and the source manage the source and channel respectively.
8138 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8144 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8147 * tests/Makefile.am:
8148 * tests/test-cleanup.c:
8149 tests: add tests directory and cleanup test
8151 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8153 * gst/rtsp-server/rtsp-media-factory-uri.c:
8154 * gst/rtsp-server/rtsp-media-factory.c:
8155 * gst/rtsp-server/rtsp-media-mapping.c:
8156 * gst/rtsp-server/rtsp-media.c:
8157 * gst/rtsp-server/rtsp-session-pool.c:
8158 * gst/rtsp-server/rtsp-session.c:
8159 server: improve debugging in various objects
8161 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8163 * gst/rtsp-server/rtsp-server.c:
8164 server: chain up to the parent finalize
8166 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
8168 * bindings/python/rtspserver-types.defs:
8169 * bindings/python/rtspserver.defs:
8170 * bindings/python/rtspserver.override:
8171 * bindings/python/test.py:
8172 gst-rtsp-server: update python bindings
8174 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8176 * gst/rtsp-server/rtsp-client.c:
8177 client: use the response from the clientstate
8178 Create the response object only once and store in the client state.
8179 Make all methods use the state response,
8181 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8183 * gst/rtsp-server/rtsp-server.c:
8184 server: use signal to keep track of clients
8185 Keep track of all the clients that the server creates and remove them when they
8186 fire the 'closed' signal.
8188 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8190 * gst/rtsp-server/rtsp-client.c:
8191 * gst/rtsp-server/rtsp-client.h:
8192 client: emit signal when closing
8194 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8196 * examples/.gitignore:
8197 * examples/Makefile.am:
8198 * examples/test-auth.c:
8199 * examples/test-video.c:
8200 * gst/rtsp-server/rtsp-auth.c:
8201 * gst/rtsp-server/rtsp-auth.h:
8202 * gst/rtsp-server/rtsp-client.c:
8203 * gst/rtsp-server/rtsp-media-factory.c:
8204 * gst/rtsp-server/rtsp-media.c:
8205 * gst/rtsp-server/rtsp-media.h:
8206 * gst/rtsp-server/rtsp-session-pool.h:
8207 * gst/rtsp-server/rtsp-session.h:
8208 media: enable per factory authorisations
8209 Allow for adding a GstRTSPAuth on the factory and media level and check
8210 permissions when accessing the factory.
8211 Add hints to the auth methods for future more fine grained authorisation.
8212 Add example application for per factory authentication.
8214 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8216 * gst/rtsp-server/rtsp-auth.c:
8217 * gst/rtsp-server/rtsp-auth.h:
8218 * gst/rtsp-server/rtsp-client.c:
8219 * gst/rtsp-server/rtsp-client.h:
8220 * gst/rtsp-server/rtsp-params.c:
8221 * gst/rtsp-server/rtsp-params.h:
8222 rtsp-server: Pass ClientState structure arround
8223 Pass the collected information for the ongoing request in a GstRTSPClientState
8224 structure that we can then pass around to simplify the method arguments. This
8225 will also be handy when we implement logging functionality.
8227 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8229 * gst/rtsp-server/rtsp-media-factory.c:
8230 * gst/rtsp-server/rtsp-media-factory.h:
8231 media-factory: add methods to configure authorisation
8233 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8235 * gst/rtsp-server/rtsp-client.c:
8236 client: unref auth in finalize
8238 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8240 * gst/rtsp-server/rtsp-server.c:
8241 server: unref auth in finalize
8243 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8245 * docs/libs/gst-rtsp-server-docs.sgml:
8246 * docs/libs/gst-rtsp-server-sections.txt:
8247 * docs/libs/gst-rtsp-server.types:
8250 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8252 * gst/rtsp-server/rtsp-server.c:
8253 * gst/rtsp-server/rtsp-server.h:
8254 server: separate create and accept
8255 Create separate create and accept methods so that subclasses can create custom
8257 Configure the server in the client object and prepare for keeping track of
8260 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8262 * gst/rtsp-server/rtsp-client.c:
8263 * gst/rtsp-server/rtsp-client.h:
8264 client: add support for setting the server.
8265 Add support for keeping a ref to the server that started this client
8268 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8270 * gst/rtsp-server/rtsp-auth.c:
8271 auth: fix memleak and add some docs
8272 Fix a memleak of the basic auth token.
8273 Add docs for the helper function
8275 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8277 * gst/rtsp-server/rtsp-auth.c:
8278 * gst/rtsp-server/rtsp-auth.h:
8279 * gst/rtsp-server/rtsp-client.c:
8280 client: delegate setup of auth to the manager
8281 Delegate the configuration of the authentication tokens to the manager object
8284 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8286 * examples/test-video.c:
8287 * gst/rtsp-server/Makefile.am:
8288 * gst/rtsp-server/rtsp-auth.c:
8289 * gst/rtsp-server/rtsp-auth.h:
8290 * gst/rtsp-server/rtsp-client.c:
8291 * gst/rtsp-server/rtsp-client.h:
8292 * gst/rtsp-server/rtsp-server.c:
8293 * gst/rtsp-server/rtsp-server.h:
8294 auth: add authentication object
8295 Add an object that can check the authorization of requests.
8296 Implement basic authentication.
8297 Add example authentication to test-video
8299 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8301 * gst/rtsp-server/rtsp-server.c:
8302 * gst/rtsp-server/rtsp-server.h:
8303 server: move includes back
8304 the includes are needed for sockaddr_in.
8306 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8308 * gst/rtsp-server/rtsp-client.c:
8309 * gst/rtsp-server/rtsp-client.h:
8310 * gst/rtsp-server/rtsp-server.c:
8311 * gst/rtsp-server/rtsp-server.h:
8312 rtsp: move network includes where they are needed
8314 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
8316 * gst/rtsp-server/rtsp-media.h:
8317 rtsp-media.h: Minor corrections in comments.
8320 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
8323 Automatic update of common submodule
8324 From e572c87 to f94d739
8326 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8330 * docs/libs/.gitignore:
8331 * examples/.gitignore:
8332 * gst/rtsp-server/.gitignore:
8335 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8337 * docs/libs/Makefile.am:
8338 docs: We don't build ps/pdf for API reference docs
8340 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8343 Automatic update of common submodule
8344 From ccbaa85 to e572c87
8346 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8349 Automatic update of common submodule
8350 From 46445ad to ccbaa85
8352 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8354 * gst/rtsp-server/Makefile.am:
8355 * gst/rtsp-server/rtsp-funnel.c:
8356 * gst/rtsp-server/rtsp-funnel.h:
8357 * gst/rtsp-server/rtsp-media.c:
8358 funnel: rename fsfunnel to rtspfunnel
8359 Rename the funnel to avoid conflicts with the farsight one.
8361 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8363 * gst/rtsp-server/Makefile.am:
8364 * gst/rtsp-server/fs-funnel.c:
8365 * gst/rtsp-server/fs-funnel.h:
8366 * gst/rtsp-server/rtsp-media.c:
8367 rtsp-media: add and use fsfunnel
8368 Add a copy of fsfunnel to the build because input-selector removed the (broken)
8369 select-all property that we need.
8371 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8373 * gst/rtsp-server/Makefile.am:
8374 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
8375 Use PKG_CONFIG_PATH specified at configure time (if any) as well
8376 for the g-ir-compiler, rather than just assuming the env var has
8379 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8386 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
8388 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8391 * gst/rtsp-server/Makefile.am:
8392 gobject-introspection: fix g-i build for uninstalled setup
8393 Requires gst-plugins-base git (> 0.10.31.2).
8395 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8397 * examples/test-uri.c:
8398 examples: add some more options and comments
8400 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8402 * gst/rtsp-server/rtsp-media-factory-uri.c:
8403 factory-uri: use right property type
8405 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8407 * gst/rtsp-server/rtsp-media-factory-uri.c:
8408 factory-uri: attempt to configure buffer-lists
8409 Attempt to configure buffer lists in the payloader for improved performance.
8411 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8413 * gst/rtsp-server/rtsp-media.c:
8414 media: attempt to configure bigger UDP buffers
8415 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
8416 send buffers with high bitrate streams.
8418 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
8420 * gst/rtsp-server/rtsp-client.c:
8421 client: use the socket length from getsockname
8422 Use the length returned by getsockname to perform the getnameinfo call because
8423 the size can depend on the socket type and platform.
8426 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8428 * docs/libs/gst-rtsp-server-docs.sgml:
8429 * docs/libs/gst-rtsp-server-sections.txt:
8430 docs: add uri factory to the docs
8432 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8434 * gst/rtsp-server/rtsp-client.c:
8435 * gst/rtsp-server/rtsp-media.h:
8438 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8440 * gst/rtsp-server/rtsp-client.c:
8441 * gst/rtsp-server/rtsp-media.c:
8442 * gst/rtsp-server/rtsp-media.h:
8443 * gst/rtsp-server/rtsp-session.c:
8444 * gst/rtsp-server/rtsp-session.h:
8445 rtsp-server: add support for buffer lists
8446 Add support for sending bufferlists received from appsink.
8449 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8451 * gst/rtsp-server/rtsp-client.c:
8452 * gst/rtsp-server/rtsp-media.c:
8453 * gst/rtsp-server/rtsp-media.h:
8454 * gst/rtsp-server/rtsp-sdp.c:
8455 media: make method to retrieve the play range
8456 Make a method to retrieve the playback range so that we can conditionally create
8457 a different range for the SDP and the PLAY requests.
8459 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8461 * gst/rtsp-server/rtsp-media.c:
8462 * gst/rtsp-server/rtsp-media.h:
8463 media: add signal to notify of state changes
8465 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8467 * gst/rtsp-server/rtsp-client.h:
8468 client: cleanup headers
8470 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8472 * gst/rtsp-server/rtsp-client.c:
8475 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8477 * gst/rtsp-server/rtsp-media-factory-uri.c:
8478 * gst/rtsp-server/rtsp-media-factory-uri.h:
8479 factory-uri: add support for gstpay
8480 Add an option to prefer gstpay over decoder + raw payloader.
8482 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8484 * gst/rtsp-server/rtsp-media-factory-uri.c:
8485 * gst/rtsp-server/rtsp-media-factory-uri.h:
8486 factory-uri: rework the autoplugger.
8487 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
8490 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8492 * gst/rtsp-server/rtsp-media-factory-uri.c:
8493 factory-uri: use better factory filter
8494 Make better payloader filter based on autoplug rank and RTP use case.
8496 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8499 Automatic update of common submodule
8500 From 169462a to 46445ad
8502 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8504 * gst/rtsp-server/rtsp-server.c:
8505 server: set SO_REUSEADDR before bind
8506 Set the SO_REUSEADDR _before_ bind() to make it actually work.
8508 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8510 * gst/rtsp-server/rtsp-media.c:
8511 * gst/rtsp-server/rtsp-media.h:
8512 media: emit prepared signal when prepared
8513 Make a 'prepared' signal and emit it when we successfully prepared the element.
8514 This signal can be used to configure the media object after it has been prepared
8517 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
8520 Automatic update of common submodule
8521 From 011bcc8 to 169462a
8523 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
8525 python an optional dependency
8526 * configure.ac: Move up valgrind and g-i checks. Make the python
8527 dependency optional, as it was before.
8529 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8531 Merge branch 'master' into 0.11
8536 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8538 * gst/rtsp-server/rtsp-media.c:
8539 media: update range when active clients changed
8540 When we changed the number of active clients, update the current range
8541 information because we want the second client connecting to a shared resource
8542 continue from where the stream currently.
8544 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8546 * gst/rtsp-server/rtsp-media-factory-uri.c:
8547 * gst/rtsp-server/rtsp-media-factory-uri.h:
8548 factory-uri: add colorspace and fix pt
8549 Rework the way we pass data to the autoplugger.
8550 When we have raw caps, plug a converter element to make pluggin to raw
8551 payloaders more successful.
8552 Make sure all dynamically plugged payloaders have a unique payload types.
8554 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8556 * examples/Makefile.am:
8557 * examples/test-uri.c:
8558 example: add example of the uri factory
8560 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8562 * gst/rtsp-server/Makefile.am:
8563 * gst/rtsp-server/rtsp-media-factory-uri.c:
8564 * gst/rtsp-server/rtsp-media-factory-uri.h:
8565 * gst/rtsp-server/rtsp-server.h:
8566 factory-uri: add a factory to stream any URI
8567 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
8570 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8572 * gst/rtsp-server/rtsp-media.c:
8573 * gst/rtsp-server/rtsp-media.h:
8574 media: ignore spurious ASYNC_DONE messages
8575 When we are dynamically adding pads, the addition of the udpsrc elements will
8576 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
8577 the real ASYNC_DONE when everything is prerolled.
8579 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8581 * gst/rtsp-server/rtsp-media-factory.c:
8582 * gst/rtsp-server/rtsp-media-factory.h:
8583 media-factory: make lock macro
8585 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
8587 * gst/rtsp-server/rtsp-client.c:
8588 rtsp-server: Remove unused variable and dead assignment
8590 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
8592 * examples/test-launch.c:
8593 * examples/test-mp4.c:
8594 * examples/test-ogg.c:
8595 * examples/test-readme.c:
8596 * examples/test-sdp.c:
8597 * examples/test-video.c:
8598 examples: Run gst-indent
8600 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
8602 * gst/rtsp-server/rtsp-client.c:
8603 * gst/rtsp-server/rtsp-media-factory.c:
8604 * gst/rtsp-server/rtsp-media-mapping.c:
8605 * gst/rtsp-server/rtsp-media.c:
8606 * gst/rtsp-server/rtsp-params.c:
8607 * gst/rtsp-server/rtsp-sdp.c:
8608 * gst/rtsp-server/rtsp-server.c:
8609 * gst/rtsp-server/rtsp-session-pool.c:
8610 * gst/rtsp-server/rtsp-session.c:
8611 rtsp-server: Run gst-indent
8612 Since it wasn't using the upstream common previously, there was no
8613 indentation check before commiting.
8615 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
8617 * gst/rtsp-server/rtsp-media-mapping.h:
8618 * gst/rtsp-server/rtsp-media.c:
8619 * gst/rtsp-server/rtsp-media.h:
8620 * gst/rtsp-server/rtsp-sdp.c:
8621 * gst/rtsp-server/rtsp-session-pool.h:
8622 * gst/rtsp-server/rtsp-session.c:
8623 * gst/rtsp-server/rtsp-session.h:
8624 rtsp-server: Some more doc fixups
8626 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8629 Makefile: Add cruft-cleaning support
8631 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8636 * docs/libs/Makefile.am:
8637 * docs/libs/gst-rtsp-server-docs.sgml:
8638 * docs/libs/gst-rtsp-server-sections.txt:
8639 * docs/libs/gst-rtsp-server.types:
8640 * docs/version.entities.in:
8641 docs: Add gtk-doc build system
8643 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8645 * gst/rtsp-server/Makefile.am:
8646 Makefile.am: Use standard GIR make behaviour
8648 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8652 autogen/configure: Bring more in sync to standard gst module behaviour
8654 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8656 * gst/rtsp-server/rtsp-media.c:
8657 media: warn and fail when gstrtpbin is not found
8659 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8662 configure: open 0.11 branch
8664 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
8668 Add common submodule
8670 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
8673 * common/Makefile.am:
8674 * common/c-to-xml.py:
8676 * common/coverage/coverage-report-entry.pl:
8677 * common/coverage/coverage-report.pl:
8678 * common/coverage/coverage-report.xsl:
8679 * common/coverage/lcov.mak:
8680 * common/gettext.patch:
8681 * common/glib-gen.mak:
8682 * common/gst-autogen.sh:
8683 * common/gst-xmlinspect.py:
8685 * common/gstdoc-scangobj:
8686 * common/gtk-doc-plugins.mak:
8687 * common/gtk-doc.mak:
8688 * common/m4/.gitignore:
8689 * common/m4/Makefile.am:
8691 * common/m4/as-ac-expand.m4:
8692 * common/m4/as-auto-alt.m4:
8693 * common/m4/as-compiler-flag.m4:
8694 * common/m4/as-compiler.m4:
8695 * common/m4/as-docbook.m4:
8696 * common/m4/as-libtool-tags.m4:
8697 * common/m4/as-libtool.m4:
8698 * common/m4/as-python.m4:
8699 * common/m4/as-scrub-include.m4:
8700 * common/m4/as-version.m4:
8701 * common/m4/ax_create_stdint_h.m4:
8702 * common/m4/check.m4:
8703 * common/m4/glib-gettext.m4:
8704 * common/m4/gst-arch.m4:
8705 * common/m4/gst-args.m4:
8706 * common/m4/gst-check.m4:
8707 * common/m4/gst-debuginfo.m4:
8708 * common/m4/gst-default.m4:
8709 * common/m4/gst-doc.m4:
8710 * common/m4/gst-error.m4:
8711 * common/m4/gst-feature.m4:
8712 * common/m4/gst-function.m4:
8713 * common/m4/gst-gettext.m4:
8714 * common/m4/gst-glib2.m4:
8715 * common/m4/gst-libxml2.m4:
8716 * common/m4/gst-plugindir.m4:
8717 * common/m4/gst-valgrind.m4:
8718 * common/m4/gtk-doc.m4:
8719 * common/m4/introspection.m4:
8721 * common/mangle-tmpl.py:
8722 * common/plugins.xsl:
8724 * common/release.mak:
8725 * common/scangobj-merge.py:
8726 * common/upload.mak:
8727 common: Remove static version
8729 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
8731 * common/m4/introspection.m4:
8732 Update introspection.m4 to match usage
8734 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8738 Remove old stuff from the README
8740 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8745 === release 0.10.7 ===
8747 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8752 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8754 * examples/test-ogg.c:
8755 test-ogg: remove parsers
8756 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
8757 buffers with timestamps. Using the parsers also seems to break things.
8759 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8761 * bindings/vala/gst-rtsp-server-0.10.vapi:
8762 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8763 Updated Vala bindings
8765 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8767 * common/m4/introspection.m4:
8769 * gst/rtsp-server/Makefile.am:
8770 Added initial gobject-introspection support
8772 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8774 * gst/rtsp-server/rtsp-media-factory.c:
8775 media-factory: don't use host for shared hash key
8776 When we generate the key to share made between connections, don't include the
8777 host used to connect so that we can share media even if between clients that
8778 connected with localhost and ones with the ip address.
8780 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8782 * bindings/vala/Makefile.am:
8783 build: fix distcheck
8785 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8787 * bindings/vala/gst-rtsp-server-0.10.vapi:
8788 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8789 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8790 Update Vala bindings
8792 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8794 * bindings/vala/Makefile.am:
8796 Fix configure checks and installation location for Vala bindings
8799 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8804 === release 0.10.6 ===
8806 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8809 configure: release 0.10.6
8811 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8813 * gst/rtsp-server/rtsp-media.c:
8814 media: help the compiler a little
8816 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8818 * gst/rtsp-server/rtsp-media.c:
8819 * gst/rtsp-server/rtsp-media.h:
8820 * gst/rtsp-server/rtsp-session.c:
8821 media: cleanup media transport before freeing
8822 Cleanup the media transport data before freeing. In particular, remove the qdata
8823 from the rtpsource object.
8825 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8827 * gst/rtsp-server/rtsp-media-factory.c:
8828 * gst/rtsp-server/rtsp-media-factory.h:
8829 * gst/rtsp-server/rtsp-media.c:
8830 * gst/rtsp-server/rtsp-media.h:
8831 media-factory: add eos-shutdown property
8832 Add an eos-shutdown property that will send an EOS to the pipeline before
8833 shutting it down. This allows for nice cleanup in case of a muxer.
8836 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8838 * gst/rtsp-server/rtsp-media.c:
8839 * gst/rtsp-server/rtsp-media.h:
8840 media: use multiudpsink send-duplicates when we can
8841 If we have a new enough multiudpsink with the send-duplicates property, use this
8842 instead of doing our own filtering. Our custom filtering code should eventually
8843 be removed when we can depend on a released -good.
8845 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8847 * gst/rtsp-server/rtsp-media.c:
8848 media: don't leak destinations
8849 Refactor and cleanup the destinations array when the stream is destroyed.
8851 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8853 * gst/rtsp-server/rtsp-media.c:
8854 * gst/rtsp-server/rtsp-media.h:
8855 media: don't add udp addresses multiple times
8856 Keep track of the udp addresses we added to udpsink and never add the same udp
8857 destination twice. This avoids duplicate packets when using multicast.
8859 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8861 * gst/rtsp-server/rtsp-server.c:
8862 server: disable use of SO_LINGER
8863 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
8864 server close()s the connection.
8866 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8868 * gst/rtsp-server/rtsp-server.c:
8869 server: use 5 second linger period in SO_LINGER
8870 Wait 5 seconds before clearing the send buffers and reseting the connection with
8871 the client when we do a close. This should be enough time to get the message to
8875 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
8877 * gst/rtsp-server/rtsp-server.c:
8878 server: use SO_LINGER
8879 SO_LINGER on the socket will make sure that any pending data on the socket is
8880 flushed ASAP and that the socket connection is reset. This makes sure that the
8881 socket can be reused immediately.
8884 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8887 README: add blurb about shared media factories
8889 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
8891 * gst/rtsp-server/rtsp-media.c:
8892 Add stdlib.h for atoi()
8894 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8896 * bindings/python/Makefile.am:
8897 * bindings/vala/Makefile.am:
8898 build: distcheck fixes
8899 Fix 'make distcheck', somewhat (it still fails because it tries to
8900 install files into /usr/share/vala/vapi/ irrespective of the
8903 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8906 configure: bump core/base requirements to released version
8907 Makes things less confusing for people.
8909 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8912 configure: fail if GStreamer core/base requirements are not met
8914 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8916 * gst/rtsp-server/rtsp-client.c:
8917 client: improve client cleanups
8918 Make sure the session does not timeout when using TCP. We need to do this
8919 because quicktime player does not send RTCP for some reason in tunneled
8921 Refactor some cleanup code.
8924 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8926 * gst/rtsp-server/rtsp-session.c:
8927 * gst/rtsp-server/rtsp-session.h:
8928 session: add support for prevent session timeouts
8929 Add an atomix counter to prevent session timeouts when we are, for example,
8932 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8934 * gst/rtsp-server/rtsp-client.c:
8935 client: fix unlink on session timeouts
8936 When our session times out, make sure we unlink all streams in this
8938 Remove the tunnelid when closing the connection.
8940 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8942 * gst/rtsp-server/rtsp-session.c:
8943 session: small cleanups
8945 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8947 * gst/rtsp-server/rtsp-client.c:
8948 client: handle lost_tunnel callbacks
8949 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
8950 hashtable so that we can reuse it for when the client reopens the POST
8952 Close the connection after a TEARDOWN.
8953 Make sure or watchid is cleared when the watch is removed.
8956 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8958 * gst/rtsp-server/rtsp-client.c:
8959 * gst/rtsp-server/rtsp-media.c:
8960 * gst/rtsp-server/rtsp-sdp.c:
8961 rtsp-server: add more support for multicast
8963 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8966 * gst/rtsp-server/rtsp-media.c:
8967 * gst/rtsp-server/rtsp-media.h:
8968 media: allow configuration of allowed lower transport
8970 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8972 * gst/rtsp-server/rtsp-client.h:
8973 * gst/rtsp-server/rtsp-media.c:
8974 * gst/rtsp-server/rtsp-media.h:
8975 * gst/rtsp-server/rtsp-sdp.c:
8976 * gst/rtsp-server/rtsp-sdp.h:
8977 * gst/rtsp-server/rtsp-server.c:
8978 rtsp: keep track of server ip and ipv6
8979 Keep track of how the client connected to the server and setup the udp ports
8980 with the same protocol.
8981 Copy the server ip address in the SDP so that clients can send RTCP back to
8984 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8986 * gst/rtsp-server/rtsp-session.c:
8989 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8991 * gst/rtsp-server/rtsp-client.c:
8992 client: use right size for malloc
8994 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8996 * gst/rtsp-server/rtsp-server.c:
8997 server: comment ipv6 server listening address
8999 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9001 * gst/rtsp-server/rtsp-media.c:
9002 media: allow for ipv6 sockets
9004 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9006 * gst/rtsp-server/rtsp-server.c:
9007 * gst/rtsp-server/rtsp-server.h:
9008 server: rework server part
9009 Allow setting a bind address, make sure we can deal with ipv6.
9010 Remove the port property and change with the service property.
9012 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9014 * gst/rtsp-server/rtsp-media.h:
9015 media: update comments a little
9017 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9019 * gst/rtsp-server/rtsp-client.c:
9020 client: make content-base better
9021 Use the URI formatting functions to make a content-base. Also make sure that
9022 there is a trailing / at the end.
9024 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9026 * gst/rtsp-server/rtsp-client.c:
9027 client: guard against invalid paths
9029 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9031 * examples/test-video.c:
9032 test: catch server bind errors
9034 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
9036 * gst/rtsp-server/rtsp-media.c:
9037 rtspmedia: emit "unprepared" if _prepare fails.
9038 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
9039 media object is removed from its factory's cache.
9041 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9043 * gst/rtsp-server/rtsp-media.c:
9044 media: collect media position when seek completes
9046 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
9048 * gst/rtsp-server/rtsp-client.c:
9049 client: call unlink_streams in client finalize
9052 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9054 * gst/rtsp-server/rtsp-media.c:
9055 media: limit the time to wait to something huge
9056 Avoid waiting forever but limit the timeout to 20 seconds.
9058 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9060 * gst/rtsp-server/rtsp-sdp.c:
9061 sdp: reindent and check for prepared status
9063 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9065 * gst/rtsp-server/rtsp-media.c:
9066 * gst/rtsp-server/rtsp-media.h:
9067 * gst/rtsp-server/rtsp-session.c:
9068 media: avoid doing _get_state() for state changes
9069 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
9070 until the media is prerolled or in error. This avoids doing a blocking call of
9071 gst_element_get_state() that can cause lockups when there is an error.
9074 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9076 * gst/rtsp-server/rtsp-media.c:
9079 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9081 * gst/rtsp-server/rtsp-media-factory.c:
9082 media-factory: better error handling
9083 Improve the error handling a bit.
9085 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9087 * gst/rtsp-server/rtsp-client.c:
9088 client: rework transport parsing
9089 Rework the transport parsing code so that we can ignore transports we don't
9090 support instead of just picking the first one we can parse.
9091 Configure a (for now hardcoded) destination for multicast transports.
9093 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9095 * gst/rtsp-server/rtsp-media.c:
9096 media: set multicast sink parameters
9097 Disable loop and automatic multicast join on the udpsink elements.
9098 Add some more debug info.
9099 Reset some state variables in the right place.
9100 Use the right port numbers for multicast.
9102 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9104 * gst/rtsp-server/rtsp-session.c:
9105 session: handle transport setup correctly
9106 Handle UDP, MCAST and TCP transport negotiation more correctly.
9107 Store the server session SSRC in the transport.
9109 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9111 * gst/rtsp-server/rtsp-client.c:
9112 rtsp-client: implement error_full
9113 Implement error_full to avoid some segfaults when the rtspconnection calls it.
9116 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9119 * gst/rtsp-server/rtsp-client.c:
9120 * gst/rtsp-server/rtsp-server.c:
9121 docs: update docs and comments
9123 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
9125 * gst/rtsp-server/rtsp-sdp.c:
9126 sdp: make server work better when behind a proxy
9128 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9130 * gst/rtsp-server/rtsp-client.c:
9131 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
9133 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9135 * gst/rtsp-server/rtsp-client.c:
9136 * gst/rtsp-server/rtsp-media-factory.c:
9137 * gst/rtsp-server/rtsp-media-mapping.c:
9138 * gst/rtsp-server/rtsp-media.c:
9139 * gst/rtsp-server/rtsp-server.c:
9140 * gst/rtsp-server/rtsp-session-pool.c:
9141 * gst/rtsp-server/rtsp-session.c:
9142 Use GStreamer's debugging subsystem
9144 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9146 * gst/rtsp-server/rtsp-media-factory.c:
9147 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
9149 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9154 === release 0.10.5 ===
9156 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9161 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9164 configure: bump required versions
9166 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
9168 * gst/rtsp-server/rtsp-client.c:
9169 client: call weak-unref on client->sessions from finalize
9172 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9174 * gst/rtsp-server/rtsp-media.c:
9175 media: Fixed crasher where caps got unref'ed too often
9177 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9180 * pkgconfig/.gitignore:
9181 * pkgconfig/Makefile.am:
9182 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
9183 Added pkg-config file to use gst-rtsp-server uninstalled
9185 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9187 * gst/rtsp-server/rtsp-media.c:
9188 media: add some docs
9190 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
9192 * gst/rtsp-server/rtsp-client.c:
9193 rtsp: Use gst_rtsp_watch_send_message().
9194 Use gst_rtsp_watch_send_message() since the old API which used
9195 gst_rtsp_watch_queue_message() has been deprecated.
9197 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9202 === release 0.10.4 ===
9204 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9209 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9211 * gst/rtsp-server/rtsp-client.c:
9212 * gst/rtsp-server/rtsp-session.c:
9213 * gst/rtsp-server/rtsp-session.h:
9214 rtsp: allocate channels in TCP mode
9215 When the client does not provide us with channels in TCP mode, allocate channels
9218 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9220 * gst/rtsp-server/rtsp-client.c:
9221 client: don't crash when tunnelid is missing
9222 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
9223 don't crash but return an error response to the client.
9226 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9228 * bindings/vala/gst-rtsp-server-0.10.vapi:
9229 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9230 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9231 bindings: update vala bindings with new method
9233 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9235 * gst/rtsp-server/rtsp-session-pool.c:
9236 * gst/rtsp-server/rtsp-session-pool.h:
9237 sessionpool: add function to filter sessions
9238 Add generic function to retrieve/remove sessions.
9240 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9243 configure: bump core/base requirements to release
9245 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9247 * gst/rtsp-server/rtsp-media.c:
9248 media: fix indentation
9250 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9252 * gst/rtsp-server/rtsp-media.c:
9253 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
9255 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9257 * gst/rtsp-server/rtsp-media.c:
9258 set state and remove elements of media in for loop
9260 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
9262 * bindings/vala/gst-rtsp-server-0.10.vapi:
9263 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9264 Added gst_rtsp_media_remove_elements function to Vala bindings
9266 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
9268 * gst/rtsp-server/rtsp-media.c:
9269 * gst/rtsp-server/rtsp-media.h:
9270 Added gst_rtsp_media_remove_elements function
9272 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
9274 * gst/rtsp-server/rtsp-media.c:
9275 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
9277 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9279 * bindings/vala/gst-rtsp-server-0.10.vapi:
9280 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9281 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9282 Updated Vala bindings
9284 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9286 * gst/rtsp-server/rtsp-media.c:
9287 * gst/rtsp-server/rtsp-media.h:
9288 Added vmethod unprepare to GstRTSPMedia
9289 The default implementation sets the state of the pipeline to GST_STATE_NULL
9291 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9293 * gst/rtsp-server/rtsp-media-factory.c:
9294 * gst/rtsp-server/rtsp-media-factory.h:
9295 Made collect_streams function public
9297 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9299 * gst/rtsp-server/rtsp-media-factory.c:
9300 * gst/rtsp-server/rtsp-media-factory.h:
9301 * gst/rtsp-server/rtsp-media.c:
9302 Added vmethod create_pipeline to GstRTSPMediaFactory
9303 The pipeline is created in this method and the GstRTSPMedia's element is added to it
9305 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9307 * gst/rtsp-server/rtsp-client.c:
9308 client: use g_source_destroy()
9309 We need to use g_source_destroy() because we might have added the source to a
9310 different main context than the default one.
9312 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9314 * gst/rtsp-server/Makefile.am:
9315 * gst/rtsp-server/rtsp-client.c:
9316 * gst/rtsp-server/rtsp-params.c:
9317 * gst/rtsp-server/rtsp-params.h:
9318 rtsp: prepare for handling GET/SET_PARAMETER
9319 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
9321 Fix return codes of handlers.
9323 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9325 * gst/rtsp-server/rtsp-media.c:
9326 media: don't leak session pads
9328 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9330 * gst/rtsp-server/rtsp-media.c:
9331 media: clean up the messages a bit
9333 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9335 * gst/rtsp-server/rtsp-sdp.c:
9336 sdp: warn and skip streams without media
9338 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9340 * bindings/vala/gst-rtsp-server-0.10.vapi:
9341 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9342 vala: Fixed typo in header file of RTSPMediaStream
9344 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9346 * gst/rtsp-server/rtsp-media.c:
9349 Make dumping RTCP stats configurable
9351 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9353 * gst/rtsp-server/rtsp-media.c:
9354 media: be less verbose and leak less
9356 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9358 * gst/rtsp-server/rtsp-media.c:
9359 media: don't leak the destination address
9361 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9363 * gst/rtsp-server/rtsp-client.c:
9364 * gst/rtsp-server/rtsp-media.c:
9365 * gst/rtsp-server/rtsp-media.h:
9366 * gst/rtsp-server/rtsp-session.c:
9367 * gst/rtsp-server/rtsp-session.h:
9368 rtsp: use RTCP to keep the session alive
9369 Use the RTCP rtcp-from stats field to find the associated session and use this
9370 to keep the session alive.
9372 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9374 * gst/rtsp-server/rtsp-session.c:
9375 session: add 5sec to the real session timeout
9376 Allow the session to live 5sec longer before really timing out. This should give
9377 clients some extra time to keep the session active.
9379 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9381 * gst/rtsp-server/rtsp-client.c:
9382 client: replay OK to GET/SET_PARAMETER
9383 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
9384 so that we return OK for those requests.
9386 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9388 * gst/rtsp-server/rtsp-media.c:
9389 * gst/rtsp-server/rtsp-media.h:
9390 media: keep track of active transports
9391 Keep track of which transport is active to avoid closing the connection too
9393 Remove the destination transport also when going to NULL.
9394 Print some stats about the SDES and other RTCP messages we receive from the
9397 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9399 * examples/.gitignore:
9400 * examples/Makefile.am:
9401 * examples/test-sdp.c:
9402 example: add SDP relay example
9404 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9406 * gst/rtsp-server/rtsp-media.c:
9407 media: also count active TCP connections
9409 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9411 * gst/rtsp-server/rtsp-media-factory.c:
9412 * gst/rtsp-server/rtsp-media.c:
9413 * gst/rtsp-server/rtsp-media.h:
9414 rtsp: add support for dynamic elements
9415 Add support for dynamic elements.
9416 Don't set live pipelines back to paused.
9418 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9420 * gst/rtsp-server/rtsp-sdp.c:
9421 sdp: don't add encoding name when absent in caps
9423 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9425 * gst/rtsp-server/rtsp-client.c:
9426 client: warn when we can't do RTP-Info
9428 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9430 * gst/rtsp-server/rtsp-media-factory.c:
9431 factory: factor out the stream construction
9433 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9435 * gst/rtsp-server/rtsp-client.c:
9436 client: only add RTP-Info when we have the info
9437 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
9440 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9445 === release 0.10.3 ===
9447 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9451 - Fixes a bug where it put the wrong verion in pkgconfig
9452 - Link RTP and RTCP sources
9454 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9456 * gst/rtsp-server/rtsp-media.c:
9457 * gst/rtsp-server/rtsp-media.h:
9458 media: link the RTP udpsrc to the session manager
9459 Link the RTP udpsrc and the appsrc to the session manager so that they don't
9460 shut down when the client sends a packet to open firewalls.
9462 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9464 * pkgconfig/gst-rtsp-server.pc.in:
9465 Don't use hard-coded version number in pkg-config file
9467 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9472 === release 0.10.2 ===
9474 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9479 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9482 * common/m4/.gitignore:
9483 * examples/.gitignore:
9484 * pkgconfig/.gitignore:
9485 add some .gitignore files
9487 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9489 * gst/rtsp-server/rtsp-media.c:
9490 media: seek to key frames
9492 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9494 * gst/rtsp-server/rtsp-media.c:
9495 media: emit the unprepared signal by id
9496 Emit the unprepared signal by id instead of name and set the media as
9499 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9501 * gst/rtsp-server/rtsp-media.c:
9502 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
9504 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9506 * gst/rtsp-server/rtsp-server.c:
9507 Added finalize function to GstRTPSPServer to unref session pool and media mapping
9509 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9511 * bindings/vala/gst-rtsp-server-0.10.vapi:
9512 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9513 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9514 Updated vala bindings
9516 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9518 * gst/rtsp-server/Makefile.am:
9519 * gst/rtsp-server/rtsp-client.c:
9520 * gst/rtsp-server/rtsp-media.c:
9521 server: use appsink and appsrc with the API
9522 Use the appsink/appsrc API instead of the signals for higher
9525 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9527 * examples/test-ogg.c:
9528 tests: set the payload type correctly
9530 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9532 * gst/rtsp-server/rtsp-media-factory.c:
9533 factory: connect to the unprepare signal
9534 Connect to the unprepare signal for non-reusable media so that we can remove
9535 them from the cache.
9537 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9539 * gst/rtsp-server/rtsp-media.c:
9540 * gst/rtsp-server/rtsp-media.h:
9541 media: add signal to notify of unprepare
9543 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9545 * gst/rtsp-server/rtsp-media.c:
9546 * gst/rtsp-server/rtsp-media.h:
9547 media: more work on making the media shared
9548 Add a reusable flag to medias, indicating that they can be reused after a state
9552 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9554 * examples/test-readme.c:
9555 examples: mark the example as shared for testing
9557 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9559 * gst/rtsp-server/rtsp-media.c:
9560 * gst/rtsp-server/rtsp-media.h:
9561 client: support shared media
9562 Always perform the state actions even if the target state of the pipeline is
9563 already correct, we still want to add/remove the transports when we are dealing
9565 Keep a counter of the number of active transports for a media so that we can use
9566 this to perform a state change when needed.
9567 Perform a state change of the pipeline only when the first transport was added
9568 or when there are no active transports.
9570 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9572 * gst/rtsp-server/rtsp-client.c:
9573 client: fix refcounting crasher
9574 Don't need to remove the weak refs in the finalize methods, they are already
9575 removed in the dispose.
9576 Don't register the callback with a DestroyNofity.
9578 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9580 * gst/rtsp-server/rtsp-client.c:
9581 Fix rtsp client refcount management in TCP mode.
9582 Don't unref a client ref we never had. Fixes an unref
9583 of an already-free client object after a client
9584 teardown request for me.
9586 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9588 * gst/rtsp-server/rtsp-session.c:
9589 docs: fix typo in API docs
9591 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9593 * gst/rtsp-server/rtsp-media.c:
9595 Keep the udp sources in playing even if we go to paused. unlock the sources when
9597 Add some more debug info.
9598 Only seek when we need to.
9599 Keep track of the position when we go to paused.
9601 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9603 * gst/rtsp-server/rtsp-client.c:
9604 * gst/rtsp-server/rtsp-media.c:
9605 * gst/rtsp-server/rtsp-media.h:
9606 Add beginnings of seeking.
9607 Parse the Range header and perform a seek on the pipeline for the requested
9608 position. It's disabled currently until I figure out what's going wrong.
9610 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9612 * gst/rtsp-server/rtsp-client.c:
9613 allow pause requests for now.
9616 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9618 * gst/rtsp-server/rtsp-client.c:
9619 Remove weak ref on the session in teardown
9620 We need to remove our weakref from the session when we do a teardown because
9621 else we close the TCP connection prematurely.
9623 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9625 * gst/rtsp-server/rtsp-client.c:
9626 * gst/rtsp-server/rtsp-client.h:
9627 * gst/rtsp-server/rtsp-session-pool.c:
9628 Do some more session cleanup
9629 Make session timeout kill the TCP connection that currently watches the
9631 Remove the client timeout property.
9633 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9635 * gst/rtsp-server/rtsp-client.c:
9636 * gst/rtsp-server/rtsp-client.h:
9637 * gst/rtsp-server/rtsp-media.c:
9638 * gst/rtsp-server/rtsp-media.h:
9639 * gst/rtsp-server/rtsp-server.c:
9640 * gst/rtsp-server/rtsp-session.c:
9641 * gst/rtsp-server/rtsp-session.h:
9643 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
9646 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9648 * examples/Makefile.am:
9649 * examples/test-launch.c:
9650 Add example server that takes launch lines
9651 Add an example server that streams any -launch line.
9653 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9655 * examples/test-readme.c:
9656 * gst/rtsp-server/rtsp-client.c:
9657 * gst/rtsp-server/rtsp-media.c:
9658 * gst/rtsp-server/rtsp-media.h:
9659 Add support for live streams
9660 Add support for live streams and ranges
9661 Start on handling TCP data transfer.
9663 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9665 * gst/rtsp-server/rtsp-media.c:
9666 Free the pipeline before other things
9669 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9671 * gst/rtsp-server/rtsp-client.c:
9672 Only free the pending tunnel if there is one
9675 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9677 * gst/rtsp-server/rtsp-client.c:
9678 * gst/rtsp-server/rtsp-client.h:
9679 * gst/rtsp-server/rtsp-media.c:
9680 rtsp-server: Add support for tunneling
9681 Add support for tunneling over HTTP.
9682 Use new connection methods to retrieve the url.
9683 Dispatch messages based on the message type instead of blindly
9684 assuming it's always a request.
9685 Keep track of the watch id so that we can remove it later.
9686 Set the media pipeline to NULL before unreffing the pipeline.
9688 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9690 * gst/rtsp-server/rtsp-client.c:
9691 * gst/rtsp-server/rtsp-client.h:
9692 Fix for channel -> watch rename in gstreamer
9693 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
9695 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9697 * gst/rtsp-server/rtsp-client.c:
9698 * gst/rtsp-server/rtsp-client.h:
9700 Use the async RTSP channels instead of spawning a new thread for each client.
9701 If a sessionid is specified in a request, fail if we don't have the session.
9703 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9705 * gst/rtsp-server/rtsp-media.c:
9706 Add better debug info
9707 Add some better debug info.
9709 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9711 * examples/test-video.c:
9713 Add support for session timeouts in the example.
9715 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9717 * gst/rtsp-server/rtsp-session-pool.c:
9718 * gst/rtsp-server/rtsp-session-pool.h:
9719 Pass GTimeVal around for performance reasons
9720 Get the current time only once and pass it around so that sessions don't have to
9721 get the current time anymore.
9722 Add experimental support for a GSource that dispatches when the session needs to
9725 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9727 * gst/rtsp-server/rtsp-session.c:
9728 * gst/rtsp-server/rtsp-session.h:
9729 Add better support for session timeouts
9730 Add a method to request the number of milliseconds when a session will timeout.
9732 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9734 * gst/rtsp-server/rtsp-media.c:
9735 * gst/rtsp-server/rtsp-media.h:
9736 Add suport for RTP manager monitoring
9737 Add the first stage in monitoring the rtp manager.
9738 Make sure we don't update the state to something we don't want.
9740 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9742 * gst/rtsp-server/rtsp-client.c:
9743 Add support for session keepalive
9744 Get and update the session timeout for all requests. get the session as early as
9747 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9749 * gst/rtsp-server/rtsp-media-factory.h:
9750 * gst/rtsp-server/rtsp-media.c:
9751 * gst/rtsp-server/rtsp-media.h:
9752 Handle media bus messages
9753 Handle media bus messages in a custom mainloop and dispatch them to the
9754 RTSPMedia objects. Let the default implementation handle some common messages.
9756 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9758 * gst/rtsp-server/rtsp-client.c:
9759 * gst/rtsp-server/rtsp-session-pool.c:
9760 * gst/rtsp-server/rtsp-session.c:
9761 Some more session timeout handling
9762 Move the session header setting code to a central place so that we always add
9763 the timeout parameter too.
9764 Handle timeouts by running the session cleanup code.
9765 Stop media before cleaning up.
9767 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9769 * gst/rtsp-server/rtsp-client.c:
9770 * gst/rtsp-server/rtsp-client.h:
9771 Add timeout property
9772 Add a timeout property ot the client and make the other properties into GObject
9775 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9777 * gst/rtsp-server/rtsp-session-pool.c:
9778 Use getters and setters in property code
9779 Use the getters and setters for the timeout property instead of locking
9782 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9784 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
9786 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9788 * gst/rtsp-server/rtsp-session-pool.c:
9789 * gst/rtsp-server/rtsp-session-pool.h:
9790 * gst/rtsp-server/rtsp-session.c:
9791 * gst/rtsp-server/rtsp-session.h:
9792 Add more timeout stuff
9793 Add method to check if a session is expired.
9794 Add method to perform cleanup on a session pool.
9796 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9798 * gst/rtsp-server/rtsp-client.c:
9799 * gst/rtsp-server/rtsp-session-pool.c:
9800 * gst/rtsp-server/rtsp-session-pool.h:
9801 * gst/rtsp-server/rtsp-session.c:
9802 * gst/rtsp-server/rtsp-session.h:
9803 Add beginnings of session timeouts and limits
9804 Add the timeout value to the Session header for unusual timeout values.
9805 Allow us to configure a limit to the amount of active sessions in a pool. Set a
9806 limit on the amount of retry we do after a sessionid collision.
9807 Add properties to the sessionid and the timeout of a session. Keep track of
9808 creation time and last access time for sessions.
9810 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9812 * gst/rtsp-server/rtsp-client.c:
9813 * gst/rtsp-server/rtsp-media.c:
9814 * gst/rtsp-server/rtsp-media.h:
9815 * gst/rtsp-server/rtsp-sdp.c:
9816 * gst/rtsp-server/rtsp-session-pool.c:
9817 * gst/rtsp-server/rtsp-session.c:
9818 * gst/rtsp-server/rtsp-session.h:
9819 Cleanup of sessions and more
9820 Fix the refcounting of media and sessions in the client. Properly clean up the
9821 session data when the client performs a teardown.
9822 Add Server header to responses.
9823 Allow for multiple uri setups in one session.
9824 Add Range header to the PLAY response and add the range attribute to the SDP
9826 Fix the session pool remove method, it used the wrong key in the hashtable. Also
9827 give the ownership of the sessionid to the session object.
9829 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9831 * gst/rtsp-server/rtsp-server.c:
9832 * gst/rtsp-server/rtsp-server.h:
9834 Rename the 'server_port' variable to simply 'port'.
9836 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9839 * gst/rtsp-server/rtsp-client.c:
9840 * gst/rtsp-server/rtsp-media.c:
9841 * gst/rtsp-server/rtsp-media.h:
9842 * gst/rtsp-server/rtsp-session.c:
9843 * gst/rtsp-server/rtsp-session.h:
9844 Rework the way we handle transports for streams
9845 Make the media accept an array of transports for the streams that we have
9846 configured for the play/pause requests.
9847 Implement server states for a client and its media.
9848 Require 0.10.22.1 (git HEAD) of gstreamer.
9850 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9852 * gst/rtsp-server/rtsp-client.c:
9853 * gst/rtsp-server/rtsp-media-factory.c:
9854 Drop const from functions dealing with urls
9855 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
9856 have the right const in them.
9858 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9860 * gst/rtsp-server/rtsp-client.c:
9861 * gst/rtsp-server/rtsp-media.c:
9862 * gst/rtsp-server/rtsp-sdp.c:
9866 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9868 * gst/rtsp-server/rtsp-client.c:
9869 * gst/rtsp-server/rtsp-media-factory.c:
9870 * gst/rtsp-server/rtsp-media.c:
9871 * gst/rtsp-server/rtsp-media.h:
9873 Don't keep a reference to the GstRTSPMedia in the stream.
9874 Free more things when freeing the GstRTSPMedia.
9876 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9879 * gst/rtsp-server/rtsp-media-factory.c:
9880 * gst/rtsp-server/rtsp-media-factory.h:
9881 * gst/rtsp-server/rtsp-media.c:
9882 * gst/rtsp-server/rtsp-media.h:
9883 * gst/rtsp-server/rtsp-server.c:
9884 * gst/rtsp-server/rtsp-server.h:
9885 More docs and small cleanups
9886 Add some more docs and update the README
9887 Cleanup some method names.
9888 Remove an unneeded idx field in the GstRTSPMediaStream
9890 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9893 * examples/Makefile.am:
9894 * examples/test-readme.c:
9895 Add a README and more example code
9896 Add a README file that contains a small introduction on how to use the server
9897 along with the example code explained in the readme.
9899 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9901 * gst/rtsp-server/rtsp-media.c:
9902 * gst/rtsp-server/rtsp-server.c:
9903 Fix some leaks and change default port
9904 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
9905 we finished the initial preroll. If we keep them locked, setting the pipeline to
9906 NULL will not stop and clean up the sources correctly.
9907 Change the default RTSP port to 8554 aka the official alternative RTSP port.
9909 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9911 * gst/rtsp-server/rtsp-session.c:
9912 * gst/rtsp-server/rtsp-session.h:
9913 Cleanups to the session object
9914 Remove some unneeded variables in the session state of a stream such as the
9915 owner media and the server transport.
9916 Get the configuration of a media stream in a session based on the media_stream
9917 in the original object instead of our cached index.
9918 Free more data in the finalize method.
9920 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9922 * gst/rtsp-server/rtsp-client.c:
9923 * gst/rtsp-server/rtsp-client.h:
9924 Cleanups and reuse media from DESCRIBE
9925 Handle thread create errors.
9926 Rename some internal methods to better match what they actually do.
9927 Handle misconfiguration of session_pool and media_mapping gracefully.
9928 Cache the DESCRIBE media and uri in the client connection and reuse them when
9929 we receive a SETUP request in the same connection for the same uri.
9930 Cleanup the client connection object.
9932 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9934 * gst/rtsp-server/rtsp-media-factory.c:
9935 * gst/rtsp-server/rtsp-media-factory.h:
9936 * gst/rtsp-server/rtsp-media.c:
9937 * gst/rtsp-server/rtsp-media.h:
9938 Add shared properties to media and factory
9939 Add the shared property to media.
9940 Implement some simple caching in the factory depending on if the media is shared
9943 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9945 * gst/rtsp-server/rtsp-client.c:
9946 Add a little comment
9947 Add some comment about the content-base header.
9949 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9951 * examples/Makefile.am:
9952 * examples/test-mp4.c:
9953 * examples/test-ogg.c:
9954 * examples/test-video.c:
9955 * gst/rtsp-server/Makefile.am:
9956 * gst/rtsp-server/rtsp-client.c:
9957 * gst/rtsp-server/rtsp-client.h:
9958 * gst/rtsp-server/rtsp-media-factory.c:
9959 * gst/rtsp-server/rtsp-media-factory.h:
9960 * gst/rtsp-server/rtsp-media.c:
9961 * gst/rtsp-server/rtsp-media.h:
9962 * gst/rtsp-server/rtsp-sdp.c:
9963 * gst/rtsp-server/rtsp-sdp.h:
9964 * gst/rtsp-server/rtsp-server.c:
9965 * gst/rtsp-server/rtsp-server.h:
9966 * gst/rtsp-server/rtsp-session.c:
9967 * gst/rtsp-server/rtsp-session.h:
9968 Reorganize things, prepare for media sharing
9969 Added various other test server examples
9970 Move the SDP message generation to a separate helper.
9971 Refactor common code for finding the session.
9972 Add content-base for realplayer compatibility
9973 Clean up request uris before processing for better vlc compatibility.
9974 Move prerolling and pipeline construction to the RTSPMedia object.
9975 Use multiudpsink for future pipeline reuse.
9977 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9983 === release 0.10.1 ===
9985 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9991 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9993 * bindings/vala/Makefile.am:
9995 Add more directories and files to the dist.
9997 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9999 * bindings/python/Makefile.am:
10000 * bindings/python/rtspserver.override:
10001 Fixed compile error of python bindings
10003 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10005 * bindings/vala/gst-rtsp-server-0.10.vapi:
10006 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
10007 Marked values as nullable accordingly
10009 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10011 * bindings/vala/gst-rtsp-server-0.10.vapi:
10012 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
10013 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
10014 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
10015 Updated Vala bindings
10017 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10019 * gst/rtsp-server/rtsp-client.c:
10020 * gst/rtsp-server/rtsp-media-mapping.c:
10021 * gst/rtsp-server/rtsp-media-mapping.h:
10022 * gst/rtsp-server/rtsp-media.h:
10023 * gst/rtsp-server/rtsp-session-pool.h:
10024 Cleanups and doc updates
10025 Add some more documentation and do some minor cleanups here and there.
10027 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10029 * gst/rtsp-server/rtsp-client.c:
10030 * gst/rtsp-server/rtsp-media-factory.c:
10031 * gst/rtsp-server/rtsp-media-factory.h:
10032 * gst/rtsp-server/rtsp-media.c:
10033 * gst/rtsp-server/rtsp-media.h:
10034 * gst/rtsp-server/rtsp-session.c:
10035 * gst/rtsp-server/rtsp-session.h:
10037 Rename GstRTSPMediaBin to GstRTSPMedia
10038 Parse the request url into a GstRTSPUri object and pass this object to the
10039 various handlers and methods that require the uri.
10041 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10045 Add some more docs and remove some old code from the example.
10047 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10049 * gst/rtsp-server/rtsp-client.c:
10050 Handle state change failures better
10051 Handle state change failures better when changing the state of the pipeline to
10054 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10056 * gst/rtsp-server/rtsp-media-factory.c:
10057 * gst/rtsp-server/rtsp-media-factory.h:
10058 Make element creation more extendible
10059 Add get_element vmethod to the default MediaFactory so that subclasses can just
10060 override that method and still use the default logic for making a MediaBin from
10063 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10066 * gst/rtsp-server/Makefile.am:
10067 * gst/rtsp-server/rtsp-client.c:
10068 * gst/rtsp-server/rtsp-client.h:
10069 * gst/rtsp-server/rtsp-media-factory.c:
10070 * gst/rtsp-server/rtsp-media-factory.h:
10071 * gst/rtsp-server/rtsp-media-mapping.c:
10072 * gst/rtsp-server/rtsp-media-mapping.h:
10073 * gst/rtsp-server/rtsp-media.c:
10074 * gst/rtsp-server/rtsp-media.h:
10075 * gst/rtsp-server/rtsp-server.c:
10076 * gst/rtsp-server/rtsp-server.h:
10077 * gst/rtsp-server/rtsp-session.c:
10078 * gst/rtsp-server/rtsp-session.h:
10079 Make the server handle arbitrary pipelines
10080 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
10081 The GstMediaBin object has a handle to a bin with elements and to a list of
10082 GstMediaStream objects that this bin produces.
10083 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
10084 with methods to register and remove those mappings.
10085 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
10086 used by the server instance.
10087 Modify the example application so that it shows how to create custom pipelines
10088 attached to a specific mount point.
10089 Various misc cleanps.
10091 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10093 * gst/rtsp-server/rtsp-server.c:
10094 * gst/rtsp-server/rtsp-server.h:
10095 Allow setting a custom media factory for a server
10097 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10099 * gst/rtsp-server/rtsp-client.c:
10100 * gst/rtsp-server/rtsp-client.h:
10101 Allow setting a custom media factory for a client.
10103 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10105 * gst/rtsp-server/Makefile.am:
10106 Add Makefile entry for the media factory
10108 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10110 * gst/rtsp-server/rtsp-media-factory.c:
10111 * gst/rtsp-server/rtsp-media-factory.h:
10112 Add media factory to map urls to media pipeline objects.
10114 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10116 * gst/rtsp-server/rtsp-media.c:
10117 * gst/rtsp-server/rtsp-media.h:
10118 Add comments. Remove unused field
10120 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10122 * gst/rtsp-server/rtsp-session-pool.c:
10123 * gst/rtsp-server/rtsp-session-pool.h:
10124 Allow custom session pools to override the session id allocation algorithms Add some comments.
10126 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10128 * gst/rtsp-server/rtsp-session.h:
10131 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10133 * gst/rtsp-server/rtsp-client.c:
10134 * gst/rtsp-server/rtsp-client.h:
10135 Move the connection code in one place Add some comments
10137 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10139 * gst/rtsp-server/rtsp-server.c:
10140 * gst/rtsp-server/rtsp-server.h:
10141 Make vmethod to create and accept new clients. Add some docs.
10143 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10145 * gst/rtsp-server/rtsp-server.c:
10146 * gst/rtsp-server/rtsp-server.h:
10147 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
10149 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10151 * gst/rtsp-server/rtsp-client.c:
10152 * gst/rtsp-server/rtsp-client.h:
10153 Name the parameters more appropriately.
10155 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10157 * gst/rtsp-server/rtsp-session-pool.c:
10158 Do some more cleanup of the session pool.
10160 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10162 * gst/rtsp-server/Makefile.am:
10163 * gst/rtsp-server/rtsp-client.c:
10164 Check if return value of gst_rtsp_session_get_media is not NULL
10166 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10168 * gst/rtsp-server/Makefile.am:
10169 Install rtsp-session and rtsp-session-pool headers
10171 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10176 * bindings/python/Makefile.am:
10177 * bindings/python/arg-types.py:
10178 * bindings/python/codegen/Makefile.am:
10179 * bindings/python/codegen/__init__.py:
10180 * bindings/python/codegen/argtypes.py:
10181 * bindings/python/codegen/code-coverage.py:
10182 * bindings/python/codegen/codegen.py:
10183 * bindings/python/codegen/definitions.py:
10184 * bindings/python/codegen/defsparser.py:
10185 * bindings/python/codegen/docextract.py:
10186 * bindings/python/codegen/docgen.py:
10187 * bindings/python/codegen/fileprefix.override:
10188 * bindings/python/codegen/fileprefixmodule.c:
10189 * bindings/python/codegen/h2def.py:
10190 * bindings/python/codegen/mergedefs.py:
10191 * bindings/python/codegen/mkskel.py:
10192 * bindings/python/codegen/override.py:
10193 * bindings/python/codegen/reversewrapper.py:
10194 * bindings/python/codegen/scmexpr.py:
10195 * bindings/python/rtspserver-types.defs:
10196 * bindings/python/rtspserver.defs:
10197 * bindings/python/rtspserver.override:
10198 * bindings/python/rtspservermodule.c:
10200 Add python bindings.
10202 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10204 * bindings/Makefile.am:
10206 Don't go into python dir when requirements for python bindings are missing
10208 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10210 * bindings/Makefile.am:
10211 * bindings/vala/Makefile.am:
10213 Install Vala bindings if vala is available
10215 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10217 * bindings/vala/gst-rtsp-server-0.10.deps:
10218 * bindings/vala/gst-rtsp-server-0.10.vapi:
10219 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
10220 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
10221 * bindings/vala/packages/gst-rtsp-server-0.10.files:
10222 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
10223 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
10224 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
10225 Regenerated Vala bindings
10227 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10229 * bindings/vala/gst-rtsp-server.vapi:
10230 * bindings/vala/packages/gst-rtsp-server.metadata:
10231 Fixed typo in included headers for vala bindings
10233 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10237 * pkgconfig/Makefile.am:
10238 * pkgconfig/gst-rtsp-server.pc.in:
10239 Added pkgconfig file
10241 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
10243 * bindings/vala/gst-rtsp-server.vapi:
10244 * bindings/vala/packages/gst-rtsp-server.excludes:
10245 * bindings/vala/packages/gst-rtsp-server.gi:
10246 * bindings/vala/packages/gst-rtsp-server.metadata:
10247 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
10249 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
10251 * bindings/vala/gst-rtsp-server.vapi:
10252 * bindings/vala/packages/gst-rtsp-server.deps:
10253 * bindings/vala/packages/gst-rtsp-server.files:
10254 * bindings/vala/packages/gst-rtsp-server.gi:
10255 * bindings/vala/packages/gst-rtsp-server.metadata:
10256 * bindings/vala/packages/gst-rtsp-server.namespace:
10257 Added Vala bindings
10259 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
10261 * gst/rtsp-server/rtsp-session.c:
10262 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
10264 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
10266 * examples/Makefile.am:
10267 * gst/rtsp-server/Makefile.am:
10268 Put GStreamer version in library name
10270 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10272 * examples/Makefile.am:
10273 * gst/rtsp-server/Makefile.am:
10274 Fix some issues to pass distcheck
10276 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10278 * gst/rtsp-server/rtsp-server.c:
10279 Added port property to GstRTSPServer class.
10281 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10286 * examples/Makefile.am:
10289 * gst/rtsp-server/Makefile.am:
10290 * gst/rtsp-server/rtsp-client.c:
10291 * gst/rtsp-server/rtsp-client.h:
10292 * gst/rtsp-server/rtsp-media.c:
10293 * gst/rtsp-server/rtsp-media.h:
10294 * gst/rtsp-server/rtsp-server.c:
10295 * gst/rtsp-server/rtsp-server.h:
10296 * gst/rtsp-server/rtsp-session-pool.c:
10297 * gst/rtsp-server/rtsp-session-pool.h:
10298 * gst/rtsp-server/rtsp-session.c:
10299 * gst/rtsp-server/rtsp-session.h:
10301 Split in library and example program
10303 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
10305 * src/rtsp-client.h:
10306 Removed obsolete variable
10308 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
10310 * src/rtsp-client.c:
10311 * src/rtsp-client.h:
10312 Removed pipeline variable GstRTSPClient, because it's only used in one function
10314 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10316 * src/rtsp-media.c:
10317 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
10319 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
10321 * src/rtsp-session.c:
10322 Initialize some more vars.
10324 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
10326 * src/rtsp-session.c:
10327 Initialize variable to avoid compiler warning.
10329 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
10332 Add a reasonable generic .gitignore