3 2016-11-01 Sebastian Dröge <slomo@coaxion.net>
8 2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com>
10 * tests/check/gst/rtspserver.c:
11 * tests/check/gst/stream.c:
12 tests: try to avoid using the same ports in different tests
13 Causes problems with client multicast tests otherwise if
14 tests are run in parallel.
15 https://bugzilla.gnome.org/show_bug.cgi?id=773640
17 2016-10-28 17:50:59 +0100 Tim-Philipp Müller <tim@centricular.com>
19 * tests/check/gst/client.c:
20 tests: client: use fail_unless_equals_foo() for better failure reporting
22 2016-09-26 11:16:04 +0200 Göran Jönsson <goranjn@axis.com>
24 * gst/rtsp-server/rtsp-client.c:
25 rtsp-client: Session filter in unwatch session
26 Call session filter with filter_session_media as paramer in
27 client_unwatch_session if using drop_backlog = FALSE.
28 In client_unwatch_session its allowed to grow the watchs backlog.
29 If using drop_backlog = FALSE and the backlog is full it will cause
30 a deadlock when setting session media state to NULL
31 if the backlog is not allowed to grow.
32 https://bugzilla.gnome.org/show_bug.cgi?id=771983
34 2016-10-20 21:40:18 +0100 Tim-Philipp Müller <tim@centricular.com>
37 meson: add fallbacks for gst modules
40 2016-09-14 17:48:39 +0300 Nikita Bobkov <NikitaDBobkov@gmail.com>
42 * gst/rtsp-server/rtsp-client.c:
43 rtsp-client: Fix factory leaking in find_media() in error cases
44 https://bugzilla.gnome.org/show_bug.cgi?id=771488
46 2016-10-06 11:47:50 -0400 Xavier Claessens <xavier.claessens@collabora.com>
48 * gst/rtsp-server/rtsp-stream.c:
49 stream: Fix randomly missing streams from SDP with dynamic elements
50 When using dynamic elements, gst_rtsp_stream_join_bin() is called from
51 "pad-added" signal. In that case priv->srcpad could already have its caps,
52 and they'll be sent to priv->send_src[0] pad. That means that when it
53 connects "notify::caps" signal, that pad could already have received its
54 caps and the signal won't be emitted anymore.
55 In that case priv->caps stay to NULL and when building the SDP that stream
56 gets ignored. Leading to missing video or audio when playing in client side.
57 https://bugzilla.gnome.org/show_bug.cgi?id=772478
59 2016-09-30 11:42:08 +0100 Tim-Philipp Müller <tim@centricular.com>
64 === release 1.9.90 ===
66 2016-09-30 13:04:12 +0300 Sebastian Dröge <sebastian@centricular.com>
72 * gst-rtsp-server.doap:
75 2016-09-17 13:17:19 +0100 Ian Jamison <ian.dev@arkver.com>
77 * gst/rtsp-server/rtsp-media-factory.c:
78 * gst/rtsp-server/rtsp-media.c:
79 * gst/rtsp-server/rtsp-stream.c:
80 rtsp-server: Hint that set_multicast_iface expects the name of the interface
81 To prevent any possibly confusion with IPs or anything else.
82 https://bugzilla.gnome.org/show_bug.cgi?id=771530
84 2016-09-18 09:58:55 -0400 Sebastian Dröge <sebastian@centricular.com>
86 * gst/rtsp-server/rtsp-media-factory.c:
87 * gst/rtsp-server/rtsp-media.c:
88 rtsp-media: Call g_free() instead of g_object_unref() on multicast-iface strings
89 https://bugzilla.gnome.org/show_bug.cgi?id=763000#c5
91 2016-09-14 11:31:15 +0200 Sebastian Dröge <sebastian@centricular.com>
94 configure: Depend on gstreamer 1.9.2.1
96 2016-09-10 20:52:31 +1000 Jan Schmidt <jan@centricular.com>
100 Automatic update of common submodule
101 From b18d820 to f980fd9
103 2016-09-10 09:58:31 +1000 Jan Schmidt <jan@centricular.com>
107 Automatic update of common submodule
108 From 6f2d209 to b18d820
110 2016-09-07 18:44:34 +0300 Sebastian Dröge <sebastian@centricular.com>
112 * gst/rtsp-server/rtsp-stream.c:
113 rtsp-stream: Remove unused _locked() variant of a function
114 It was added during refactoring.
116 2016-09-07 10:21:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
118 * gst/rtsp-server/rtsp-stream.c:
119 stream: cosmetic cleanup
120 https://bugzilla.gnome.org/show_bug.cgi?id=766612
122 2016-09-07 10:16:19 -0400 Xavier Claessens <xavier.claessens@collabora.com>
124 * gst/rtsp-server/rtsp-stream.c:
125 stream: Compare IP addresses case insensitive in more places
126 https://bugzilla.gnome.org/show_bug.cgi?id=766612
128 2016-09-07 10:12:18 -0400 Xavier Claessens <xavier.claessens@collabora.com>
131 * gst/rtsp-server/rtsp-stream.c:
132 stream: Fix leaked joined_bin
133 There is no need to keep a strong ref on it, and _leave_bin() was
134 setting it to NULL before calling g_clear_object() so it was leaked.
135 https://bugzilla.gnome.org/show_bug.cgi?id=766612
137 2016-09-06 19:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
139 * gst/rtsp-server/rtsp-stream.c:
140 rtsp-stream: Compare IP address strings case insensitive
141 Otherwise IPv6 addresses might fail this comparision.
143 2016-09-06 19:10:21 +0300 Sebastian Dröge <sebastian@centricular.com>
145 * gst/rtsp-server/rtsp-stream.c:
146 rtsp-stream: Bind multicast sockets to ANY as before
147 https://bugzilla.gnome.org/show_bug.cgi?id=766612#c48
149 2016-09-05 18:31:36 +0300 Kseniia <vasilchukkseniia@gmail.com>
151 * gst/rtsp-server/rtsp-session.c:
152 rtsp-session: Fix segfault when doing keep-alive after removing the session
153 If keep-alive happens after removing the session but before finalizing the
154 stream transport, we would segfault.
155 https://bugzilla.gnome.org/show_bug.cgi?id=750544
157 2016-09-05 18:04:50 +0300 Sebastian Dröge <sebastian@centricular.com>
159 * gst/rtsp-server/rtsp-stream.c:
160 rtsp-stream: Always create multicast UDP elements if the protocol flag is set
161 Adding them later will cause deadlocks due to
162 1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
163 2) adding the multicast sink
164 3) waiting for it to get data to preroll again
165 3) never happens because the queues after the tee are full.
167 2016-09-05 16:32:57 +0300 Sebastian Dröge <sebastian@centricular.com>
169 * gst/rtsp-server/rtsp-stream.c:
170 rtsp-stream: Fix up various multicast related issues
172 2016-09-05 13:40:59 +0300 Sebastian Dröge <sebastian@centricular.com>
174 * tests/check/gst/stream.c:
175 tests: Fix compilation
177 2016-07-28 15:33:05 -0400 Xavier Claessens <xavier.claessens@collabora.com>
179 * gst/rtsp-server/rtsp-client.c:
180 * gst/rtsp-server/rtsp-stream.c:
181 * tests/check/gst/stream.c:
182 stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast
183 This is basically reverting changes introduced in commit f62a9a7,
184 because it was introducing various regressions:
185 - It introduces a leak of udpsrc elements that got wrongly fixed by adding
186 an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
187 ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
188 - If a mcast client connects, it creates a new socket in SETUP to try to respect
189 the destination/port given by the client in the transport, and overrides the
190 socket already set on the udpsink element. That means that if we already had a
191 client connected, the source address on the udp packets it receives suddenly
193 - If a 2nd mcast client connects, the destination/port in its transport is
194 ignored but its transport wasn't updated.
195 What this patch does:
196 - Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
197 - Always have a tee+queue when udp is enabled. This could be optimized
198 again in a later patch, but is more complicated. If no unicast clients
199 connects then those elements are useless, this could be also optimized
201 - When mcast transport is added, it creates a new set of udpsrc/udpsink,
202 seperated from those for unicast clients. Since we already support only
203 one mcast address, we also create only one set of elements.
204 https://bugzilla.gnome.org/show_bug.cgi?id=766612
206 2016-07-28 15:20:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
208 * gst/rtsp-server/rtsp-stream.c:
209 stream: factor our plug_src function
210 https://bugzilla.gnome.org/show_bug.cgi?id=766612
212 2016-07-21 21:46:16 -0400 Xavier Claessens <xavier.claessens@collabora.com>
214 * gst/rtsp-server/rtsp-stream.c:
215 stream: factor out plug_sink function
216 https://bugzilla.gnome.org/show_bug.cgi?id=766612
218 2016-07-20 23:05:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
220 * gst/rtsp-server/rtsp-stream.c:
221 stream: small documentation clarification
222 https://bugzilla.gnome.org/show_bug.cgi?id=766612
224 2016-07-20 15:35:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
226 * gst/rtsp-server/rtsp-stream.c:
227 stream: rename addr_v4/6 to mcast_addr_v4/6 for clarity
228 https://bugzilla.gnome.org/show_bug.cgi?id=766612
230 2016-07-14 11:10:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
232 * gst/rtsp-server/rtsp-stream.c:
233 stream: Keep a ref on joined bin
234 https://bugzilla.gnome.org/show_bug.cgi?id=766612
236 2016-07-20 15:11:32 -0400 Xavier Claessens <xavier.claessens@collabora.com>
238 * gst/rtsp-server/rtsp-stream.c:
240 https://bugzilla.gnome.org/show_bug.cgi?id=766612
242 2016-07-20 23:18:23 -0400 Xavier Claessens <xavier.claessens@collabora.com>
244 * gst/rtsp-server/rtsp-stream.c:
245 stream: small fix in error code path
246 https://bugzilla.gnome.org/show_bug.cgi?id=766612
248 2016-07-20 20:09:57 -0400 Xavier Claessens <xavier.claessens@collabora.com>
250 * gst/rtsp-server/rtsp-stream.c:
251 Revert "rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc"
252 This partly reverts commit cba045e1b19fad6e689e10206f57903e15f1229a,
253 but keeps unit tests.
254 https://bugzilla.gnome.org/show_bug.cgi?id=766612
256 2016-09-01 12:33:00 +0300 Sebastian Dröge <sebastian@centricular.com>
261 === release 1.9.2 ===
263 2016-09-01 12:32:51 +0300 Sebastian Dröge <sebastian@centricular.com>
269 * gst-rtsp-server.doap:
272 2016-01-27 01:03:52 +0000 Tim-Philipp Müller <tim@centricular.com>
275 * examples/meson.build:
277 * gst/rtsp-server/meson.build:
278 * gst/rtsp-sink/meson.build:
280 * pkgconfig/meson.build:
281 * tests/check/meson.build:
283 Add support for Meson as alternative/parallel build system
284 https://github.com/mesonbuild/meson
286 2016-08-26 21:56:13 +0200 Josep Torra <n770galaxy@gmail.com>
289 * tests/check/Makefile.am:
290 build: silence error about pthread for 'make check' in osx
291 Fixes "clang: error: argument unused during compilation: '-pthread'"
293 2015-09-25 15:04:00 +0000 Nikita Bobkov <NikitaDBobkov@gmail.com>
295 * gst/rtsp-server/rtsp-client.c:
296 rtsp-client: Fix leaking of media in error cases
297 With additional fixes by Kseniya Vasilchuk <vasilchukkseniia@gmail.com>
298 and myself to make the media refcounting a bit easier to follow.
299 https://bugzilla.gnome.org/show_bug.cgi?id=755632
301 2016-08-02 15:08:22 +0300 Sebastian Dröge <sebastian@centricular.com>
303 * gst/rtsp-server/rtsp-client.c:
304 rtsp-client: Fix leaking of session in error cases
305 https://bugzilla.gnome.org/show_bug.cgi?id=755632
307 2016-07-11 21:16:04 +0200 Stefan Sauer <ensonic@users.sf.net>
310 Automatic update of common submodule
311 From f363b32 to f49c55e
313 2016-07-06 13:51:15 +0300 Sebastian Dröge <sebastian@centricular.com>
318 === release 1.9.1 ===
320 2016-07-06 13:28:12 +0300 Sebastian Dröge <sebastian@centricular.com>
326 * gst-rtsp-server.doap:
329 2016-06-24 02:02:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
332 configure: Need to add -DGST_STATIC_COMPILATION when building only statically
333 https://bugzilla.gnome.org/show_bug.cgi?id=767463
335 2016-06-21 11:49:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
338 Automatic update of common submodule
339 From ac2f647 to f363b32
341 2016-04-14 22:56:11 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
343 * gst/rtsp-server/rtsp-sdp.c:
344 * gst/rtsp-server/rtsp-sdp.h:
345 * gst/rtsp-server/rtsp-stream.c:
346 * gst/rtsp-server/rtsp-stream.h:
347 sdp: add rollover counters for all sender SSRC
348 We add different crypto sessions in MIKEY, one for each sender
349 SSRC. Currently, all of them will have the same security policy, 0.
350 The rollover counters are obtained from the srtpenc element using the
352 https://bugzilla.gnome.org/show_bug.cgi?id=730539
354 2016-06-07 20:44:42 +0100 Tim-Philipp Müller <tim@centricular.com>
356 * gst/rtsp-server/rtsp-media-factory.h:
357 * gst/rtsp-server/rtsp-server.h:
360 2016-05-25 10:28:43 +0100 Tim-Philipp Müller <tim@centricular.com>
362 * gst/rtsp-server/Makefile.am:
363 g-i: pass compiler env to g-ir-scanner
364 It's what introspection.mak does as well. Should
365 fix spurious build failures on gnome-continuous
366 (caused by g-ir-scanner getting compiler details
367 via python which is broken in some environments
368 so passing the compiler details bypasses that).
370 2016-05-18 16:48:44 +0100 Ian <ian.arkver.dev@gmail.com>
372 * gst/rtsp-server/rtsp-session.c:
373 rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header
374 This works with rtspsrc and live555, but fails with e.g. ffmpeg.
375 https://bugzilla.gnome.org/show_bug.cgi?id=766619
377 2016-03-07 14:48:38 +0100 Edward Hervey <bilboed@bilboed.com>
379 * gst/rtsp-sink/gstrtspclientsink.c:
380 rtspclientsink: Check return value of sscanf
381 And just make sure we always have 0/0 if we have an error
384 2016-04-25 08:55:25 -0400 Jake Foytik <jake.foytik@ipconfigure.com>
386 * gst/rtsp-server/rtsp-stream.c:
387 * tests/check/gst/rtspserver.c:
388 * tests/check/gst/stream.c:
389 rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
390 - Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
391 - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
392 - Create unit test for shared media.
393 https://bugzilla.gnome.org/show_bug.cgi?id=764744
395 2016-04-11 10:55:23 +0300 Sebastian Dröge <sebastian@centricular.com>
397 * gst/rtsp-server/rtsp-stream.c:
398 rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
399 For IPv6 addresses, binding to a multicast group does not work on Linux
400 either. Always bind to ANY and then later join the multicast group.
401 https://bugzilla.gnome.org/show_bug.cgi?id=764679
403 2016-04-14 10:05:02 +0100 Julien Isorce <j.isorce@samsung.com>
406 Automatic update of common submodule
407 From 6f2d209 to ac2f647
409 2016-04-06 10:09:46 +0200 Patricia Muscalu <patricia@axis.com>
411 * gst/rtsp-server/rtsp-thread-pool.c:
412 rtsp-thread-pool: explained why GSource is a part of ThreadImpl
413 Clarified why it is necessary to add source information to
414 GstRTSPThreadImpl. See the reported bug in GLib:
415 https://bugzilla.gnome.org/show_bug.cgi?id=720186
416 for more information.
417 https://bugzilla.gnome.org/show_bug.cgi?id=761702
419 2016-04-04 12:58:38 +0300 Sebastian Dröge <sebastian@centricular.com>
421 * examples/Makefile.am:
422 examples: Clean up CFLAGS/LDADD even more
423 The internal .la should come first and is part of LDADD, as is
426 2016-04-04 12:39:39 +0300 Sebastian Dröge <sebastian@centricular.com>
428 * examples/Makefile.am:
429 examples: Clean up CFLAGS/LDADD to link with the correct versions of all libraries
431 2016-04-03 12:06:29 +0300 Sebastian Dröge <sebastian@centricular.com>
433 * gst/rtsp-server/Makefile.am:
434 rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS)
436 2015-12-30 18:39:05 +0200 Sebastian Dröge <sebastian@centricular.com>
438 * gst/rtsp-server/rtsp-client.c:
439 * gst/rtsp-server/rtsp-media-factory.c:
440 * gst/rtsp-server/rtsp-media-factory.h:
441 * gst/rtsp-server/rtsp-media.c:
442 * gst/rtsp-server/rtsp-media.h:
443 * gst/rtsp-server/rtsp-sdp.c:
444 * gst/rtsp-server/rtsp-stream.c:
445 * gst/rtsp-server/rtsp-stream.h:
446 rtsp-server: Implement clock signalling according to RFC7273
447 For NTP and PTP clocks we signal the actual clock that is used and signal
448 the direct media clock offset.
449 For all other clocks we at least signal that it's the local sender clock.
450 This allows receivers to know which clock was used to generate the media and
451 its RTP timestamps. Receivers can then implement network synchronization,
452 either absolute or at least relative by getting the sender clock rate directly
453 via NTP/PTP instead of estimating it from RTP timestamps and packet receive
455 https://bugzilla.gnome.org/show_bug.cgi?id=760005
457 2016-03-02 19:42:58 +0200 Sebastian Dröge <sebastian@centricular.com>
459 * gst/rtsp-sink/gstrtspclientsink.c:
460 rtspclientsink: Add support for setting the multicast interface
461 https://bugzilla.gnome.org/show_bug.cgi?id=763000
463 2016-03-02 19:42:13 +0200 Sebastian Dröge <sebastian@centricular.com>
465 * gst/rtsp-server/rtsp-media-factory.c:
466 * gst/rtsp-server/rtsp-media-factory.h:
467 * gst/rtsp-server/rtsp-media.c:
468 * gst/rtsp-server/rtsp-media.h:
469 * gst/rtsp-server/rtsp-stream.c:
470 * gst/rtsp-server/rtsp-stream.h:
471 rtsp-media: Add support for setting the multicast interface
472 https://bugzilla.gnome.org/show_bug.cgi?id=763000
474 2016-03-07 08:50:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
476 * gst/rtsp-sink/gstrtspclientsink.c:
477 rtspclientsink: use new gst_element_class_add_static_pad_template()
478 https://bugzilla.gnome.org/show_bug.cgi?id=763196
480 2016-03-24 13:33:43 +0200 Sebastian Dröge <sebastian@centricular.com>
485 === release 1.8.0 ===
487 2016-03-24 13:00:35 +0200 Sebastian Dröge <sebastian@centricular.com>
493 * gst-rtsp-server.doap:
496 2016-03-16 23:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
498 * gst/rtsp-server/rtsp-stream.c:
499 rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
500 This would get us NO_PREROLL in the bin again and break seeking.
501 Thanks to Carlos Rafael Giani for helping to debug this!
502 https://bugzilla.gnome.org/show_bug.cgi?id=740509
504 === release 1.7.91 ===
506 2016-03-15 12:26:13 +0200 Sebastian Dröge <sebastian@centricular.com>
512 * gst-rtsp-server.doap:
515 2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
517 * gst/rtsp-server/rtsp-stream.c:
518 rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
519 Without this, RECORD pipelines are broken because
520 a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
521 added later. Previously it was there earlier and due to NO_PREROLL caused the
522 pipeline to preroll immediately
523 b) the udpsrc for the pipeline is added later and never set to PLAYING state,
524 as the corresponding code previously was only for PLAY pipelines.
525 https://bugzilla.gnome.org/show_bug.cgi?id=763281
527 2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
529 * gst/rtsp-server/rtsp-stream.c:
530 rtsp-stream: Fix typo in the docstring
531 gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
533 2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
535 * gst/rtsp-server/rtsp-stream.c:
536 rtsp-stream: Disable multicast loopback for all our sockets
537 On Windows this is a receiver-side setting, on Linux a sender-side setting. As
538 we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
539 loopback setting on the socket... while udpsink does which unfortunately has
540 no effect here on Windows but on Linux.
541 https://bugzilla.gnome.org/show_bug.cgi?id=757488
543 2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
545 * tests/check/gst/stream.c:
546 stream tests: added new tests
547 Test a case when the address pool only contains multicast addresses
548 and the client is requesting unicast udp.
549 Added tests for multicast ports allocation.
550 https://bugzilla.gnome.org/show_bug.cgi?id=757488
552 2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
554 * gst/rtsp-server/rtsp-stream.c:
555 rtsp-stream: Only bind multicast sockets to ANY on Windows
556 On Linux it is still needed to bind to the multicast address
557 to filter out random other packets, while on Windows binding
558 to multicast addresses just fails.
560 2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
562 * gst/rtsp-server/rtsp-stream.c:
563 rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
564 Otherwise we fail to allocate UDP ports if the pool only contains multicast
565 addresses, which is something that used to work before. For unicast addresses
566 if the pool contains none, we just allocate them as if there is no pool at
568 https://bugzilla.gnome.org/show_bug.cgi?id=757488
570 2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
572 * gst/rtsp-server/rtsp-client.c:
573 * gst/rtsp-server/rtsp-stream.c:
574 rtsp-server: Fix indentation
576 2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
578 * gst/rtsp-server/rtsp-stream.c:
579 rtsp-stream: Don't bind the sockets to multicast addresses
580 This works on Linux but fails completely on Windows. You're supposed
581 to bind to ANY and then join the multicast group.
582 https://bugzilla.gnome.org/show_bug.cgi?id=757488
584 === release 1.7.90 ===
586 2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
592 * gst-rtsp-server.doap:
595 2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
598 Automatic update of common submodule
599 From b64f03f to 6f2d209
601 2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
603 * gst/rtsp-sink/gstrtspclientsink.c:
604 * tests/check/gst/rtspclientsink.c:
605 rtspsink: Fix some leaks in rtspclientsink and the unit test.
606 https://bugzilla.gnome.org/show_bug.cgi?id=762525
608 2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
610 * tests/check/gst/media.c:
611 * tests/check/gst/rtspclientsink.c:
612 * tests/check/gst/rtspserver.c:
613 * tests/check/gst/stream.c:
614 tests: unit test fixes
615 Removed port allocation test from the media suite.
616 The port allocation failure is now in the stream suite.
618 Make sure that the media is suspended after the DESCRIBE request
619 before reconfiguring the UDP sinks.
621 In the RECORD case we have to set async property to false
622 for the appsink element in the test in order to make sure
623 that the media pipeline doesn't hang in start_preroll().
624 https://bugzilla.gnome.org/show_bug.cgi?id=757488
626 2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
628 * gst/rtsp-server/rtsp-client.c:
629 * gst/rtsp-server/rtsp-stream.c:
630 * gst/rtsp-server/rtsp-stream.h:
631 rtsp-stream: postpone UDP socket allocation until SETUP
632 Postpone the allocation of the UDP sockets until we know
633 what transport has been chosen by the client.
634 Both unicast and multicast UDP sources are created in one
636 https://bugzilla.gnome.org/show_bug.cgi?id=757488
638 2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
640 * gst/rtsp-server/rtsp-stream.c:
641 rtsp-stream: postpone the creation of the UDP sources
642 Code refactoring: allocate the UDP ports after the sender and
643 the reciver parts have been created.
644 We postpone the creation of the UDP sources until the UDP
645 ports have been allocated.
646 https://bugzilla.gnome.org/show_bug.cgi?id=757488
648 2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
650 * gst/rtsp-server/rtsp-stream.c:
651 rtsp-stream: added function for setting UDP sources to PLAYING state
652 Code refactoring: Introduced a function for setting UDP sources
654 https://bugzilla.gnome.org/show_bug.cgi?id=757488
656 2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
658 * gst/rtsp-server/rtsp-stream.c:
659 rtsp-stream: added function for creating and configuring UDP sources
660 Code refactoring: create and configure UDP sources in a separate function.
661 https://bugzilla.gnome.org/show_bug.cgi?id=757488
663 2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
665 * gst/rtsp-server/rtsp-stream.c:
666 rtsp-stream: added function for RTP/RTCP socket configuration
667 Code refactoring: configure RTP and RTCP sockets for UDP sinks
668 in a separate function.
669 https://bugzilla.gnome.org/show_bug.cgi?id=757488
671 2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
673 * gst/rtsp-server/rtsp-stream.c:
674 rtsp-stream: added function for creating and configuring UDP sinks
675 Code refactoring: create and configure UDP sinks in a separate function.
676 https://bugzilla.gnome.org/show_bug.cgi?id=757488
678 2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
680 * gst/rtsp-server/rtsp-stream.c:
681 rtsp-stream: added helper function for creating the sender/receiver parts
682 Code refactoring: introduced helper function for creating
683 the receiver and the sender parts of the streaming pipeline.
684 https://bugzilla.gnome.org/show_bug.cgi?id=757488
686 2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
691 === release 1.7.2 ===
693 2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
699 * gst-rtsp-server.doap:
702 2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
704 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
705 uninstalled.pc: add support for non libtool build systems
706 Currently the .la path is provided which requires to use libtool as
707 mentioned in the GStreamer manual section-helloworld-compilerun.html.
708 It is fine as long as the application is built using libtool.
709 So currently it is not possible to compile a GStreamer application
710 within gst-uninstalled with CMake or other build system different
712 This patch allows to do the following in gst-uninstalled env:
713 gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
714 gstreamer-rtsp-server-1.0)
715 Previously it required to prepend libtool --mode=link
716 https://bugzilla.gnome.org/show_bug.cgi?id=720778
718 2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
720 * gst/rtsp-sink/gstrtspclientsink.c:
721 rtspclientsink: remove check for impossible condition
722 Goto error label checks stream to see if it needs to be unreferenced before
723 returning, but this goto jumps happens before the stream is ever set, so it
724 will always be NULL in this error label.
727 2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
729 * gst/rtsp-sink/gstrtspclientsink.c:
730 rtspclientsink: clean switch statements
731 Coverity demands for fallthrough statements to be clearly commented,
732 to distinguish from accidental fall throughs. And it also needs all
733 cases to finish with a break, even if the break is never going to be
734 executed like in the case of a continue jump.
738 2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
740 * tests/check/Makefile.am:
741 tests: extend the AM_TESTS_ENVIRONMENT from check.mak
742 To get the CK_DEFAULT_TIMEOUT defined for all tests
743 Also removes a 120 seconds timeout that was set as default
744 explicitly in this module
745 https://bugzilla.gnome.org/show_bug.cgi?id=761472
747 2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
751 Automatic update of common submodule
752 From 86e4663 to b64f03f
754 2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
756 * gst/rtsp-server/rtsp-media.c:
757 rtsp-media: fix state_lock not locked again when preroll fails
758 https://bugzilla.gnome.org/show_bug.cgi?id=761399
760 2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
763 configure: Move plugin specific flags below all the others
764 They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
765 -no-undefined. And -no-undefined is required on Windows to build DLLs.
767 2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
769 * gst/rtsp-sink/gstrtspclientsink.c:
770 rtspclientsink: Simplify slightly using new -base API
771 Use the new Mikey and SDP API in the base plugins libs
772 to simplify some code.
773 https://bugzilla.gnome.org/show_bug.cgi?id=758180
775 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
780 * gst/rtsp-sink/Makefile.am:
781 * gst/rtsp-sink/gstrtspclientsink.c:
782 * gst/rtsp-sink/gstrtspclientsink.h:
783 * gst/rtsp-sink/plugin.c:
784 * tests/check/Makefile.am:
785 * tests/check/gst/rtspclientsink.c:
786 rtspsink: Add rtspclientsink element
787 Add an rtspclientsink element that accepts streams for which
788 there is a registered payloader and sends them to
789 an RTSP server using RECORD.
790 Sending is synchronised to the pipeline clock. Payload-types
791 are automatically selected. The 'new-payloader' signal is fired
792 for custom configuration of payloaders when they are created.
793 Can now stream a movie like this:
795 ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
796 decodebin name=depay1 ! audioconvert ! autoaudiosink )"
798 gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
799 queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
800 https://bugzilla.gnome.org/show_bug.cgi?id=758180
802 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
804 * gst/rtsp-server/rtsp-stream.c:
805 * gst/rtsp-server/rtsp-stream.h:
806 rtsp-stream: Add functions for using rtsp-stream from the client
807 Add a boolean to indicate that the rtsp-stream is running on the
808 'client' side of an RTSP connection, for sending streams via
809 RECORD. In that case, the roles of the client/server ports
810 in transport setup are swapped.
811 https://bugzilla.gnome.org/show_bug.cgi?id=758180
813 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
815 * gst/rtsp-server/rtsp-sdp.c:
816 * gst/rtsp-server/rtsp-sdp.h:
817 rtsp-sdp: Add gst_rtsp_sdp_from_stream()
818 A new function that adds info from a GstRTSPStream into an SDP message.
819 https://bugzilla.gnome.org/show_bug.cgi?id=758180
821 2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
823 * gst/rtsp-server/rtsp-media.c:
824 rtsp-media: Fix mutex beeing unlocked while they should be locked
825 https://bugzilla.gnome.org/show_bug.cgi?id=761226
827 2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
829 * gst/rtsp-server/rtsp-media-factory.c:
830 rtsp-media-factory: add missing break in "clock" property setter
833 2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
835 * gst/rtsp-server/rtsp-stream.c:
836 rtsp-stream: fixed assert during update transport
837 When RTSP server trying update transport during multicast, it throws an
838 assert. The assert is thrown because it is trying to get the parent of
839 an non-existing funnel element.
840 https://bugzilla.gnome.org/show_bug.cgi?id=760150
842 2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
844 * gst/rtsp-server/rtsp-permissions.h:
845 * gst/rtsp-server/rtsp-thread-pool.h:
846 * gst/rtsp-server/rtsp-token.h:
847 docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
848 gtk-doc can handle static inline functions just fine these days,
849 there's no need for this stuff any more.
851 2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
853 * gst/rtsp-server/rtsp-media.c:
854 * gst/rtsp-server/rtsp-sdp.c:
855 sdp: replace duplicated codes to call new base sdp apis
856 https://bugzilla.gnome.org/show_bug.cgi?id=745880
858 2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
860 * examples/test-netclock.c:
861 test-netclock: Use the new API to configure a clock directly
863 2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
865 * gst/rtsp-server/rtsp-media-factory.c:
866 * gst/rtsp-server/rtsp-media-factory.h:
867 * gst/rtsp-server/rtsp-media.c:
868 * gst/rtsp-server/rtsp-media.h:
869 rtsp-media: Add API to directly configure a clock on the media pipelines
871 2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
873 * gst/rtsp-server/rtsp-media.c:
874 rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
876 2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
878 * gst/rtsp-server/rtsp-media-factory.c:
879 rtsp-media-factory: Add FIXME for 2.0
881 2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
883 * gst/rtsp-server/rtsp-stream.c:
884 rtsp-stream: Fix indentation
886 2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
888 * gst/rtsp-server/rtsp-media.c:
889 rtsp-media: Do not prepare media after media times out
890 Deferred calls to start_prepare() can be deferred past the point until
891 which wait_preroll() and by proxy gst_rtsp_media_get_status() is
892 prepared to wait. Previously there was no lock and no check for this
893 situation. This meant that a media could be prepared and unprepared
894 simultaneously by two different threads. Now a lock is in place and a
895 suitable check is done.
896 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
898 2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
900 * gst/rtsp-server/rtsp-client.c:
901 * gst/rtsp-server/rtsp-media-factory.c:
902 * gst/rtsp-server/rtsp-media-factory.h:
903 * gst/rtsp-server/rtsp-media.c:
904 * gst/rtsp-server/rtsp-media.h:
905 rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
906 Without TEARDOWN it might be desireable to keep the media running and continue
907 sending data to the client, even if the RTSP connection itself is
909 Only do this for session medias that have only UDP transports. If there's at
910 least on TCP transport, it will stop working and cause problems when the
911 connection is disconnected.
912 https://bugzilla.gnome.org/show_bug.cgi?id=758999
914 2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
919 === release 1.7.1 ===
921 2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
927 * gst-rtsp-server.doap:
930 2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
933 configure: Make -Bsymbolic check work with clang.
934 Update the -Bsymbolic check with the version glib has. This version
936 https://bugzilla.gnome.org/show_bug.cgi?id=759713
938 2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
940 * gst/rtsp-server/rtsp-session-pool.c:
941 rtsp-session-pool: Avoid dollar sign ($) in session ids
942 Live555 in VLC strips off dollar signs and then gets very confused,
943 we don't loose too much entropy by just skipping it.
945 2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
947 * gst/rtsp-server/rtsp-address-pool.h:
948 * gst/rtsp-server/rtsp-auth.h:
949 * gst/rtsp-server/rtsp-client.h:
950 * gst/rtsp-server/rtsp-media-factory-uri.h:
951 * gst/rtsp-server/rtsp-media-factory.h:
952 * gst/rtsp-server/rtsp-media.h:
953 * gst/rtsp-server/rtsp-mount-points.h:
954 * gst/rtsp-server/rtsp-permissions.h:
955 * gst/rtsp-server/rtsp-server.h:
956 * gst/rtsp-server/rtsp-session-media.h:
957 * gst/rtsp-server/rtsp-session-pool.h:
958 * gst/rtsp-server/rtsp-session.h:
959 * gst/rtsp-server/rtsp-stream-transport.h:
960 * gst/rtsp-server/rtsp-stream.h:
961 * gst/rtsp-server/rtsp-thread-pool.h:
962 * gst/rtsp-server/rtsp-token.h:
963 rtsp-server: Add g_autoptr() support to all types
964 https://bugzilla.gnome.org/show_bug.cgi?id=754464
966 2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
968 * gst/rtsp-server/rtsp-stream.c:
969 rtsp-stream: fixed valgrind error
970 Fixed the valgrind error in unit test. The UDP source created during
971 gst_rtsp_stream_join_bin() was not released while destroying the rtp
973 https://bugzilla.gnome.org/show_bug.cgi?id=759010
975 2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
979 Automatic update of common submodule
980 From b319909 to 86e4663
982 2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
984 * gst/rtsp-server/rtsp-client.c:
985 rtsp-client: suspend media during setup request
986 SETUP request from clients needs to suspend the media to clear the
987 prerolled buffers. Otherwise it will not affect the prerolled buffer
988 and the prerolled buffers will be incorrect (for example block-size
989 from setup request will not affect the prerolled buffer unless the
991 https://bugzilla.gnome.org/show_bug.cgi?id=758268
993 2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
995 * gst/rtsp-server/rtsp-stream.c:
996 rtsp-stream: create stream pipeline based on transport
997 Based on the protocol, create the rtsp stream pipeline. If only TCP or
998 only UDP is set as the transport protocol, it will not add the extra tee
999 or queue element to the pipeline. Both these elements will be added, if
1000 it supports both TCP and UDP protocols. This improves the pipeline
1001 performance when one protocol is present.
1002 https://bugzilla.gnome.org/show_bug.cgi?id=758179
1004 2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
1006 * gst/rtsp-server/rtsp-stream.c:
1007 rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
1008 Adding them when not needed will start some logic inside rtpbin that might be
1009 problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
1010 would start up a rtpjitterbuffer and behave in weird ways.
1011 We still set up the UDP sources for RTP receiving for a sender media to be
1012 able to receive any packets sent by the client for NAT traversal. They will
1013 all go to a fakesink though.
1014 Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
1015 NO_PREROLL, which will cause deadlocks when seeking the media as it will never
1016 receive ASYNC_DONE after a seek.
1017 https://bugzilla.gnome.org/show_bug.cgi?id=758319
1019 2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
1021 * gst/rtsp-server/rtsp-stream.c:
1022 rtsp-stream: Disable multicast loopback for the multicast udp sources too
1023 On POSIX this setting is for sender sockets, on Windows for receiver sockets.
1024 Previously we were only setting this for sender sockets, which caused looped
1025 back packets to be received on Windows if a multicast transport was used.
1027 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1029 * examples/test-record-auth.c:
1030 * examples/test-record.c:
1031 examples: Actually use the provided port in the record examples
1033 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1035 * examples/test-record-auth.c:
1036 test-record-auth: Add the option to build in TLS support
1038 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1040 * examples/test-auth.c:
1041 test-auth: Use an 'anonymous' user for unauthenticated default
1042 There's a comment on one of the resources that 'user' and 'admin'
1043 shouldn't even be able to see it, but they can if the default
1044 token is 'admin2', since that gives them access anyway.
1046 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1048 * examples/.gitignore:
1049 * examples/Makefile.am:
1050 * examples/test-record-auth.c:
1051 Add test-record-auth example
1053 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
1055 * gst/rtsp-server/rtsp-client.c:
1056 * tests/check/gst/client.c:
1057 rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
1059 2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
1061 * gst/rtsp-server/rtsp-server.c:
1062 rtsp-server: Change the logic so we don't pop a NULL context
1063 When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
1064 will sometimes fail. This call is made before any context is pushed
1065 resulting in an attempt to pop a NULL context.
1066 https://bugzilla.gnome.org/show_bug.cgi?id=757949
1068 2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
1070 * tests/check/gst/rtspserver.c:
1071 rtspserver: Add udp-mcast transport SETUP test
1072 Refactor utility functions in the test file so they can handle
1073 more than UDP and TCP as lower transport.
1074 https://bugzilla.gnome.org/show_bug.cgi?id=756969
1076 2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
1078 * gst/rtsp-server/rtsp-stream.c:
1079 rtsp-stream: Always unref return value of gst_object_get_parent()
1080 Fixes a leak of a GstBin in the udp-mcast case.
1081 https://bugzilla.gnome.org/show_bug.cgi?id=756968
1083 2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
1086 Automatic update of common submodule
1087 From b99800a to b319909
1089 2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
1092 Use new GST_ENABLE_EXTRA_CHECKS #define
1093 https://bugzilla.gnome.org/show_bug.cgi?id=756870
1095 2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
1098 Automatic update of common submodule
1099 From 6babecd to b99800a
1101 2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
1104 Update GLib dependency to 2.40.0
1106 2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1108 * examples/test-mp4.c:
1109 * gst/rtsp-server/rtsp-stream.c:
1110 stream: listen to sender ssrc signals
1111 https://bugzilla.gnome.org/show_bug.cgi?id=746747
1113 2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
1116 common: update for new suppression
1117 Makes check-valgrind pass with glib 2.46
1119 2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1121 * gst/rtsp-server/rtsp-media.c:
1122 rtsp-media: Take reference to media that will be prepared
1123 default_prepare() takes a transfer-none reference GstRTSPMedia object.
1124 Later on a g_idle_source_new() is created and a pointer to the media
1125 object is passed as user data. If the media is freed before the idle
1126 source is dispatched the media object pointer is invalid, but the idle
1127 source callback expects it to still be valid. To fix this a reference to
1128 the media object is taken when registering the source callback function
1129 and a corresponding release of the reference is done when the souce is
1131 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
1133 2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
1135 * examples/test-launch.c:
1136 * examples/test-mp4.c:
1137 * examples/test-ogg.c:
1138 * examples/test-record.c:
1139 * examples/test-uri.c:
1140 rtsp-server: Fix memory leaks when context parse fails
1141 When g_option_context_parse fails, context and error variables are not getting free'd
1142 which results in memory leaks. Free'ing the same.
1143 And replacing g_error_free with g_clear_error, which checks if the error being passed
1144 is not NULL and sets the variable to NULL on free'ing.
1145 https://bugzilla.gnome.org/show_bug.cgi?id=753863
1147 2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
1152 === release 1.6.0 ===
1154 2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
1160 * gst-rtsp-server.doap:
1163 === release 1.5.91 ===
1165 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
1171 * gst-rtsp-server.doap:
1174 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
1176 * docs/libs/gst-rtsp-server-sections.txt:
1177 * gst/rtsp-server/rtsp-stream.c:
1178 stream: fix docs for recently-added get/set_buffer_size API
1179 https://bugzilla.gnome.org/show_bug.cgi?id=749095
1181 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
1183 * gst/rtsp-server/rtsp-media.c:
1184 rtsp-media: Don't crash on encrypted RTX SDP
1185 In parse_keymgmt(), don't mutate the input string that's been passed
1186 as const, especially since we might need the original value again if
1187 the same key info applies to multiple streams (RTX, for example).
1188 https://bugzilla.gnome.org/show_bug.cgi?id=754753
1190 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
1192 * examples/test-mp4.c:
1193 test-mp4: Support filenames with spaces in them. Error out on too few arguments
1195 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
1197 * examples/test-record.c:
1198 test-record: Check parameter count and print out help
1199 If no launch pipeline was supplied, print out some help
1201 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
1203 * gst/rtsp-server/rtsp-media.c:
1204 * gst/rtsp-server/rtsp-stream.c:
1205 * gst/rtsp-server/rtsp-stream.h:
1206 rtsp-stream: Implement UDP buffer size setting.
1207 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
1209 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
1210 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
1212 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
1214 * gst/rtsp-server/rtsp-media.h:
1215 rtsp-media: Fix small typo causing gtk-doc to complain
1217 === release 1.5.90 ===
1219 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
1225 * gst-rtsp-server.doap:
1228 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1230 * gst/rtsp-server/rtsp-media-factory.c:
1231 media-factory: get port number through gst_rtsp_url_get_port
1232 https://bugzilla.gnome.org/show_bug.cgi?id=753473
1234 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
1236 * tests/check/gst/media.c:
1237 media-test: Removing unnecessary assertion
1238 https://bugzilla.gnome.org/show_bug.cgi?id=753385
1240 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1242 * gst/rtsp-server/rtsp-server.c:
1243 Document that source keeps a ref on server until it's destroyed
1244 https://bugzilla.gnome.org/show_bug.cgi?id=749227
1246 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1248 * tests/check/gst/media.c:
1249 media-test: Test for multiple dynamic payload
1250 https://bugzilla.gnome.org/show_bug.cgi?id=753385
1252 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1254 * gst/rtsp-server/rtsp-media.c:
1255 media: Only add fakesink once per pipeline
1256 The intention is to prevent going PLAYING state before pads are created.
1257 If there was mutilple dynamic payload, it would leak few fakesink and
1258 actually prevent from ever reaching playing state.
1259 https://bugzilla.gnome.org/show_bug.cgi?id=753385
1261 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1263 * gst/rtsp-server/rtsp-media.c:
1264 Revert "rtsp-media: Only add 1 fakesink per pipeline"
1265 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
1267 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1269 * gst/rtsp-server/rtsp-media.c:
1270 rtsp-media: Only add 1 fakesink per pipeline
1271 There should be only one fakesink per pipeline, not per dynpay. This
1272 would lead to element naming clash.
1274 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
1276 * gst/rtsp-server/rtsp-media.c:
1277 rtsp-media: assertion error due to wrong condition check
1278 In media to caps function, reserved_keys array is being used for variable i,
1279 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
1280 changed it to variable j
1281 https://bugzilla.gnome.org/show_bug.cgi?id=753009
1283 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
1285 * gst/rtsp-server/rtsp-media.c:
1286 rtsp-media: Strip keys from the fmtp that we use internally in our caps
1287 Skip keys from the fmtp, which we already use ourselves for the
1288 caps. Some software is adding random things like clock-rate into
1289 the fmtp, and we would otherwise here set a string-typed clock-rate
1290 in the caps... and thus fail to create valid RTP caps
1291 https://bugzilla.gnome.org/show_bug.cgi?id=753009
1293 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1295 * gst/rtsp-server/rtsp-thread-pool.c:
1296 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
1297 https://bugzilla.gnome.org/show_bug.cgi?id=752640
1299 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
1302 Automatic update of common submodule
1303 From f74b2df to 9aed1d7
1305 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
1310 === release 1.5.2 ===
1312 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
1318 * gst-rtsp-server.doap:
1321 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
1323 * gst/rtsp-server/rtsp-client.c:
1324 * gst/rtsp-server/rtsp-client.h:
1325 * tests/check/gst/client.c:
1326 rtsp-client: allow application to decide what requirements are supported
1327 Add "check-requirements" signal and vfunc to allow application
1328 (and subclasses) to check the requirements.
1329 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
1330 https://bugzilla.gnome.org/show_bug.cgi?id=749417
1332 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1335 Automatic update of common submodule
1336 From 6015d26 to f74b2df
1338 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
1340 * gst/rtsp-server/rtsp-media.c:
1341 rtsp-media: Always use real payloader when creating streams
1342 A bin that contains the real payloader might be used as payloader. In this
1343 case we have to get the real payloader for the various properties it provides.
1344 Example use cases for this are bins that payload some media and then have
1345 additional elements that add metadata or RTP extension headers to the stream.
1346 https://bugzilla.gnome.org/show_bug.cgi?id=750800
1348 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
1350 * examples/test-netclock-client.c:
1351 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
1353 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
1355 * examples/test-netclock-client.c:
1356 * examples/test-netclock.c:
1357 test-netclock: Use new ntp-time-source property on rtpbin
1358 Select the clock time to be used as NTP time source. This allows proper
1359 synchronization between receivers, independent of sharing base times, and just
1360 requires them to use the same clock.
1362 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
1364 * examples/test-netclock-client.c:
1365 * examples/test-netclock.c:
1366 test-netclock: Setting the same base time on sender and receiver is not necessary
1367 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
1369 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1371 * gst/rtsp-server/rtsp-stream.c:
1372 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
1373 https://bugzilla.gnome.org/show_bug.cgi?id=750764
1375 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1377 * docs/libs/gst-rtsp-server.types:
1378 docs: add missing types
1379 https://bugzilla.gnome.org/show_bug.cgi?id=750764
1381 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1383 * docs/libs/gst-rtsp-server-sections.txt:
1384 docs: add missing apis
1385 https://bugzilla.gnome.org/show_bug.cgi?id=750764
1387 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
1389 * examples/test-netclock-client.c:
1390 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
1392 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1394 * docs/libs/gst-rtsp-server-sections.txt:
1395 * gst/rtsp-server/rtsp-auth.c:
1396 * gst/rtsp-server/rtsp-auth.h:
1397 GstRTSPAuth: Add client certificate authentication support
1398 https://bugzilla.gnome.org/show_bug.cgi?id=750471
1400 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
1402 * examples/test-netclock-client.c:
1403 test-netclock-client: Use new GstClock API to wait for clock synchronization
1405 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
1407 * examples/test-netclock-client.c:
1408 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
1409 A mainloop is needed to get glimagesink to display something on OSX, and
1410 the source-setup signal just makes things a little bit easier.
1412 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
1415 Automatic update of common submodule
1416 From d9a3353 to 6015d26
1418 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
1421 Automatic update of common submodule
1422 From d37af32 to d9a3353
1424 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
1427 Automatic update of common submodule
1428 From 21ba2e5 to d37af32
1430 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
1433 Automatic update of common submodule
1434 From c408583 to 21ba2e5
1436 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
1438 * docs/libs/Makefile.am:
1439 docs: remove variables that we define in the snippet from common
1440 This is syncing our Makefile.am with upstream gtkdoc.
1442 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
1445 Automatic update of common submodule
1446 From 44a3517 to c408583
1448 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
1453 === release 1.5.1 ===
1455 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
1461 * gst-rtsp-server.doap:
1464 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
1466 * gst/rtsp-server/rtsp-client.c:
1467 rtsp-client: No flush during Teardown.
1468 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
1469 backlog is empty it can happen that just a part of a message will be
1470 sent and rest is in backlog queue. If then flush during teardown
1471 just a part of message will be sent.This can lead to client miss
1472 teardown response since it expect to get the last part of message.
1473 The flushing during teardown was introduced to fix a deadlock that now
1474 is fixed more generally in handle_request by temporary setting backlog
1476 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
1478 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
1480 * tests/check/Makefile.am:
1481 tests: Use AM_TESTS_ENVIRONMENT
1482 Needed by the new automake test runner and the
1483 current version of the common submodule.
1485 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
1487 * gst/rtsp-server/rtsp-media.h:
1488 * gst/rtsp-server/rtsp-stream.h:
1489 rtsp-server: Use single-include rtsp header to make sure we get all definitions
1491 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
1493 * gst/rtsp-server/rtsp-media.c:
1494 rtsp-media: Mark some more functions static
1496 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
1498 * gst/rtsp-server/rtsp-media.c:
1499 rtsp-media: Only unblock the media in suspend() when actually changing the state
1500 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
1502 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
1504 * examples/test-video-rtx.c:
1505 examples: Use AVPF profile for the RTX example
1507 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
1509 * gst/rtsp-server/rtsp-sdp.c:
1510 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
1512 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1514 * gst/rtsp-server/rtsp-stream.c:
1515 rtsp-stream: get valid clock-rate from last-sample
1516 clock-rate in last-sample's caps is integer, not unsigned.
1517 To get this value properly, variable needs to be type-casted to int.
1518 https://bugzilla.gnome.org/show_bug.cgi?id=747614
1520 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
1524 autogen.sh: only run autopoint if gettext requested in configure.ac
1525 Not just because there happens to be a po directory.
1526 https://bugzilla.gnome.org/show_bug.cgi?id=748058
1528 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
1531 Revert "configure.ac: uncomment gettext version setup"
1532 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
1533 We don't need a gettext setup here and there's no po
1534 directory either, so no reason why autopoint would be
1535 run in the first place.
1536 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
1538 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
1540 * examples/test-multicast.c:
1541 * examples/test-multicast2.c:
1542 * examples/test-sdp.c:
1543 * examples/test-video-rtx.c:
1544 * examples/test-video.c:
1545 * tests/test-cleanup.c:
1546 * tests/test-reuse.c:
1547 Fix timeout function signatures across tests and examples
1549 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
1551 * tests/check/Makefile.am:
1552 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
1553 Make sure the test environment is set up.
1554 https://bugzilla.gnome.org//show_bug.cgi?id=747624
1556 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
1559 configure: bump automake requirement to 1.14 and autoconf to 2.69
1560 This is only required for builds from git, people can still
1561 build tarballs if they only have older autotools.
1562 https://bugzilla.gnome.org//show_bug.cgi?id=747624
1564 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
1567 configure.ac: uncomment gettext version setup
1568 Fixes autogen.sh. It would run autopoint, which would complain
1569 that it could not find the gettext version in configure.ac.
1570 https://bugzilla.gnome.org/show_bug.cgi?id=748058
1572 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1574 * examples/test-video-rtx.c:
1575 test-video-rtx: set exact payload type to PCMA payloader
1576 Setting wrong payload type causes failure to do retransmission through audio stream
1577 https://bugzilla.gnome.org/show_bug.cgi?id=747839
1579 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1581 * gst/rtsp-server/rtsp-media.c:
1582 * gst/rtsp-server/rtsp-stream.c:
1583 * gst/rtsp-server/rtsp-stream.h:
1584 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
1585 Because of duplicated g_signal_connect for request-aux-sender signal,
1586 wrong stream pointer is passed to the signal handler.
1587 Instead of passing each stream, pass stream array and get the relevant stream.
1588 https://bugzilla.gnome.org/show_bug.cgi?id=747839
1590 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
1594 Update autogen.sh to latest version from common
1595 Fixes build after aclocal_check etc. helpers have been removed.
1597 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
1600 Automatic update of common submodule
1601 From bc76a8b to c8fb372
1603 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
1605 * gst/rtsp-server/rtsp-stream.c:
1606 rtsp-stream: Limit the queues to 1 buffer
1607 We only need them to be able to pre-roll, queueing up more data here
1608 is only going to harm latency and memory usage.
1610 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
1612 * gst/rtsp-server/rtsp-stream.c:
1613 rtsp-stream: Update comment and ASCII art to the latest code
1614 We have a queue in front of the udpsink too to prevent the pipeline from
1617 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
1619 * gst/rtsp-server/rtsp-stream.c:
1620 rtsp-media: Properly return first rtptime
1621 Instead we where returning first GstBuffer timestamp. This would result
1622 in clock skew and unwanted behaviour in RTSP playback.
1623 https://bugzilla.gnome.org/show_bug.cgi?id=746479
1625 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
1627 * gst/rtsp-server/rtsp-stream.c:
1628 rtsp-stream: Don't leave buffer mapped
1629 If the seq is NULL, the RTP buffer was left mapped. We should always
1632 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
1637 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
1639 * gst/rtsp-server/rtsp-media-factory.c:
1640 * tests/check/gst/client.c:
1641 Fix double semicolons
1643 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
1645 * gst/rtsp-server/rtsp-stream.c:
1646 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
1647 This gives more accurate values than asking the payloader. There might be
1648 queueing happening between the payloader and the sink.
1649 https://bugzilla.gnome.org/show_bug.cgi?id=745704
1651 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
1653 * gst/rtsp-server/rtsp-media.c:
1654 rtsp-media: Don't seek for PLAY if the position will not change
1655 https://bugzilla.gnome.org/show_bug.cgi?id=745704
1657 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
1659 * gst/rtsp-server/rtsp-media.c:
1660 rtsp-media: Don't include payload type in the caps for framesize
1661 When the sdp media attribute framesize are converted to caps
1662 the <payload> should not be included.
1663 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
1664 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
1666 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
1668 * gst/rtsp-server/rtsp-sdp.c:
1669 rtsp-sdp: add payload type to the sdp framesize attribute
1670 The sdp framesize attribute is desribed in RFC6064. It is specified
1671 for payloading of H263 and has the following form
1672 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
1673 should be added to the caps in a payloader and the <payload type> should
1674 be added by the rtsp-server.
1675 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
1677 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
1679 * examples/test-uri.c:
1680 examples: test-uri: fix tainted variable
1681 Insignificant but this keeps Coverity happy.
1684 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
1686 * examples/.gitignore:
1687 * examples/Makefile.am:
1688 * examples/test-netclock-client.c:
1689 * examples/test-netclock.c:
1690 examples: Add a simple example of network synch for live streams.
1691 An example server and client that works for synchronising live streams
1692 only - as it can't support pause/play.
1694 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
1696 * gst/rtsp-server/rtsp-media-factory.c:
1697 * gst/rtsp-server/rtsp-media-factory.h:
1698 rtsp-media-factory: Add functions to set/get the media gtype
1699 Allow specifying the GType of a GstRtspMedia subclass to create
1700 as a simpler way to get the factory to create a custom
1701 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
1703 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
1705 * gst/rtsp-server/rtsp-media.c:
1706 rtsp-media: fix double unlock in _get_buffer_size()
1707 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
1708 because of double g_mutex_unlock () usage.
1709 https://bugzilla.gnome.org/show_bug.cgi?id=745434
1711 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
1713 * gst/rtsp-server/rtsp-session-pool.c:
1714 * gst/rtsp-server/rtsp-session.c:
1715 * gst/rtsp-server/rtsp-session.h:
1716 rtsp-session: Use monotonic time for RTSP session timeout
1717 Changed RTSP session timeout handling to monotonic time
1718 and deprecating the API for current system time.
1719 This fixes timeouts when the system time changes.
1720 https://bugzilla.gnome.org/show_bug.cgi?id=743346
1722 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
1724 * gst/rtsp-server/rtsp-client.c:
1725 * gst/rtsp-server/rtsp-media.c:
1726 rtsp-client: Only error out in PLAY if seeking actually failed
1727 If the media was just not seekable, we continue from whatever position we are
1728 and let the client decide if that is what is wanted or not.
1729 Only if the actual seek failed, we can't really recover and should error out.
1731 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
1733 * gst/rtsp-server/rtsp-stream.c:
1734 rtsp-stream: Add necessary queues between tee and multiudpsink
1735 https://bugzilla.gnome.org/show_bug.cgi?id=744379
1737 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
1739 * gst/rtsp-server/rtsp-client.c:
1740 * gst/rtsp-server/rtsp-media.c:
1741 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
1742 Instead error out properly the same way as if the SEEKING query already
1745 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
1747 * gst/rtsp-server/rtsp-stream.h:
1748 rtsp-stream: minor code formatting fix
1750 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
1752 * gst/rtsp-server/rtsp-media.c:
1753 rtsp-media: fix logic for collect_streams
1754 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
1755 all streams it knows if it got any, and can check if the transport mode is OK.
1758 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
1760 * gst/rtsp-server/rtsp-media.c:
1761 rtsp-media: Don't set the transport mode based on what elements we find
1762 Just print a warning if the one that was set before disagrees with what
1763 elements we found. It must already be set to something before as this
1764 function is called after we received the SDP from ANNOUNCE in RECORD mode,
1765 and we would reject ANNOUNCE if the RECORD flag was not set.
1767 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
1769 * tests/check/gst/rtspserver.c:
1770 tests: rtspserver: rename shadowed variable
1771 We have two different 'sink' variables here,
1772 rename one of them for clarity.
1774 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
1776 * gst/rtsp-server/rtsp-client.c:
1777 rtsp-client: fix awkward if clause
1779 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
1781 * examples/test-uri.c:
1782 examples: test-uri: improve uri argument handling and accept file names
1783 Print an error if the argument passed is not a URI and can't
1784 be converted into one, or no arguments have been provided.
1786 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
1788 * examples/test-uri.c:
1789 examples: test-uri: don't remove mount point after 10 seconds
1790 It's very irritating when trying to test stuff repeatedly
1791 and serves no real purpose other than showing that it can
1794 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
1796 * examples/.gitignore:
1797 examples: add new test-record to .gitignore
1799 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
1801 * examples/test-record.c:
1802 * gst/rtsp-server/rtsp-client.c:
1803 * gst/rtsp-server/rtsp-media-factory.c:
1804 * gst/rtsp-server/rtsp-media-factory.h:
1805 * gst/rtsp-server/rtsp-media.c:
1806 * gst/rtsp-server/rtsp-media.h:
1807 * tests/check/gst/rtspserver.c:
1808 rtsp-media: Use flags to distinguish between PLAY and RECORD media
1810 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
1812 * examples/test-record.c:
1813 test-record: Set latency for playback-style example to 2s instead of 200ms
1815 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
1817 * tests/check/gst/rtspserver.c:
1818 tests: add some unit tests for ANNOUNCE and RECORD
1819 https://bugzilla.gnome.org/show_bug.cgi?id=743175
1821 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
1823 * gst/rtsp-server/rtsp-client.c:
1824 rtsp-client: fix a couple of leaks in handle_announce
1826 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
1828 * gst/rtsp-server/rtsp-media-factory.c:
1829 * gst/rtsp-server/rtsp-media-factory.h:
1830 * gst/rtsp-server/rtsp-media.c:
1831 * gst/rtsp-server/rtsp-media.h:
1832 rtsp-media: Expose latency setting for setting the rtpbin latency
1834 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
1836 * examples/test-record.c:
1837 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
1839 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
1841 * gst/rtsp-server/rtsp-stream.c:
1842 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
1844 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
1846 * examples/Makefile.am:
1847 * examples/test-record.c:
1848 * gst/rtsp-server/rtsp-client.c:
1849 * gst/rtsp-server/rtsp-client.h:
1850 * gst/rtsp-server/rtsp-media-factory.c:
1851 * gst/rtsp-server/rtsp-media-factory.h:
1852 * gst/rtsp-server/rtsp-media.c:
1853 * gst/rtsp-server/rtsp-media.h:
1854 * gst/rtsp-server/rtsp-session-media.c:
1855 * gst/rtsp-server/rtsp-stream.c:
1856 * gst/rtsp-server/rtsp-stream.h:
1857 Add initial support for RECORD
1858 We currently only support media that is RECORD or PLAY only, not both at once.
1859 https://bugzilla.gnome.org/show_bug.cgi?id=743175
1861 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
1863 * gst/rtsp-server/rtsp-stream.c:
1864 rtsp-stream: RTCP and RTP transport cache cookies seperated
1865 RTCP packets were not sent because the same tr_cache_cookie was used for
1866 both RTP and RTCP. So only one of the tr_cache lists were populated
1867 depending on which one was sent first. If the tr_cache list is not
1868 populated then no packets can be sent. Most often this happened to be
1869 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
1870 resulted in both the tr_cache_lists to be populated regardless of which
1872 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
1874 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
1876 * gst/rtsp-server/rtsp-stream.c:
1877 rtsp-stream: fix false compiler warning
1878 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
1880 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
1882 * gst/rtsp-server/rtsp-client.c:
1883 rtsp-client: log interleaved data received
1885 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
1887 * gst/rtsp-server/rtsp-client.c:
1888 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
1890 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
1892 * gst/rtsp-server/rtsp-client.c:
1893 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
1895 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
1897 * gst/rtsp-server/rtsp-client.c:
1898 rtsp-client: Use a random session ID in the SDP
1899 RFC4566 Section 5.2 says that it should make the username, session id,
1900 nettype, addrtype and unicast address tuple globally unique. Always using
1901 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
1902 Instead let's create a 64 bit random number, which at least brings us
1903 closer to the goal of global uniqueness.
1904 https://tools.ietf.org/html/rfc4566#section-5.2
1906 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
1908 * examples/test-launch.c:
1909 * examples/test-mp4.c:
1910 * examples/test-ogg.c:
1911 * examples/test-uri.c:
1912 examples: Don't call gst_init() and gst_get_option_group()
1913 The latter calls the former at the appropriate time.
1915 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
1917 * gst/rtsp-server/rtsp-client.c:
1918 rtsp-client: Drop trailing \0 of RTSP DATA messages
1919 We add a trailing \0 in GstRTSPConnection to make parsing of
1920 string message bodies easier (e.g. the SDP from DESCRIBE) but
1921 for actual data this means we have to drop it or otherwise
1922 create invalid data.
1924 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
1926 * gst/rtsp-server/rtsp-stream.c:
1927 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
1928 Fixes crash when two threads access handle_new_sample() at the same
1929 time, one for RTP, one for RTCP.
1930 Otherwise, when iterating over the transports cache, it might be modified by
1931 another thread at the same time if the transports cookie has changed.
1932 https://bugzilla.gnome.org/show_bug.cgi?id=742954
1934 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
1936 * gst/rtsp-server/rtsp-stream.c:
1937 rtsp-stream: Set format=TIME on our app sources for TCP
1939 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
1941 * gst/rtsp-server/rtsp-session-pool.c:
1942 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
1943 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
1944 RFC 2326 states that session IDs may consist of alphanumeric as well as
1945 the safe characters $-_.+ -- N.B. the percent character is not allowed.
1946 Previously the session ID was URI-escaped, this meant that any character
1947 which was not alphanumeric or any of the characters +-._~ would be
1948 percent encoded. While the RFC (surprisingly) mentions that linear white
1949 space in session IDs should be URI-escaped, it does not say anything
1950 about other characters. Moreover no white space is allowed in the
1951 session ID. Finally the percent character which is the result of
1952 URI-escaping is not allowed in a session ID.
1953 So there is no reason to do any URI-escaping, and now it is removed.
1954 https://bugzilla.gnome.org/show_bug.cgi?id=742869
1956 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
1959 Automatic update of common submodule
1960 From f2c6b95 to bc76a8b
1962 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
1965 Fix 'make check' from top-level directory
1967 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
1969 * examples/test-launch.c:
1970 * examples/test-mp4.c:
1971 * examples/test-ogg.c:
1972 * examples/test-uri.c:
1973 examples: Add command-line parsing and take a 'port' argument
1974 This allows users to run multiple servers on different ports for testing.
1975 Only done for examples that actually take arguments and hence are capable of
1976 outputting different streams for each instance on each port.
1977 https://bugzilla.gnome.org/show_bug.cgi?id=742115
1979 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
1981 * gst/rtsp-server/rtsp-client.c:
1982 * gst/rtsp-server/rtsp-client.h:
1983 rtsp-client: Add a send_message default signal handler
1984 This allows subclasses to easily hook into the response sending
1985 mechanism without doing everything from a signal, which seems
1986 awkward from subclasses.
1988 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
1991 Automatic update of common submodule
1992 From ef1ffdc to f2c6b95
1994 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1998 configure: add --disable-examples switch
1999 https://bugzilla.gnome.org/show_bug.cgi?id=741678
2001 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
2003 * examples/.gitignore:
2004 * examples/Makefile.am:
2005 * examples/test-video-rtx.c:
2006 examples: add a retransmisison example implementing RFC4588
2007 Currently only SSRC-multiplexed rtx streams are supported
2009 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
2011 * gst/rtsp-server/rtsp-stream.c:
2012 rtsp-stream: Fix some minor memory leaks
2014 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
2016 * gst/rtsp-server/rtsp-media.c:
2017 rtsp-media: Some minor cleanup
2019 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
2021 * gst/rtsp-server/rtsp-stream.c:
2022 rtsp-stream: Fix compiler warnings
2023 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
2024 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
2026 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
2027 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
2030 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
2032 * docs/libs/gst-rtsp-server-sections.txt:
2033 * gst/rtsp-server/rtsp-media-factory.c:
2034 * gst/rtsp-server/rtsp-media-factory.h:
2035 * gst/rtsp-server/rtsp-media.c:
2036 * gst/rtsp-server/rtsp-media.h:
2037 * gst/rtsp-server/rtsp-sdp.c:
2038 * gst/rtsp-server/rtsp-stream.c:
2039 * gst/rtsp-server/rtsp-stream.h:
2040 media: implement ssrc-multiplexed retransmission support
2041 based off RFC 4588 and the server-rtpaux example in -good
2043 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
2045 * gst/rtsp-server/rtsp-client.c:
2046 * gst/rtsp-server/rtsp-stream-transport.c:
2047 * gst/rtsp-server/rtsp-stream.c:
2048 rtsp: Ref transports in hash table.
2049 Also ref streams for transports.
2050 This solves a crash when reciving a rtcp after teardown but before
2052 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
2054 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
2057 Automatic update of common submodule
2058 From 7bb2bce to ef1ffdc
2060 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
2062 * gst/rtsp-server/rtsp-client.c:
2063 client: refactor cleanup of cached media
2065 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
2067 * tests/check/gst/client.c:
2069 The session leak is now fixed, lets remove those FIXME comments.
2071 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
2073 * tests/check/gst/rtspserver.c:
2074 tests: Test to setup two sessions on one connection
2075 https://bugzilla.gnome.org/show_bug.cgi?id=739112
2077 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
2079 * tests/check/gst/rtspserver.c:
2080 tests: Test setup with tcp transport
2081 https://bugzilla.gnome.org/show_bug.cgi?id=739112
2083 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
2085 * gst/rtsp-server/rtsp-client.c:
2086 client: Configure transport after creating session media
2087 The default implementation of configure_client_transport() in
2088 rtsp-client uses the session media when it chooses channels for
2089 interleaved traffic.
2090 https://bugzilla.gnome.org/show_bug.cgi?id=739112
2092 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
2094 * gst/rtsp-server/rtsp-client.c:
2095 * gst/rtsp-server/rtsp-session-media.c:
2096 client: Stop caching media in client when doing setup
2097 If the media has been managed by a session media, it should not be
2098 cached in the client any longer. The GstRTSPSessionMedia object is now
2099 responsible for unpreparing the GstRTSPMedia object using
2100 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
2102 https://bugzilla.gnome.org/show_bug.cgi?id=739112
2104 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2106 * gst/rtsp-server/rtsp-stream.c:
2107 rtsp-stream: unref srtp decoder when leaving bin
2108 https://bugzilla.gnome.org/show_bug.cgi?id=739481
2110 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2112 * gst/rtsp-server/rtsp-client.c:
2113 rtsp-client: mikey memory leaks
2114 https://bugzilla.gnome.org/show_bug.cgi?id=739383
2116 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
2119 Automatic update of common submodule
2120 From 84d06cd to 7bb2bce
2122 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
2125 Parallelise 'make check-valgrind'
2127 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
2130 Automatic update of common submodule
2131 From a8c8939 to 84d06cd
2133 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
2136 Automatic update of common submodule
2137 From 36388a1 to a8c8939
2139 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
2141 * gst/rtsp-server/rtsp-media.c:
2142 rtsp-media: deactivate media when shutting down from paused
2143 This was only done when going directly from playing.
2144 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
2146 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2148 * gst/rtsp-server/rtsp-client.c:
2149 * gst/rtsp-server/rtsp-context.h:
2150 rtsp-client: add stream transport to context
2151 We add the stream transport to the context so we can get the configured
2152 client stream transport in the setup request signal.
2153 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
2155 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2157 * gst/rtsp-server/rtsp-stream.c:
2158 stream: release lock even not all transports have been removed
2159 We don't want to keep the lock even we return FALSE because not all the
2160 transports have been removed. This could lead into a deadlock.
2161 https://bugzilla.gnome.org/show_bug.cgi?id=737797
2163 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
2165 * gst/rtsp-server/rtsp-sdp.c:
2166 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
2167 These were renamed in GstRTPBasePayload in 1.0
2169 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2171 * gst/rtsp-server/rtsp-client.c:
2172 client: set session media to NULL without the lock
2173 We need to set session medias to NULL without the client lock otherwise
2174 we can end up in a deadlock if another thread is waiting for the lock
2175 and media unprepare is also waiting for that thread to end.
2176 https://bugzilla.gnome.org/show_bug.cgi?id=737690
2178 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
2180 * gst/rtsp-server/rtsp-media.c:
2181 rtsp-media: Set state to UNPREPARING in all cases
2183 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
2185 * gst/rtsp-server/rtsp-media.c:
2186 media: set state to unpreparing when unprepare is initiated
2187 https://bugzilla.gnome.org/show_bug.cgi?id=737675
2189 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
2191 * gst/rtsp-server/rtsp-client.c:
2192 rtsp-client: Remove backlog limit while processings requests
2193 If the backlog limit is kept two cases of deadlocks may be
2194 encountered when streaming over TCP. Without the backlog
2195 limit this deadlocks can not happen, at the expence of
2197 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
2199 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
2201 * gst/rtsp-server/rtsp-client.c:
2202 rtsp-client: do not free main context before rtsp watch
2203 https://bugzilla.gnome.org/show_bug.cgi?id=737110
2205 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
2207 * tests/check/gst/rtspserver.c:
2208 tests: Extend unit test timeout to accomodate for valgrind
2209 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
2211 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
2213 * gst/rtsp-server/rtsp-client.c:
2214 * gst/rtsp-server/rtsp-session.c:
2215 * gst/rtsp-server/rtsp-stream-transport.c:
2216 rtsp-*: Treat sending packets to clients as keepalive
2217 As long as gst-rtsp-server can successfully send RTP/RTCP data to
2218 clients then the client must be reading. This change makes the server
2219 timeout the connection if the client stops reading.
2220 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
2222 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
2224 * gst/rtsp-server/rtsp-client.c:
2225 rtsp-client: Allow backlog to grow while expiring session
2226 Allow the send backlog in the RTSP watch to grow to unlimited size while
2227 attempting to bring the media pipeline to NULL due to a session
2228 expiring. Without this change the appsink element cannot change state
2229 because it is blocked while rendering data in the new_sample callback.
2230 This callback will block until it has successfully put the data into the
2231 send backlog. There is a chance that the send backlog is full at this
2232 point which means that the callback may block for a long time, possibly
2233 forever. Therefore the media pipeline may also be prevented from
2234 changing state for a long time.
2235 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
2237 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
2239 * gst/rtsp-server/rtsp-client.c:
2240 rtsp-client: Make old compilers happy
2241 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
2242 Just in case that guint8 doesn't fit in a pointer. Just in case ...
2244 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
2246 * gst/rtsp-server/rtsp-client.c:
2247 client: raise the backlog limits before pausing
2248 We need to raise the backlog limits before pausing the pipeline or else
2249 the appsink might be blocking in the render method in wait_backlog() and
2250 we would deadlock waiting for paused.
2251 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
2253 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
2255 * gst/rtsp-server/rtsp-client.c:
2256 client: make define for the WATCH_BACKLOG
2257 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
2259 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
2261 * gst/rtsp-server/rtsp-client.c:
2262 client: simplify session transport handling
2263 link/unlink of the transport in a session was done to keep track of all
2264 TCP transports and to send RTP/RTCP data to the streams. We can simplify
2265 that by putting all the TCP transports in a hashtable indexed with the
2267 We also don't need to link/unlink the transports when we pause/resume
2268 the streams. The same effect is already achieved when we pause/play the
2269 media. Indeed, when we pause the media, the transport is removed from
2270 the media and the callbacks will not be called anymore.
2271 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
2273 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
2275 * gst/rtsp-server/rtsp-stream-transport.c:
2276 * gst/rtsp-server/rtsp-stream-transport.h:
2277 stream-transport: make method to handle received data
2278 Make a method to handle the data received on a channel. It sends the
2279 data to the stream of the transport on the RTP or RTCP pads based on
2282 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
2284 * examples/test-mp4.c:
2285 test: add example of dumping RTCP reports
2287 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
2289 * gst/rtsp-server/rtsp-media.c:
2290 * gst/rtsp-server/rtsp-stream.c:
2291 * gst/rtsp-server/rtsp-stream.h:
2292 rtsp-media: Make sure that sequence numbers are monotonic after pause
2293 The sequence number is not monotonic for RTP packets after pause. The
2294 reason is basepayloader generates a randon sequence number when the
2295 pipeline goes from ready to pause. With this fix generation of sequence
2296 number will be monotonic when going from pause to play request.
2297 https://bugzilla.gnome.org/show_bug.cgi?id=736017
2299 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
2301 * gst/rtsp-server/rtsp-client.c:
2302 rtsp-client: Protect saved clients watch with a mutex
2303 Fixes a crash when close() is called while merging clients
2304 in handle_tunnel(). In that case close() would destroy the
2305 watch while it is still being used in handle_tunnel().
2306 https://bugzilla.gnome.org/show_bug.cgi?id=735570
2308 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
2310 * gst/rtsp-server/rtsp-stream.c:
2311 rtsp-stream: Remove the multicast group udp sources when removing from the bin
2313 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
2315 * gst/rtsp-server/rtsp-media.c:
2316 * gst/rtsp-server/rtsp-stream.c:
2317 * gst/rtsp-server/rtsp-stream.h:
2318 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
2319 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
2320 seeking and will always continue counting the time. This leads to
2321 the NPT after a backwards seek to be something completely different
2322 to the actual seek position.
2323 https://bugzilla.gnome.org/show_bug.cgi?id=732644
2325 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
2327 * examples/test-appsrc.c:
2328 examples: fix another reference leak
2329 gst_rtsp_media_get_element() returns a new ref.
2331 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2333 * examples/test-appsrc.c:
2334 examples: unref element after usage
2335 gst_bin_get_by_name_recurse_up() returns an element
2336 reference that must be unreffed after usage.
2337 https://bugzilla.gnome.org/show_bug.cgi?id=734546
2339 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
2341 * gst/rtsp-server/rtsp-media.c:
2342 signals: Fix copy-pasto in target-state signal offset
2344 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
2348 Makefile: Add usage of build-checks step
2349 Allows building checks without running them
2351 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
2353 * gst/rtsp-server/rtsp-stream.c:
2354 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
2355 When a UDP multicast transport is used it is expected that the server listens
2356 for RTP and RTCP packets on the multicast group with the corresponding port.
2357 Without this we will never get RTCP packets from clients in multicast mode.
2358 https://bugzilla.gnome.org/show_bug.cgi?id=732238
2360 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
2365 === release 1.4.0 ===
2367 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
2373 * gst-rtsp-server.doap:
2376 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
2378 * gst/rtsp-server/rtsp-media.h:
2379 media: correct misspelled words in description
2380 https://bugzilla.gnome.org/show_bug.cgi?id=733244
2382 === release 1.3.91 ===
2384 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
2390 * gst-rtsp-server.doap:
2393 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
2395 * docs/libs/gst-rtsp-server-sections.txt:
2398 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
2400 * gst/rtsp-server/rtsp-server.c:
2401 server: implement client REMOVE filter
2403 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
2405 * gst/rtsp-server/rtsp-client.c:
2406 * gst/rtsp-server/rtsp-client.h:
2407 client: expose _close() method
2408 Expose a previously internal close method to close the client
2411 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
2413 * gst/rtsp-server/rtsp-session-pool.c:
2414 session-pool: signal session-removed outside of the lock
2415 Release the lock before emiting the session-removed signal.
2417 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
2419 * gst/rtsp-server/rtsp-client.c:
2420 * gst/rtsp-server/rtsp-server.c:
2421 * gst/rtsp-server/rtsp-session-pool.c:
2422 * gst/rtsp-server/rtsp-session.c:
2423 * gst/rtsp-server/rtsp-stream.c:
2424 filter: Release lock in filter functions
2425 Release the object lock before calling the filter functions. We need to
2426 keep a cookie to detect when the list changed during the filter
2427 callback. We also keep a hashtable to make sure we only call the filter
2428 function once for each object in case of concurrent modification.
2429 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
2431 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
2433 * gst/rtsp-server/rtsp-client.c:
2434 client: check if watch is set in handle_teardown()
2435 The unit tests run without a watch
2437 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
2439 * tests/check/gst/client.c:
2440 client tests: send teardown to cleanup session
2442 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
2444 * tests/check/gst/rtspserver.c:
2445 server tests: send teardown to cleanup session
2447 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
2449 * gst/rtsp-server/rtsp-client.c:
2450 client: keep ref to client for the session removed handler
2451 This extra ref will be dropped when all client sessions have been
2452 removed. A session is removed when a client sends teardown, closes its
2453 endpoint of the TCP connection or the sessions expires.
2454 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
2456 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
2458 * gst/rtsp-server/rtsp-client.c:
2459 * gst/rtsp-server/rtsp-session.c:
2460 * tests/check/gst/client.c:
2461 client: manage media in session as a last step
2462 Once we manage a media in a session, we can't unmanage it anymore
2463 without destroying it. Therefore, first check everything before we
2464 manage the media, otherwise if something is wrong we have no way to
2466 If we created a new session and something went wrong, remove the session
2467 again. Fixes a leak in the unit test.
2469 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
2471 * examples/test-mp4.c:
2472 * examples/test-ogg.c:
2473 examples: print 'stream ready at url' for mp4 and ogg example
2475 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
2477 * gst/rtsp-server/rtsp-client.c:
2478 * gst/rtsp-server/rtsp-sdp.c:
2479 rtsp: fix for MIKEY api change
2481 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
2483 * gst/rtsp-server/rtsp-client.c:
2484 client: free watch context only once
2485 The watch context is freed when the source is destroyed. Avoids
2486 a CRITICAL when we try to unref the context twice.
2488 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
2490 * gst/rtsp-server/rtsp-client.c:
2493 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
2495 * gst/rtsp-server/rtsp-client.c:
2496 client: protect sessions with lock
2497 Protect the list of sessions with the lock.
2498 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
2500 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
2502 * gst/rtsp-server/rtsp-client.c:
2503 Client: keep a ref to the session
2504 Don't just keep a weak ref to the session objects but use a hard ref. We
2505 will be notified when a session is removed from the pool (expired) with
2506 the new session-removed signal.
2507 Don't automatically close the RTSP connection when all the sessions of
2508 a client are removed, a client can continue to operate and it can create
2509 a new session if it wants. If you want to remove the client from the
2510 server, you have to use gst_rtsp_server_client_filter() now.
2511 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
2512 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
2514 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
2516 * gst/rtsp-server/rtsp-session-pool.c:
2517 * gst/rtsp-server/rtsp-session-pool.h:
2518 session-pool: add session-removed signal
2519 Add a signal to be notified when a session is removed from the pool.
2521 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
2523 * gst/rtsp-server/Makefile.am:
2524 * gst/rtsp-server/rtsp-server.h:
2525 Make rtsp-server.h a single-include header, use it for G-I
2526 https://bugzilla.gnome.org/show_bug.cgi?id=732411
2528 === release 1.3.90 ===
2530 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
2536 * gst-rtsp-server.doap:
2539 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
2541 * gst/rtsp-server/rtsp-stream.c:
2542 stream: crypto can be NULL
2544 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
2546 * gst/rtsp-server/rtsp-client.c:
2547 * gst/rtsp-server/rtsp-media.c:
2548 * gst/rtsp-server/rtsp-mount-points.c:
2549 introspection: add missing allow-none annotations
2550 https://bugzilla.gnome.org/show_bug.cgi?id=730952
2552 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
2554 * gst/rtsp-server/rtsp-address-pool.c:
2555 * gst/rtsp-server/rtsp-media.c:
2556 * gst/rtsp-server/rtsp-session-media.c:
2557 * gst/rtsp-server/rtsp-session-pool.c:
2558 * gst/rtsp-server/rtsp-stream-transport.c:
2559 * gst/rtsp-server/rtsp-stream.c:
2560 * gst/rtsp-server/rtsp-token.c:
2561 introspection: add (nullable) annotations to return values
2562 https://bugzilla.gnome.org/show_bug.cgi?id=730952
2564 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
2566 * gst/rtsp-server/rtsp-client.c:
2567 * gst/rtsp-server/rtsp-stream.c:
2568 gi: improve annotations
2569 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
2571 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
2573 * gst/rtsp-server/rtsp-client.c:
2574 * gst/rtsp-server/rtsp-media-factory.c:
2575 * gst/rtsp-server/rtsp-media.c:
2576 * gst/rtsp-server/rtsp-server.c:
2577 signals: use generic marshal function
2578 Use the generic C marshal function.
2579 Use more explicit type instead of G_TYPE_POINTER
2581 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
2583 * gst/rtsp-server/rtsp-context.h:
2584 context: add type macro
2586 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
2588 * gst/rtsp-server/rtsp-client.c:
2589 * gst/rtsp-server/rtsp-sdp.c:
2590 * gst/rtsp-server/rtsp-sdp.h:
2591 sdp: hide key length defines
2592 They don't have a namespace.
2594 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
2599 === release 1.3.3 ===
2601 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
2607 * gst-rtsp-server.doap:
2610 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2612 * gst/rtsp-server/rtsp-client.c:
2613 * gst/rtsp-server/rtsp-sdp.c:
2614 * gst/rtsp-server/rtsp-sdp.h:
2615 mikey: add different key length parameters
2616 Add encryption and authentication key length parameters to MIKEY. For
2617 the encoders, the key lengths are obtained from the cipher and auth
2618 algorithms set in the caps. For the decoders, they are obtained while
2619 parsing the key management from the client.
2620 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
2622 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
2624 * tests/check/gst/stream.c:
2625 stream tests: Make sure we get right multicast address from stream
2626 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
2628 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
2630 * gst/rtsp-server/rtsp-client.c:
2631 client: ref the context until rtsp watch is alive
2632 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
2634 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
2636 * gst/rtsp-server/rtsp-client.c:
2637 client: Destroy the rtsp watch after connection close
2639 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
2641 * gst/rtsp-server/rtsp-media.c:
2642 media: fix confusing comment
2644 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
2646 * gst/rtsp-server/rtsp-session.c:
2647 rtsp-session: Timeout in header.
2648 Adding the possbilty to always have timout in header.
2649 This is configurabe with setting "timeout-always-visible".
2650 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
2652 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
2657 === release 1.3.2 ===
2659 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
2666 * gst-rtsp-server.doap:
2669 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
2672 Automatic update of common submodule
2673 From 211fa5f to 1f5d3c3
2675 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
2677 * gst/rtsp-server/rtsp-client.c:
2678 client: store TCP ports in transport
2679 Store the TCP ports in the transport when we are doing RTSP over TCP.
2680 This way, we can easily get to the ports from the transport.
2681 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
2683 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2685 * gst/rtsp-server/rtsp-stream.c:
2686 stream: add signals for new RTP/RTCP encoders
2687 New signals to allow the user to configure the dynamically created
2689 https://bugzilla.gnome.org/show_bug.cgi?id=730228
2691 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
2693 * gst/rtsp-server/rtsp-media.c:
2694 * gst/rtsp-server/rtsp-media.h:
2695 media: Make suspend()/unsuspend() virtual
2696 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
2698 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2700 * gst/rtsp-server/rtsp-client.c:
2701 client: fix send-message signal marshaller
2702 Use generic marshalling for the send-message signal. It has
2703 two POINTER arguments, not just one.
2704 https://bugzilla.gnome.org/show_bug.cgi?id=729900
2706 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
2708 * tests/check/gst/media.c:
2709 tests: add and remove pads only once
2710 In this test we simulate a dynamic pad by watching the caps event.
2711 Because of renegotiation in the base payloader now, this caps is sent
2712 multiple times but we can only deal with 1 invocation, use a variable to
2713 only 'add and remove' the pad once.
2715 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
2717 * tests/check/gst/rtspserver.c:
2718 tests: add unit test for correct handling of Require headers
2719 https://bugzilla.gnome.org/show_bug.cgi?id=729426
2721 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
2723 * gst/rtsp-server/rtsp-client.c:
2724 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
2725 Servers must handle Require headers and must report a failure
2726 if they don't handle any of the Required options, see RFC 2326,
2727 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
2728 https://bugzilla.gnome.org/show_bug.cgi?id=729426
2730 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
2735 === release 1.3.1 ===
2737 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
2743 * gst-rtsp-server.doap:
2746 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
2749 Automatic update of common submodule
2750 From bcb1518 to 211fa5f
2752 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
2757 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
2759 * tests/check/gst/sessionmedia.c:
2760 tests: fix memory leak in sessionmedia unit test
2762 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
2764 * gst/rtsp-server/rtsp-client.c:
2765 client: emit a signal before sending a message
2766 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
2768 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
2770 * gst/rtsp-server/rtsp-client.c:
2771 client: pass context to send_message
2772 Pass the current context to send_message, we will need it later.
2774 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
2776 * gst/rtsp-server/rtsp-client.c:
2777 client: fix typo in comment
2779 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
2781 * gst/rtsp-server/rtsp-media.c:
2782 media: Do not stop thread twice if default_prepare() fails
2784 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
2786 * gst/rtsp-server/rtsp-client.c:
2787 client: set the watch to flushing before going to NULL
2788 First set the watch to flushing so that we unblock any current and
2789 future attempt to send data on the watch, Then set the pipeline to
2791 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
2793 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
2795 * gst/rtsp-server/rtsp-session-pool.c:
2796 * tests/check/gst/sessionpool.c:
2797 rtsp-session-pool: Fixes annotation
2798 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
2799 in the sessionpool test.
2800 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
2802 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
2804 * gst/rtsp-server/rtsp-media.c:
2805 * gst/rtsp-server/rtsp-media.h:
2806 media: make media_prepare virtual
2807 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
2809 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
2811 * gst/rtsp-server/rtsp-media.c:
2812 * tests/check/gst/media.c:
2813 media: stop the thread in more error cases
2815 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
2817 * gst/rtsp-server/rtsp-media.c:
2818 * tests/check/gst/media.c:
2819 media: allow NULL as the thread
2820 Use the default context whan passing a NULL thread.
2822 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
2824 * gst/rtsp-server/rtsp-client.c:
2825 rtsp-client: indent cleanup
2826 Coverity was moaning about unreachable code, and I think it was just
2827 confused by { being before the label. We'll see if it pops up again.
2830 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
2832 * gst/rtsp-server/rtsp-client.c:
2833 * gst/rtsp-server/rtsp-media.c:
2834 client: Add drop-backlog property
2835 When we have too many messages queued for a client (currently hardcoded
2836 to 100) we overflow and drop the messages. Add a drop-backlog property
2837 to control this behaviour. Setting this property to FALSE will retry
2838 to send the messages to the client by waiting for more room in the
2840 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
2842 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
2844 * gst/rtsp-server/rtsp-client.c:
2845 client: support for POST before GET when setting up a tunnel
2847 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
2849 * gst/rtsp-server/rtsp-client.c:
2850 client: remove watch of the second client after http tunnel setup
2851 The second client will be freed after the HTTP tunnel has been set up.
2852 Make sure it's RTSP watch is never dispatched again.
2853 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
2855 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
2857 * gst/rtsp-server/rtsp-media.c:
2858 * tests/check/gst/media.c:
2859 media: Make media_prepare() fail if port allocation fails
2860 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
2862 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
2864 * tests/check/gst/media.c:
2865 media test: cleanup the thread pool in tests
2867 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
2869 * gst/rtsp-server/rtsp-media.c:
2870 * tests/check/gst/media.c:
2871 rtsp-media: Unblock blocked streams in unprepare
2872 The streams will be blocked when a live media is prepared.
2873 The streams should be unblocked in gst_rtsp_media_unprepare.
2874 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
2876 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
2878 * gst/rtsp-server/rtsp-media.c:
2879 media: release the state lock when going to NULL
2880 Set our state to UNPREPARING and release the state-lock before
2881 setting the pipeline to the NULL state. This way, any pad-added
2882 callback will be able to take the state-lock and check that we are now
2883 unpreparing instead of deadlocking.
2884 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
2886 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
2888 * gst/rtsp-server/rtsp-media.c:
2889 media: protect status with lock
2890 Make sure we only update the status with the lock.
2892 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
2894 * gst/rtsp-server/rtsp-client.c:
2895 * gst/rtsp-server/rtsp-sdp.c:
2896 rtsp: update for MIKEY API changes
2898 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
2900 * gst/rtsp-server/rtsp-client.c:
2901 client: parse the mikey response from the client
2902 Parse the mikey response from the client and update the policy for
2905 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
2907 * gst/rtsp-server/rtsp-stream.c:
2908 * gst/rtsp-server/rtsp-stream.h:
2909 stream: add method to set crypto info
2910 Make a method to configure the crypto information of a stream.
2911 Set udpsrc in READY instead of PAUSED so that we can configure caps
2914 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
2916 * gst/rtsp-server/rtsp-client.c:
2917 client: cleanup error paths
2919 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
2921 * gst/rtsp-server/rtsp-media.c:
2924 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
2926 * examples/test-video.c:
2927 test: enable SRTP only on RTSPS
2928 We only want to enable SRTP when doing rtsp over TLS so that we can
2929 exchange the keys in a secure way.
2931 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
2933 * examples/test-video.c:
2934 test: print an error on failure
2936 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
2939 * examples/test-video.c:
2940 * gst/rtsp-server/rtsp-sdp.c:
2941 * gst/rtsp-server/rtsp-stream.c:
2942 * tests/check/Makefile.am:
2943 stream: add SRTP support
2944 Install srtp encoder and decoder elements in rtpbin
2947 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2949 * tests/check/Makefile.am:
2950 * tests/check/gst/sessionpool.c:
2951 tests: Add unit tests for sessionpool
2952 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
2954 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2956 * tests/check/gst/threadpool.c:
2957 tests: Improve code coverage of rtsp-threadpool tests
2958 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
2960 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2962 * tests/check/gst/sessionmedia.c:
2963 tests: Improve code coverage for rtsp-session-media
2964 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
2966 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2968 gobject-introspection: Add annotations to support language bindings
2969 In addition a few cosmetic changes:
2970 * Adjust the order of arguments
2971 * Fix typo: occured -> occurred
2972 * Fix indentation after Return:-clauses
2973 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
2975 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2977 * gst/rtsp-server/rtsp-stream.c:
2978 rtsp-stream: Don't mix IPv4 and IPv6 addresses
2979 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
2981 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
2983 * gst/rtsp-server/rtsp-stream.c:
2984 stream: take caps after the session manager
2985 Take the caps for the SDP after they leave the rtpbin so that we can
2986 also get the properties added by rtpbin elements.
2988 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
2990 * gst/rtsp-server/rtsp-stream.c:
2991 stream: release lock while pushing out packets
2992 Keep a cache of the transports and use this to iterate the transport
2993 while pushing packets. This allows us to release the lock early.
2994 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
2996 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
2998 * gst/rtsp-server/rtsp-client.c:
2999 * gst/rtsp-server/rtsp-client.h:
3000 rtsp-client: vmethod for modifying tunnel GET response
3001 Add a vmethod tunnel_http_response where the response to the HTTP GET
3002 for tunneled connections can be modified.
3003 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
3005 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
3007 * gst/rtsp-server/rtsp-sdp.c:
3008 sdp: make 1 media line per profile
3009 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
3010 line in the SDP for each profile. The client is then supposed to pick
3011 one of the profiles in the SETUP request. Because the m= lines have the
3012 same pt, the client also knows that only 1 option is possible.
3014 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
3016 * gst/rtsp-server/rtsp-media-factory.c:
3017 * gst/rtsp-server/rtsp-media-factory.h:
3018 * gst/rtsp-server/rtsp-media.c:
3019 factory: add profile property and pass to media and streams
3021 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
3023 * examples/test-multicast.c:
3024 * gst/rtsp-server/rtsp-sdp.c:
3025 sdp: pass multicast connection for multicast-only stream
3026 Pass the multicast address of the stream in the connection info in the
3027 SDP so that clients try a multicast connection first.
3028 Only allow multicast connections in the test-multicast example. Also
3029 increase the TTL a little.
3031 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3034 .gitignore: Ignore gcov intermediate files
3035 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
3037 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
3039 * gst/rtsp-server/rtsp-stream.c:
3040 stream: release some locks in error cases
3042 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3044 docs: Enable and fix gtk-doc warnings
3045 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
3046 * addresspool/mediafactory: Add missing annotation colon
3047 * stream: Annotate return value
3048 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
3050 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
3053 Automatic update of common submodule
3054 From fe1672e to bcb1518
3056 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
3059 Automatic update of common submodule
3060 From 1a07da9 to fe1672e
3062 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
3064 * examples/Makefile.am:
3065 examples: use LDADD for libs instead of LDFLAGS
3067 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
3070 configure: make sure releases are in .doap file
3072 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
3074 * examples/test-cgroups.c:
3075 examples: test-cgroups: don't put code with side effects into g_assert()
3076 The g_assert() might get compiled out with the right
3077 compiler/preprocessor flags.
3079 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
3081 * examples/.gitignore:
3082 examples: add cgroup test binary to .gitignore
3084 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
3086 * examples/test-cgroups.c:
3087 examples: fix cgroup test build
3088 Fixes build failure caused by compiler warning:
3089 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
3091 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
3094 .gitignore: ignore temp files created in the course of 'make check'
3096 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
3098 * gst/rtsp-server/rtsp-media.c:
3099 rtsp-media: don't loose frames handling new PLAY request
3100 If client supplied a range check if the range specifies the start point.
3101 If not, then do an accurate seek to the current position. If a start
3102 point was specified do do a key unit seek to make sure the streaming
3103 starts with decodeable frames.
3104 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
3106 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
3108 * gst/rtsp-server/rtsp-media.c:
3109 Revert "media: only flush when setting a new start position"
3110 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
3111 We need to do the flush in all cases, demuxer block currently for
3114 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
3116 * gst/rtsp-server/rtsp-media.c:
3117 media: only flush when setting a new start position
3118 Only flush the pipeline when we change the start position with
3120 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
3122 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
3124 * gst/rtsp-server/rtsp-stream.c:
3125 stream: set ttl-mc before adding the socket
3126 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
3127 never be set on socket.
3128 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
3130 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
3132 * gst/rtsp-server/rtsp-media.c:
3133 media: stop thread if media is already prepared
3134 in gst_rtsp_media_prepare() the thread is not used if media is already
3135 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
3137 https://bugzilla.gnome.org/show_bug.cgi?id=724182
3139 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
3142 build: Ship gst-rtsp-server.doap file
3144 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
3146 * tests/check/gst/rtspserver.c:
3147 tests: Fix another compiler warning with gcc
3149 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
3151 * gst/rtsp-server/rtsp-client.c:
3152 * gst/rtsp-server/rtsp-mount-points.c:
3153 * gst/rtsp-server/rtsp-stream.c:
3154 * tests/check/gst/client.c:
3155 rtsp-server: Fix lots of compiler warnings with clang
3157 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
3160 * gst-rtsp-server.doap:
3161 * tests/Makefile.am:
3162 configure: Synchronise with the configure scripts of the other modules
3164 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
3167 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
3169 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
3171 * gst/rtsp-server/rtsp-media.c:
3172 * gst/rtsp-server/rtsp-stream.c:
3173 Revert "rtsp-server: support build against last stable release"
3174 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
3175 Let us require 1.2.3 now, which is going to be released in a few
3178 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
3180 * gst/rtsp-server/rtsp-session-media.c:
3181 * gst/rtsp-server/rtsp-stream-transport.c:
3182 session: improve RTP-Info
3183 Ignore streams that can't generate RTP-Info instead of failing.
3184 Don't return the empty string when all streams are unconfigured but
3185 return NULL so that we don't generate and empty RTP-Info header.
3186 Improve docs a little.
3188 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
3190 * gst/rtsp-server/rtsp-session-media.c:
3191 Don't free rtpinfo GString when it is NULL
3192 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
3194 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
3196 * gst/rtsp-server/rtsp-media.c:
3197 media: only set keyframe flag when modifying start
3198 Only set the keyframe flag when we modify the start position. The
3199 keyframe flag should probably be ignored when no change is requested but
3200 until we can claim this is all documented properly and all demuxer
3201 implement this, avoid setting the flag.
3202 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
3204 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
3206 * gst/rtsp-server/rtsp-thread-pool.c:
3207 thread-pool: Unref source after mainloop has quit to avoid races in GLib
3208 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
3210 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
3212 * gst/rtsp-server/rtsp-stream.c:
3213 stream: handle NULL seqnum and rtptime arguments
3215 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
3217 * gst/rtsp-server/rtsp-thread-pool.c:
3218 * tests/check/gst/threadpool.c:
3219 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
3220 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
3222 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
3224 * gst/rtsp-server/rtsp-stream.c:
3225 stream: add fallback for missing stats property
3226 Use a fallback when the payloader does not have a stats property
3227 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
3229 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
3232 Automatic update of common submodule
3233 From f7bc1c3 to 1a07da9
3235 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
3237 * gst/rtsp-server/rtsp-stream.c:
3238 stream: don't leak stats structure
3239 Don't leak the stats structure and deal with NULL stats.
3241 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
3243 * gst/rtsp-server/rtsp-stream.c:
3244 stream: Get rtpinfo properties atomically from payloader
3245 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
3247 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
3249 * gst/rtsp-server/rtsp-media.c:
3250 media: refactor state change functions and signals
3251 Make functions to set the target state and the pipeline state and emit
3252 the signals from those functions.
3254 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
3256 * gst/rtsp-server/rtsp-media.c:
3257 * gst/rtsp-server/rtsp-media.h:
3258 media: add signal to notify of pending state changes
3260 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
3262 * gst/rtsp-server/rtsp-media.c:
3263 * gst/rtsp-server/rtsp-stream.c:
3264 rtsp-server: support build against last stable release
3265 Until 1.2.3 is out with the new get_type function and we
3268 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
3270 * gst/rtsp-server/rtsp-stream.c:
3271 stream: fix compilation
3273 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
3275 * gst/rtsp-server/rtsp-media.c:
3276 * gst/rtsp-server/rtsp-media.h:
3277 * gst/rtsp-server/rtsp-stream.c:
3278 * gst/rtsp-server/rtsp-stream.h:
3279 stream: add property to configure profiles
3281 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
3283 * gst/rtsp-server/rtsp-client.c:
3284 client: let stream check supported transport
3285 Delegate the check if a transport is allowed to the stream.
3286 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
3288 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
3290 * gst/rtsp-server/rtsp-stream.c:
3291 * gst/rtsp-server/rtsp-stream.h:
3292 stream: add method to check supported transport
3293 Add a method to check if a transport is supported
3295 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
3298 configure.ac: Only check for gstreamer-check, not check
3299 We include check in gstreamer-check since quite some time now.
3301 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
3303 * gst/rtsp-server/rtsp-session-media.c:
3304 * gst/rtsp-server/rtsp-stream-transport.c:
3305 * gst/rtsp-server/rtsp-stream.c:
3306 * gst/rtsp-server/rtsp-stream.h:
3307 stream: return clock-rate from get_rtpinfo
3308 And use it to correct the rtptime to the requested start-time.
3309 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
3311 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
3313 * gst/rtsp-server/rtsp-session-media.c:
3314 * gst/rtsp-server/rtsp-stream-transport.c:
3315 * gst/rtsp-server/rtsp-stream-transport.h:
3316 session-media: calculate start-time
3318 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
3320 * gst/rtsp-server/rtsp-stream-transport.c:
3321 * gst/rtsp-server/rtsp-stream.c:
3322 * gst/rtsp-server/rtsp-stream.h:
3323 stream: also return the running-time
3324 Return the running-time in the rtpinfo as well.
3326 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
3328 * gst/rtsp-server/rtsp-client.c:
3329 * gst/rtsp-server/rtsp-session-media.c:
3330 * gst/rtsp-server/rtsp-session-media.h:
3331 * gst/rtsp-server/rtsp-stream-transport.c:
3332 * gst/rtsp-server/rtsp-stream-transport.h:
3333 session-media: let the session-media make the RTPInfo
3334 Add method to create the RTPInfo for a stream-transport.
3335 Add method to create the RTPInfo for all stream-transports in a
3337 Use the session-media RTPInfo code in client. This allows us to refactor
3338 another method to link the TCP callbacks.
3340 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
3342 mount-points: sort sequence before g_sequence_lookup
3343 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
3344 sort sequence if dirty, otherwise lookup will fail.
3345 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
3347 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
3350 configure: rename package from gst-rtsp to gst-rtsp-server
3351 To match git module name and avoid confusion with the
3352 rtsp lib in gst-plugins-base and rtsp plugin in -good.
3354 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
3357 configure: bump core/base/good requirement to 1.2.0
3358 Bump to released stable version and make implicit
3359 requirements explicit.
3361 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
3366 Fix broken gettext setup which is not used anyway
3368 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
3371 Automatic update of common submodule
3372 From dbedaa0 to d48bed3
3374 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
3376 * gst/rtsp-server/rtsp-client.c:
3377 * gst/rtsp-server/rtsp-media.c:
3378 * gst/rtsp-server/rtsp-media.h:
3379 media: add setup_sdp vmethod
3380 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
3381 gst_rtsp_media_setup_sdp.
3382 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
3384 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
3386 * gst/rtsp-server/rtsp-stream.c:
3387 rtsp-stream: Check return value of sscanf
3388 streamid is only valid if sscanf matched something.
3390 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
3392 * gst/rtsp-server/rtsp-client.c:
3393 rtsp-client: Fix iteration
3394 Wouldn't even enter the code block otherwise (i++ was used as the check
3395 and not the postfix).
3397 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
3399 * gst/rtsp-server/rtsp-client.c:
3400 * gst/rtsp-server/rtsp-client.h:
3401 client: add vmethod to configure media and streams
3402 Implement a vmethod that can be used to configure the media and the
3403 streams based on the current context. Handle the blocksize handling in
3404 the default handler.
3405 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
3407 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
3410 Make git ignore more unit test binaries
3412 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
3414 * gst/rtsp-server/rtsp-address-pool.h:
3415 * gst/rtsp-server/rtsp-auth.h:
3416 * gst/rtsp-server/rtsp-client.h:
3417 * gst/rtsp-server/rtsp-context.h:
3418 * gst/rtsp-server/rtsp-media-factory-uri.h:
3419 * gst/rtsp-server/rtsp-media-factory.h:
3420 * gst/rtsp-server/rtsp-media.h:
3421 * gst/rtsp-server/rtsp-mount-points.h:
3422 * gst/rtsp-server/rtsp-server.h:
3423 * gst/rtsp-server/rtsp-session-media.h:
3424 * gst/rtsp-server/rtsp-session-pool.h:
3425 * gst/rtsp-server/rtsp-session.h:
3426 * gst/rtsp-server/rtsp-stream-transport.h:
3427 * gst/rtsp-server/rtsp-stream.h:
3428 * gst/rtsp-server/rtsp-thread-pool.h:
3429 * gst/rtsp-server/rtsp-token.h:
3430 rtsp-server: add padding to many public structures
3431 Not mini objects though, since they are not subclassable
3432 anyway, nor kept on the stack or inlined in a structure.
3434 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
3436 media: add new create_rtpbin vmethod
3437 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
3438 https://bugzilla.gnome.org/show_bug.cgi?id=719734
3440 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
3442 * tests/check/gst/media.c:
3443 tests: fix memory leak, free test's thread pool
3444 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
3446 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
3448 * gst/rtsp-server/rtsp-stream-transport.c:
3449 stream-transport: free url in finalize
3451 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
3453 * gst/rtsp-server/rtsp-media.c:
3454 media: also do state change in suspended state
3456 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
3458 * gst/rtsp-server/rtsp-client.c:
3459 * gst/rtsp-server/rtsp-media.c:
3460 media: also handle prepare and range in suspended state
3461 When we are suspended, we are already prepared.
3462 We can get the range in the suspended state.
3464 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
3466 * tests/check/Makefile.am:
3467 * tests/check/gst/sessionmedia.c:
3468 check: add test for uri in setup
3469 Added unit tests for the new functionality in GstRTSPStreamTransport.
3470 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
3472 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
3474 * gst/rtsp-server/rtsp-client.c:
3475 client: store setup uri and use in PLAY response
3476 Store the uri used when doing the setup and use that in the PLAY
3478 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
3480 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
3482 * gst/rtsp-server/rtsp-stream-transport.c:
3483 * gst/rtsp-server/rtsp-stream-transport.h:
3484 stream-transport: add method to get/set url
3486 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
3488 * gst/rtsp-server/rtsp-client.c:
3489 client: suspend after SDP and unsuspend before PLAYING
3490 Based on patches by Ognyan Tonchev <ognyan@axis.com>
3491 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
3493 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
3495 * gst/rtsp-server/rtsp-media-factory.c:
3496 * gst/rtsp-server/rtsp-media-factory.h:
3497 * gst/rtsp-server/rtsp-media.c:
3498 * gst/rtsp-server/rtsp-media.h:
3499 * gst/rtsp-server/rtsp-session-media.c:
3500 * gst/rtsp-server/rtsp-session.c:
3501 * tests/check/gst/media.c:
3502 * tests/check/gst/mediafactory.c:
3503 media: add suspend modes
3504 Add support for different suspend modes. The stream is suspended right after
3505 producing the SDP and after PAUSE. Different suspend modes are available that
3506 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
3507 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
3508 state and RESET will bring the pipeline to the NULL state.
3509 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
3510 this means that the pipeline needs to be prerolled again.
3511 Base on patches by Ognyan Tonchev <ognyan@axis.com>
3512 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3514 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
3516 * gst/rtsp-server/rtsp-media.c:
3517 media: start live streams in blocked state
3518 Start live streams in the blocked state and make them preroll using the
3519 messages. This ensure that no data is played by the sink until we explicitly
3520 unblock the stream right before going to PLAYING.
3521 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3523 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
3525 * gst/rtsp-server/rtsp-media.c:
3526 media: refactor starting and waiting for preroll
3527 Based on patches from Ognyan Tonchev <ognyan@axis.com>
3528 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3530 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
3532 * gst/rtsp-server/rtsp-stream.c:
3533 * gst/rtsp-server/rtsp-stream.h:
3534 stream: add API to block streams
3535 Add an API to block on the streams and make it post a message.
3536 Based on patch by Ognyan Tonchev <ognyan@axis.com>
3537 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3539 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
3541 * docs/libs/Makefile.am:
3542 docs: Specify the override file
3543 Even if it's empty (for now) it avoids make distcheck complaining
3545 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
3547 * gst/rtsp-server/rtsp-media.c:
3548 media: move default implementations to where they are used
3550 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
3552 * gst/rtsp-server/rtsp-media.c:
3553 media: take the right lock in gst_rtsp_media_set_pipeline_state()
3554 We need to take the state_lock when calling this method.
3556 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
3558 * gst/rtsp-server/rtsp-media.c:
3559 media: handle add-added on non-bins too
3560 Handle dynamic payloaders that are not bins, as used in the unit-test.
3562 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3564 * gst/rtsp-server/rtsp-media-factory.c:
3565 * gst/rtsp-server/rtsp-media-factory.h:
3566 * gst/rtsp-server/rtsp-media.c:
3567 rtsp-media/-factory: Fix request pad name comments
3568 These must be escaped for gtk-doc to parse the comments without warnings.
3570 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
3572 rtsp-media: remove transports if media is in error status
3573 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
3574 trying to change to GST_STATE_NULL and media is in error status, we
3575 remove all transports.
3576 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
3578 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
3580 * gst/rtsp-server/rtsp-media.c:
3581 rtsp-media: use element metadata to find payloader
3582 Use the element metadata to find the payloader instead of checking
3584 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
3586 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
3588 rtsp-stream: add getter for payload type
3589 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
3590 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
3591 element and create the stream with this one instead of the dynpay%d
3593 https://bugzilla.gnome.org/show_bug.cgi?id=712396
3595 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3597 * gst/rtsp-server/rtsp-client.c:
3598 * gst/rtsp-server/rtsp-context.h:
3599 * gst/rtsp-server/rtsp-media.c:
3600 * gst/rtsp-server/rtsp-mount-points.c:
3601 * gst/rtsp-server/rtsp-server.c:
3602 * gst/rtsp-server/rtsp-token.c:
3603 rtsp-*: Refer to NULL as a constant in comments
3605 https://bugzilla.gnome.org/show_bug.cgi?id=714988
3607 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3609 rtsp-*: Fix type name typos in comments
3610 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
3611 * rtsp-auth: Refer to part of constant name as text
3612 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
3613 * rtsp-session-media: Fix GstRTSPSessionMedia typo
3614 * rtsp-stream: Fix typo when refering to GstBin
3615 https://bugzilla.gnome.org/show_bug.cgi?id=714988
3617 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3620 * docs/libs/gst-rtsp-server-docs.sgml:
3621 * docs/libs/gst-rtsp-server-sections.txt:
3622 docs: Improve documentation
3623 * Include annotation-glossary to quiet gtk-doc
3624 * Rename remaining ClientState -> Context
3625 * Rename object hierarchy file
3626 * Remove stale chapter references
3627 * Add missing function and object references
3628 * Include missing GstRTSPAddressPoolResult
3629 https://bugzilla.gnome.org/show_bug.cgi?id=714988
3631 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
3633 * gst/rtsp-server/rtsp-client.c:
3634 * gst/rtsp-server/rtsp-server.c:
3635 * gst/rtsp-server/rtsp-session-pool.c:
3636 * gst/rtsp-server/rtsp-session.c:
3637 * gst/rtsp-server/rtsp-stream.c:
3638 rtsp-server: sprinkle some allow-none annotations for g-i
3640 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
3642 * gst/rtsp-server/rtsp-stream.c:
3643 * gst/rtsp-server/rtsp-stream.h:
3644 stream: add method to filter transports
3645 Add a method to safely iterate and collect the stream transports
3646 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
3648 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
3650 * gst/rtsp-server/rtsp-client.c:
3651 * gst/rtsp-server/rtsp-server.c:
3652 * gst/rtsp-server/rtsp-session-pool.c:
3653 * gst/rtsp-server/rtsp-session.c:
3654 rtsp: allow NULL func in filters
3655 Passing a null function make the filters return a list of
3658 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
3660 * gst/rtsp-server/rtsp-address-pool.c:
3661 * tests/check/gst/addresspool.c:
3662 address-pool: fix address increment
3663 Use a guint instead of guint8 to increment the address. It's still not
3664 completely correct because a guint might not be able to hold the complete
3665 address range, but that's an enhacement for later.
3666 Add unit test to test improved behaviour.
3667 https://bugzilla.gnome.org/show_bug.cgi?id=708237
3669 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
3671 * gst/rtsp-server/rtsp-client.c:
3672 * tests/check/gst/client.c:
3673 client: allow absolute path in requests
3674 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
3676 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
3678 * gst/rtsp-server/rtsp-client.c:
3679 * gst/rtsp-server/rtsp-client.h:
3680 client: make make_path_from_uri a vmethod
3682 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
3684 * docs/libs/gst-rtsp-server-sections.txt:
3685 * gst/rtsp-server/rtsp-stream.c:
3686 * gst/rtsp-server/rtsp-stream.h:
3687 * tests/check/Makefile.am:
3688 * tests/check/gst/stream.c:
3689 stream: Add functions to get rtp and rtcp sockets
3690 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
3692 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
3694 * gst/rtsp-server/rtsp-context.c:
3695 * gst/rtsp-server/rtsp-context.h:
3696 context: defing a GType for the context
3697 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
3699 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
3701 * gst/rtsp-server/Makefile.am:
3702 * gst/rtsp-server/rtsp-auth.c:
3703 * gst/rtsp-server/rtsp-context.c:
3704 * gst/rtsp-server/rtsp-media.c:
3705 * gst/rtsp-server/rtsp-mount-points.c:
3706 * gst/rtsp-server/rtsp-server.h:
3707 * gst/rtsp-server/rtsp-session-media.c:
3708 * gst/rtsp-server/rtsp-session.c:
3709 * gst/rtsp-server/rtsp-stream.c:
3710 Fixed several GIR warnings
3712 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
3714 * gst/rtsp-server/rtsp-auth.c:
3717 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3719 * tests/check/Makefile.am:
3720 * tests/check/gst/token.c:
3721 tests: Add unit tests for token
3722 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
3724 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3726 * gst/rtsp-server/rtsp-token.c:
3727 token: Validate args for gst_rtsp_token_is_allowed
3728 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
3730 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3732 * gst/rtsp-server/rtsp-token.c:
3733 token: Fix bug when creating empty token
3734 We always want to have a valid GstStructure in the token.
3735 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
3737 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
3739 * gst/rtsp-server/rtsp-thread-pool.c:
3740 thread-pool: avoid race in shutdown
3741 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
3742 don't actually stop the mainloop ever. Solve this race by adding an idle source
3743 to the mainloop that calls the _quit. This way we immediately exit the mainloop
3744 if quit was called before we started it.
3746 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3748 * tests/check/Makefile.am:
3749 * tests/check/gst/permissions.c:
3750 tests: Add unit tests for permissions
3751 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
3753 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3755 * tests/check/gst/mediafactory.c:
3756 tests: Test mediafactory permissions
3757 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
3759 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3761 * gst/rtsp-server/rtsp-permissions.c:
3762 permissions: Fix refcounting when adding/removing roles
3763 Previously a role that was removed was unreffed twice, and when
3764 replacing an existing role the replaced role was freed while still being
3765 referenced. Both bugs are now fixed.
3766 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
3768 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3770 * tests/check/gst/media.c:
3771 * tests/check/gst/mediafactory.c:
3772 * tests/check/gst/rtspserver.c:
3773 tests: Check gst_rtsp_url_parse return value
3774 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
3776 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
3779 Automatic update of common submodule
3780 From 865aa20 to dbedaa0
3782 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
3784 * gst/rtsp-server/rtsp-server.c:
3785 rtsp-server: Fix socket leak
3786 https://bugzilla.gnome.org/show_bug.cgi?id=710088
3788 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
3790 * gst/rtsp-server/rtsp-session-pool.c:
3791 rtsp-session-pool: Make sure session IDs are properly URI-escaped
3792 https://bugzilla.gnome.org/show_bug.cgi?id=643812
3794 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
3796 * examples/.gitignore:
3797 * examples/test-video.c:
3798 examples: fix compilation when WITH_AUTH is defined
3799 https://bugzilla.gnome.org/show_bug.cgi?id=710228
3801 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
3804 gitignore: Add new test binary
3806 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
3808 * tests/check/Makefile.am:
3809 * tests/check/gst/threadpool.c:
3810 thread-pool: Add unit test for the thread pools
3811 https://bugzilla.gnome.org/show_bug.cgi?id=710228
3813 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
3815 * gst/rtsp-server/rtsp-thread-pool.c:
3816 thread-pool: Fix thread leak when reusing threads
3817 https://bugzilla.gnome.org/show_bug.cgi?id=709730
3819 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
3821 * gst/rtsp-server/rtsp-server.c:
3822 * tests/check/gst/rtspserver.c:
3823 tests: fixed racy behavior in rtspserver tests
3824 https://bugzilla.gnome.org/show_bug.cgi?id=710078
3826 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3828 * tests/check/gst/addresspool.c:
3829 tests: Improve address pool unit tests
3830 Add a range with mixed IPV4 and IPV6 addresses to pool.
3831 Get an IPV4 address from an IPV6-only pool.
3832 Get an IPV6 address from an IPV4-only pool.
3833 Reserve a IPV6 address from an IPV4-only pool.
3834 Check for unicast addresses in multicast-only pool.
3835 Check for unicast addresses in uni-/multicast-mixed pool.
3836 https://bugzilla.gnome.org/show_bug.cgi?id=710128
3838 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3840 * gst/rtsp-server/rtsp-client.c:
3841 client: append query string in PAUSE/PLAY/TEARDOWN as well
3843 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
3845 * gst/rtsp-server/rtsp-client.c:
3846 client: Add query to control path
3847 If the SETUP url contains a query it must be appended to the control
3848 path so that it matches any already created stream in the media. The
3849 query will also be appended to the session media path.
3851 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3853 * gst/rtsp-server/rtsp-media.c:
3854 rtsp-media: remove old line
3856 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
3858 * gst/rtsp-server/rtsp-stream.c:
3859 stream: Correct control comparison
3860 https://bugzilla.gnome.org/show_bug.cgi?id=709176
3862 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3864 * gst/rtsp-server/rtsp-media.c:
3865 media: Check dynamically if the pipeline supports seeking
3866 We should not depend on whether or not the pipeline state change
3867 returned NO_PREROLL or not. A media could dynamically change its
3868 element and switch from seekable to non seekable so it's best to test
3869 the seekable nature of the pipeline dynamically when we try to do a seek.
3871 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3873 * gst/rtsp-server/rtsp-media.c:
3874 media: Return FALSE if seeking is not supported
3876 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3878 * gst/rtsp-server/rtsp-media.c:
3879 rtsp-media: don't seek accurate by default
3880 Accurate seeking is perhaps a little overkill in the most common situation and
3881 causes some formats (mp3) over slow media to seek extremely slowly.
3883 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
3885 * tests/check/gst/rtspserver.c:
3886 tests: fix unit test
3887 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
3889 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
3891 * gst/rtsp-server/rtsp-client.c:
3892 client: Reply 400 if media cannot be constructed
3893 Reply 400 Bad Request instead of 503 Service Unavailable if media
3894 cannot be constructed in SETUP.
3895 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
3897 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
3899 * gst/rtsp-server/rtsp-client.c:
3900 client: Send setup reply once only
3901 If find_media() failed in handle_setup_request() two replies was sent.
3902 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
3904 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
3907 Automatic update of common submodule
3908 From 6b03ba7 to 865aa20
3910 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
3912 * gst/rtsp-server/rtsp-server.c:
3913 server: Emit client-connected signal earlier
3914 Emit client-connected before the client ref is given to a GSource,
3915 otherwise client-connected can be emitted after the client object has
3918 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
3920 * gst/rtsp-server/rtsp-address-pool.c:
3921 * gst/rtsp-server/rtsp-address-pool.h:
3922 * gst/rtsp-server/rtsp-stream.c:
3923 * tests/check/gst/addresspool.c:
3924 addresspool: return reason of failure
3925 Let gst_rtsp_address_pool_reserve_address() return the reason why
3926 the address could not be reserved.
3927 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
3929 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
3932 autogen.sh: Sync behaviour with other GStreamer modules
3933 Allows building from outside of tree amongst other things
3935 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
3938 Automatic update of common submodule
3939 From b613661 to 6b03ba7
3941 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
3944 Automatic update of common submodule
3945 From 74a6857 to b613661
3947 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
3950 Automatic update of common submodule
3951 From 01a7a46 to 74a6857
3953 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
3955 * gst/rtsp-server/rtsp-client.c:
3956 client: Do not read beyond end of path string
3957 If the setup was done without a control url, make sure we don't try to read the
3958 non-existing control string and crash.
3960 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3962 * gst/rtsp-server/rtsp-client.c:
3963 client: Fix RTPInfo header
3964 Refactor the method to make the content_base.
3965 Use the content-base and the control url to construct the RTPInfo
3968 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3970 * gst/rtsp-server/rtsp-client.c:
3971 client: map url to path only in describe
3972 Only map the request url to a path in the DESCRIBE method. The SDP then
3973 contains the base and control urls that should be used to SETUP/PAUSE/
3974 PLAY/TEARDOWN the media.
3976 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3978 * gst/rtsp-server/rtsp-client.c:
3979 Revert "client: map URL to path in requests"
3980 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
3981 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
3982 contains the base and control urls which are used in the SETUP, PLAY,
3983 PAUSE and TEARDOWN requests.
3985 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3987 * gst/rtsp-server/rtsp-client.c:
3988 client: map URL to path in requests
3990 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3992 * gst/rtsp-server/rtsp-client.c:
3993 * gst/rtsp-server/rtsp-mount-points.c:
3994 * gst/rtsp-server/rtsp-mount-points.h:
3995 mount-points: make vmethod to make path from uri
3996 Make a vmethod to transform an url into a path. The path is then used to lookup
3997 the factory. This makes it possible to also use other bits of the url, such as
3998 the query parameters, to locate the factory.
4000 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
4002 * gst/rtsp-server/rtsp-thread-pool.c:
4003 * gst/rtsp-server/rtsp-thread-pool.h:
4004 thread-pool: Add cleanup to wait for the threadpool to finish
4005 Also fix race condition if two threads are asking for the first
4006 thread from the thread pool at once. This would case two internal
4007 GThreadPools to be created.
4008 https://bugzilla.gnome.org/show_bug.cgi?id=707753
4010 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
4012 * gst/rtsp-server/rtsp-client.c:
4013 * tests/check/gst/client.c:
4014 client: free threadpool
4015 https://bugzilla.gnome.org/show_bug.cgi?id=707638
4017 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
4019 * tests/check/gst/mountpoints.c:
4020 mountpoints tests: unref matched factories
4021 https://bugzilla.gnome.org/show_bug.cgi?id=707638
4023 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
4025 * tests/check/gst/media.c:
4026 media tests: unref thread pool and caps
4027 https://bugzilla.gnome.org/show_bug.cgi?id=707638
4029 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
4031 * gst/rtsp-server/rtsp-auth.c:
4032 * gst/rtsp-server/rtsp-media-factory.c:
4033 * gst/rtsp-server/rtsp-media.c:
4034 auth, media, media-factory: unref permissions
4035 https://bugzilla.gnome.org/show_bug.cgi?id=707638
4037 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4039 * examples/Makefile.am:
4040 Makefile: add rule for appsrc example
4042 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4044 * examples/test-appsrc.c:
4045 tests: add appsrc example
4046 Add an example on how to use appsrc to feed the server pipeline with data.
4048 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
4050 * gst/rtsp-server/rtsp-client.c:
4051 rtsp-client: remove query part from content-base string
4052 Make sure that after the control url has been resolved, it's
4053 not a part of the query-string.
4054 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
4056 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4058 * gst/rtsp-server/rtsp-client.c:
4059 client: don't check url in response
4060 There is no url or method in the response to check
4062 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4064 * gst/rtsp-server/rtsp-client.c:
4065 * gst/rtsp-server/rtsp-client.h:
4066 Add handle-response signal for when we receive a GET_PARAMETER response
4068 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4070 * gst/rtsp-server/rtsp-server.c:
4071 Fix gst_rtsp_server_client_filter, using wrong variable type
4073 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
4075 * gst/rtsp-server/rtsp-media-factory-uri.c:
4076 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
4077 For AAC we need to check for framed=true instead of parsed=true.
4078 https://bugzilla.gnome.org/show_bug.cgi?id=701384
4080 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4082 * gst/rtsp-server/rtsp-stream.c:
4083 stream: optimize pipeline for protocols
4084 When TCP is not an allowed protocol for the stream, avoid creating the
4085 appsrc/appsink/queue and tee elements.
4087 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4089 * gst/rtsp-server/rtsp-media.c:
4090 media: set protocols on streams
4092 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4094 * gst/rtsp-server/rtsp-client.c:
4095 client: use protocols supported by stream
4097 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4099 * gst/rtsp-server/rtsp-media-factory.c:
4100 * gst/rtsp-server/rtsp-media.c:
4101 * gst/rtsp-server/rtsp-stream.c:
4102 media-factory: allow all protocols
4104 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4106 * gst/rtsp-server/rtsp-media.c:
4107 media: configure protocols in new streams
4109 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4111 * gst/rtsp-server/rtsp-stream.c:
4112 * gst/rtsp-server/rtsp-stream.h:
4113 stream: add protocols property
4115 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4117 * gst/rtsp-server/rtsp-media.c:
4118 rtsp-media: send state in "new-state" signal
4119 https://bugzilla.gnome.org/show_bug.cgi?id=705110
4121 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
4124 build: add subdir-objects to AM_INIT_AUTOMAKE
4125 Fixes warnings with automake 1.14
4126 https://bugzilla.gnome.org/show_bug.cgi?id=705350
4128 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4130 * docs/libs/gst-rtsp-server-sections.txt:
4131 * gst/rtsp-server/rtsp-client.c:
4132 * gst/rtsp-server/rtsp-server.c:
4133 * gst/rtsp-server/rtsp-server.h:
4134 server: add method to iterate clients of server
4136 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4138 * gst/rtsp-server/rtsp-media.c:
4139 * gst/rtsp-server/rtsp-media.h:
4140 Add vmethod for rtsp-media subclass to access rtpbin
4142 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4144 * gst/rtsp-server/rtsp-client.h:
4145 small documentation fix
4147 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4149 * gst/rtsp-server/rtsp-client.c:
4150 Do not take range header if range is invalid
4152 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4154 * docs/libs/gst-rtsp-server-sections.txt:
4155 * gst/rtsp-server/rtsp-media.c:
4156 media: add docs for new method
4158 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4160 * gst/rtsp-server/rtsp-media.c:
4161 * gst/rtsp-server/rtsp-media.h:
4162 Add API to rtsp-media set the pipeline's state
4164 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4166 * gst/rtsp-server/rtsp-media.c:
4167 Update current position/duration when gst_rtsp_media_get_range_string is called
4169 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4171 * examples/test-cgroups.c:
4172 tests: add some more docs
4174 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4176 * examples/test-cgroups.c:
4177 * gst/rtsp-server/Makefile.am:
4178 * gst/rtsp-server/rtsp-auth.c:
4179 * gst/rtsp-server/rtsp-auth.h:
4180 * gst/rtsp-server/rtsp-client.c:
4181 * gst/rtsp-server/rtsp-client.h:
4182 * gst/rtsp-server/rtsp-context.c:
4183 * gst/rtsp-server/rtsp-context.h:
4184 * gst/rtsp-server/rtsp-params.c:
4185 * gst/rtsp-server/rtsp-params.h:
4186 * gst/rtsp-server/rtsp-server.c:
4187 * gst/rtsp-server/rtsp-thread-pool.c:
4188 * gst/rtsp-server/rtsp-thread-pool.h:
4189 * tests/check/gst/client.c:
4190 ClientState -> Context
4191 Rename the clientstate to context and put the code in a separate file.
4193 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4195 * examples/test-auth.c:
4196 * gst/rtsp-server/rtsp-auth.c:
4197 * gst/rtsp-server/rtsp-auth.h:
4198 auth: add support for default token
4199 The default token is used when the user is not authenticated and can be used to
4200 give minimal permissions.
4202 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4204 * examples/test-auth.c:
4205 * gst/rtsp-server/rtsp-auth.c:
4206 auth: use defines when possible
4208 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4210 * gst/rtsp-server/rtsp-address-pool.c:
4211 address-pool: improve docs
4213 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4215 * gst/rtsp-server/rtsp-permissions.c:
4216 permissions: add the role to the copy
4218 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
4220 * gst/rtsp-server/rtsp-permissions.c:
4221 permissions: Also copy the roles
4223 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
4225 * gst/rtsp-server/rtsp-permissions.c:
4226 permissions: Make it build
4228 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4230 * gst/rtsp-server/rtsp-address-pool.h:
4233 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4235 * docs/libs/gst-rtsp-server-sections.txt:
4236 * gst/rtsp-server/rtsp-auth.c:
4237 * gst/rtsp-server/rtsp-auth.h:
4238 * gst/rtsp-server/rtsp-media.c:
4239 * gst/rtsp-server/rtsp-session-media.c:
4240 * gst/rtsp-server/rtsp-stream-transport.c:
4241 * gst/rtsp-server/rtsp-stream-transport.h:
4242 * gst/rtsp-server/rtsp-stream.c:
4243 * tests/check/gst/client.c:
4246 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4248 * docs/libs/gst-rtsp-server-sections.txt:
4249 * gst/rtsp-server/rtsp-address-pool.c:
4250 * gst/rtsp-server/rtsp-address-pool.h:
4251 * tests/check/gst/addresspool.c:
4252 * tests/check/gst/rtspserver.c:
4253 address-pool: cleanups
4254 Remove redundant method, improve docs.
4256 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4258 * docs/libs/gst-rtsp-server-sections.txt:
4259 * gst/rtsp-server/rtsp-auth.h:
4260 * gst/rtsp-server/rtsp-permissions.c:
4261 * gst/rtsp-server/rtsp-permissions.h:
4262 * gst/rtsp-server/rtsp-token.c:
4265 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4267 * gst/rtsp-server/rtsp-permissions.c:
4268 permissions: implement _remove_role
4270 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4272 * gst/rtsp-server/rtsp-permissions.c:
4273 permissions: update docs
4275 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4277 * tests/check/gst/client.c:
4278 tests: simplify tests
4279 Client settings are now disabled by default so we don't need an auth
4280 module to disable them.
4282 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4284 * gst/rtsp-server/rtsp-auth.c:
4285 auth: add default authorizations
4286 When no auth module is specified, use our table of defaults to look up the
4287 default value of the check instead of always allowing everything. This was
4288 we can disallow client settings by default.
4290 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4293 README: update readme
4295 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4297 * gst/rtsp-server/rtsp-thread-pool.c:
4298 * gst/rtsp-server/rtsp-thread-pool.h:
4299 thread-pool: add more docs
4301 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4303 * gst/rtsp-server/rtsp-thread-pool.c:
4304 * gst/rtsp-server/rtsp-thread-pool.h:
4305 thread-pool: fix race in thread reuse
4306 If we try to reuse a thread right after we made it stop, we end up using a
4307 stopped thread. Catch this case and only reuse threads that are not stopping.
4309 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4311 * gst/rtsp-server/rtsp-server.c:
4312 server: add small debug
4314 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4316 * tests/check/gst/client.c:
4318 Add some permissions to media so we can use the auth and enable
4321 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4323 * gst/rtsp-server/rtsp-client.c:
4324 client: support pushed context in handle_request
4325 If we already have a pushed state, reuse it and add our own things. This makes
4326 it easier to write tests.
4328 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4330 * gst/rtsp-server/rtsp-auth.c:
4331 auth: don't auth on methods
4332 Don't authorize on methods anymore but on the resources that we
4333 try to access, this is more flexible.
4334 Move the authorization checks to where they are needed and let the
4335 check return the response on error.
4337 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4339 * gst/rtsp-server/rtsp-mount-points.c:
4340 mount-points: add some debug
4342 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4344 * tests/check/gst/client.c:
4345 tests: almost fix test
4347 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4349 * gst/rtsp-server/rtsp-auth.c:
4350 * gst/rtsp-server/rtsp-auth.h:
4351 * gst/rtsp-server/rtsp-client.c:
4352 * gst/rtsp-server/rtsp-client.h:
4353 * gst/rtsp-server/rtsp-server.c:
4354 * gst/rtsp-server/rtsp-server.h:
4355 auth: let the auth module check client_settings
4356 Let the auth module decide if client settings are allowed for the
4359 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4361 * gst/rtsp-server/rtsp-token.c:
4362 * gst/rtsp-server/rtsp-token.h:
4363 token: add method to check boolean permission
4365 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4367 * examples/test-auth.c:
4368 * examples/test-cgroups.c:
4369 * gst/rtsp-server/rtsp-token.c:
4370 * gst/rtsp-server/rtsp-token.h:
4371 token: simplify token constructor
4372 Use variable arguments to make easier API.
4374 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4376 * examples/test-auth.c:
4377 * examples/test-cgroups.c:
4378 * gst/rtsp-server/rtsp-media-factory.c:
4379 * gst/rtsp-server/rtsp-media-factory.h:
4380 media-factory: add convenience API for factory
4382 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4384 * examples/test-auth.c:
4385 * examples/test-cgroups.c:
4386 * gst/rtsp-server/rtsp-permissions.c:
4387 * gst/rtsp-server/rtsp-permissions.h:
4388 permissions: simplify API a little
4389 Avoid passing GstStructure in the add_role method, use varargs instead
4390 to construct the structure behind the scenes. We can then also use the
4391 structure name as the role and simplify some more logic.
4393 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4395 * gst/rtsp-server/rtsp-auth.c:
4398 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4400 * gst/rtsp-server/rtsp-auth.c:
4401 * gst/rtsp-server/rtsp-auth.h:
4402 * gst/rtsp-server/rtsp-client.c:
4403 auth: handle unauthorized response
4404 Move handling of the unauthorized response to the auth module, it can add
4405 the appropriate headers to request authorization for the required method
4406 much better than the client.
4408 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4410 * gst/rtsp-server/rtsp-client.c:
4411 * gst/rtsp-server/rtsp-client.h:
4412 client: allow for sending any message, not only requests
4413 Change the _send_request() method to _send_message() so that we
4414 can both send requests and replies.
4416 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4418 * docs/libs/gst-rtsp-server-sections.txt:
4419 * gst/rtsp-server/rtsp-server.h:
4422 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4424 * examples/test-video.c:
4425 * gst/rtsp-server/rtsp-auth.c:
4426 * gst/rtsp-server/rtsp-auth.h:
4427 * gst/rtsp-server/rtsp-server.c:
4428 * gst/rtsp-server/rtsp-server.h:
4429 auth: move TLS handling to auth module
4430 Remove the TLS settings on the server and move it to the auth module because
4431 that is where security related bits go.
4433 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4435 * gst/rtsp-server/rtsp-client.c:
4436 * gst/rtsp-server/rtsp-client.h:
4437 client: add state push/pop
4439 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4441 * gst/rtsp-server/rtsp-client.c:
4442 * gst/rtsp-server/rtsp-client.h:
4443 client: add connection to state
4445 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4447 * gst/rtsp-server/rtsp-mount-points.c:
4448 mount-points: fix debug
4450 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4452 * tests/check/gst/media.c:
4453 tests: fix media test
4455 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4457 * gst/rtsp-server/rtsp-thread-pool.c:
4458 thread-pool: we don't require a state
4460 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4462 * gst/rtsp-server/rtsp-server.c:
4463 server: let context ref the server
4464 So that we don't risk losing the server object early anc crash.
4466 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4468 * tests/check/gst/client.c:
4469 tests: fix client test
4471 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4474 * docs/libs/gst-rtsp-server-docs.sgml:
4475 * docs/libs/gst-rtsp-server-sections.txt:
4476 * gst/rtsp-server/rtsp-address-pool.c:
4477 * gst/rtsp-server/rtsp-auth.c:
4478 * gst/rtsp-server/rtsp-client.c:
4479 * gst/rtsp-server/rtsp-client.h:
4480 * gst/rtsp-server/rtsp-media-factory-uri.c:
4481 * gst/rtsp-server/rtsp-media-factory.c:
4482 * gst/rtsp-server/rtsp-media-factory.h:
4483 * gst/rtsp-server/rtsp-media.c:
4484 * gst/rtsp-server/rtsp-mount-points.c:
4485 * gst/rtsp-server/rtsp-params.c:
4486 * gst/rtsp-server/rtsp-permissions.c:
4487 * gst/rtsp-server/rtsp-sdp.c:
4488 * gst/rtsp-server/rtsp-server.c:
4489 * gst/rtsp-server/rtsp-server.h:
4490 * gst/rtsp-server/rtsp-session-media.c:
4491 * gst/rtsp-server/rtsp-session-pool.c:
4492 * gst/rtsp-server/rtsp-session.c:
4493 * gst/rtsp-server/rtsp-stream-transport.c:
4494 * gst/rtsp-server/rtsp-stream.c:
4495 * gst/rtsp-server/rtsp-thread-pool.c:
4496 * gst/rtsp-server/rtsp-token.c:
4499 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4501 * gst/rtsp-server/rtsp-session-pool.c:
4502 * gst/rtsp-server/rtsp-session-pool.h:
4503 session-pool: make vmethod to create a session
4504 Make a vmethod to create a sessions so that subclasses can create
4505 custom session objects
4507 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4509 * gst/rtsp-server/rtsp-auth.c:
4510 * gst/rtsp-server/rtsp-media-factory.h:
4511 * gst/rtsp-server/rtsp-media.h:
4512 * gst/rtsp-server/rtsp-mount-points.h:
4513 * gst/rtsp-server/rtsp-session-pool.h:
4514 * gst/rtsp-server/rtsp-stream.h:
4517 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4519 * docs/libs/gst-rtsp-server-docs.sgml:
4520 * docs/libs/gst-rtsp-server-sections.txt:
4521 * gst/rtsp-server/rtsp-address-pool.c:
4522 * gst/rtsp-server/rtsp-address-pool.h:
4523 * gst/rtsp-server/rtsp-auth.c:
4524 * gst/rtsp-server/rtsp-client.h:
4525 * gst/rtsp-server/rtsp-media-factory.h:
4526 * gst/rtsp-server/rtsp-media.c:
4527 * gst/rtsp-server/rtsp-media.h:
4528 * gst/rtsp-server/rtsp-permissions.c:
4529 * gst/rtsp-server/rtsp-permissions.h:
4530 * gst/rtsp-server/rtsp-server.h:
4531 * gst/rtsp-server/rtsp-session-media.c:
4532 * gst/rtsp-server/rtsp-session-media.h:
4533 * gst/rtsp-server/rtsp-session-pool.h:
4534 * gst/rtsp-server/rtsp-session.h:
4535 * gst/rtsp-server/rtsp-stream-transport.h:
4536 * gst/rtsp-server/rtsp-stream.c:
4537 * gst/rtsp-server/rtsp-thread-pool.h:
4540 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4543 * examples/Makefile.am:
4544 configure: compile cgroup example conditionally
4545 Only compile the cgroup example when we have libcgroup
4547 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4550 * examples/Makefile.am:
4551 * examples/test-cgroups.c:
4552 examples: add cgroups example
4554 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4556 * tests/check/gst/rtspserver.c:
4557 tests: fix compilation
4559 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4561 * gst/rtsp-server/rtsp-thread-pool.c:
4562 thread-pool: fix vmethod invocation
4564 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4566 * gst/rtsp-server/rtsp-thread-pool.c:
4567 * gst/rtsp-server/rtsp-thread-pool.h:
4568 thread-pool: store thread type in thread
4570 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4572 * gst/rtsp-server/rtsp-client.c:
4573 client: pass thread from pool to media _prepare
4574 Get a thread from the configured threadpool and pass it to the prepare method of
4577 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4579 * gst/rtsp-server/rtsp-media.c:
4580 * gst/rtsp-server/rtsp-media.h:
4581 media: Accept a thread in _prepare
4582 Remove out own threadpool handling and use the provided thread and
4583 maincontext for the bus messages and the state changes.
4585 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4587 * gst/rtsp-server/rtsp-server.c:
4588 server: configure client thread pool
4590 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4592 * gst/rtsp-server/rtsp-client.c:
4593 * gst/rtsp-server/rtsp-client.h:
4594 client: add method to configure thread pool
4596 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4598 * gst/rtsp-server/rtsp-client.h:
4599 * gst/rtsp-server/rtsp-server.c:
4600 * gst/rtsp-server/rtsp-server.h:
4601 server: use thread pool
4602 Use the thread pool instead of doing our own thing.
4604 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4606 * gst/rtsp-server/Makefile.am:
4607 * gst/rtsp-server/rtsp-thread-pool.c:
4608 * gst/rtsp-server/rtsp-thread-pool.h:
4609 thread-pool: add object to manage threads
4610 Add an object to manage the client and media threads.
4612 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4614 * gst/rtsp-server/rtsp-auth.c:
4615 auth: debug authorization check
4617 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4619 * gst/rtsp-server/rtsp-media.c:
4620 media: start media pipeline in context
4621 Start the media pipeline in the provided context (or our default one
4622 when NULL). This makes sure that we run the bus thread in this context and that
4623 all media threads are children of this context.
4625 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4627 * gst/rtsp-server/rtsp-media-factory.c:
4628 factory: pass permissions to media by default
4630 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4632 * examples/test-auth.c:
4633 test: add permissions to auth test
4634 Ass some permissions to the media factory in the test.
4636 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4638 * gst/rtsp-server/rtsp-auth.c:
4639 * gst/rtsp-server/rtsp-auth.h:
4640 * gst/rtsp-server/rtsp-client.c:
4641 auth: simplify auth checks
4642 Remove client from methods, it's now in the state
4643 Perform the check specified by the string, use the information from the
4644 thread local context.
4646 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4648 * gst/rtsp-server/rtsp-client.c:
4649 * gst/rtsp-server/rtsp-client.h:
4650 client: add state to current thread
4651 Add the client to the ClientState object.
4652 Place the ClientState on the current thread.
4654 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4656 * gst/rtsp-server/rtsp-media-factory.c:
4657 * gst/rtsp-server/rtsp-media-factory.h:
4658 * gst/rtsp-server/rtsp-media.c:
4659 * gst/rtsp-server/rtsp-media.h:
4660 media: make it possible to set permissions
4661 Make it possible to set permissions on media and media factory objects
4663 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4665 * gst/rtsp-server/Makefile.am:
4666 * gst/rtsp-server/rtsp-permissions.c:
4667 * gst/rtsp-server/rtsp-permissions.h:
4668 permissions: add permissions object
4669 Add a mini object to store permissions based on a role.
4671 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4673 * examples/test-auth.c:
4674 * gst/rtsp-server/rtsp-auth.c:
4675 * gst/rtsp-server/rtsp-auth.h:
4676 * gst/rtsp-server/rtsp-client.c:
4677 auth: add auth checks
4678 Add an enum with auth checks and implement the checks in the auth object.
4679 Perform the checks from the client.
4681 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4683 * examples/test-auth.c:
4684 * gst/rtsp-server/rtsp-auth.c:
4685 * gst/rtsp-server/rtsp-auth.h:
4686 * gst/rtsp-server/rtsp-client.h:
4687 auth: use the token after authentication
4688 After we authenticated a user, keep the Token around in the state.
4690 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4692 * gst/rtsp-server/rtsp-client.c:
4693 * gst/rtsp-server/rtsp-media.c:
4694 * gst/rtsp-server/rtsp-media.h:
4695 * tests/check/gst/media.c:
4696 media: add optional context for bus messages
4697 Add an optional mainloop to _prepare that will handle the bus messages instead
4698 of always using the shared mainloop.
4700 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4702 * gst/rtsp-server/Makefile.am:
4703 * gst/rtsp-server/rtsp-token.c:
4704 * gst/rtsp-server/rtsp-token.h:
4705 token: add authorization token
4706 Add a simply miniobject that contains the authorizations. The object contains a
4707 GstStructure that hold all authorization fields. When a user is authenticated,
4708 the auth module will create a Token for the user. The token is then used to
4709 check what operations the user is allowed to do and various other configuration
4712 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4714 * examples/test-auth.c:
4715 * gst/rtsp-server/rtsp-auth.c:
4716 * gst/rtsp-server/rtsp-auth.h:
4717 * gst/rtsp-server/rtsp-client.c:
4718 * gst/rtsp-server/rtsp-client.h:
4719 * gst/rtsp-server/rtsp-media-factory.c:
4720 * gst/rtsp-server/rtsp-media-factory.h:
4721 * gst/rtsp-server/rtsp-media.c:
4722 * gst/rtsp-server/rtsp-media.h:
4723 auth: remove auth from media and factory
4724 Remove the auth object from media and factory. We want to have the RTSPClient
4725 authenticate and authorize resources, there is no need to place another auth
4726 manager on the media/factory.
4728 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4730 * examples/test-auth.c:
4731 * gst/rtsp-server/rtsp-auth.c:
4732 * gst/rtsp-server/rtsp-auth.h:
4733 * gst/rtsp-server/rtsp-client.h:
4734 auth: add support for multiple basic auth tokens
4735 Make it possible to add multiple basic authorisation tokens to one authorization
4736 object. Associate with each token an authorization group that will define what
4737 capabilities are allowed.
4739 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4741 * gst/rtsp-server/rtsp-client.c:
4742 client: error out on non-aggregate control
4743 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
4745 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4747 * gst/rtsp-server/rtsp-client.c:
4748 client: rework setup request a little
4749 Cache the media in DESCRIBE based on the longest matching path with the uri
4750 that we can find in the mount points.
4751 Rework the setup request a little to get the media from the session or from
4752 the longest matching path, this way we can derive the control string as
4753 everything after the path instead of hardcoding it.
4754 Find the stream based on the control string and only open a session when all
4757 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4759 * gst/rtsp-server/rtsp-media.c:
4760 * gst/rtsp-server/rtsp-media.h:
4761 media: add method to find a stream by control url
4763 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4765 * gst/rtsp-server/rtsp-stream.c:
4766 * gst/rtsp-server/rtsp-stream.h:
4767 stream: add method to check control url of stream
4769 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4771 * gst/rtsp-server/rtsp-client.c:
4772 * gst/rtsp-server/rtsp-session-media.c:
4773 * gst/rtsp-server/rtsp-session-media.h:
4774 * gst/rtsp-server/rtsp-session.c:
4775 * gst/rtsp-server/rtsp-session.h:
4776 session: use path matching for session media
4777 Use a path string instead of a uri to lookup session media in the sessions. Also
4778 use path matching to find the largest possible path that matches.
4780 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4782 * gst/rtsp-server/rtsp-client.c:
4783 * gst/rtsp-server/rtsp-mount-points.c:
4784 * gst/rtsp-server/rtsp-mount-points.h:
4785 * tests/check/gst/mountpoints.c:
4786 mount-points: remove useless vmethod
4787 Making lookups in the mount points should not be done with a URL, if there is a
4788 mapping to be done from URL to mount points, we'll need to do it somewhere
4791 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4793 * gst/rtsp-server/rtsp-mount-points.c:
4794 * gst/rtsp-server/rtsp-mount-points.h:
4795 * tests/check/gst/mountpoints.c:
4796 mount-points: improve mount point searching
4797 Use a GSequence to keep track of the mount points.
4798 Match a URL to the longest matching registered mount point. This should be the
4799 URL to perform aggreagate control and the remainder is the stream specific
4801 Add some unit tests for this.
4803 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
4805 * gst/rtsp-server/Makefile.am:
4806 rtsp-server: Allow building of static library
4808 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4810 * tests/check/gst/mediafactory.c:
4811 tests: fix compilation
4813 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4815 * gst/rtsp-server/rtsp-sdp.c:
4816 sdp: get control string from stream
4817 Use the control string as configured in the stream.
4819 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4821 * gst/rtsp-server/rtsp-stream.c:
4822 * gst/rtsp-server/rtsp-stream.h:
4823 stream: add methods and property to set control string
4825 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4827 * gst/rtsp-server/rtsp-client.c:
4829 Rename variables for clarity
4830 Keep media in state when we can
4832 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4834 * gst/rtsp-server/rtsp-client.c:
4835 * gst/rtsp-server/rtsp-stream.c:
4836 * gst/rtsp-server/rtsp-stream.h:
4837 stream: add more support for IPv6
4838 Rename _get_address to _get_multicast_address in GstRTSPStream to
4839 make it clear that this function only deals with multicast.
4840 Make it possible to have both an IPv4 and IPv6 multicast address on
4841 a stream. Give the client an IPv4 or IPv6 address depending on the
4842 address it used to connect to the server.
4843 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
4845 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4847 * gst/rtsp-server/rtsp-client.c:
4850 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4852 * gst/rtsp-server/rtsp-stream.c:
4853 stream: handle failed port allocation
4854 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
4855 can't allocate any family at all. Also keep track of what port families we
4857 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
4859 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4861 * gst/rtsp-server/rtsp-stream.c:
4862 stream: improve docs
4864 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4866 * gst/rtsp-server/rtsp-stream-transport.c:
4867 stream-transport: remove old if 0 block
4869 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
4871 * tests/check/gst/client.c:
4873 gst_rtsp_client_get_uri() has been removed
4874 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
4876 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4878 * gst/rtsp-server/rtsp-client.c:
4879 * gst/rtsp-server/rtsp-client.h:
4880 client: add method to filter managed sessions
4881 Add a method to filter the sessions managed by this client connection.
4882 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
4884 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4886 * gst/rtsp-server/rtsp-client.c:
4887 * gst/rtsp-server/rtsp-client.h:
4888 client: remove _get_uri() method
4889 Remove the get_uri() method on the client. A client has no uri, the uri
4890 property is an internal property to manage the last cached media for
4893 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4895 * gst/rtsp-server/rtsp-media-factory.h:
4896 media-factory: fix typo
4898 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
4900 * gst/rtsp-server/rtsp-media.c:
4901 rtsp-media: Do not leak the query in default_query_stop
4902 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
4904 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4906 * gst/rtsp-server/rtsp-media.c:
4907 media: don't unlock when conversion fails
4908 Don't unlock the state lock when conversion fails because it was not locked.
4910 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4912 * gst/rtsp-server/rtsp-media.c:
4913 * gst/rtsp-server/rtsp-media.h:
4914 Add query_position and query_stop vmethods to rtsp-media
4916 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4918 * gst/rtsp-server/rtsp-media.c:
4919 Fix typo in property install for rtsp-media's time-provider
4921 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4923 * gst/rtsp-server/rtsp-client.c:
4924 * gst/rtsp-server/rtsp-client.h:
4925 client: clean some variables
4926 Clean some variables and add some guards to _send_request()
4928 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4930 * gst/rtsp-server/rtsp-client.c:
4931 * gst/rtsp-server/rtsp-client.h:
4932 Add gst_rtsp_client_send_request API
4933 This makes it possible to send arbitrary messages to a client, such as
4934 SET_PARAMETER or GET_PARAMETER
4936 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4938 * gst/rtsp-server/rtsp-media.c:
4939 * gst/rtsp-server/rtsp-media.h:
4940 media: add _get_element() method
4941 Add method to get the element used when creating the media.
4942 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
4944 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4946 * gst/rtsp-server/rtsp-media.c:
4949 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4951 * gst/rtsp-server/rtsp-stream.c:
4952 * gst/rtsp-server/rtsp-stream.h:
4953 stream: allow access to the rtp session
4954 https://bugzilla.gnome.org/show_bug.cgi?id=703004
4956 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
4958 * gst/rtsp-server/rtsp-stream.c:
4959 * gst/rtsp-server/rtsp-stream.h:
4960 dscp qos support in gst-rtsp-stream
4961 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
4963 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4965 * tests/check/gst/rtspserver.c:
4967 Actually do what the comment says. Also keep the old code around, not sure what
4968 should happen when you get a 454 from a TEARDOWN, does it close the connection?
4969 it currently doesn't.
4971 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4973 * gst/rtsp-server/rtsp-client.c:
4974 client: also watch newly created session
4975 When we newly created a session, start watching it immediately instead of
4976 on the next request.
4978 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
4980 * tests/check/gst/client.c:
4981 tests: add unit test for new-session
4982 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
4984 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4986 * gst/rtsp-server/rtsp-client.c:
4987 client: emit new-session when new session is created
4988 Only emit new-session when we created a new session for a client, not when a
4989 client picked up a previous session.
4990 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
4992 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
4994 * gst/rtsp-server/rtsp-client.c:
4995 client: handle asterisk as path in requests
4996 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
4998 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5000 * gst/rtsp-server/rtsp-media.c:
5001 media: handle segment query format mismatch
5002 It's possible that the segment query returns with a different format than what
5003 we asked for, handle this case also.
5005 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
5007 * gst/rtsp-server/rtsp-media.c:
5008 media: use segment stop in collect_media_stats
5009 Use segment stop instead of duration as range end point.
5010 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
5012 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
5014 * gst/rtsp-server/rtsp-media.c:
5015 * tests/check/gst/media.c:
5016 rtsp-media: Do not leak the element in take_pipeline
5017 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
5019 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
5021 * gst/rtsp-server/rtsp-client.c:
5022 * gst/rtsp-server/rtsp-client.h:
5023 rtsp-client: Make configure_client_transport virtual
5024 This patch makes configure_client_transport virtual. The functionality is
5025 needed to handle some weird clients sending multicast transport settings as url
5027 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
5029 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
5031 * gst/rtsp-server/rtsp-client.c:
5032 * gst/rtsp-server/rtsp-client.h:
5033 rtsp-client: Make param_set and param_get virtual
5034 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
5036 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
5038 * gst/rtsp-server/rtsp-client.c:
5039 * gst/rtsp-server/rtsp-media.c:
5040 * gst/rtsp-server/rtsp-media.h:
5041 media: convert_range replaces get_range_times
5042 get_range_times worked for handling UTC ranges for seeks, but we also
5043 need to convert back from NPT to the requested unit in
5044 get_range_string. convert_range is now used for both.
5045 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
5047 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5049 * gst/rtsp-server/rtsp-client.c:
5050 * gst/rtsp-server/rtsp-sdp.c:
5051 * gst/rtsp-server/rtsp-sdp.h:
5052 sdp: cleanup sdp info
5053 We don't need to pass the proto, we can more easily check a boolean.
5054 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
5056 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
5058 * gst/rtsp-server/rtsp-sdp.c:
5059 use 0.0.0.0 or :: for c= line instead of server address
5061 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
5063 * gst/rtsp-server/rtsp-client.c:
5064 use local address, not remote, in SDP
5065 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
5067 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5070 Automatic update of common submodule
5071 From 098c0d7 to 01a7a46
5073 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
5075 * gst/rtsp-server/rtsp-media.c:
5076 * gst/rtsp-server/rtsp-media.h:
5077 media: possibility to override range time conversion
5078 Make it possible to override the conversion from GstRTSPTimeRange to
5079 GstClockTimes, that is done before seeking on the media
5080 pipeline. Overriding can be useful for UTC ranges, where the default
5081 conversion gives nanoseconds since 1900.
5082 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
5084 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
5086 * gst/rtsp-server/rtsp-server.c:
5087 * gst/rtsp-server/rtsp-server.h:
5088 rtsp-server: Expose the use_client_settings API
5089 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
5091 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
5093 * gst/rtsp-server/rtsp-client.c:
5094 * gst/rtsp-server/rtsp-stream.c:
5095 * gst/rtsp-server/rtsp-stream.h:
5096 rtspstream: handle both ipv4 and ipv6 clients
5097 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
5099 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5101 * gst/rtsp-server/rtsp-sdp.c:
5102 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
5103 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
5104 We already have a way to place extra attributes in the SDP by using a string
5105 property with prefix x- or a- in the caps.
5107 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5109 * gst/rtsp-server/rtsp-sdp.c:
5110 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
5111 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
5112 We already have a way to place extra attributes in the SDP, just make a string
5113 property in the payloader with a- or x- prefix.
5115 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5117 * gst/rtsp-server/rtsp-sdp.c:
5118 rtsp: place a- and x- properties as attributes
5119 application/x-rtp has properties with a- and x- prefixes that should be
5120 placed as attributes in the SDP for the media instead of being added to the
5123 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5125 * examples/Makefile.am:
5126 * examples/test-video.c:
5127 example: add TLS example
5129 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5131 * gst/rtsp-server/rtsp-server.c:
5132 * gst/rtsp-server/rtsp-server.h:
5133 server: add support for TLS
5134 Add methods to set and get a TLS certificate.
5135 Add vmethod to configure a new connection. By default, configure the TLS
5136 certificate in a new connection if needed.
5138 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5140 * gst/rtsp-server/rtsp-server.c:
5141 * gst/rtsp-server/rtsp-server.h:
5142 server: remove accept_client vmethod
5143 This vmethod is not very useful so remove it.
5145 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5147 * gst/rtsp-server/rtsp-server.c:
5148 server: don't crash on NULL GError
5150 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
5152 * gst/rtsp-server/rtsp-session-pool.c:
5153 rtsp-session-pool: corrected session timeout detection
5154 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
5156 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5158 * gst/rtsp-server/rtsp-client.c:
5159 client: improve debug
5161 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5163 * gst/rtsp-server/rtsp-client.c:
5164 * gst/rtsp-server/rtsp-client.h:
5165 * gst/rtsp-server/rtsp-server.c:
5166 server: refactor connection setup
5167 Let the server accept the socket connection and construct a GstRTSPConnection
5168 from it. Remove the code from the client and let the client only deal with
5169 a fully configure GstRTSPConnection object.
5170 We will need this later when the server will configure the connection for
5173 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5175 * gst/rtsp-server/rtsp-stream.c:
5176 stream: keep the transport object alive
5177 Keep the transport object alive while we have it as qdata on the
5180 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
5182 * gst/rtsp-server/rtsp-client.c:
5183 * gst/rtsp-server/rtsp-server.c:
5184 rtsp-server: Do not crash on nmapping of server
5185 * generate error when gst_rtsp_connection_accept fails
5186 * do not stop accepting incoming connections because
5187 accepting a client fails
5188 https://bugzilla.gnome.org/show_bug.cgi?id=701072
5190 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
5192 * gst/rtsp-server/rtsp-client.c:
5193 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
5194 https://bugzilla.gnome.org/show_bug.cgi?id=700953
5196 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
5198 * gst/rtsp-server/rtsp-sdp.c:
5199 rtsp-sdp: Parse framerate caps field and set SDP attribute
5200 The SDP attribute and its format is described in RFC4566.
5201 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
5203 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
5205 * gst/rtsp-server/rtsp-sdp.c:
5206 rtsp-sdp: Parse width/height from caps and set SDP attribute
5207 The SDP attribute and its format is described in RFC6064.
5208 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
5210 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
5212 * gst/rtsp-server/rtsp-sdp.c:
5213 * tests/check/gst/client.c:
5214 rtsp-sdp: add bandwidth line
5215 https://bugzilla.gnome.org/show_bug.cgi?id=699220
5217 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5220 Automatic update of common submodule
5221 From 5edcd85 to 098c0d7
5223 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
5225 * tests/check/gst/media.c:
5226 tests: add dynamic payloader prepare/unprepare check
5228 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5230 * gst/rtsp-server/rtsp-media.c:
5231 media: release lock when removing fakesink
5233 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5235 * gst/rtsp-server/rtsp-stream.c:
5236 stream: set elements to NULL before removing
5237 When removing a stream, set the elements to NULL first. This avoids
5238 element-is-not-in-NULL-state errors when we dispose the elements.
5240 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
5243 Automatic update of common submodule
5244 From 3cb3d3c to 5edcd85
5246 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5248 * gst/rtsp-server/rtsp-media.c:
5249 * gst/rtsp-server/rtsp-media.h:
5250 media: listen to pad-removed signals
5251 Listen to the pad-removed signal and remove the stream associated with the
5253 Add signal to be notified of the removed pad.
5254 Remove the fakesink in unprepare()
5255 Fix signatures of the signal methods
5257 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5259 * examples/test-sdp.c:
5260 tests: add example of reusable pipelines
5262 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
5264 * gst/rtsp-server/rtsp-stream.c:
5265 * gst/rtsp-server/rtsp-stream.h:
5266 stream: add method to get the srcpad
5268 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
5270 * tests/check/gst/media.c:
5271 check: add media prepare/unprepare test
5272 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
5274 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
5276 * gst/rtsp-server/rtsp-media.c:
5277 media: disconnect from signal handlers in unprepare()
5278 We connected to the pad-added and no-more-pads signals in prepare() so
5279 we need to disconnect from them in unprepare().
5280 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
5282 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
5284 * gst/rtsp-server/rtsp-media.c:
5285 media: don't free streams array
5286 Don't free the streams array in the unprepare() method, they were not
5288 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
5290 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
5292 * gst/rtsp-server/rtsp-media.c:
5293 media: don't unref the pipeline in unprepare
5294 Unprepare() should undo what prepare() does. Because the pipeline is
5295 not created in prepare(), we should not unref it in unprepare()
5297 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
5299 * gst/rtsp-server/rtsp-stream.c:
5300 stream: clear session and caps for reuse
5301 Set the session and caps to NULL after unref otherwise we might unref
5303 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
5305 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
5307 * gst/rtsp-server/rtsp-client.c:
5308 client: send out teardown signal before tearing down
5309 The advantage is that in the signal handler you get direct access to
5310 information about what streams are about to get torn down (in the
5311 GstRTSPClientState).
5312 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
5314 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
5316 * gst/rtsp-server/rtsp-client.c:
5317 * gst/rtsp-server/rtsp-client.h:
5318 client: expose connection
5319 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
5321 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
5324 Automatic update of common submodule
5325 From aed87ae to 3cb3d3c
5327 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5329 * gst/rtsp-server/rtsp-media.c:
5330 * gst/rtsp-server/rtsp-media.h:
5331 * gst/rtsp-server/rtsp-session-media.c:
5332 * gst/rtsp-server/rtsp-session-media.h:
5333 media: add method to get the base_time of the pipeline
5334 Together with a shared clock, this base-time could eventually be sent to
5335 the client so that it can reconstruct the exact running-time of the clock
5338 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5340 * gst/rtsp-server/Makefile.am:
5341 * gst/rtsp-server/rtsp-media.c:
5342 * gst/rtsp-server/rtsp-media.h:
5343 * gst/rtsp-server/rtsp-sdp.c:
5344 media: add GstNetTimeProvider support
5345 Add a property to let the media provide a GstNetTimeProvider for its clock.
5346 Make methods to get the clock and nettimeprovider
5347 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
5348 provider and also the current time of the clock. This should make it possible
5349 for (GStreamer) clients to slave their clock to the server clock.
5351 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
5354 Automatic update of common submodule
5355 From 04c7a1e to aed87ae
5357 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5359 * gst/rtsp-server/rtsp-media.c:
5360 media: wait for buffering to complete
5361 Wait for buffering to complete before changing the state to the target state.
5363 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5365 * gst/rtsp-server/rtsp-media.c:
5366 media: small cleanup
5368 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
5370 * tests/check/gst/rtspserver.c:
5371 tests: remove extra unref in test_setup_non_existing_stream
5372 The unref is not needed anymore, teardown runs without it.
5373 https://bugzilla.gnome.org/show_bug.cgi?id=696542
5375 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
5377 * tests/check/gst/rtspserver.c:
5378 tests: GSocketService cleanup in test_bind_already_in_use
5379 Use g_socket_service_stop so the rtspserver test stops listening for
5380 incoming connections in test_bind_already_in_use.
5381 https://bugzilla.gnome.org/show_bug.cgi?id=696541
5383 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
5385 * gst/rtsp-server/rtsp-media-factory.c:
5386 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
5387 Instead use a GWeakRef which is safe to use
5388 This is a known GLib bug, see:
5389 https://bugzilla.gnome.org/show_bug.cgi?id=667145
5391 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
5393 * gst/rtsp-server/rtsp-client.c:
5394 * gst/rtsp-server/rtsp-media.c:
5395 * gst/rtsp-server/rtsp-media.h:
5396 * gst/rtsp-server/rtsp-sdp.c:
5397 * tests/check/gst/media.c:
5398 * tests/check/gst/rtspserver.c:
5399 rtsp-media/client: Reply to PLAY request with same type of Range
5400 Remember the type of Range from the PLAY request and use the same type for
5403 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
5405 * gst/rtsp-server/rtsp-client.c:
5406 * gst/rtsp-server/rtsp-client.h:
5407 * tests/check/gst/client.c:
5408 rtsp-client: expose uri
5410 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
5412 * tests/check/gst/mediafactory.c:
5413 tests: Hold ref while creating second media
5414 To test if the media aren't shared, make sure we keep the first one while creating a second
5415 otherwise the same memory address may be reused.
5417 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
5420 configure: remove out-of-date comment
5422 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
5425 .gitignore: ignore more build files
5427 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
5429 * tests/check/Makefile.am:
5430 tests: use right _LIBS variable for gst-plugins-base libs
5432 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5434 * tests/check/Makefile.am:
5435 check: add librtp to libs
5437 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
5439 * tests/check/gst/rtspserver.c:
5440 tests: Add test to check selecting a port the server will send from
5442 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
5444 * tests/check/gst/rtspserver.c:
5445 tests: Make sure packets are actually received
5447 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
5449 * gst/rtsp-server/rtsp-stream.c:
5450 stream: Select unicast address from pool if appropriate
5452 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
5454 * gst/rtsp-server/rtsp-stream.c:
5455 stream: Properties are always there in Gst 1.0
5457 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
5459 * tests/check/gst/addresspool.c:
5460 tests: Add tests for unicast addresses in pool
5462 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
5464 * gst/rtsp-server/rtsp-address-pool.c:
5465 * tests/check/gst/addresspool.c:
5466 address-pool: Verify that multicast addresses are used for multicast and vice-versa
5468 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
5470 * docs/libs/gst-rtsp-server-sections.txt:
5471 * gst/rtsp-server/rtsp-address-pool.c:
5472 * gst/rtsp-server/rtsp-address-pool.h:
5473 * gst/rtsp-server/rtsp-stream.c:
5474 * tests/check/gst/addresspool.c:
5475 address-pool: Add unicast addresses
5477 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
5480 * gst/rtsp-server/rtsp-server.c:
5481 * tests/check/gst/rtspserver.c:
5482 rtsp-server: Limit the number of threads per server instance
5483 If we exceed the maximum, just round robin the clients over the existing
5486 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
5488 * gst/rtsp-server/rtsp-server.c:
5489 rtsp-server: No need to store the GMainContext in the client context
5491 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
5493 * tests/check/gst/rtspserver.c:
5494 tests: Add test for client disconnection
5496 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
5498 * tests/check/gst/rtspserver.c:
5499 tests: Test client and session timeouts with multiple threads
5501 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
5503 * gst/rtsp-server/rtsp-address-pool.c:
5504 * gst/rtsp-server/rtsp-auth.c:
5505 * gst/rtsp-server/rtsp-client.c:
5506 * gst/rtsp-server/rtsp-media-factory-uri.c:
5507 * gst/rtsp-server/rtsp-media-factory.c:
5508 * gst/rtsp-server/rtsp-media.c:
5509 * gst/rtsp-server/rtsp-mount-points.c:
5510 * gst/rtsp-server/rtsp-server.c:
5511 * gst/rtsp-server/rtsp-session-media.c:
5512 * gst/rtsp-server/rtsp-session-pool.c:
5513 * gst/rtsp-server/rtsp-session.c:
5514 Document locking and its order
5516 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
5518 * tests/check/gst/rtspserver.c:
5519 tests: Test that slow DESCRIBE don't block other clients
5521 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
5523 * tests/check/gst/client.c:
5524 tests: Add tests for client-requested multicast address
5526 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
5528 * docs/libs/gst-rtsp-server-sections.txt:
5529 docs: Put the various functions in the right sections
5531 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
5533 * docs/libs/gst-rtsp-server-docs.sgml:
5534 * docs/libs/gst-rtsp-server-sections.txt:
5535 * gst/rtsp-server/rtsp-address-pool.c:
5536 * gst/rtsp-server/rtsp-address-pool.h:
5537 docs: Generate docs for GstRTSPAddressPool
5539 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
5541 * gst/rtsp-server/rtsp-client.c:
5542 * gst/rtsp-server/rtsp-stream.c:
5543 * gst/rtsp-server/rtsp-stream.h:
5544 client: Check client provided addresses against the address pool
5546 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
5548 * gst/rtsp-server/rtsp-address-pool.c:
5549 * gst/rtsp-server/rtsp-address-pool.h:
5550 * tests/check/gst/addresspool.c:
5551 address-pool: Add API to request a specific address from the pool
5552 Also add relevant unit tests.
5554 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
5556 * tests/check/gst/mediafactory.c:
5557 tests: Check the passing around of a RTSPAddressPool
5558 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
5559 way down to the stream.
5561 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
5563 * tests/check/gst/addresspool.c:
5564 tests: Add more tests for the address pool
5566 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
5568 * gst/rtsp-server/rtsp-address-pool.c:
5569 address-pool: Fix off by one error
5570 When splitting a port range, the port after a skip is not part of range.
5572 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
5575 Automatic update of common submodule
5576 From 2de221c to 04c7a1e
5578 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
5581 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
5582 AM_CONFIG_HEADER was removed in automake 1.13
5583 https://bugzilla.gnome.org/show_bug.cgi?id=693368
5585 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
5588 Automatic update of common submodule
5589 From a942293 to 2de221c
5591 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5593 * gst/rtsp-server/rtsp-client.c:
5594 client: make sure the watch exists while sending data
5595 Protect the send_func with a lock. This allows us to wait for sending
5596 to complete before changing the send_func and user_data. We add an
5597 extra ref to the watch to make sure that it remains valid during
5599 When closing the connection, set the send_func to NULL
5600 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
5602 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5604 * tests/check/Makefile.am:
5605 tests: use GST_*_1_0 environment variables everywhere
5606 The _1_0 suffixed environment variables override the
5607 non-suffixed ones, so if we're in an environment that
5608 sets the _1_0 suffixed ones, such as jhbuild, we need
5609 to set those to make sure ours actually always get
5612 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5615 Automatic update of common submodule
5616 From acb04d9 to a942293
5618 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5620 * gst/rtsp-server/rtsp-client.c:
5621 rtsp-client: set the client backlog
5622 Set the client backlog to a reasonable default
5624 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
5626 * gst/rtsp-server/rtsp-media.c:
5627 rtsp-media: Make the element a constructor parameter
5628 https://bugzilla.gnome.org/show_bug.cgi?id=689594
5630 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5632 * docs/libs/Makefile.am:
5633 docs: Link with gcov library when gcov is enabled
5634 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
5636 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5638 * gst/rtsp-server/rtsp-media.c:
5639 media: match prepare with unprepare
5640 Really unprepare when there were an equal amount of prepare calls.
5642 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5644 * gst/rtsp-server/rtsp-media.c:
5645 media: media has to be unprepared in finalize
5646 Because unprepare takes away the last ref on the media.
5648 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5650 * gst/rtsp-server/rtsp-client.c:
5651 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
5652 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
5653 We can't use the refcount to trigger unprepare because it is the unprepare call
5654 that removes the last refcount after all messages are consumed. What we should
5655 probably do is make a prepared refcount and only unprepare when the refcount
5658 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5660 * gst/rtsp-server/rtsp-media.c:
5661 media: let the source unref the last media ref
5662 the last ref to the media is held by the source so we don't need to add more ref
5663 and unrefs, we simply destroy the media when the source is gone.
5665 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5667 * gst/rtsp-server/rtsp-media.c:
5668 media: improve debug
5670 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5672 * gst/rtsp-server/rtsp-media.c:
5674 Make sure we are in the right state when collecting the position and duration.
5675 Only make ourselves PREPARED when we were previously PREPARING.
5677 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5679 * gst/rtsp-server/rtsp-media.c:
5680 media: use g_object_ref/unref for GObjects
5682 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
5684 * gst/rtsp-server/rtsp-client.c:
5685 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
5686 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
5687 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
5688 isn't being used anymore.
5690 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
5692 * gst/rtsp-server/rtsp-media.c:
5693 Fix compiler warning
5695 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
5697 * gst/rtsp-server/rtsp-media-factory-uri.c:
5698 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
5700 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5702 * gst/rtsp-server/rtsp-session-media.h:
5705 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5707 * gst/rtsp-server/rtsp-media.c:
5708 * tests/check/gst/media.c:
5709 media: avoid element leak
5711 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5713 * gst/rtsp-server/rtsp-media.c:
5714 media: require an element in media constructor
5716 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5718 * gst/rtsp-server/rtsp-client.c:
5719 Revert "client: TEARDOWN brings that state to Init again"
5720 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
5721 The object is already disposed, there is no point in setting the state.
5723 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5725 * gst/rtsp-server/rtsp-client.c:
5726 client: TEARDOWN brings that state to Init again
5728 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5730 * docs/libs/gst-rtsp-server-sections.txt:
5731 * examples/test-auth.c:
5732 * gst/rtsp-server/rtsp-auth.c:
5733 * gst/rtsp-server/rtsp-auth.h:
5734 * gst/rtsp-server/rtsp-client.c:
5735 * gst/rtsp-server/rtsp-client.h:
5736 * gst/rtsp-server/rtsp-media-factory-uri.c:
5737 * gst/rtsp-server/rtsp-media-factory-uri.h:
5738 * gst/rtsp-server/rtsp-media-factory.c:
5739 * gst/rtsp-server/rtsp-media-factory.h:
5740 * gst/rtsp-server/rtsp-media.c:
5741 * gst/rtsp-server/rtsp-media.h:
5742 * gst/rtsp-server/rtsp-mount-points.c:
5743 * gst/rtsp-server/rtsp-mount-points.h:
5744 * gst/rtsp-server/rtsp-sdp.c:
5745 * gst/rtsp-server/rtsp-server.c:
5746 * gst/rtsp-server/rtsp-server.h:
5747 * gst/rtsp-server/rtsp-session-media.c:
5748 * gst/rtsp-server/rtsp-session-media.h:
5749 * gst/rtsp-server/rtsp-session-pool.c:
5750 * gst/rtsp-server/rtsp-session-pool.h:
5751 * gst/rtsp-server/rtsp-session.c:
5752 * gst/rtsp-server/rtsp-session.h:
5753 * gst/rtsp-server/rtsp-stream-transport.c:
5754 * gst/rtsp-server/rtsp-stream-transport.h:
5755 * gst/rtsp-server/rtsp-stream.c:
5756 * gst/rtsp-server/rtsp-stream.h:
5757 * tests/check/gst/media.c:
5758 rtsp: make object details private
5759 Make all object details private
5760 Add methods to access private bits
5762 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5764 * tests/check/Makefile.am:
5765 * tests/check/gst/media.c:
5766 tests: add media tests
5768 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5770 * gst/rtsp-server/rtsp-media.c:
5771 media: check if prepared for some methods
5772 Check that the media object is prepared before doing seek and getting the
5773 current position etc.
5774 Add some g_return checks.
5776 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5778 * tests/check/Makefile.am:
5779 * tests/check/gst/mediafactory.c:
5780 tests: add mediafactory test
5782 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5784 * gst/rtsp-server/rtsp-stream.c:
5785 stream: improve debug
5787 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5789 * gst/rtsp-server/rtsp-media.c:
5790 * gst/rtsp-server/rtsp-media.h:
5791 media: unref pipeline in finalize to avoid leaking it
5793 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5795 * gst/rtsp-server/rtsp-media-factory-uri.c:
5796 * gst/rtsp-server/rtsp-media.c:
5797 rtsp: use gst_object_unref on GstObjects
5799 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5801 * gst/rtsp-server/rtsp-media-factory.c:
5802 media-factory: require an url
5804 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5806 * examples/test-uri.c:
5807 examples: fix include
5809 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5811 * gst/rtsp-server/rtsp-server.h:
5812 server: remove unused include
5814 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5816 * tests/check/Makefile.am:
5817 * tests/check/gst/mountpoints.c:
5818 tests: add test for mountpoints
5820 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5822 * gst/rtsp-server/rtsp-client.c:
5823 client: fix factory leak
5824 Keep the factory in the state object only for authorization checks and make
5825 sure we unref it on failure. Also don't keep invalid objects in the state
5828 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5830 * gst/rtsp-server/rtsp-mount-points.c:
5831 mounts: add g_return_if guards
5833 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5835 * tests/check/gst/client.c:
5836 tests: add more tests
5838 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5840 * gst/rtsp-server/rtsp-client.c:
5841 client: improve debug
5843 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5845 * gst/rtsp-server/rtsp-client.c:
5846 client: improve debug and fix leaks
5847 Cleanup the uri and session when there is a bad request.
5849 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5854 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5856 * tests/check/gst/client.c:
5857 test: add test for session in options request
5859 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5861 * gst/rtsp-server/rtsp-client.c:
5862 client: use 454 when session can't be found
5863 We should use 454 when a session can't be found because there was no session
5864 pool configured in the server. This is not a server configuration problem
5865 because the server on which the request is done might not be the same one that
5866 will keep the sessions for us and so it does not need to support sessions.
5868 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5870 * gst/rtsp-server/rtsp-client.c:
5871 client: only free connection when there is one
5872 It's possible that the client doesn't have a connection when we try to free it.
5874 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5876 * tests/check/Makefile.am:
5877 * tests/check/gst/client.c:
5878 tests: add unit test for the client object
5880 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5882 * gst/rtsp-server/rtsp-client.c:
5883 client: small cleanup
5885 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5887 * gst/rtsp-server/rtsp-client.h:
5888 client: remove unused include
5890 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5892 * gst/rtsp-server/rtsp-client.c:
5893 client: fix compilation
5895 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5897 * gst/rtsp-server/rtsp-client.c:
5898 client: call destroy without the lock
5900 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5902 * gst/rtsp-server/rtsp-client.c:
5903 * gst/rtsp-server/rtsp-client.h:
5904 client: make the client usable without a socket
5905 Make a method to let the client handle a message and a callback when the client
5906 wants us to send a response message back. This makes it possible to also use the
5907 client object without the sockets, which should make it easier to test.
5909 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5911 * gst/rtsp-server/rtsp-client.c:
5912 * gst/rtsp-server/rtsp-client.h:
5913 client: small cleanup
5915 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5917 * docs/libs/gst-rtsp-server-sections.txt:
5918 * gst/rtsp-server/rtsp-client.c:
5919 * gst/rtsp-server/rtsp-client.h:
5920 * gst/rtsp-server/rtsp-server.c:
5921 client: remove reference to server
5922 We don't need to keep a ref to the server
5924 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5926 * gst/rtsp-server/rtsp-client.c:
5927 * gst/rtsp-server/rtsp-client.h:
5929 Also add some g_return_if()
5931 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5933 * gst/rtsp-server/rtsp-client.c:
5934 client: log more errors
5936 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5938 * gst/rtsp-server/rtsp-client.c:
5939 client: fix compilation
5941 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5943 * gst/rtsp-server/rtsp-client.c:
5944 * gst/rtsp-server/rtsp-client.h:
5945 client: add generic close-after-send support
5946 Add a property to send_response() to close the connection after the response has
5947 been sent to the client.
5949 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5952 * docs/libs/gst-rtsp-server-docs.sgml:
5953 * docs/libs/gst-rtsp-server-sections.txt:
5954 * docs/libs/gst-rtsp-server.types:
5955 * examples/test-auth.c:
5956 * examples/test-launch.c:
5957 * examples/test-mp4.c:
5958 * examples/test-multicast.c:
5959 * examples/test-multicast2.c:
5960 * examples/test-ogg.c:
5961 * examples/test-readme.c:
5962 * examples/test-sdp.c:
5963 * examples/test-uri.c:
5964 * examples/test-video.c:
5965 * gst/rtsp-server/Makefile.am:
5966 * gst/rtsp-server/rtsp-auth.h:
5967 * gst/rtsp-server/rtsp-client.c:
5968 * gst/rtsp-server/rtsp-client.h:
5969 * gst/rtsp-server/rtsp-media-mapping.c:
5970 * gst/rtsp-server/rtsp-media-mapping.h:
5971 * gst/rtsp-server/rtsp-mount-points.c:
5972 * gst/rtsp-server/rtsp-mount-points.h:
5973 * gst/rtsp-server/rtsp-server.c:
5974 * gst/rtsp-server/rtsp-server.h:
5975 * gst/rtsp-server/rtsp-session-media.c:
5976 * gst/rtsp-server/rtsp-session-pool.c:
5977 * gst/rtsp-server/rtsp-session-pool.h:
5978 * tests/check/gst/rtspserver.c:
5979 MediaMapping -> MountPoints
5980 Describes better what the object manages.
5982 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5985 configure: bump required version of -base
5987 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5989 * gst/rtsp-server/rtsp-media.c:
5992 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5994 * gst/rtsp-server/rtsp-media.c:
5995 * gst/rtsp-server/rtsp-media.h:
5996 media: support more Range formats
5997 Use the new -base methods to convert the Range string into a seek start and stop
6000 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6002 * examples/test-launch.c:
6003 examples: fix whitespace
6005 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6007 * examples/test-auth.c:
6008 test-auth: add example of how to remove sessions
6009 Add an example of the session filter api.
6011 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6013 * examples/test-uri.c:
6014 test-uri: remove mapping example
6016 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6018 * examples/test-uri.c:
6019 test-uri: fix callback signature
6021 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6023 * gst/rtsp-server/rtsp-media-factory.c:
6024 factory: keep ref to factory while media active
6025 While the media from a factory is alive, keep a ref to the factory.
6026 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
6028 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6030 * gst/rtsp-server/rtsp-media-factory-uri.c:
6031 factory-uri: add some debug
6033 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6035 * gst/rtsp-server/rtsp-stream.c:
6036 stream: set udp sources to PLAYING
6037 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
6038 so that it doesn't cause our pipeline to produce ASYNC-DONE.
6040 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6042 * gst/rtsp-server/rtsp-media-factory-uri.c:
6043 factory-uri: take ref to factory
6044 Take a ref to the factory that we place in our list.
6046 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6048 * tests/Makefile.am:
6049 * tests/test-reuse.c:
6050 test: add test for server reuse
6051 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
6053 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
6055 * gst/rtsp-server/rtsp-server.c:
6056 server: start and stop multiple times
6057 Stop listening on the RTSP port when the GSource is removed, so clients
6058 can't connect and the server can be started again.
6059 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
6061 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6063 * gst/rtsp-server/rtsp-server.c:
6064 server: fix small leak
6066 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6068 * gst/rtsp-server/rtsp-media.c:
6069 media: unref source in finish_unprepare
6070 The source is created in prepare, unref it in finish_unprepare.
6071 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
6073 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
6075 * gst/rtsp-server/rtsp-client.c:
6076 * gst/rtsp-server/rtsp-media.c:
6077 rtsp-media: remove bus watch before finalizing
6078 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
6079 * An extra media ref is added for the bus watch. This extra ref is unreffed by
6080 the GDestroyNotify function.
6081 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
6082 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
6083 gst_rtsp_media_unprepare before unreffing the media.
6084 This way, the bus watch will be removed before the media is finalized.
6085 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
6087 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
6089 * gst/rtsp-server/rtsp-client.c:
6090 * gst/rtsp-server/rtsp-client.h:
6091 client: wait until the TEARDOWN response is sent to close the connection
6092 Responses can be sent async so we need to wait until the TEARDOWN response has
6093 been written before we close the connection to the client. This avoids the risk
6094 of writing/polling closed sockets.
6095 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
6097 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
6099 * gst/rtsp-server/rtsp-stream.c:
6100 rtsp-stream: plug socket leak
6101 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
6103 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
6106 Automatic update of common submodule
6107 From 6bb6951 to a72faea
6109 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
6111 * gst/rtsp-server/rtsp-media-factory-uri.c:
6112 rtsp-server: don't use deprecated API
6114 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
6116 * gst/rtsp-server/rtsp-client.c:
6117 rtsp-client: fix unused-but-set-variable compiler warning
6118 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
6120 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6123 * docs/libs/gst-rtsp-server-sections.txt:
6124 * gst/rtsp-server/rtsp-client.c:
6127 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6129 * examples/Makefile.am:
6130 * examples/test-multicast2.c:
6131 examples: add another multicast example
6132 Add an example for how to configure separate multicast ranges for each media
6135 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6137 * examples/test-multicast.c:
6140 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6142 * gst/rtsp-server/rtsp-client.c:
6143 * gst/rtsp-server/rtsp-media.c:
6144 * gst/rtsp-server/rtsp-session-media.c:
6145 * gst/rtsp-server/rtsp-session-media.h:
6146 * gst/rtsp-server/rtsp-stream-transport.c:
6147 * gst/rtsp-server/rtsp-stream-transport.h:
6148 stream: use the address managed by the stream
6149 Use the address managed by the stream for multicast. This allows us to have 1
6150 multicast address for each stream.
6151 Because the address is now managed by the stream we don't have to pass it around
6153 Set the address pool on the streams.
6155 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6157 * gst/rtsp-server/rtsp-client.c:
6158 * gst/rtsp-server/rtsp-media.c:
6159 * gst/rtsp-server/rtsp-stream.c:
6162 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6164 * gst/rtsp-server/rtsp-media.c:
6165 * gst/rtsp-server/rtsp-media.h:
6166 media: add signal for new streams
6167 This allows applications to listen for new streams and configure properties on
6168 them, like the address pool.
6170 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6172 * gst/rtsp-server/rtsp-media.c:
6173 media: configure address pool in new streams
6175 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6177 * gst/rtsp-server/rtsp-stream.c:
6178 * gst/rtsp-server/rtsp-stream.h:
6179 stream: add methods to deal with address pool
6180 Add methods to get and set the address pool for the stream
6181 Add method to allocate and get the multicast addresses for this stream.
6183 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6185 * docs/libs/gst-rtsp-server-sections.txt:
6186 * gst/rtsp-server/rtsp-media.c:
6187 * gst/rtsp-server/rtsp-media.h:
6188 media: remove MTU property
6189 It is a stream property
6191 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6193 * gst/rtsp-server/rtsp-client.c:
6194 client: set blocksize only on stream
6195 Set the blocksize only on the current stream.
6197 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6199 * gst/rtsp-server/rtsp-stream.c:
6200 stream: share src and sink sockets
6201 the allocated socket is in the used-socket property, not socket.
6203 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6205 * gst/rtsp-server/rtsp-address-pool.c:
6206 * gst/rtsp-server/rtsp-address-pool.h:
6207 * gst/rtsp-server/rtsp-client.c:
6208 * gst/rtsp-server/rtsp-session-media.c:
6209 * gst/rtsp-server/rtsp-session-media.h:
6210 * gst/rtsp-server/rtsp-stream-transport.c:
6211 * gst/rtsp-server/rtsp-stream-transport.h:
6212 * tests/check/gst/addresspool.c:
6213 rtsp: make address-pool return an address object
6214 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
6215 store more info in the structure and allows us to more easily return the address
6216 to the right pool when no longer needed.
6217 Pass the address to the StreamTransport so that we can return it to the pool
6218 when the stream transport is freed or changed.
6220 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6222 * examples/Makefile.am:
6223 * examples/test-multicast.c:
6224 examples: add multicast example
6225 Show how to set up the multicast address pool so that media can be
6226 server with multicast.
6228 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6230 * gst/rtsp-server/rtsp-client.c:
6231 * gst/rtsp-server/rtsp-media-factory.c:
6232 * gst/rtsp-server/rtsp-media-factory.h:
6233 * gst/rtsp-server/rtsp-media.c:
6234 * gst/rtsp-server/rtsp-media.h:
6235 rtsp: use AddressPool
6236 Remove the multicast_group property.
6237 Use the configured addresspool to allocate multicast addresses.
6239 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6241 * gst/rtsp-server/rtsp-address-pool.c:
6242 * gst/rtsp-server/rtsp-address-pool.h:
6243 address-pool: add clear method
6245 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6247 * gst/rtsp-server/rtsp-address-pool.c:
6248 address-pool: small cleanups
6250 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6252 * tests/check/Makefile.am:
6253 * tests/check/gst/addresspool.c:
6254 tests: add addresspool unit test
6256 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6258 * gst/rtsp-server/Makefile.am:
6259 * gst/rtsp-server/rtsp-address-pool.c:
6260 * gst/rtsp-server/rtsp-address-pool.h:
6261 address-pool: add object to manage multicast addresses
6262 Make an object that can manage a rage of multicast addresses and ports.
6264 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6266 * gst/rtsp-server/rtsp-server.c:
6267 server: set default max-threads property
6269 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6271 * gst/rtsp-server/rtsp-media.c:
6272 media: wait for concurrent _prepare
6273 If a prepare is busy, wait for the result.
6275 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6277 * gst/rtsp-server/rtsp-media.c:
6278 media: add lock around message handler
6279 We don't want to dispatch messages while we are still processing the result of
6282 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6284 * gst/rtsp-server/rtsp-media.c:
6285 * gst/rtsp-server/rtsp-media.h:
6286 media: add lock to protect state changes
6288 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6290 * gst/rtsp-server/rtsp-stream.c:
6291 * gst/rtsp-server/rtsp-stream.h:
6294 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6296 * gst/rtsp-server/rtsp-stream-transport.c:
6297 * gst/rtsp-server/rtsp-stream-transport.h:
6298 * gst/rtsp-server/rtsp-stream.c:
6299 stream-transport: add keep-alive method
6301 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6303 * gst/rtsp-server/rtsp-stream-transport.c:
6304 * gst/rtsp-server/rtsp-stream-transport.h:
6305 * gst/rtsp-server/rtsp-stream.c:
6306 stream-transport: add method to handle RTP/RTCP
6307 Call new methods instead of poking into the structures directly.
6309 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6311 * gst/rtsp-server/rtsp-session-media.c:
6312 * gst/rtsp-server/rtsp-session-media.h:
6313 session-media: add locking
6315 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6317 * gst/rtsp-server/rtsp-session.c:
6318 * gst/rtsp-server/rtsp-session.h:
6319 session: add locking
6321 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6323 * gst/rtsp-server/rtsp-server.c:
6324 server: free old socket
6326 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6328 * gst/rtsp-server/rtsp-media-mapping.c:
6329 * gst/rtsp-server/rtsp-media-mapping.h:
6330 mapping: add locking
6332 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6334 * gst/rtsp-server/rtsp-media-factory.c:
6335 media-factory: add locking
6337 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6339 * gst/rtsp-server/rtsp-auth.c:
6340 * gst/rtsp-server/rtsp-auth.h:
6343 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6345 * gst/rtsp-server/rtsp-server.c:
6346 * gst/rtsp-server/rtsp-server.h:
6347 server: add max-thread property
6349 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6351 * gst/rtsp-server/rtsp-server.c:
6352 * gst/rtsp-server/rtsp-server.h:
6353 server: use a threadpool for the mainloops
6355 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6357 * gst/rtsp-server/rtsp-client.c:
6358 * gst/rtsp-server/rtsp-client.h:
6359 client: rename method
6360 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
6361 don't really create the client from the socket, we use the socket for the
6364 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6366 * gst/rtsp-server/rtsp-client.c:
6367 * gst/rtsp-server/rtsp-client.h:
6368 * gst/rtsp-server/rtsp-server.c:
6369 server: rework maincontext handling in clients
6370 Make a separate method to attach a client to a MainContext.
6371 Let the server decide in what GMainContext the client will operate and give this
6372 context to the client in attach. Then the server can later decide to use a
6373 separate thread for each client or just use the mainthread.
6375 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6377 * gst/rtsp-server/rtsp-client.c:
6378 * gst/rtsp-server/rtsp-session.c:
6379 * gst/rtsp-server/rtsp-session.h:
6380 session: move session header code in session object
6382 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
6386 * examples/test-auth.c:
6387 * examples/test-launch.c:
6388 * examples/test-mp4.c:
6389 * examples/test-ogg.c:
6390 * examples/test-readme.c:
6391 * examples/test-sdp.c:
6392 * examples/test-uri.c:
6393 * examples/test-video.c:
6394 * gst/rtsp-server/rtsp-auth.c:
6395 * gst/rtsp-server/rtsp-auth.h:
6396 * gst/rtsp-server/rtsp-client.c:
6397 * gst/rtsp-server/rtsp-client.h:
6398 * gst/rtsp-server/rtsp-media-factory-uri.c:
6399 * gst/rtsp-server/rtsp-media-factory-uri.h:
6400 * gst/rtsp-server/rtsp-media-factory.c:
6401 * gst/rtsp-server/rtsp-media-factory.h:
6402 * gst/rtsp-server/rtsp-media-mapping.c:
6403 * gst/rtsp-server/rtsp-media-mapping.h:
6404 * gst/rtsp-server/rtsp-media.c:
6405 * gst/rtsp-server/rtsp-media.h:
6406 * gst/rtsp-server/rtsp-params.c:
6407 * gst/rtsp-server/rtsp-params.h:
6408 * gst/rtsp-server/rtsp-sdp.c:
6409 * gst/rtsp-server/rtsp-sdp.h:
6410 * gst/rtsp-server/rtsp-server.c:
6411 * gst/rtsp-server/rtsp-server.h:
6412 * gst/rtsp-server/rtsp-session-media.c:
6413 * gst/rtsp-server/rtsp-session-media.h:
6414 * gst/rtsp-server/rtsp-session-pool.c:
6415 * gst/rtsp-server/rtsp-session-pool.h:
6416 * gst/rtsp-server/rtsp-session.c:
6417 * gst/rtsp-server/rtsp-session.h:
6418 * gst/rtsp-server/rtsp-stream-transport.c:
6419 * gst/rtsp-server/rtsp-stream-transport.h:
6420 * gst/rtsp-server/rtsp-stream.c:
6421 * gst/rtsp-server/rtsp-stream.h:
6422 * tests/check/gst/rtspserver.c:
6423 * tests/test-cleanup.c:
6426 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6428 * gst/rtsp-server/rtsp-media.c:
6429 * gst/rtsp-server/rtsp-session-media.c:
6430 * gst/rtsp-server/rtsp-session.c:
6431 rtsp-server: added annotations to indicate type of ownership transfer of return values
6432 https://bugzilla.gnome.org/show_bug.cgi?id=680777
6434 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
6437 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
6439 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
6442 * bindings/Makefile.am:
6443 * bindings/vala/Makefile.am:
6444 * bindings/vala/gst-rtsp-server-0.10.deps:
6445 * bindings/vala/gst-rtsp-server-0.10.vapi:
6446 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
6447 * bindings/vala/packages/gst-rtsp-server-0.10.files:
6448 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6449 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6450 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
6452 bindings: remove vala bindings
6453 They'll be reunited with the other GStreamer bindings
6454 https://bugzilla.gnome.org/show_bug.cgi?id=680777
6456 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6458 * gst/rtsp-server/rtsp-client.c:
6459 * gst/rtsp-server/rtsp-session-media.c:
6460 * gst/rtsp-server/rtsp-session-media.h:
6461 * gst/rtsp-server/rtsp-stream-transport.c:
6462 * gst/rtsp-server/rtsp-stream-transport.h:
6463 rtsp: only create transport when needed
6464 Only create the StreamTransport when configured.
6466 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6468 * gst/rtsp-server/rtsp-client.c:
6469 client: small cleanup
6471 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6473 * gst/rtsp-server/rtsp-client.c:
6474 * gst/rtsp-server/rtsp-client.h:
6475 * gst/rtsp-server/rtsp-stream-transport.c:
6476 * gst/rtsp-server/rtsp-stream-transport.h:
6477 rtsp: refactor configuration of transport
6478 Move the configuration of the transport to a place where it makes
6481 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6483 * gst/rtsp-server/rtsp-client.c:
6484 client: refactor transport parsing
6486 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6488 * gst/rtsp-server/rtsp-client.c:
6489 client: refuse to change the MTU on shared media
6490 If we change the MTU of chared media, it changes for all clients.
6491 We don't want to set the MTU to something large for clients that
6494 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6496 * examples/test-mp4.c:
6497 * gst/rtsp-server/rtsp-media.c:
6498 small fixes to docs and debug
6500 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6502 * gst/rtsp-server/rtsp-stream.c:
6503 stream: transports must already have been removed
6505 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6507 * gst/rtsp-server/rtsp-media.c:
6508 * gst/rtsp-server/rtsp-stream.c:
6509 * gst/rtsp-server/rtsp-stream.h:
6510 stream: improve join and leave of the pipeline
6512 Do the cleanup properly
6515 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6517 * gst/rtsp-server/rtsp-media.c:
6518 media: move unprepare below default implementation
6519 Makes it easier to find the default implementation
6521 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6523 * gst/rtsp-server/rtsp-media.c:
6524 media: signal unprepared when we actually finish
6526 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6528 * gst/rtsp-server/rtsp-media.c:
6529 media: no need to unlock, unprepare does that when needed
6531 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6533 * docs/libs/gst-rtsp-server-sections.txt:
6534 * gst/rtsp-server/rtsp-media-factory.h:
6535 * gst/rtsp-server/rtsp-media-mapping.c:
6536 * gst/rtsp-server/rtsp-media.h:
6537 * gst/rtsp-server/rtsp-params.c:
6538 * gst/rtsp-server/rtsp-server.c:
6539 * gst/rtsp-server/rtsp-session-pool.h:
6540 * gst/rtsp-server/rtsp-session.c:
6541 * gst/rtsp-server/rtsp-session.h:
6542 * gst/rtsp-server/rtsp-stream-transport.h:
6543 * gst/rtsp-server/rtsp-stream.h:
6546 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6548 * gst/rtsp-server/rtsp-client.c:
6549 * gst/rtsp-server/rtsp-media-mapping.h:
6550 * gst/rtsp-server/rtsp-media.c:
6551 * gst/rtsp-server/rtsp-media.h:
6552 * gst/rtsp-server/rtsp-server.h:
6553 * gst/rtsp-server/rtsp-stream.c:
6554 * gst/rtsp-server/rtsp-stream.h:
6555 rtsp: fix MTU setting
6556 Fix setting of the MTU. There is no need for a vmethod.
6558 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6563 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6566 configure: bump version number after refactoring
6568 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6570 * gst/rtsp-server/Makefile.am:
6571 * gst/rtsp-server/rtsp-client.c:
6572 * gst/rtsp-server/rtsp-client.h:
6573 * gst/rtsp-server/rtsp-media-factory-uri.c:
6574 * gst/rtsp-server/rtsp-media-factory.c:
6575 * gst/rtsp-server/rtsp-media-factory.h:
6576 * gst/rtsp-server/rtsp-media.c:
6577 * gst/rtsp-server/rtsp-media.h:
6578 * gst/rtsp-server/rtsp-sdp.c:
6579 * gst/rtsp-server/rtsp-session-media.c:
6580 * gst/rtsp-server/rtsp-session-media.h:
6581 * gst/rtsp-server/rtsp-session.c:
6582 * gst/rtsp-server/rtsp-session.h:
6583 * gst/rtsp-server/rtsp-stream-transport.c:
6584 * gst/rtsp-server/rtsp-stream-transport.h:
6585 * gst/rtsp-server/rtsp-stream.c:
6586 * gst/rtsp-server/rtsp-stream.h:
6587 rtsp: massive refactoring
6588 Make GObjects from the remaining simple structures.
6589 Remove GstRTSPSessionStream, it's not needed.
6590 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
6591 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
6592 a GstRTSPStream should be transported to a client.
6593 Rename GstRTSPMediaFactory::get_element -> create_element because that
6594 more accurately describes what it does.
6595 Make nice methods instead of poking in the structures.
6596 Move some methods inside the relevant object source code.
6597 Use GPtrArray to store objects instead of plain arrays, it is more
6598 natural and allows us to more easily clean up.
6599 Move the allocation of udp ports to the Stream object. The Stream object
6600 contains the elements needed to stream the media to a client.
6601 Improve the prepare and unprepare methods. Unprepare should now undo
6602 everything prepare did. Improve also async unprepare when doing EOS on
6603 shutdown. Make sure we always unprepare correctly.
6605 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
6607 * gst/rtsp-server/rtsp-client.c:
6608 rtsp-client: Unref server address clients connected to
6609 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
6611 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
6613 * gst/rtsp-server/rtsp-server.c:
6614 rtsp-server: don't ref server socket if it is NULL
6615 Fixes test_bind_already_in_use unit test again after commit 6a497440.
6616 https://bugzilla.gnome.org/show_bug.cgi?id=686644
6618 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
6620 * tests/check/Makefile.am:
6621 tests: Add libgio link dependency
6622 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
6624 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6626 * gst/rtsp-server/rtsp-media-mapping.c:
6627 * gst/rtsp-server/rtsp-media-mapping.h:
6628 rtsp-media-mapping: rename find_media vfunc to find_factory
6629 The virtual method and class method should have the same name
6630 so it is correctly represented in GIR file
6631 https://bugzilla.gnome.org/show_bug.cgi?id=680777
6633 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6635 * gst/rtsp-server/rtsp-auth.c:
6636 * gst/rtsp-server/rtsp-client.c:
6637 * gst/rtsp-server/rtsp-media-factory-uri.c:
6638 * gst/rtsp-server/rtsp-media-factory.c:
6639 * gst/rtsp-server/rtsp-media-mapping.c:
6640 * gst/rtsp-server/rtsp-media.c:
6641 * gst/rtsp-server/rtsp-server.c:
6642 * gst/rtsp-server/rtsp-session-pool.c:
6643 * gst/rtsp-server/rtsp-session.c:
6644 rtsp-server: fixed comments and GIR annotations
6645 https://bugzilla.gnome.org/show_bug.cgi?id=680777
6647 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
6649 * gst/rtsp-server/rtsp-media-mapping.c:
6650 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
6652 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
6654 * gst/rtsp-server/rtsp-server.c:
6655 rtsp-server: allow binding on port 0 (binds on a random port)
6657 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
6659 * gst/rtsp-server/rtsp-server.c:
6660 * gst/rtsp-server/rtsp-server.h:
6661 rtsp-server: add bound-port property
6662 bound-port can be used to retrieve the port number when the server is bound on
6663 port 0, which binds on a random port.
6665 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
6667 * gst/rtsp-server/rtsp-media-factory.c:
6668 * gst/rtsp-server/rtsp-media-factory.h:
6669 rtsp-media-factory: make ::get_element overridable by GI bindings
6670 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
6671 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
6672 as the invoker for ::get_element(), making it overridable by GI generated
6675 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
6677 * gst/rtsp-server/rtsp-media-factory-uri.c:
6678 rtsp-media-factory-uri: don't autoplug parsers in a loop
6679 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
6682 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
6684 * gst/rtsp-server/Makefile.am:
6685 Explicitly link against gio. Fix link error on mac.
6687 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
6689 * gst/rtsp-server/rtsp-session.c:
6690 session: add ttl to the transport header in SETUP
6691 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
6693 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
6695 * gst/rtsp-server/rtsp-client.c:
6696 * gst/rtsp-server/rtsp-client.h:
6697 * gst/rtsp-server/rtsp-media.c:
6698 client: Use client transport settings for multicast if allowed.
6699 This patch makes it possible for the client to send transport settings for
6700 multicast (destination && ttl). Client settings must be explicitly allowed or
6701 the server will use its own settings.
6702 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
6704 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
6707 Automatic update of common submodule
6708 From 6c0b52c to 6bb6951
6710 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
6712 * gst/rtsp-server/rtsp-client.c:
6713 rtsp-client: do not destroy the rtsp watch
6714 Don't destroy the client watch while dispatching. The rtsp watch is
6715 automatically destroyed after the rtsp watch function closed() has
6717 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
6719 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
6722 Automatic update of common submodule
6723 From 4f962f7 to 6c0b52c
6725 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
6727 * gst/rtsp-server/rtsp-media.c:
6728 media: fix check for seekability
6730 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6732 * gst/rtsp-server/rtsp-client.c:
6733 client: use more GIO
6734 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
6736 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6738 * gst/rtsp-server/rtsp-server.c:
6739 server: remove obsolete includes
6741 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
6743 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
6744 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
6745 be available in "on_new_ssrc". The transports are added in
6746 gst_rtsp_media_set_state when going to PLAYING state. However,
6747 "on_new_ssrc" might be called before this happens.
6748 https://bugzilla.gnome.org/show_bug.cgi?id=683304
6750 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
6752 * gst/rtsp-server/rtsp-client.c:
6753 * gst/rtsp-server/rtsp-client.h:
6754 rtsp-client: add signals for rtsp requests (fixes #683287)
6756 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
6758 * gst/rtsp-server/rtsp-client.c:
6759 * gst/rtsp-server/rtsp-client.h:
6760 add new-session signal to rtsp-client (fixes #683058)
6762 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
6765 Automatic update of common submodule
6766 From 668acee to 4f962f7
6768 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
6770 * gst/rtsp-server/rtsp-server.c:
6771 * tests/check/gst/rtspserver.c:
6772 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
6773 Do not assume that *error is set in g_socket_address_enumerator_next.
6774 Added test_bind_already_in_use unit-test.
6775 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
6777 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
6780 Automatic update of common submodule
6781 From 94ccf4c to 668acee
6783 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
6785 * gst/rtsp-server/rtsp-client.c:
6786 * gst/rtsp-server/rtsp-client.h:
6787 rtsp-client: make create_sdp virtual method
6788 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
6790 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6793 Automatic update of common submodule
6794 From 98e386f to 94ccf4c
6796 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6798 * gst/rtsp-server/rtsp-client.c:
6801 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
6803 * gst/rtsp-server/rtsp-client.c:
6804 * gst/rtsp-server/rtsp-client.h:
6805 * gst/rtsp-server/rtsp-server.c:
6806 * gst/rtsp-server/rtsp-server.h:
6807 rtsp-server: use an existing socket to establish HTTP tunnel
6808 Make it possible to transfer a socket from an HTTP server to be used as
6809 an RTSP over HTTP tunnel.
6811 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
6813 * gst/rtsp-server/rtsp-client.c:
6814 * gst/rtsp-server/rtsp-media.c:
6815 * gst/rtsp-server/rtsp-media.h:
6816 rtsp: Handle the blocksize parameter
6817 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
6819 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
6821 * tests/check/Makefile.am:
6822 * tests/check/gst/rtspserver.c:
6823 Have unit test get header from source dir, not installed dir
6824 This makes compilation of unit tests work in a build directory other
6825 than the source directory.
6826 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
6828 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
6830 * gst/rtsp-server/rtsp-media.c:
6831 rtsp-media: update for gst_element_make_from_uri() changes
6833 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
6836 * tests/Makefile.am:
6837 * tests/check/Makefile.am:
6838 * tests/check/gst/rtspserver.c:
6840 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
6842 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
6844 * gst/rtsp-server/rtsp-media.c:
6845 rtsp-media: don't collect media stats when going to NULL
6846 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
6848 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6850 * gst/rtsp-server/rtsp-client.c:
6851 client: don't leak transports
6853 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
6855 * gst/rtsp-server/rtsp-client.c:
6856 rtsp-client: free transport on no_stream in SETUP handler
6858 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
6860 * gst/rtsp-server/rtsp-client.c:
6861 rtsp-client: changed session media iteration
6862 In client_unlink_session: now don't iterate in session->medias
6863 list where items are removed by gst_rtsp_session_release_media.
6864 Instead, repeatedly remove the first item.
6866 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
6868 * gst/rtsp-server/rtsp-client.c:
6869 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
6870 GstRTSPSessionMedia is not a GObject type. When the
6871 GstRTSPSession is freed, it will free the media.
6873 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
6875 * gst/rtsp-server/rtsp-media-factory.c:
6876 factory: plug pad leak in collect_streams
6877 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
6878 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
6879 will take one reference, and the other reference will otherwise
6882 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
6885 configure: suppress some warnings when debug is disabled
6886 Warnings about unused variables should be suppressed if core has the
6887 debug system disabled.
6888 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
6890 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6892 * docs/libs/Makefile.am:
6893 docs: fix build in uninstalled setup
6894 Include gst-plugins-base libs properly.
6896 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
6898 * docs/libs/gst-rtsp-server.types:
6899 docs: include headers defining rtsp-server object types
6900 Fixes compiler warnings during docs build.
6901 https://bugzilla.gnome.org/show_bug.cgi?id=676824
6903 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
6906 configure: Add warning flags for compiler when configuring
6907 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
6909 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6912 Automatic update of common submodule
6913 From 03a0e57 to 98e386f
6915 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6918 Automatic update of common submodule
6919 From 1fab359 to 03a0e57
6921 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
6923 * gst/rtsp-server/rtsp-client.c:
6924 client: fix GSocketAddress leak in gst_rtsp_client_accept
6925 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
6927 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6930 Automatic update of common submodule
6931 From f1b5a96 to 1fab359
6933 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6936 Automatic update of common submodule
6937 From 92b7266 to f1b5a96
6939 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6942 Automatic update of common submodule
6943 From ec1c4a8 to 92b7266
6945 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6948 Automatic update of common submodule
6949 From 3429ba6 to ec1c4a8
6951 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
6953 * gst/rtsp-server/rtsp-auth.c:
6954 * gst/rtsp-server/rtsp-client.c:
6955 * gst/rtsp-server/rtsp-media-factory-uri.c:
6956 * gst/rtsp-server/rtsp-server.c:
6957 rtsp: fix compiler warnings
6958 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
6960 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6963 Automatic update of common submodule
6964 From dc70203 to 3429ba6
6966 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6968 * gst/rtsp-server/rtsp-client.c:
6969 * gst/rtsp-server/rtsp-media-factory.c:
6970 * gst/rtsp-server/rtsp-media-factory.h:
6971 * gst/rtsp-server/rtsp-media.c:
6972 * gst/rtsp-server/rtsp-media.h:
6973 * gst/rtsp-server/rtsp-server.c:
6974 * gst/rtsp-server/rtsp-server.h:
6975 * gst/rtsp-server/rtsp-session-pool.c:
6976 * gst/rtsp-server/rtsp-session-pool.h:
6977 rtsp-server: port to new thread API
6979 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6982 Automatic update of common submodule
6983 From 6db25be to dc70203
6985 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6987 * gst/rtsp-server/rtsp-auth.c:
6988 * gst/rtsp-server/rtsp-auth.h:
6989 * gst/rtsp-server/rtsp-client.c:
6990 rtsp-server: Fix compilation and compiler warnings
6992 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6996 * gst/rtsp-server/Makefile.am:
6997 configure: Modernize autotools setup a bit
6998 Also we now only create tar.bz2 and tar.xz tarballs.
7000 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7003 Automatic update of common submodule
7004 From 464fe15 to 6db25be
7006 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7009 Automatic update of common submodule
7010 From 7fda524 to 464fe15
7012 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7015 * docs/libs/Makefile.am:
7016 * docs/version.entities.in:
7018 * gst/rtsp-server/Makefile.am:
7019 * pkgconfig/Makefile.am:
7020 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
7021 * pkgconfig/gstreamer-rtsp-server.pc.in:
7022 * tests/Makefile.am:
7023 rtsp-server: Update versioning
7025 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7027 Merge remote-tracking branch 'origin/0.10'
7029 gst/rtsp-server/rtsp-session-pool.c
7031 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7033 * gst/rtsp-server/rtsp-session-pool.c:
7034 rtsp-server: Don't use deprecated GLib API
7036 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7038 Replace master with 0.11
7040 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7042 Merge branch 'master' into 0.11
7044 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7046 Merge branch 'master' into 0.11
7048 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
7051 A couple minor typo fixes
7053 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7055 * gst/rtsp-server/rtsp-media.c:
7056 media: fix state of the appqueue
7058 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7060 * gst/rtsp-server/rtsp-media-factory-uri.c:
7061 factory: use videoconvert
7063 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7065 * gst/rtsp-server/rtsp-media-factory-uri.c:
7066 factory: change to new style caps
7068 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7070 * gst/rtsp-server/rtsp-client.c:
7071 * gst/rtsp-server/rtsp-client.h:
7072 * gst/rtsp-server/rtsp-media-factory-uri.c:
7073 * gst/rtsp-server/rtsp-media.c:
7074 * gst/rtsp-server/rtsp-server.c:
7075 * gst/rtsp-server/rtsp-server.h:
7076 * gst/rtsp-server/rtsp-session-pool.c:
7077 rtsp-server: port to GIO
7080 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7083 configure: fix build
7085 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7088 docs: fix for gst_rtsp_server_set_port() -> _set_service()
7089 https://bugzilla.gnome.org/show_bug.cgi?id=666548
7091 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7094 * examples/Makefile.am:
7095 First rule of gst-rtsp-server club: don't talk about gst-phonon
7097 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7100 * pkgconfig/Makefile.am:
7101 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
7102 * pkgconfig/gstreamer-rtsp-server.pc.in:
7103 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
7104 For consistency with all other modules.
7106 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7108 * gst/rtsp-server/rtsp-client.c:
7109 rtsp-client: update for new map API
7111 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7114 * bindings/Makefile.am:
7115 * bindings/python/Makefile.am:
7116 * bindings/python/arg-types.py:
7117 * bindings/python/codegen/Makefile.am:
7118 * bindings/python/codegen/__init__.py:
7119 * bindings/python/codegen/argtypes.py:
7120 * bindings/python/codegen/code-coverage.py:
7121 * bindings/python/codegen/codegen.py:
7122 * bindings/python/codegen/definitions.py:
7123 * bindings/python/codegen/defsparser.py:
7124 * bindings/python/codegen/docextract.py:
7125 * bindings/python/codegen/docgen.py:
7126 * bindings/python/codegen/fileprefix.override:
7127 * bindings/python/codegen/fileprefixmodule.c:
7128 * bindings/python/codegen/h2def.py:
7129 * bindings/python/codegen/mergedefs.py:
7130 * bindings/python/codegen/mkskel.py:
7131 * bindings/python/codegen/override.py:
7132 * bindings/python/codegen/reversewrapper.py:
7133 * bindings/python/codegen/scmexpr.py:
7134 * bindings/python/rtspserver-types.defs:
7135 * bindings/python/rtspserver.defs:
7136 * bindings/python/rtspserver.override:
7137 * bindings/python/rtspservermodule.c:
7138 * bindings/python/test.py:
7140 python: remove pygst-based python bindings
7141 pygi is the future, apparently.
7143 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
7146 Automatic update of common submodule
7147 From c463bc0 to 7fda524
7149 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7152 Automatic update of common submodule
7153 From 2a59016 to c463bc0
7155 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7158 Automatic update of common submodule
7159 From 0807187 to 2a59016
7161 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7164 Automatic update of common submodule
7165 From 11f0cd5 to 0807187
7167 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7169 * examples/test-auth.c:
7170 example: update for new caps
7172 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7174 * examples/test-video.c:
7175 * gst/rtsp-server/rtsp-client.c:
7176 * gst/rtsp-server/rtsp-media-factory-uri.c:
7177 * gst/rtsp-server/rtsp-media.c:
7178 * gst/rtsp-server/rtsp-media.h:
7179 * gst/rtsp-server/rtsp-session.c:
7180 * gst/rtsp-server/rtsp-session.h:
7181 rtsp-server: port some more to 0.11
7183 Remove bufferlist stuff
7185 Add queue before appsink now that preroll-queue-len is gone.
7186 Update for request pad changes.
7188 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7190 Merge branch 'master' into 0.11
7192 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
7194 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7195 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
7196 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
7198 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
7200 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7201 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
7202 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
7204 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7206 Merge branch 'master' into 0.11
7208 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7210 * gst/rtsp-server/rtsp-media.c:
7211 * gst/rtsp-server/rtsp-media.h:
7212 media: add a seekable boolean
7213 Maintain the seekable state with a new variable instead of reusing the
7216 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
7218 * gst/rtsp-server/rtsp-media.c:
7219 Disallow seek in live media
7221 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7223 Merge branch 'master' into 0.11
7225 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
7227 * gst/rtsp-server/rtsp-server.c:
7228 #ifdef statements for windows socket creation were missing
7230 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
7233 Automatic update of common submodule
7234 From a39eb83 to 11f0cd5
7236 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
7239 Automatic update of common submodule
7240 From 605cd9a to a39eb83
7242 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7244 Merge branch 'master' into 0.11
7246 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7248 * gst/rtsp-server/rtsp-client.c:
7249 client: use method to access property
7251 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7253 * gst/rtsp-server/rtsp-media-factory.c:
7254 * gst/rtsp-server/rtsp-media-factory.h:
7255 media-factory: add protocols property
7256 Add a property to configure the allowed protocols in the media created from the
7259 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7261 * gst/rtsp-server/rtsp-media-factory.c:
7262 * gst/rtsp-server/rtsp-media-factory.h:
7263 media-factory: add media-configure signal
7264 Add signal to allow the application to configure the media after it was created
7267 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7269 * gst/rtsp-server/rtsp-client.c:
7270 client: use method to access property
7272 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7274 * gst/rtsp-server/rtsp-media-factory.c:
7275 * gst/rtsp-server/rtsp-media-factory.h:
7276 media-factory: add protocols property
7277 Add a property to configure the allowed protocols in the media created from the
7280 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7282 * gst/rtsp-server/rtsp-media-factory.c:
7283 * gst/rtsp-server/rtsp-media-factory.h:
7284 media-factory: add media-configure signal
7285 Add signal to allow the application to configure the media after it was created
7288 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7290 Merge branch 'master' into 0.11
7292 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7294 * gst/rtsp-server/rtsp-client.c:
7295 client: use media multicast group
7297 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7299 * gst/rtsp-server/rtsp-media-factory.h:
7300 * gst/rtsp-server/rtsp-server.h:
7301 * gst/rtsp-server/rtsp-session-pool.h:
7302 * gst/rtsp-server/rtsp-session.h:
7305 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
7307 * gst/rtsp-server/rtsp-client.c:
7308 * gst/rtsp-server/rtsp-sdp.h:
7309 sdp: copy and free the server ip address
7310 Copy and free the server ip address to make memory management easier later.
7312 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7314 * gst/rtsp-server/rtsp-media-factory.c:
7315 media-factory: configure multicast in media
7317 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7319 * gst/rtsp-server/rtsp-media.c:
7320 * gst/rtsp-server/rtsp-media.h:
7321 media: add property for multicast group
7322 Add a property to configure the multicast group in the media.
7323 Based on patches from Marc Leeman and Robert Krakora.
7325 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7327 * gst/rtsp-server/rtsp-media-factory.c:
7328 * gst/rtsp-server/rtsp-media-factory.h:
7329 media-factory: add property for multicast group
7330 Add a property to configure the multicast group in the media factory.
7331 Based on patches from Marc Leeman and Robert Krakora.
7333 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7335 * gst/rtsp-server/rtsp-client.c:
7336 client: do configuration of transport in one place
7337 Move the configuration of the transport destination address to where we also
7338 configure the other bits.
7340 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7342 * gst/rtsp-server/rtsp-client.c:
7343 client: use media multicast group
7345 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7347 * gst/rtsp-server/rtsp-media-factory.h:
7348 * gst/rtsp-server/rtsp-server.h:
7349 * gst/rtsp-server/rtsp-session-pool.h:
7350 * gst/rtsp-server/rtsp-session.h:
7353 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
7355 * gst/rtsp-server/rtsp-client.c:
7356 * gst/rtsp-server/rtsp-sdp.h:
7357 sdp: copy and free the server ip address
7358 Copy and free the server ip address to make memory management easier later.
7360 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7362 * gst/rtsp-server/rtsp-media-factory.c:
7363 media-factory: configure multicast in media
7365 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7367 * gst/rtsp-server/rtsp-media.c:
7368 * gst/rtsp-server/rtsp-media.h:
7369 media: add property for multicast group
7370 Add a property to configure the multicast group in the media.
7371 Based on patches from Marc Leeman and Robert Krakora.
7373 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7375 * gst/rtsp-server/rtsp-media-factory.c:
7376 * gst/rtsp-server/rtsp-media-factory.h:
7377 media-factory: add property for multicast group
7378 Add a property to configure the multicast group in the media factory.
7379 Based on patches from Marc Leeman and Robert Krakora.
7381 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7383 * gst/rtsp-server/rtsp-client.c:
7384 client: do configuration of transport in one place
7385 Move the configuration of the transport destination address to where we also
7386 configure the other bits.
7388 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7390 Merge branch 'master' into 0.11
7392 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
7394 * gst/rtsp-server/rtsp-client.c:
7395 client: destroy pipeline on client disconnect with no prior TEARDOWN.
7396 The problem occurs when the client abruptly closes the connection without
7397 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
7398 server is where the pipeline gets torn down. Since this handler is not called,
7399 the pipeline remains and is up and running. Subsequent clients get their own
7400 pipelines and if the do not issue TEARDOWNs then those pipelines will also
7401 remain up and running. This is a resource leak.
7403 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7405 Merge branch 'master' into 0.11
7407 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
7409 * gst/rtsp-server/rtsp-media-factory.c:
7410 * gst/rtsp-server/rtsp-media-factory.h:
7411 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
7412 For example, it can be used to retrieve source elements like appsrc, in a more
7413 convenient way than subclassing get_element.
7415 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7417 Merge branch 'master' into 0.11
7419 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
7421 * gst/rtsp-server/rtsp-server.c:
7422 rtsp-server: hold on to reference while using object
7424 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7426 * gst/rtsp-server/rtsp-media.c:
7429 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7432 configure: use unstable api
7434 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
7436 * gst/rtsp-server/rtsp-client.c:
7437 client: fix reference counting
7439 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
7441 * gst/rtsp-server/rtsp-client.c:
7442 * gst/rtsp-server/rtsp-media.c:
7443 fix compiler warnings about unused variables
7445 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
7447 * examples/test-launch.c:
7448 * examples/test-readme.c:
7449 * examples/test-uri.c:
7450 * examples/test-video.c:
7451 examples: tell rtsp uri when ready
7453 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
7456 Automatic update of common submodule
7457 From 69b981f to 605cd9a
7459 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7461 * gst/rtsp-server/rtsp-client.c:
7462 client: update for buffer API change
7464 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7466 * gst/rtsp-server/Makefile.am:
7467 Makefile.am: 0.10 => @GST_MAJORMINOR@
7469 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7471 * gst/rtsp-server/rtsp-media-factory-uri.c:
7472 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
7474 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7476 * gst/rtsp-server/.gitignore:
7477 .gitignore: 0.10 => 0.11
7479 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7481 * gst/rtsp-server/Makefile.am:
7482 Makefile.am: 0.10 => @GST_MAJORMINOR@
7484 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7486 Merge branch 'master' into 0.11
7488 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
7491 Automatic update of common submodule
7492 From 9e5bbd5 to 69b981f
7494 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
7497 Automatic update of common submodule
7498 From fd35073 to 9e5bbd5
7500 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
7503 Automatic update of common submodule
7504 From 46dfcea to fd35073
7506 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7508 * gst/rtsp-server/rtsp-media-factory-uri.c:
7509 * gst/rtsp-server/rtsp-media.c:
7510 media: port to new caps API
7512 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7514 Merge branch 'master' into 0.11
7516 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
7518 * bindings/vala/gst-rtsp-server-0.10.vapi:
7519 Updated Vala bindings.
7520 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
7522 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
7524 * gst/rtsp-server/rtsp-server.c:
7525 * gst/rtsp-server/rtsp-server.h:
7526 Add a signal for newly connected clients.
7527 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
7529 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
7531 * bindings/python/rtspserver.override:
7532 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
7534 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7536 * gst/rtsp-server/Makefile.am:
7537 * gst/rtsp-server/rtsp-client.c:
7538 * gst/rtsp-server/rtsp-funnel.c:
7539 * gst/rtsp-server/rtsp-funnel.h:
7540 * gst/rtsp-server/rtsp-media.c:
7541 rtsp-server: port to 0.11
7543 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7548 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7550 Merge branch 'master' into 0.11
7555 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7558 Automatic update of common submodule
7559 From c3cafe1 to 46dfcea
7561 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
7563 * bindings/python/Makefile.am:
7564 * bindings/python/rtspserver.defs:
7565 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
7567 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
7569 * bindings/python/arg-types.py:
7570 python bindings: add GstRTSPUrlParam
7571 Needed to implement MediaFactory virtual proxies
7573 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
7575 * bindings/python/arg-types.py:
7576 python bindings: fix returning GstRTSPUrl types
7578 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
7580 * bindings/python/arg-types.py:
7581 python bindings: add arg type for GstRTSPUrl
7583 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
7585 * bindings/python/rtspserver.defs:
7586 python bindings: fix the definition of MediaFactory.collect_stream
7588 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
7591 Automatic update of common submodule
7592 From 1ccbe09 to c3cafe1
7594 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7597 Automatic update of common submodule
7598 From 193b717 to 1ccbe09
7600 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
7603 Automatic update of common submodule
7604 From b77e2bf to 193b717
7606 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7609 build: Include lcov.mak to allow test coverage report generation
7611 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7614 Automatic update of common submodule
7615 From d8814b6 to b77e2bf
7617 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7620 Automatic update of common submodule
7621 From 6aaa286 to d8814b6
7623 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
7626 Automatic update of common submodule
7627 From 6aec6b9 to 6aaa286
7629 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
7632 autogen: wingo signed comment
7634 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
7636 * gst/rtsp-server/rtsp-session-pool.c:
7637 session: use full charset for RTSP session ID
7638 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
7639 session ID more difficult.
7640 https://bugzilla.gnome.org/show_bug.cgi?id=643812
7642 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7644 * gst/rtsp-server/Makefile.am:
7645 rtsp-server: Don't install the funnel header
7647 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
7650 Automatic update of common submodule
7651 From 1de7f6a to 6aec6b9
7653 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7656 configure: require core/base 0.10.31
7657 Needed at least for gst_plugin_feature_rank_compare_func().
7659 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
7662 Automatic update of common submodule
7663 From f94d739 to 1de7f6a
7665 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7667 * gst/rtsp-server/rtsp-media.c:
7668 media: remove more unused code
7670 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7672 * gst/rtsp-server/rtsp-media.c:
7673 * gst/rtsp-server/rtsp-media.h:
7674 media: remove duplicate filtering
7675 Remove the duplicate filtering code now that we have a released -good version.
7676 Give a warning instead.
7678 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7680 * gst/rtsp-server/rtsp-media-factory.c:
7681 * gst/rtsp-server/rtsp-media.c:
7682 media: fix default buffer size
7684 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7686 * gst/rtsp-server/rtsp-media-factory.c:
7687 * gst/rtsp-server/rtsp-media-factory.h:
7688 media-factory: add property to configure the buffer-size
7689 Add a property to configure the kernel UDP buffer size.
7691 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7693 * gst/rtsp-server/rtsp-media.c:
7694 * gst/rtsp-server/rtsp-media.h:
7695 media: add property to configure kernel buffer sizes
7696 Add a property to configure the kernel UDP buffer size.
7698 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7701 configure: set PYGOBJECT_REQ before using it
7702 https://bugzilla.gnome.org/show_bug.cgi?id=640641
7704 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7707 docs: recursive into sub-directories on 'make upload'
7709 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7711 * docs/libs/gst-rtsp-server-docs.sgml:
7712 * docs/version.entities.in:
7713 docs: mention full version these docs are for, not just major-minor
7715 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7720 === release 0.10.8 ===
7722 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7727 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7729 * gst/rtsp-server/rtsp-server.c:
7730 rtsp-server: clarify docs a little
7732 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7734 * gst/rtsp-server/rtsp-media.c:
7735 media: init debug category before starting thread
7737 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7739 * gst/rtsp-server/rtsp-auth.c:
7740 auth: add realm to make it more spec compliant
7742 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7744 * gst/rtsp-server/rtsp-server.c:
7745 * gst/rtsp-server/rtsp-server.h:
7748 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7750 * examples/test-video.c:
7751 example: improve example docs a little
7753 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7755 * gst/rtsp-server/rtsp-server.c:
7756 server: ensure the watch has a ref to the server
7758 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7760 * gst/rtsp-server/rtsp-server.c:
7761 server: simpify channel function
7763 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7765 * gst/rtsp-server/rtsp-server.c:
7766 * gst/rtsp-server/rtsp-server.h:
7767 server: simplify management of channel and source
7768 We don't need to keep around the channel and source objects. Let the mainloop
7769 and the source manage the source and channel respectively.
7771 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7777 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7780 * tests/Makefile.am:
7781 * tests/test-cleanup.c:
7782 tests: add tests directory and cleanup test
7784 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7786 * gst/rtsp-server/rtsp-media-factory-uri.c:
7787 * gst/rtsp-server/rtsp-media-factory.c:
7788 * gst/rtsp-server/rtsp-media-mapping.c:
7789 * gst/rtsp-server/rtsp-media.c:
7790 * gst/rtsp-server/rtsp-session-pool.c:
7791 * gst/rtsp-server/rtsp-session.c:
7792 server: improve debugging in various objects
7794 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7796 * gst/rtsp-server/rtsp-server.c:
7797 server: chain up to the parent finalize
7799 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
7801 * bindings/python/rtspserver-types.defs:
7802 * bindings/python/rtspserver.defs:
7803 * bindings/python/rtspserver.override:
7804 * bindings/python/test.py:
7805 gst-rtsp-server: update python bindings
7807 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7809 * gst/rtsp-server/rtsp-client.c:
7810 client: use the response from the clientstate
7811 Create the response object only once and store in the client state.
7812 Make all methods use the state response,
7814 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7816 * gst/rtsp-server/rtsp-server.c:
7817 server: use signal to keep track of clients
7818 Keep track of all the clients that the server creates and remove them when they
7819 fire the 'closed' signal.
7821 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7823 * gst/rtsp-server/rtsp-client.c:
7824 * gst/rtsp-server/rtsp-client.h:
7825 client: emit signal when closing
7827 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7829 * examples/.gitignore:
7830 * examples/Makefile.am:
7831 * examples/test-auth.c:
7832 * examples/test-video.c:
7833 * gst/rtsp-server/rtsp-auth.c:
7834 * gst/rtsp-server/rtsp-auth.h:
7835 * gst/rtsp-server/rtsp-client.c:
7836 * gst/rtsp-server/rtsp-media-factory.c:
7837 * gst/rtsp-server/rtsp-media.c:
7838 * gst/rtsp-server/rtsp-media.h:
7839 * gst/rtsp-server/rtsp-session-pool.h:
7840 * gst/rtsp-server/rtsp-session.h:
7841 media: enable per factory authorisations
7842 Allow for adding a GstRTSPAuth on the factory and media level and check
7843 permissions when accessing the factory.
7844 Add hints to the auth methods for future more fine grained authorisation.
7845 Add example application for per factory authentication.
7847 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7849 * gst/rtsp-server/rtsp-auth.c:
7850 * gst/rtsp-server/rtsp-auth.h:
7851 * gst/rtsp-server/rtsp-client.c:
7852 * gst/rtsp-server/rtsp-client.h:
7853 * gst/rtsp-server/rtsp-params.c:
7854 * gst/rtsp-server/rtsp-params.h:
7855 rtsp-server: Pass ClientState structure arround
7856 Pass the collected information for the ongoing request in a GstRTSPClientState
7857 structure that we can then pass around to simplify the method arguments. This
7858 will also be handy when we implement logging functionality.
7860 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7862 * gst/rtsp-server/rtsp-media-factory.c:
7863 * gst/rtsp-server/rtsp-media-factory.h:
7864 media-factory: add methods to configure authorisation
7866 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7868 * gst/rtsp-server/rtsp-client.c:
7869 client: unref auth in finalize
7871 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7873 * gst/rtsp-server/rtsp-server.c:
7874 server: unref auth in finalize
7876 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7878 * docs/libs/gst-rtsp-server-docs.sgml:
7879 * docs/libs/gst-rtsp-server-sections.txt:
7880 * docs/libs/gst-rtsp-server.types:
7883 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7885 * gst/rtsp-server/rtsp-server.c:
7886 * gst/rtsp-server/rtsp-server.h:
7887 server: separate create and accept
7888 Create separate create and accept methods so that subclasses can create custom
7890 Configure the server in the client object and prepare for keeping track of
7893 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7895 * gst/rtsp-server/rtsp-client.c:
7896 * gst/rtsp-server/rtsp-client.h:
7897 client: add support for setting the server.
7898 Add support for keeping a ref to the server that started this client
7901 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7903 * gst/rtsp-server/rtsp-auth.c:
7904 auth: fix memleak and add some docs
7905 Fix a memleak of the basic auth token.
7906 Add docs for the helper function
7908 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7910 * gst/rtsp-server/rtsp-auth.c:
7911 * gst/rtsp-server/rtsp-auth.h:
7912 * gst/rtsp-server/rtsp-client.c:
7913 client: delegate setup of auth to the manager
7914 Delegate the configuration of the authentication tokens to the manager object
7917 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7919 * examples/test-video.c:
7920 * gst/rtsp-server/Makefile.am:
7921 * gst/rtsp-server/rtsp-auth.c:
7922 * gst/rtsp-server/rtsp-auth.h:
7923 * gst/rtsp-server/rtsp-client.c:
7924 * gst/rtsp-server/rtsp-client.h:
7925 * gst/rtsp-server/rtsp-server.c:
7926 * gst/rtsp-server/rtsp-server.h:
7927 auth: add authentication object
7928 Add an object that can check the authorization of requests.
7929 Implement basic authentication.
7930 Add example authentication to test-video
7932 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7934 * gst/rtsp-server/rtsp-server.c:
7935 * gst/rtsp-server/rtsp-server.h:
7936 server: move includes back
7937 the includes are needed for sockaddr_in.
7939 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7941 * gst/rtsp-server/rtsp-client.c:
7942 * gst/rtsp-server/rtsp-client.h:
7943 * gst/rtsp-server/rtsp-server.c:
7944 * gst/rtsp-server/rtsp-server.h:
7945 rtsp: move network includes where they are needed
7947 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
7949 * gst/rtsp-server/rtsp-media.h:
7950 rtsp-media.h: Minor corrections in comments.
7953 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
7956 Automatic update of common submodule
7957 From e572c87 to f94d739
7959 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7963 * docs/libs/.gitignore:
7964 * examples/.gitignore:
7965 * gst/rtsp-server/.gitignore:
7968 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7970 * docs/libs/Makefile.am:
7971 docs: We don't build ps/pdf for API reference docs
7973 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7976 Automatic update of common submodule
7977 From ccbaa85 to e572c87
7979 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7982 Automatic update of common submodule
7983 From 46445ad to ccbaa85
7985 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7987 * gst/rtsp-server/Makefile.am:
7988 * gst/rtsp-server/rtsp-funnel.c:
7989 * gst/rtsp-server/rtsp-funnel.h:
7990 * gst/rtsp-server/rtsp-media.c:
7991 funnel: rename fsfunnel to rtspfunnel
7992 Rename the funnel to avoid conflicts with the farsight one.
7994 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7996 * gst/rtsp-server/Makefile.am:
7997 * gst/rtsp-server/fs-funnel.c:
7998 * gst/rtsp-server/fs-funnel.h:
7999 * gst/rtsp-server/rtsp-media.c:
8000 rtsp-media: add and use fsfunnel
8001 Add a copy of fsfunnel to the build because input-selector removed the (broken)
8002 select-all property that we need.
8004 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8006 * gst/rtsp-server/Makefile.am:
8007 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
8008 Use PKG_CONFIG_PATH specified at configure time (if any) as well
8009 for the g-ir-compiler, rather than just assuming the env var has
8012 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8019 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
8021 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8024 * gst/rtsp-server/Makefile.am:
8025 gobject-introspection: fix g-i build for uninstalled setup
8026 Requires gst-plugins-base git (> 0.10.31.2).
8028 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8030 * examples/test-uri.c:
8031 examples: add some more options and comments
8033 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8035 * gst/rtsp-server/rtsp-media-factory-uri.c:
8036 factory-uri: use right property type
8038 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8040 * gst/rtsp-server/rtsp-media-factory-uri.c:
8041 factory-uri: attempt to configure buffer-lists
8042 Attempt to configure buffer lists in the payloader for improved performance.
8044 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8046 * gst/rtsp-server/rtsp-media.c:
8047 media: attempt to configure bigger UDP buffers
8048 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
8049 send buffers with high bitrate streams.
8051 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
8053 * gst/rtsp-server/rtsp-client.c:
8054 client: use the socket length from getsockname
8055 Use the length returned by getsockname to perform the getnameinfo call because
8056 the size can depend on the socket type and platform.
8059 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8061 * docs/libs/gst-rtsp-server-docs.sgml:
8062 * docs/libs/gst-rtsp-server-sections.txt:
8063 docs: add uri factory to the docs
8065 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8067 * gst/rtsp-server/rtsp-client.c:
8068 * gst/rtsp-server/rtsp-media.h:
8071 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8073 * gst/rtsp-server/rtsp-client.c:
8074 * gst/rtsp-server/rtsp-media.c:
8075 * gst/rtsp-server/rtsp-media.h:
8076 * gst/rtsp-server/rtsp-session.c:
8077 * gst/rtsp-server/rtsp-session.h:
8078 rtsp-server: add support for buffer lists
8079 Add support for sending bufferlists received from appsink.
8082 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8084 * gst/rtsp-server/rtsp-client.c:
8085 * gst/rtsp-server/rtsp-media.c:
8086 * gst/rtsp-server/rtsp-media.h:
8087 * gst/rtsp-server/rtsp-sdp.c:
8088 media: make method to retrieve the play range
8089 Make a method to retrieve the playback range so that we can conditionally create
8090 a different range for the SDP and the PLAY requests.
8092 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8094 * gst/rtsp-server/rtsp-media.c:
8095 * gst/rtsp-server/rtsp-media.h:
8096 media: add signal to notify of state changes
8098 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8100 * gst/rtsp-server/rtsp-client.h:
8101 client: cleanup headers
8103 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8105 * gst/rtsp-server/rtsp-client.c:
8108 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8110 * gst/rtsp-server/rtsp-media-factory-uri.c:
8111 * gst/rtsp-server/rtsp-media-factory-uri.h:
8112 factory-uri: add support for gstpay
8113 Add an option to prefer gstpay over decoder + raw payloader.
8115 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8117 * gst/rtsp-server/rtsp-media-factory-uri.c:
8118 * gst/rtsp-server/rtsp-media-factory-uri.h:
8119 factory-uri: rework the autoplugger.
8120 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
8123 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8125 * gst/rtsp-server/rtsp-media-factory-uri.c:
8126 factory-uri: use better factory filter
8127 Make better payloader filter based on autoplug rank and RTP use case.
8129 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8132 Automatic update of common submodule
8133 From 169462a to 46445ad
8135 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8137 * gst/rtsp-server/rtsp-server.c:
8138 server: set SO_REUSEADDR before bind
8139 Set the SO_REUSEADDR _before_ bind() to make it actually work.
8141 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8143 * gst/rtsp-server/rtsp-media.c:
8144 * gst/rtsp-server/rtsp-media.h:
8145 media: emit prepared signal when prepared
8146 Make a 'prepared' signal and emit it when we successfully prepared the element.
8147 This signal can be used to configure the media object after it has been prepared
8150 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
8153 Automatic update of common submodule
8154 From 011bcc8 to 169462a
8156 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
8158 python an optional dependency
8159 * configure.ac: Move up valgrind and g-i checks. Make the python
8160 dependency optional, as it was before.
8162 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8164 Merge branch 'master' into 0.11
8169 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8171 * gst/rtsp-server/rtsp-media.c:
8172 media: update range when active clients changed
8173 When we changed the number of active clients, update the current range
8174 information because we want the second client connecting to a shared resource
8175 continue from where the stream currently.
8177 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8179 * gst/rtsp-server/rtsp-media-factory-uri.c:
8180 * gst/rtsp-server/rtsp-media-factory-uri.h:
8181 factory-uri: add colorspace and fix pt
8182 Rework the way we pass data to the autoplugger.
8183 When we have raw caps, plug a converter element to make pluggin to raw
8184 payloaders more successful.
8185 Make sure all dynamically plugged payloaders have a unique payload types.
8187 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8189 * examples/Makefile.am:
8190 * examples/test-uri.c:
8191 example: add example of the uri factory
8193 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8195 * gst/rtsp-server/Makefile.am:
8196 * gst/rtsp-server/rtsp-media-factory-uri.c:
8197 * gst/rtsp-server/rtsp-media-factory-uri.h:
8198 * gst/rtsp-server/rtsp-server.h:
8199 factory-uri: add a factory to stream any URI
8200 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
8203 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8205 * gst/rtsp-server/rtsp-media.c:
8206 * gst/rtsp-server/rtsp-media.h:
8207 media: ignore spurious ASYNC_DONE messages
8208 When we are dynamically adding pads, the addition of the udpsrc elements will
8209 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
8210 the real ASYNC_DONE when everything is prerolled.
8212 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8214 * gst/rtsp-server/rtsp-media-factory.c:
8215 * gst/rtsp-server/rtsp-media-factory.h:
8216 media-factory: make lock macro
8218 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
8220 * gst/rtsp-server/rtsp-client.c:
8221 rtsp-server: Remove unused variable and dead assignment
8223 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
8225 * examples/test-launch.c:
8226 * examples/test-mp4.c:
8227 * examples/test-ogg.c:
8228 * examples/test-readme.c:
8229 * examples/test-sdp.c:
8230 * examples/test-video.c:
8231 examples: Run gst-indent
8233 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
8235 * gst/rtsp-server/rtsp-client.c:
8236 * gst/rtsp-server/rtsp-media-factory.c:
8237 * gst/rtsp-server/rtsp-media-mapping.c:
8238 * gst/rtsp-server/rtsp-media.c:
8239 * gst/rtsp-server/rtsp-params.c:
8240 * gst/rtsp-server/rtsp-sdp.c:
8241 * gst/rtsp-server/rtsp-server.c:
8242 * gst/rtsp-server/rtsp-session-pool.c:
8243 * gst/rtsp-server/rtsp-session.c:
8244 rtsp-server: Run gst-indent
8245 Since it wasn't using the upstream common previously, there was no
8246 indentation check before commiting.
8248 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
8250 * gst/rtsp-server/rtsp-media-mapping.h:
8251 * gst/rtsp-server/rtsp-media.c:
8252 * gst/rtsp-server/rtsp-media.h:
8253 * gst/rtsp-server/rtsp-sdp.c:
8254 * gst/rtsp-server/rtsp-session-pool.h:
8255 * gst/rtsp-server/rtsp-session.c:
8256 * gst/rtsp-server/rtsp-session.h:
8257 rtsp-server: Some more doc fixups
8259 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8262 Makefile: Add cruft-cleaning support
8264 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8269 * docs/libs/Makefile.am:
8270 * docs/libs/gst-rtsp-server-docs.sgml:
8271 * docs/libs/gst-rtsp-server-sections.txt:
8272 * docs/libs/gst-rtsp-server.types:
8273 * docs/version.entities.in:
8274 docs: Add gtk-doc build system
8276 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8278 * gst/rtsp-server/Makefile.am:
8279 Makefile.am: Use standard GIR make behaviour
8281 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8285 autogen/configure: Bring more in sync to standard gst module behaviour
8287 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8289 * gst/rtsp-server/rtsp-media.c:
8290 media: warn and fail when gstrtpbin is not found
8292 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8295 configure: open 0.11 branch
8297 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
8301 Add common submodule
8303 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
8306 * common/Makefile.am:
8307 * common/c-to-xml.py:
8309 * common/coverage/coverage-report-entry.pl:
8310 * common/coverage/coverage-report.pl:
8311 * common/coverage/coverage-report.xsl:
8312 * common/coverage/lcov.mak:
8313 * common/gettext.patch:
8314 * common/glib-gen.mak:
8315 * common/gst-autogen.sh:
8316 * common/gst-xmlinspect.py:
8318 * common/gstdoc-scangobj:
8319 * common/gtk-doc-plugins.mak:
8320 * common/gtk-doc.mak:
8321 * common/m4/.gitignore:
8322 * common/m4/Makefile.am:
8324 * common/m4/as-ac-expand.m4:
8325 * common/m4/as-auto-alt.m4:
8326 * common/m4/as-compiler-flag.m4:
8327 * common/m4/as-compiler.m4:
8328 * common/m4/as-docbook.m4:
8329 * common/m4/as-libtool-tags.m4:
8330 * common/m4/as-libtool.m4:
8331 * common/m4/as-python.m4:
8332 * common/m4/as-scrub-include.m4:
8333 * common/m4/as-version.m4:
8334 * common/m4/ax_create_stdint_h.m4:
8335 * common/m4/check.m4:
8336 * common/m4/glib-gettext.m4:
8337 * common/m4/gst-arch.m4:
8338 * common/m4/gst-args.m4:
8339 * common/m4/gst-check.m4:
8340 * common/m4/gst-debuginfo.m4:
8341 * common/m4/gst-default.m4:
8342 * common/m4/gst-doc.m4:
8343 * common/m4/gst-error.m4:
8344 * common/m4/gst-feature.m4:
8345 * common/m4/gst-function.m4:
8346 * common/m4/gst-gettext.m4:
8347 * common/m4/gst-glib2.m4:
8348 * common/m4/gst-libxml2.m4:
8349 * common/m4/gst-plugindir.m4:
8350 * common/m4/gst-valgrind.m4:
8351 * common/m4/gtk-doc.m4:
8352 * common/m4/introspection.m4:
8354 * common/mangle-tmpl.py:
8355 * common/plugins.xsl:
8357 * common/release.mak:
8358 * common/scangobj-merge.py:
8359 * common/upload.mak:
8360 common: Remove static version
8362 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
8364 * common/m4/introspection.m4:
8365 Update introspection.m4 to match usage
8367 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8371 Remove old stuff from the README
8373 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8378 === release 0.10.7 ===
8380 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8385 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8387 * examples/test-ogg.c:
8388 test-ogg: remove parsers
8389 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
8390 buffers with timestamps. Using the parsers also seems to break things.
8392 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8394 * bindings/vala/gst-rtsp-server-0.10.vapi:
8395 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8396 Updated Vala bindings
8398 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8400 * common/m4/introspection.m4:
8402 * gst/rtsp-server/Makefile.am:
8403 Added initial gobject-introspection support
8405 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8407 * gst/rtsp-server/rtsp-media-factory.c:
8408 media-factory: don't use host for shared hash key
8409 When we generate the key to share made between connections, don't include the
8410 host used to connect so that we can share media even if between clients that
8411 connected with localhost and ones with the ip address.
8413 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8415 * bindings/vala/Makefile.am:
8416 build: fix distcheck
8418 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8420 * bindings/vala/gst-rtsp-server-0.10.vapi:
8421 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8422 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8423 Update Vala bindings
8425 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8427 * bindings/vala/Makefile.am:
8429 Fix configure checks and installation location for Vala bindings
8432 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8437 === release 0.10.6 ===
8439 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8442 configure: release 0.10.6
8444 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8446 * gst/rtsp-server/rtsp-media.c:
8447 media: help the compiler a little
8449 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8451 * gst/rtsp-server/rtsp-media.c:
8452 * gst/rtsp-server/rtsp-media.h:
8453 * gst/rtsp-server/rtsp-session.c:
8454 media: cleanup media transport before freeing
8455 Cleanup the media transport data before freeing. In particular, remove the qdata
8456 from the rtpsource object.
8458 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8460 * gst/rtsp-server/rtsp-media-factory.c:
8461 * gst/rtsp-server/rtsp-media-factory.h:
8462 * gst/rtsp-server/rtsp-media.c:
8463 * gst/rtsp-server/rtsp-media.h:
8464 media-factory: add eos-shutdown property
8465 Add an eos-shutdown property that will send an EOS to the pipeline before
8466 shutting it down. This allows for nice cleanup in case of a muxer.
8469 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8471 * gst/rtsp-server/rtsp-media.c:
8472 * gst/rtsp-server/rtsp-media.h:
8473 media: use multiudpsink send-duplicates when we can
8474 If we have a new enough multiudpsink with the send-duplicates property, use this
8475 instead of doing our own filtering. Our custom filtering code should eventually
8476 be removed when we can depend on a released -good.
8478 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8480 * gst/rtsp-server/rtsp-media.c:
8481 media: don't leak destinations
8482 Refactor and cleanup the destinations array when the stream is destroyed.
8484 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8486 * gst/rtsp-server/rtsp-media.c:
8487 * gst/rtsp-server/rtsp-media.h:
8488 media: don't add udp addresses multiple times
8489 Keep track of the udp addresses we added to udpsink and never add the same udp
8490 destination twice. This avoids duplicate packets when using multicast.
8492 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8494 * gst/rtsp-server/rtsp-server.c:
8495 server: disable use of SO_LINGER
8496 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
8497 server close()s the connection.
8499 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8501 * gst/rtsp-server/rtsp-server.c:
8502 server: use 5 second linger period in SO_LINGER
8503 Wait 5 seconds before clearing the send buffers and reseting the connection with
8504 the client when we do a close. This should be enough time to get the message to
8508 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
8510 * gst/rtsp-server/rtsp-server.c:
8511 server: use SO_LINGER
8512 SO_LINGER on the socket will make sure that any pending data on the socket is
8513 flushed ASAP and that the socket connection is reset. This makes sure that the
8514 socket can be reused immediately.
8517 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8520 README: add blurb about shared media factories
8522 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
8524 * gst/rtsp-server/rtsp-media.c:
8525 Add stdlib.h for atoi()
8527 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8529 * bindings/python/Makefile.am:
8530 * bindings/vala/Makefile.am:
8531 build: distcheck fixes
8532 Fix 'make distcheck', somewhat (it still fails because it tries to
8533 install files into /usr/share/vala/vapi/ irrespective of the
8536 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8539 configure: bump core/base requirements to released version
8540 Makes things less confusing for people.
8542 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8545 configure: fail if GStreamer core/base requirements are not met
8547 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8549 * gst/rtsp-server/rtsp-client.c:
8550 client: improve client cleanups
8551 Make sure the session does not timeout when using TCP. We need to do this
8552 because quicktime player does not send RTCP for some reason in tunneled
8554 Refactor some cleanup code.
8557 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8559 * gst/rtsp-server/rtsp-session.c:
8560 * gst/rtsp-server/rtsp-session.h:
8561 session: add support for prevent session timeouts
8562 Add an atomix counter to prevent session timeouts when we are, for example,
8565 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8567 * gst/rtsp-server/rtsp-client.c:
8568 client: fix unlink on session timeouts
8569 When our session times out, make sure we unlink all streams in this
8571 Remove the tunnelid when closing the connection.
8573 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8575 * gst/rtsp-server/rtsp-session.c:
8576 session: small cleanups
8578 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8580 * gst/rtsp-server/rtsp-client.c:
8581 client: handle lost_tunnel callbacks
8582 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
8583 hashtable so that we can reuse it for when the client reopens the POST
8585 Close the connection after a TEARDOWN.
8586 Make sure or watchid is cleared when the watch is removed.
8589 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8591 * gst/rtsp-server/rtsp-client.c:
8592 * gst/rtsp-server/rtsp-media.c:
8593 * gst/rtsp-server/rtsp-sdp.c:
8594 rtsp-server: add more support for multicast
8596 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8599 * gst/rtsp-server/rtsp-media.c:
8600 * gst/rtsp-server/rtsp-media.h:
8601 media: allow configuration of allowed lower transport
8603 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8605 * gst/rtsp-server/rtsp-client.h:
8606 * gst/rtsp-server/rtsp-media.c:
8607 * gst/rtsp-server/rtsp-media.h:
8608 * gst/rtsp-server/rtsp-sdp.c:
8609 * gst/rtsp-server/rtsp-sdp.h:
8610 * gst/rtsp-server/rtsp-server.c:
8611 rtsp: keep track of server ip and ipv6
8612 Keep track of how the client connected to the server and setup the udp ports
8613 with the same protocol.
8614 Copy the server ip address in the SDP so that clients can send RTCP back to
8617 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8619 * gst/rtsp-server/rtsp-session.c:
8622 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8624 * gst/rtsp-server/rtsp-client.c:
8625 client: use right size for malloc
8627 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8629 * gst/rtsp-server/rtsp-server.c:
8630 server: comment ipv6 server listening address
8632 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8634 * gst/rtsp-server/rtsp-media.c:
8635 media: allow for ipv6 sockets
8637 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8639 * gst/rtsp-server/rtsp-server.c:
8640 * gst/rtsp-server/rtsp-server.h:
8641 server: rework server part
8642 Allow setting a bind address, make sure we can deal with ipv6.
8643 Remove the port property and change with the service property.
8645 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8647 * gst/rtsp-server/rtsp-media.h:
8648 media: update comments a little
8650 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8652 * gst/rtsp-server/rtsp-client.c:
8653 client: make content-base better
8654 Use the URI formatting functions to make a content-base. Also make sure that
8655 there is a trailing / at the end.
8657 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8659 * gst/rtsp-server/rtsp-client.c:
8660 client: guard against invalid paths
8662 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8664 * examples/test-video.c:
8665 test: catch server bind errors
8667 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
8669 * gst/rtsp-server/rtsp-media.c:
8670 rtspmedia: emit "unprepared" if _prepare fails.
8671 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
8672 media object is removed from its factory's cache.
8674 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8676 * gst/rtsp-server/rtsp-media.c:
8677 media: collect media position when seek completes
8679 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
8681 * gst/rtsp-server/rtsp-client.c:
8682 client: call unlink_streams in client finalize
8685 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8687 * gst/rtsp-server/rtsp-media.c:
8688 media: limit the time to wait to something huge
8689 Avoid waiting forever but limit the timeout to 20 seconds.
8691 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8693 * gst/rtsp-server/rtsp-sdp.c:
8694 sdp: reindent and check for prepared status
8696 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8698 * gst/rtsp-server/rtsp-media.c:
8699 * gst/rtsp-server/rtsp-media.h:
8700 * gst/rtsp-server/rtsp-session.c:
8701 media: avoid doing _get_state() for state changes
8702 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
8703 until the media is prerolled or in error. This avoids doing a blocking call of
8704 gst_element_get_state() that can cause lockups when there is an error.
8707 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8709 * gst/rtsp-server/rtsp-media.c:
8712 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8714 * gst/rtsp-server/rtsp-media-factory.c:
8715 media-factory: better error handling
8716 Improve the error handling a bit.
8718 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8720 * gst/rtsp-server/rtsp-client.c:
8721 client: rework transport parsing
8722 Rework the transport parsing code so that we can ignore transports we don't
8723 support instead of just picking the first one we can parse.
8724 Configure a (for now hardcoded) destination for multicast transports.
8726 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8728 * gst/rtsp-server/rtsp-media.c:
8729 media: set multicast sink parameters
8730 Disable loop and automatic multicast join on the udpsink elements.
8731 Add some more debug info.
8732 Reset some state variables in the right place.
8733 Use the right port numbers for multicast.
8735 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8737 * gst/rtsp-server/rtsp-session.c:
8738 session: handle transport setup correctly
8739 Handle UDP, MCAST and TCP transport negotiation more correctly.
8740 Store the server session SSRC in the transport.
8742 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8744 * gst/rtsp-server/rtsp-client.c:
8745 rtsp-client: implement error_full
8746 Implement error_full to avoid some segfaults when the rtspconnection calls it.
8749 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8752 * gst/rtsp-server/rtsp-client.c:
8753 * gst/rtsp-server/rtsp-server.c:
8754 docs: update docs and comments
8756 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
8758 * gst/rtsp-server/rtsp-sdp.c:
8759 sdp: make server work better when behind a proxy
8761 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8763 * gst/rtsp-server/rtsp-client.c:
8764 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
8766 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8768 * gst/rtsp-server/rtsp-client.c:
8769 * gst/rtsp-server/rtsp-media-factory.c:
8770 * gst/rtsp-server/rtsp-media-mapping.c:
8771 * gst/rtsp-server/rtsp-media.c:
8772 * gst/rtsp-server/rtsp-server.c:
8773 * gst/rtsp-server/rtsp-session-pool.c:
8774 * gst/rtsp-server/rtsp-session.c:
8775 Use GStreamer's debugging subsystem
8777 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8779 * gst/rtsp-server/rtsp-media-factory.c:
8780 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
8782 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8787 === release 0.10.5 ===
8789 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8794 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8797 configure: bump required versions
8799 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
8801 * gst/rtsp-server/rtsp-client.c:
8802 client: call weak-unref on client->sessions from finalize
8805 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8807 * gst/rtsp-server/rtsp-media.c:
8808 media: Fixed crasher where caps got unref'ed too often
8810 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8813 * pkgconfig/.gitignore:
8814 * pkgconfig/Makefile.am:
8815 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
8816 Added pkg-config file to use gst-rtsp-server uninstalled
8818 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8820 * gst/rtsp-server/rtsp-media.c:
8821 media: add some docs
8823 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
8825 * gst/rtsp-server/rtsp-client.c:
8826 rtsp: Use gst_rtsp_watch_send_message().
8827 Use gst_rtsp_watch_send_message() since the old API which used
8828 gst_rtsp_watch_queue_message() has been deprecated.
8830 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8835 === release 0.10.4 ===
8837 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8842 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8844 * gst/rtsp-server/rtsp-client.c:
8845 * gst/rtsp-server/rtsp-session.c:
8846 * gst/rtsp-server/rtsp-session.h:
8847 rtsp: allocate channels in TCP mode
8848 When the client does not provide us with channels in TCP mode, allocate channels
8851 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8853 * gst/rtsp-server/rtsp-client.c:
8854 client: don't crash when tunnelid is missing
8855 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
8856 don't crash but return an error response to the client.
8859 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8861 * bindings/vala/gst-rtsp-server-0.10.vapi:
8862 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8863 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8864 bindings: update vala bindings with new method
8866 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8868 * gst/rtsp-server/rtsp-session-pool.c:
8869 * gst/rtsp-server/rtsp-session-pool.h:
8870 sessionpool: add function to filter sessions
8871 Add generic function to retrieve/remove sessions.
8873 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8876 configure: bump core/base requirements to release
8878 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8880 * gst/rtsp-server/rtsp-media.c:
8881 media: fix indentation
8883 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8885 * gst/rtsp-server/rtsp-media.c:
8886 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
8888 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8890 * gst/rtsp-server/rtsp-media.c:
8891 set state and remove elements of media in for loop
8893 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
8895 * bindings/vala/gst-rtsp-server-0.10.vapi:
8896 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8897 Added gst_rtsp_media_remove_elements function to Vala bindings
8899 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
8901 * gst/rtsp-server/rtsp-media.c:
8902 * gst/rtsp-server/rtsp-media.h:
8903 Added gst_rtsp_media_remove_elements function
8905 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
8907 * gst/rtsp-server/rtsp-media.c:
8908 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
8910 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8912 * bindings/vala/gst-rtsp-server-0.10.vapi:
8913 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8914 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8915 Updated Vala bindings
8917 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8919 * gst/rtsp-server/rtsp-media.c:
8920 * gst/rtsp-server/rtsp-media.h:
8921 Added vmethod unprepare to GstRTSPMedia
8922 The default implementation sets the state of the pipeline to GST_STATE_NULL
8924 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8926 * gst/rtsp-server/rtsp-media-factory.c:
8927 * gst/rtsp-server/rtsp-media-factory.h:
8928 Made collect_streams function public
8930 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8932 * gst/rtsp-server/rtsp-media-factory.c:
8933 * gst/rtsp-server/rtsp-media-factory.h:
8934 * gst/rtsp-server/rtsp-media.c:
8935 Added vmethod create_pipeline to GstRTSPMediaFactory
8936 The pipeline is created in this method and the GstRTSPMedia's element is added to it
8938 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8940 * gst/rtsp-server/rtsp-client.c:
8941 client: use g_source_destroy()
8942 We need to use g_source_destroy() because we might have added the source to a
8943 different main context than the default one.
8945 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8947 * gst/rtsp-server/Makefile.am:
8948 * gst/rtsp-server/rtsp-client.c:
8949 * gst/rtsp-server/rtsp-params.c:
8950 * gst/rtsp-server/rtsp-params.h:
8951 rtsp: prepare for handling GET/SET_PARAMETER
8952 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
8954 Fix return codes of handlers.
8956 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8958 * gst/rtsp-server/rtsp-media.c:
8959 media: don't leak session pads
8961 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8963 * gst/rtsp-server/rtsp-media.c:
8964 media: clean up the messages a bit
8966 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8968 * gst/rtsp-server/rtsp-sdp.c:
8969 sdp: warn and skip streams without media
8971 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8973 * bindings/vala/gst-rtsp-server-0.10.vapi:
8974 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8975 vala: Fixed typo in header file of RTSPMediaStream
8977 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8979 * gst/rtsp-server/rtsp-media.c:
8982 Make dumping RTCP stats configurable
8984 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8986 * gst/rtsp-server/rtsp-media.c:
8987 media: be less verbose and leak less
8989 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8991 * gst/rtsp-server/rtsp-media.c:
8992 media: don't leak the destination address
8994 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8996 * gst/rtsp-server/rtsp-client.c:
8997 * gst/rtsp-server/rtsp-media.c:
8998 * gst/rtsp-server/rtsp-media.h:
8999 * gst/rtsp-server/rtsp-session.c:
9000 * gst/rtsp-server/rtsp-session.h:
9001 rtsp: use RTCP to keep the session alive
9002 Use the RTCP rtcp-from stats field to find the associated session and use this
9003 to keep the session alive.
9005 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9007 * gst/rtsp-server/rtsp-session.c:
9008 session: add 5sec to the real session timeout
9009 Allow the session to live 5sec longer before really timing out. This should give
9010 clients some extra time to keep the session active.
9012 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9014 * gst/rtsp-server/rtsp-client.c:
9015 client: replay OK to GET/SET_PARAMETER
9016 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
9017 so that we return OK for those requests.
9019 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9021 * gst/rtsp-server/rtsp-media.c:
9022 * gst/rtsp-server/rtsp-media.h:
9023 media: keep track of active transports
9024 Keep track of which transport is active to avoid closing the connection too
9026 Remove the destination transport also when going to NULL.
9027 Print some stats about the SDES and other RTCP messages we receive from the
9030 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9032 * examples/.gitignore:
9033 * examples/Makefile.am:
9034 * examples/test-sdp.c:
9035 example: add SDP relay example
9037 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9039 * gst/rtsp-server/rtsp-media.c:
9040 media: also count active TCP connections
9042 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9044 * gst/rtsp-server/rtsp-media-factory.c:
9045 * gst/rtsp-server/rtsp-media.c:
9046 * gst/rtsp-server/rtsp-media.h:
9047 rtsp: add support for dynamic elements
9048 Add support for dynamic elements.
9049 Don't set live pipelines back to paused.
9051 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9053 * gst/rtsp-server/rtsp-sdp.c:
9054 sdp: don't add encoding name when absent in caps
9056 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9058 * gst/rtsp-server/rtsp-client.c:
9059 client: warn when we can't do RTP-Info
9061 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9063 * gst/rtsp-server/rtsp-media-factory.c:
9064 factory: factor out the stream construction
9066 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9068 * gst/rtsp-server/rtsp-client.c:
9069 client: only add RTP-Info when we have the info
9070 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
9073 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9078 === release 0.10.3 ===
9080 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9084 - Fixes a bug where it put the wrong verion in pkgconfig
9085 - Link RTP and RTCP sources
9087 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9089 * gst/rtsp-server/rtsp-media.c:
9090 * gst/rtsp-server/rtsp-media.h:
9091 media: link the RTP udpsrc to the session manager
9092 Link the RTP udpsrc and the appsrc to the session manager so that they don't
9093 shut down when the client sends a packet to open firewalls.
9095 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9097 * pkgconfig/gst-rtsp-server.pc.in:
9098 Don't use hard-coded version number in pkg-config file
9100 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9105 === release 0.10.2 ===
9107 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9112 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9115 * common/m4/.gitignore:
9116 * examples/.gitignore:
9117 * pkgconfig/.gitignore:
9118 add some .gitignore files
9120 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9122 * gst/rtsp-server/rtsp-media.c:
9123 media: seek to key frames
9125 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9127 * gst/rtsp-server/rtsp-media.c:
9128 media: emit the unprepared signal by id
9129 Emit the unprepared signal by id instead of name and set the media as
9132 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9134 * gst/rtsp-server/rtsp-media.c:
9135 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
9137 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9139 * gst/rtsp-server/rtsp-server.c:
9140 Added finalize function to GstRTPSPServer to unref session pool and media mapping
9142 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9144 * bindings/vala/gst-rtsp-server-0.10.vapi:
9145 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9146 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9147 Updated vala bindings
9149 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9151 * gst/rtsp-server/Makefile.am:
9152 * gst/rtsp-server/rtsp-client.c:
9153 * gst/rtsp-server/rtsp-media.c:
9154 server: use appsink and appsrc with the API
9155 Use the appsink/appsrc API instead of the signals for higher
9158 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9160 * examples/test-ogg.c:
9161 tests: set the payload type correctly
9163 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9165 * gst/rtsp-server/rtsp-media-factory.c:
9166 factory: connect to the unprepare signal
9167 Connect to the unprepare signal for non-reusable media so that we can remove
9168 them from the cache.
9170 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9172 * gst/rtsp-server/rtsp-media.c:
9173 * gst/rtsp-server/rtsp-media.h:
9174 media: add signal to notify of unprepare
9176 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9178 * gst/rtsp-server/rtsp-media.c:
9179 * gst/rtsp-server/rtsp-media.h:
9180 media: more work on making the media shared
9181 Add a reusable flag to medias, indicating that they can be reused after a state
9185 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9187 * examples/test-readme.c:
9188 examples: mark the example as shared for testing
9190 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9192 * gst/rtsp-server/rtsp-media.c:
9193 * gst/rtsp-server/rtsp-media.h:
9194 client: support shared media
9195 Always perform the state actions even if the target state of the pipeline is
9196 already correct, we still want to add/remove the transports when we are dealing
9198 Keep a counter of the number of active transports for a media so that we can use
9199 this to perform a state change when needed.
9200 Perform a state change of the pipeline only when the first transport was added
9201 or when there are no active transports.
9203 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9205 * gst/rtsp-server/rtsp-client.c:
9206 client: fix refcounting crasher
9207 Don't need to remove the weak refs in the finalize methods, they are already
9208 removed in the dispose.
9209 Don't register the callback with a DestroyNofity.
9211 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9213 * gst/rtsp-server/rtsp-client.c:
9214 Fix rtsp client refcount management in TCP mode.
9215 Don't unref a client ref we never had. Fixes an unref
9216 of an already-free client object after a client
9217 teardown request for me.
9219 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9221 * gst/rtsp-server/rtsp-session.c:
9222 docs: fix typo in API docs
9224 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9226 * gst/rtsp-server/rtsp-media.c:
9228 Keep the udp sources in playing even if we go to paused. unlock the sources when
9230 Add some more debug info.
9231 Only seek when we need to.
9232 Keep track of the position when we go to paused.
9234 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9236 * gst/rtsp-server/rtsp-client.c:
9237 * gst/rtsp-server/rtsp-media.c:
9238 * gst/rtsp-server/rtsp-media.h:
9239 Add beginnings of seeking.
9240 Parse the Range header and perform a seek on the pipeline for the requested
9241 position. It's disabled currently until I figure out what's going wrong.
9243 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9245 * gst/rtsp-server/rtsp-client.c:
9246 allow pause requests for now.
9249 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9251 * gst/rtsp-server/rtsp-client.c:
9252 Remove weak ref on the session in teardown
9253 We need to remove our weakref from the session when we do a teardown because
9254 else we close the TCP connection prematurely.
9256 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9258 * gst/rtsp-server/rtsp-client.c:
9259 * gst/rtsp-server/rtsp-client.h:
9260 * gst/rtsp-server/rtsp-session-pool.c:
9261 Do some more session cleanup
9262 Make session timeout kill the TCP connection that currently watches the
9264 Remove the client timeout property.
9266 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9268 * gst/rtsp-server/rtsp-client.c:
9269 * gst/rtsp-server/rtsp-client.h:
9270 * gst/rtsp-server/rtsp-media.c:
9271 * gst/rtsp-server/rtsp-media.h:
9272 * gst/rtsp-server/rtsp-server.c:
9273 * gst/rtsp-server/rtsp-session.c:
9274 * gst/rtsp-server/rtsp-session.h:
9276 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
9279 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9281 * examples/Makefile.am:
9282 * examples/test-launch.c:
9283 Add example server that takes launch lines
9284 Add an example server that streams any -launch line.
9286 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9288 * examples/test-readme.c:
9289 * gst/rtsp-server/rtsp-client.c:
9290 * gst/rtsp-server/rtsp-media.c:
9291 * gst/rtsp-server/rtsp-media.h:
9292 Add support for live streams
9293 Add support for live streams and ranges
9294 Start on handling TCP data transfer.
9296 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9298 * gst/rtsp-server/rtsp-media.c:
9299 Free the pipeline before other things
9302 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9304 * gst/rtsp-server/rtsp-client.c:
9305 Only free the pending tunnel if there is one
9308 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9310 * gst/rtsp-server/rtsp-client.c:
9311 * gst/rtsp-server/rtsp-client.h:
9312 * gst/rtsp-server/rtsp-media.c:
9313 rtsp-server: Add support for tunneling
9314 Add support for tunneling over HTTP.
9315 Use new connection methods to retrieve the url.
9316 Dispatch messages based on the message type instead of blindly
9317 assuming it's always a request.
9318 Keep track of the watch id so that we can remove it later.
9319 Set the media pipeline to NULL before unreffing the pipeline.
9321 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9323 * gst/rtsp-server/rtsp-client.c:
9324 * gst/rtsp-server/rtsp-client.h:
9325 Fix for channel -> watch rename in gstreamer
9326 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
9328 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9330 * gst/rtsp-server/rtsp-client.c:
9331 * gst/rtsp-server/rtsp-client.h:
9333 Use the async RTSP channels instead of spawning a new thread for each client.
9334 If a sessionid is specified in a request, fail if we don't have the session.
9336 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9338 * gst/rtsp-server/rtsp-media.c:
9339 Add better debug info
9340 Add some better debug info.
9342 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9344 * examples/test-video.c:
9346 Add support for session timeouts in the example.
9348 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9350 * gst/rtsp-server/rtsp-session-pool.c:
9351 * gst/rtsp-server/rtsp-session-pool.h:
9352 Pass GTimeVal around for performance reasons
9353 Get the current time only once and pass it around so that sessions don't have to
9354 get the current time anymore.
9355 Add experimental support for a GSource that dispatches when the session needs to
9358 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9360 * gst/rtsp-server/rtsp-session.c:
9361 * gst/rtsp-server/rtsp-session.h:
9362 Add better support for session timeouts
9363 Add a method to request the number of milliseconds when a session will timeout.
9365 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9367 * gst/rtsp-server/rtsp-media.c:
9368 * gst/rtsp-server/rtsp-media.h:
9369 Add suport for RTP manager monitoring
9370 Add the first stage in monitoring the rtp manager.
9371 Make sure we don't update the state to something we don't want.
9373 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9375 * gst/rtsp-server/rtsp-client.c:
9376 Add support for session keepalive
9377 Get and update the session timeout for all requests. get the session as early as
9380 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9382 * gst/rtsp-server/rtsp-media-factory.h:
9383 * gst/rtsp-server/rtsp-media.c:
9384 * gst/rtsp-server/rtsp-media.h:
9385 Handle media bus messages
9386 Handle media bus messages in a custom mainloop and dispatch them to the
9387 RTSPMedia objects. Let the default implementation handle some common messages.
9389 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9391 * gst/rtsp-server/rtsp-client.c:
9392 * gst/rtsp-server/rtsp-session-pool.c:
9393 * gst/rtsp-server/rtsp-session.c:
9394 Some more session timeout handling
9395 Move the session header setting code to a central place so that we always add
9396 the timeout parameter too.
9397 Handle timeouts by running the session cleanup code.
9398 Stop media before cleaning up.
9400 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9402 * gst/rtsp-server/rtsp-client.c:
9403 * gst/rtsp-server/rtsp-client.h:
9404 Add timeout property
9405 Add a timeout property ot the client and make the other properties into GObject
9408 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9410 * gst/rtsp-server/rtsp-session-pool.c:
9411 Use getters and setters in property code
9412 Use the getters and setters for the timeout property instead of locking
9415 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9417 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
9419 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9421 * gst/rtsp-server/rtsp-session-pool.c:
9422 * gst/rtsp-server/rtsp-session-pool.h:
9423 * gst/rtsp-server/rtsp-session.c:
9424 * gst/rtsp-server/rtsp-session.h:
9425 Add more timeout stuff
9426 Add method to check if a session is expired.
9427 Add method to perform cleanup on a session pool.
9429 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9431 * gst/rtsp-server/rtsp-client.c:
9432 * gst/rtsp-server/rtsp-session-pool.c:
9433 * gst/rtsp-server/rtsp-session-pool.h:
9434 * gst/rtsp-server/rtsp-session.c:
9435 * gst/rtsp-server/rtsp-session.h:
9436 Add beginnings of session timeouts and limits
9437 Add the timeout value to the Session header for unusual timeout values.
9438 Allow us to configure a limit to the amount of active sessions in a pool. Set a
9439 limit on the amount of retry we do after a sessionid collision.
9440 Add properties to the sessionid and the timeout of a session. Keep track of
9441 creation time and last access time for sessions.
9443 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9445 * gst/rtsp-server/rtsp-client.c:
9446 * gst/rtsp-server/rtsp-media.c:
9447 * gst/rtsp-server/rtsp-media.h:
9448 * gst/rtsp-server/rtsp-sdp.c:
9449 * gst/rtsp-server/rtsp-session-pool.c:
9450 * gst/rtsp-server/rtsp-session.c:
9451 * gst/rtsp-server/rtsp-session.h:
9452 Cleanup of sessions and more
9453 Fix the refcounting of media and sessions in the client. Properly clean up the
9454 session data when the client performs a teardown.
9455 Add Server header to responses.
9456 Allow for multiple uri setups in one session.
9457 Add Range header to the PLAY response and add the range attribute to the SDP
9459 Fix the session pool remove method, it used the wrong key in the hashtable. Also
9460 give the ownership of the sessionid to the session object.
9462 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9464 * gst/rtsp-server/rtsp-server.c:
9465 * gst/rtsp-server/rtsp-server.h:
9467 Rename the 'server_port' variable to simply 'port'.
9469 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9472 * gst/rtsp-server/rtsp-client.c:
9473 * gst/rtsp-server/rtsp-media.c:
9474 * gst/rtsp-server/rtsp-media.h:
9475 * gst/rtsp-server/rtsp-session.c:
9476 * gst/rtsp-server/rtsp-session.h:
9477 Rework the way we handle transports for streams
9478 Make the media accept an array of transports for the streams that we have
9479 configured for the play/pause requests.
9480 Implement server states for a client and its media.
9481 Require 0.10.22.1 (git HEAD) of gstreamer.
9483 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9485 * gst/rtsp-server/rtsp-client.c:
9486 * gst/rtsp-server/rtsp-media-factory.c:
9487 Drop const from functions dealing with urls
9488 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
9489 have the right const in them.
9491 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9493 * gst/rtsp-server/rtsp-client.c:
9494 * gst/rtsp-server/rtsp-media.c:
9495 * gst/rtsp-server/rtsp-sdp.c:
9499 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9501 * gst/rtsp-server/rtsp-client.c:
9502 * gst/rtsp-server/rtsp-media-factory.c:
9503 * gst/rtsp-server/rtsp-media.c:
9504 * gst/rtsp-server/rtsp-media.h:
9506 Don't keep a reference to the GstRTSPMedia in the stream.
9507 Free more things when freeing the GstRTSPMedia.
9509 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9512 * gst/rtsp-server/rtsp-media-factory.c:
9513 * gst/rtsp-server/rtsp-media-factory.h:
9514 * gst/rtsp-server/rtsp-media.c:
9515 * gst/rtsp-server/rtsp-media.h:
9516 * gst/rtsp-server/rtsp-server.c:
9517 * gst/rtsp-server/rtsp-server.h:
9518 More docs and small cleanups
9519 Add some more docs and update the README
9520 Cleanup some method names.
9521 Remove an unneeded idx field in the GstRTSPMediaStream
9523 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9526 * examples/Makefile.am:
9527 * examples/test-readme.c:
9528 Add a README and more example code
9529 Add a README file that contains a small introduction on how to use the server
9530 along with the example code explained in the readme.
9532 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9534 * gst/rtsp-server/rtsp-media.c:
9535 * gst/rtsp-server/rtsp-server.c:
9536 Fix some leaks and change default port
9537 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
9538 we finished the initial preroll. If we keep them locked, setting the pipeline to
9539 NULL will not stop and clean up the sources correctly.
9540 Change the default RTSP port to 8554 aka the official alternative RTSP port.
9542 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9544 * gst/rtsp-server/rtsp-session.c:
9545 * gst/rtsp-server/rtsp-session.h:
9546 Cleanups to the session object
9547 Remove some unneeded variables in the session state of a stream such as the
9548 owner media and the server transport.
9549 Get the configuration of a media stream in a session based on the media_stream
9550 in the original object instead of our cached index.
9551 Free more data in the finalize method.
9553 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9555 * gst/rtsp-server/rtsp-client.c:
9556 * gst/rtsp-server/rtsp-client.h:
9557 Cleanups and reuse media from DESCRIBE
9558 Handle thread create errors.
9559 Rename some internal methods to better match what they actually do.
9560 Handle misconfiguration of session_pool and media_mapping gracefully.
9561 Cache the DESCRIBE media and uri in the client connection and reuse them when
9562 we receive a SETUP request in the same connection for the same uri.
9563 Cleanup the client connection object.
9565 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9567 * gst/rtsp-server/rtsp-media-factory.c:
9568 * gst/rtsp-server/rtsp-media-factory.h:
9569 * gst/rtsp-server/rtsp-media.c:
9570 * gst/rtsp-server/rtsp-media.h:
9571 Add shared properties to media and factory
9572 Add the shared property to media.
9573 Implement some simple caching in the factory depending on if the media is shared
9576 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9578 * gst/rtsp-server/rtsp-client.c:
9579 Add a little comment
9580 Add some comment about the content-base header.
9582 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9584 * examples/Makefile.am:
9585 * examples/test-mp4.c:
9586 * examples/test-ogg.c:
9587 * examples/test-video.c:
9588 * gst/rtsp-server/Makefile.am:
9589 * gst/rtsp-server/rtsp-client.c:
9590 * gst/rtsp-server/rtsp-client.h:
9591 * gst/rtsp-server/rtsp-media-factory.c:
9592 * gst/rtsp-server/rtsp-media-factory.h:
9593 * gst/rtsp-server/rtsp-media.c:
9594 * gst/rtsp-server/rtsp-media.h:
9595 * gst/rtsp-server/rtsp-sdp.c:
9596 * gst/rtsp-server/rtsp-sdp.h:
9597 * gst/rtsp-server/rtsp-server.c:
9598 * gst/rtsp-server/rtsp-server.h:
9599 * gst/rtsp-server/rtsp-session.c:
9600 * gst/rtsp-server/rtsp-session.h:
9601 Reorganize things, prepare for media sharing
9602 Added various other test server examples
9603 Move the SDP message generation to a separate helper.
9604 Refactor common code for finding the session.
9605 Add content-base for realplayer compatibility
9606 Clean up request uris before processing for better vlc compatibility.
9607 Move prerolling and pipeline construction to the RTSPMedia object.
9608 Use multiudpsink for future pipeline reuse.
9610 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9616 === release 0.10.1 ===
9618 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9624 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9626 * bindings/vala/Makefile.am:
9628 Add more directories and files to the dist.
9630 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9632 * bindings/python/Makefile.am:
9633 * bindings/python/rtspserver.override:
9634 Fixed compile error of python bindings
9636 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9638 * bindings/vala/gst-rtsp-server-0.10.vapi:
9639 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9640 Marked values as nullable accordingly
9642 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9644 * bindings/vala/gst-rtsp-server-0.10.vapi:
9645 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
9646 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9647 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9648 Updated Vala bindings
9650 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9652 * gst/rtsp-server/rtsp-client.c:
9653 * gst/rtsp-server/rtsp-media-mapping.c:
9654 * gst/rtsp-server/rtsp-media-mapping.h:
9655 * gst/rtsp-server/rtsp-media.h:
9656 * gst/rtsp-server/rtsp-session-pool.h:
9657 Cleanups and doc updates
9658 Add some more documentation and do some minor cleanups here and there.
9660 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9662 * gst/rtsp-server/rtsp-client.c:
9663 * gst/rtsp-server/rtsp-media-factory.c:
9664 * gst/rtsp-server/rtsp-media-factory.h:
9665 * gst/rtsp-server/rtsp-media.c:
9666 * gst/rtsp-server/rtsp-media.h:
9667 * gst/rtsp-server/rtsp-session.c:
9668 * gst/rtsp-server/rtsp-session.h:
9670 Rename GstRTSPMediaBin to GstRTSPMedia
9671 Parse the request url into a GstRTSPUri object and pass this object to the
9672 various handlers and methods that require the uri.
9674 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9678 Add some more docs and remove some old code from the example.
9680 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9682 * gst/rtsp-server/rtsp-client.c:
9683 Handle state change failures better
9684 Handle state change failures better when changing the state of the pipeline to
9687 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9689 * gst/rtsp-server/rtsp-media-factory.c:
9690 * gst/rtsp-server/rtsp-media-factory.h:
9691 Make element creation more extendible
9692 Add get_element vmethod to the default MediaFactory so that subclasses can just
9693 override that method and still use the default logic for making a MediaBin from
9696 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9699 * gst/rtsp-server/Makefile.am:
9700 * gst/rtsp-server/rtsp-client.c:
9701 * gst/rtsp-server/rtsp-client.h:
9702 * gst/rtsp-server/rtsp-media-factory.c:
9703 * gst/rtsp-server/rtsp-media-factory.h:
9704 * gst/rtsp-server/rtsp-media-mapping.c:
9705 * gst/rtsp-server/rtsp-media-mapping.h:
9706 * gst/rtsp-server/rtsp-media.c:
9707 * gst/rtsp-server/rtsp-media.h:
9708 * gst/rtsp-server/rtsp-server.c:
9709 * gst/rtsp-server/rtsp-server.h:
9710 * gst/rtsp-server/rtsp-session.c:
9711 * gst/rtsp-server/rtsp-session.h:
9712 Make the server handle arbitrary pipelines
9713 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
9714 The GstMediaBin object has a handle to a bin with elements and to a list of
9715 GstMediaStream objects that this bin produces.
9716 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
9717 with methods to register and remove those mappings.
9718 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
9719 used by the server instance.
9720 Modify the example application so that it shows how to create custom pipelines
9721 attached to a specific mount point.
9722 Various misc cleanps.
9724 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9726 * gst/rtsp-server/rtsp-server.c:
9727 * gst/rtsp-server/rtsp-server.h:
9728 Allow setting a custom media factory for a server
9730 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9732 * gst/rtsp-server/rtsp-client.c:
9733 * gst/rtsp-server/rtsp-client.h:
9734 Allow setting a custom media factory for a client.
9736 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9738 * gst/rtsp-server/Makefile.am:
9739 Add Makefile entry for the media factory
9741 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9743 * gst/rtsp-server/rtsp-media-factory.c:
9744 * gst/rtsp-server/rtsp-media-factory.h:
9745 Add media factory to map urls to media pipeline objects.
9747 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9749 * gst/rtsp-server/rtsp-media.c:
9750 * gst/rtsp-server/rtsp-media.h:
9751 Add comments. Remove unused field
9753 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9755 * gst/rtsp-server/rtsp-session-pool.c:
9756 * gst/rtsp-server/rtsp-session-pool.h:
9757 Allow custom session pools to override the session id allocation algorithms Add some comments.
9759 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9761 * gst/rtsp-server/rtsp-session.h:
9764 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9766 * gst/rtsp-server/rtsp-client.c:
9767 * gst/rtsp-server/rtsp-client.h:
9768 Move the connection code in one place Add some comments
9770 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9772 * gst/rtsp-server/rtsp-server.c:
9773 * gst/rtsp-server/rtsp-server.h:
9774 Make vmethod to create and accept new clients. Add some docs.
9776 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9778 * gst/rtsp-server/rtsp-server.c:
9779 * gst/rtsp-server/rtsp-server.h:
9780 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
9782 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9784 * gst/rtsp-server/rtsp-client.c:
9785 * gst/rtsp-server/rtsp-client.h:
9786 Name the parameters more appropriately.
9788 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9790 * gst/rtsp-server/rtsp-session-pool.c:
9791 Do some more cleanup of the session pool.
9793 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9795 * gst/rtsp-server/Makefile.am:
9796 * gst/rtsp-server/rtsp-client.c:
9797 Check if return value of gst_rtsp_session_get_media is not NULL
9799 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9801 * gst/rtsp-server/Makefile.am:
9802 Install rtsp-session and rtsp-session-pool headers
9804 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9809 * bindings/python/Makefile.am:
9810 * bindings/python/arg-types.py:
9811 * bindings/python/codegen/Makefile.am:
9812 * bindings/python/codegen/__init__.py:
9813 * bindings/python/codegen/argtypes.py:
9814 * bindings/python/codegen/code-coverage.py:
9815 * bindings/python/codegen/codegen.py:
9816 * bindings/python/codegen/definitions.py:
9817 * bindings/python/codegen/defsparser.py:
9818 * bindings/python/codegen/docextract.py:
9819 * bindings/python/codegen/docgen.py:
9820 * bindings/python/codegen/fileprefix.override:
9821 * bindings/python/codegen/fileprefixmodule.c:
9822 * bindings/python/codegen/h2def.py:
9823 * bindings/python/codegen/mergedefs.py:
9824 * bindings/python/codegen/mkskel.py:
9825 * bindings/python/codegen/override.py:
9826 * bindings/python/codegen/reversewrapper.py:
9827 * bindings/python/codegen/scmexpr.py:
9828 * bindings/python/rtspserver-types.defs:
9829 * bindings/python/rtspserver.defs:
9830 * bindings/python/rtspserver.override:
9831 * bindings/python/rtspservermodule.c:
9833 Add python bindings.
9835 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9837 * bindings/Makefile.am:
9839 Don't go into python dir when requirements for python bindings are missing
9841 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9843 * bindings/Makefile.am:
9844 * bindings/vala/Makefile.am:
9846 Install Vala bindings if vala is available
9848 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9850 * bindings/vala/gst-rtsp-server-0.10.deps:
9851 * bindings/vala/gst-rtsp-server-0.10.vapi:
9852 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
9853 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
9854 * bindings/vala/packages/gst-rtsp-server-0.10.files:
9855 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9856 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9857 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
9858 Regenerated Vala bindings
9860 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9862 * bindings/vala/gst-rtsp-server.vapi:
9863 * bindings/vala/packages/gst-rtsp-server.metadata:
9864 Fixed typo in included headers for vala bindings
9866 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9870 * pkgconfig/Makefile.am:
9871 * pkgconfig/gst-rtsp-server.pc.in:
9872 Added pkgconfig file
9874 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
9876 * bindings/vala/gst-rtsp-server.vapi:
9877 * bindings/vala/packages/gst-rtsp-server.excludes:
9878 * bindings/vala/packages/gst-rtsp-server.gi:
9879 * bindings/vala/packages/gst-rtsp-server.metadata:
9880 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
9882 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
9884 * bindings/vala/gst-rtsp-server.vapi:
9885 * bindings/vala/packages/gst-rtsp-server.deps:
9886 * bindings/vala/packages/gst-rtsp-server.files:
9887 * bindings/vala/packages/gst-rtsp-server.gi:
9888 * bindings/vala/packages/gst-rtsp-server.metadata:
9889 * bindings/vala/packages/gst-rtsp-server.namespace:
9892 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
9894 * gst/rtsp-server/rtsp-session.c:
9895 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
9897 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
9899 * examples/Makefile.am:
9900 * gst/rtsp-server/Makefile.am:
9901 Put GStreamer version in library name
9903 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9905 * examples/Makefile.am:
9906 * gst/rtsp-server/Makefile.am:
9907 Fix some issues to pass distcheck
9909 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9911 * gst/rtsp-server/rtsp-server.c:
9912 Added port property to GstRTSPServer class.
9914 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9919 * examples/Makefile.am:
9922 * gst/rtsp-server/Makefile.am:
9923 * gst/rtsp-server/rtsp-client.c:
9924 * gst/rtsp-server/rtsp-client.h:
9925 * gst/rtsp-server/rtsp-media.c:
9926 * gst/rtsp-server/rtsp-media.h:
9927 * gst/rtsp-server/rtsp-server.c:
9928 * gst/rtsp-server/rtsp-server.h:
9929 * gst/rtsp-server/rtsp-session-pool.c:
9930 * gst/rtsp-server/rtsp-session-pool.h:
9931 * gst/rtsp-server/rtsp-session.c:
9932 * gst/rtsp-server/rtsp-session.h:
9934 Split in library and example program
9936 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
9938 * src/rtsp-client.h:
9939 Removed obsolete variable
9941 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
9943 * src/rtsp-client.c:
9944 * src/rtsp-client.h:
9945 Removed pipeline variable GstRTSPClient, because it's only used in one function
9947 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9950 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
9952 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
9954 * src/rtsp-session.c:
9955 Initialize some more vars.
9957 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
9959 * src/rtsp-session.c:
9960 Initialize variable to avoid compiler warning.
9962 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
9965 Add a reasonable generic .gitignore