3 2016-09-30 Sebastian Dröge <slomo@coaxion.net>
8 2016-09-17 13:17:19 +0100 Ian Jamison <ian.dev@arkver.com>
10 * gst/rtsp-server/rtsp-media-factory.c:
11 * gst/rtsp-server/rtsp-media.c:
12 * gst/rtsp-server/rtsp-stream.c:
13 rtsp-server: Hint that set_multicast_iface expects the name of the interface
14 To prevent any possibly confusion with IPs or anything else.
15 https://bugzilla.gnome.org/show_bug.cgi?id=771530
17 2016-09-18 09:58:55 -0400 Sebastian Dröge <sebastian@centricular.com>
19 * gst/rtsp-server/rtsp-media-factory.c:
20 * gst/rtsp-server/rtsp-media.c:
21 rtsp-media: Call g_free() instead of g_object_unref() on multicast-iface strings
22 https://bugzilla.gnome.org/show_bug.cgi?id=763000#c5
24 2016-09-14 11:31:15 +0200 Sebastian Dröge <sebastian@centricular.com>
27 configure: Depend on gstreamer 1.9.2.1
29 2016-09-10 20:52:31 +1000 Jan Schmidt <jan@centricular.com>
33 Automatic update of common submodule
34 From b18d820 to f980fd9
36 2016-09-10 09:58:31 +1000 Jan Schmidt <jan@centricular.com>
40 Automatic update of common submodule
41 From 6f2d209 to b18d820
43 2016-09-07 18:44:34 +0300 Sebastian Dröge <sebastian@centricular.com>
45 * gst/rtsp-server/rtsp-stream.c:
46 rtsp-stream: Remove unused _locked() variant of a function
47 It was added during refactoring.
49 2016-09-07 10:21:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
51 * gst/rtsp-server/rtsp-stream.c:
52 stream: cosmetic cleanup
53 https://bugzilla.gnome.org/show_bug.cgi?id=766612
55 2016-09-07 10:16:19 -0400 Xavier Claessens <xavier.claessens@collabora.com>
57 * gst/rtsp-server/rtsp-stream.c:
58 stream: Compare IP addresses case insensitive in more places
59 https://bugzilla.gnome.org/show_bug.cgi?id=766612
61 2016-09-07 10:12:18 -0400 Xavier Claessens <xavier.claessens@collabora.com>
64 * gst/rtsp-server/rtsp-stream.c:
65 stream: Fix leaked joined_bin
66 There is no need to keep a strong ref on it, and _leave_bin() was
67 setting it to NULL before calling g_clear_object() so it was leaked.
68 https://bugzilla.gnome.org/show_bug.cgi?id=766612
70 2016-09-06 19:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
72 * gst/rtsp-server/rtsp-stream.c:
73 rtsp-stream: Compare IP address strings case insensitive
74 Otherwise IPv6 addresses might fail this comparision.
76 2016-09-06 19:10:21 +0300 Sebastian Dröge <sebastian@centricular.com>
78 * gst/rtsp-server/rtsp-stream.c:
79 rtsp-stream: Bind multicast sockets to ANY as before
80 https://bugzilla.gnome.org/show_bug.cgi?id=766612#c48
82 2016-09-05 18:31:36 +0300 Kseniia <vasilchukkseniia@gmail.com>
84 * gst/rtsp-server/rtsp-session.c:
85 rtsp-session: Fix segfault when doing keep-alive after removing the session
86 If keep-alive happens after removing the session but before finalizing the
87 stream transport, we would segfault.
88 https://bugzilla.gnome.org/show_bug.cgi?id=750544
90 2016-09-05 18:04:50 +0300 Sebastian Dröge <sebastian@centricular.com>
92 * gst/rtsp-server/rtsp-stream.c:
93 rtsp-stream: Always create multicast UDP elements if the protocol flag is set
94 Adding them later will cause deadlocks due to
95 1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
96 2) adding the multicast sink
97 3) waiting for it to get data to preroll again
98 3) never happens because the queues after the tee are full.
100 2016-09-05 16:32:57 +0300 Sebastian Dröge <sebastian@centricular.com>
102 * gst/rtsp-server/rtsp-stream.c:
103 rtsp-stream: Fix up various multicast related issues
105 2016-09-05 13:40:59 +0300 Sebastian Dröge <sebastian@centricular.com>
107 * tests/check/gst/stream.c:
108 tests: Fix compilation
110 2016-07-28 15:33:05 -0400 Xavier Claessens <xavier.claessens@collabora.com>
112 * gst/rtsp-server/rtsp-client.c:
113 * gst/rtsp-server/rtsp-stream.c:
114 * tests/check/gst/stream.c:
115 stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast
116 This is basically reverting changes introduced in commit f62a9a7,
117 because it was introducing various regressions:
118 - It introduces a leak of udpsrc elements that got wrongly fixed by adding
119 an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
120 ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
121 - If a mcast client connects, it creates a new socket in SETUP to try to respect
122 the destination/port given by the client in the transport, and overrides the
123 socket already set on the udpsink element. That means that if we already had a
124 client connected, the source address on the udp packets it receives suddenly
126 - If a 2nd mcast client connects, the destination/port in its transport is
127 ignored but its transport wasn't updated.
128 What this patch does:
129 - Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
130 - Always have a tee+queue when udp is enabled. This could be optimized
131 again in a later patch, but is more complicated. If no unicast clients
132 connects then those elements are useless, this could be also optimized
134 - When mcast transport is added, it creates a new set of udpsrc/udpsink,
135 seperated from those for unicast clients. Since we already support only
136 one mcast address, we also create only one set of elements.
137 https://bugzilla.gnome.org/show_bug.cgi?id=766612
139 2016-07-28 15:20:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
141 * gst/rtsp-server/rtsp-stream.c:
142 stream: factor our plug_src function
143 https://bugzilla.gnome.org/show_bug.cgi?id=766612
145 2016-07-21 21:46:16 -0400 Xavier Claessens <xavier.claessens@collabora.com>
147 * gst/rtsp-server/rtsp-stream.c:
148 stream: factor out plug_sink function
149 https://bugzilla.gnome.org/show_bug.cgi?id=766612
151 2016-07-20 23:05:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
153 * gst/rtsp-server/rtsp-stream.c:
154 stream: small documentation clarification
155 https://bugzilla.gnome.org/show_bug.cgi?id=766612
157 2016-07-20 15:35:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
159 * gst/rtsp-server/rtsp-stream.c:
160 stream: rename addr_v4/6 to mcast_addr_v4/6 for clarity
161 https://bugzilla.gnome.org/show_bug.cgi?id=766612
163 2016-07-14 11:10:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
165 * gst/rtsp-server/rtsp-stream.c:
166 stream: Keep a ref on joined bin
167 https://bugzilla.gnome.org/show_bug.cgi?id=766612
169 2016-07-20 15:11:32 -0400 Xavier Claessens <xavier.claessens@collabora.com>
171 * gst/rtsp-server/rtsp-stream.c:
173 https://bugzilla.gnome.org/show_bug.cgi?id=766612
175 2016-07-20 23:18:23 -0400 Xavier Claessens <xavier.claessens@collabora.com>
177 * gst/rtsp-server/rtsp-stream.c:
178 stream: small fix in error code path
179 https://bugzilla.gnome.org/show_bug.cgi?id=766612
181 2016-07-20 20:09:57 -0400 Xavier Claessens <xavier.claessens@collabora.com>
183 * gst/rtsp-server/rtsp-stream.c:
184 Revert "rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc"
185 This partly reverts commit cba045e1b19fad6e689e10206f57903e15f1229a,
186 but keeps unit tests.
187 https://bugzilla.gnome.org/show_bug.cgi?id=766612
189 2016-09-01 12:33:00 +0300 Sebastian Dröge <sebastian@centricular.com>
194 === release 1.9.2 ===
196 2016-09-01 12:32:51 +0300 Sebastian Dröge <sebastian@centricular.com>
202 * gst-rtsp-server.doap:
205 2016-01-27 01:03:52 +0000 Tim-Philipp Müller <tim@centricular.com>
208 * examples/meson.build:
210 * gst/rtsp-server/meson.build:
211 * gst/rtsp-sink/meson.build:
213 * pkgconfig/meson.build:
214 * tests/check/meson.build:
216 Add support for Meson as alternative/parallel build system
217 https://github.com/mesonbuild/meson
219 2016-08-26 21:56:13 +0200 Josep Torra <n770galaxy@gmail.com>
222 * tests/check/Makefile.am:
223 build: silence error about pthread for 'make check' in osx
224 Fixes "clang: error: argument unused during compilation: '-pthread'"
226 2015-09-25 15:04:00 +0000 Nikita Bobkov <NikitaDBobkov@gmail.com>
228 * gst/rtsp-server/rtsp-client.c:
229 rtsp-client: Fix leaking of media in error cases
230 With additional fixes by Kseniya Vasilchuk <vasilchukkseniia@gmail.com>
231 and myself to make the media refcounting a bit easier to follow.
232 https://bugzilla.gnome.org/show_bug.cgi?id=755632
234 2016-08-02 15:08:22 +0300 Sebastian Dröge <sebastian@centricular.com>
236 * gst/rtsp-server/rtsp-client.c:
237 rtsp-client: Fix leaking of session in error cases
238 https://bugzilla.gnome.org/show_bug.cgi?id=755632
240 2016-07-11 21:16:04 +0200 Stefan Sauer <ensonic@users.sf.net>
243 Automatic update of common submodule
244 From f363b32 to f49c55e
246 2016-07-06 13:51:15 +0300 Sebastian Dröge <sebastian@centricular.com>
251 === release 1.9.1 ===
253 2016-07-06 13:28:12 +0300 Sebastian Dröge <sebastian@centricular.com>
259 * gst-rtsp-server.doap:
262 2016-06-24 02:02:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
265 configure: Need to add -DGST_STATIC_COMPILATION when building only statically
266 https://bugzilla.gnome.org/show_bug.cgi?id=767463
268 2016-06-21 11:49:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
271 Automatic update of common submodule
272 From ac2f647 to f363b32
274 2016-04-14 22:56:11 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
276 * gst/rtsp-server/rtsp-sdp.c:
277 * gst/rtsp-server/rtsp-sdp.h:
278 * gst/rtsp-server/rtsp-stream.c:
279 * gst/rtsp-server/rtsp-stream.h:
280 sdp: add rollover counters for all sender SSRC
281 We add different crypto sessions in MIKEY, one for each sender
282 SSRC. Currently, all of them will have the same security policy, 0.
283 The rollover counters are obtained from the srtpenc element using the
285 https://bugzilla.gnome.org/show_bug.cgi?id=730539
287 2016-06-07 20:44:42 +0100 Tim-Philipp Müller <tim@centricular.com>
289 * gst/rtsp-server/rtsp-media-factory.h:
290 * gst/rtsp-server/rtsp-server.h:
293 2016-05-25 10:28:43 +0100 Tim-Philipp Müller <tim@centricular.com>
295 * gst/rtsp-server/Makefile.am:
296 g-i: pass compiler env to g-ir-scanner
297 It's what introspection.mak does as well. Should
298 fix spurious build failures on gnome-continuous
299 (caused by g-ir-scanner getting compiler details
300 via python which is broken in some environments
301 so passing the compiler details bypasses that).
303 2016-05-18 16:48:44 +0100 Ian <ian.arkver.dev@gmail.com>
305 * gst/rtsp-server/rtsp-session.c:
306 rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header
307 This works with rtspsrc and live555, but fails with e.g. ffmpeg.
308 https://bugzilla.gnome.org/show_bug.cgi?id=766619
310 2016-03-07 14:48:38 +0100 Edward Hervey <bilboed@bilboed.com>
312 * gst/rtsp-sink/gstrtspclientsink.c:
313 rtspclientsink: Check return value of sscanf
314 And just make sure we always have 0/0 if we have an error
317 2016-04-25 08:55:25 -0400 Jake Foytik <jake.foytik@ipconfigure.com>
319 * gst/rtsp-server/rtsp-stream.c:
320 * tests/check/gst/rtspserver.c:
321 * tests/check/gst/stream.c:
322 rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
323 - Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
324 - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
325 - Create unit test for shared media.
326 https://bugzilla.gnome.org/show_bug.cgi?id=764744
328 2016-04-11 10:55:23 +0300 Sebastian Dröge <sebastian@centricular.com>
330 * gst/rtsp-server/rtsp-stream.c:
331 rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
332 For IPv6 addresses, binding to a multicast group does not work on Linux
333 either. Always bind to ANY and then later join the multicast group.
334 https://bugzilla.gnome.org/show_bug.cgi?id=764679
336 2016-04-14 10:05:02 +0100 Julien Isorce <j.isorce@samsung.com>
339 Automatic update of common submodule
340 From 6f2d209 to ac2f647
342 2016-04-06 10:09:46 +0200 Patricia Muscalu <patricia@axis.com>
344 * gst/rtsp-server/rtsp-thread-pool.c:
345 rtsp-thread-pool: explained why GSource is a part of ThreadImpl
346 Clarified why it is necessary to add source information to
347 GstRTSPThreadImpl. See the reported bug in GLib:
348 https://bugzilla.gnome.org/show_bug.cgi?id=720186
349 for more information.
350 https://bugzilla.gnome.org/show_bug.cgi?id=761702
352 2016-04-04 12:58:38 +0300 Sebastian Dröge <sebastian@centricular.com>
354 * examples/Makefile.am:
355 examples: Clean up CFLAGS/LDADD even more
356 The internal .la should come first and is part of LDADD, as is
359 2016-04-04 12:39:39 +0300 Sebastian Dröge <sebastian@centricular.com>
361 * examples/Makefile.am:
362 examples: Clean up CFLAGS/LDADD to link with the correct versions of all libraries
364 2016-04-03 12:06:29 +0300 Sebastian Dröge <sebastian@centricular.com>
366 * gst/rtsp-server/Makefile.am:
367 rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS)
369 2015-12-30 18:39:05 +0200 Sebastian Dröge <sebastian@centricular.com>
371 * gst/rtsp-server/rtsp-client.c:
372 * gst/rtsp-server/rtsp-media-factory.c:
373 * gst/rtsp-server/rtsp-media-factory.h:
374 * gst/rtsp-server/rtsp-media.c:
375 * gst/rtsp-server/rtsp-media.h:
376 * gst/rtsp-server/rtsp-sdp.c:
377 * gst/rtsp-server/rtsp-stream.c:
378 * gst/rtsp-server/rtsp-stream.h:
379 rtsp-server: Implement clock signalling according to RFC7273
380 For NTP and PTP clocks we signal the actual clock that is used and signal
381 the direct media clock offset.
382 For all other clocks we at least signal that it's the local sender clock.
383 This allows receivers to know which clock was used to generate the media and
384 its RTP timestamps. Receivers can then implement network synchronization,
385 either absolute or at least relative by getting the sender clock rate directly
386 via NTP/PTP instead of estimating it from RTP timestamps and packet receive
388 https://bugzilla.gnome.org/show_bug.cgi?id=760005
390 2016-03-02 19:42:58 +0200 Sebastian Dröge <sebastian@centricular.com>
392 * gst/rtsp-sink/gstrtspclientsink.c:
393 rtspclientsink: Add support for setting the multicast interface
394 https://bugzilla.gnome.org/show_bug.cgi?id=763000
396 2016-03-02 19:42:13 +0200 Sebastian Dröge <sebastian@centricular.com>
398 * gst/rtsp-server/rtsp-media-factory.c:
399 * gst/rtsp-server/rtsp-media-factory.h:
400 * gst/rtsp-server/rtsp-media.c:
401 * gst/rtsp-server/rtsp-media.h:
402 * gst/rtsp-server/rtsp-stream.c:
403 * gst/rtsp-server/rtsp-stream.h:
404 rtsp-media: Add support for setting the multicast interface
405 https://bugzilla.gnome.org/show_bug.cgi?id=763000
407 2016-03-07 08:50:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
409 * gst/rtsp-sink/gstrtspclientsink.c:
410 rtspclientsink: use new gst_element_class_add_static_pad_template()
411 https://bugzilla.gnome.org/show_bug.cgi?id=763196
413 2016-03-24 13:33:43 +0200 Sebastian Dröge <sebastian@centricular.com>
418 === release 1.8.0 ===
420 2016-03-24 13:00:35 +0200 Sebastian Dröge <sebastian@centricular.com>
426 * gst-rtsp-server.doap:
429 2016-03-16 23:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
431 * gst/rtsp-server/rtsp-stream.c:
432 rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
433 This would get us NO_PREROLL in the bin again and break seeking.
434 Thanks to Carlos Rafael Giani for helping to debug this!
435 https://bugzilla.gnome.org/show_bug.cgi?id=740509
437 === release 1.7.91 ===
439 2016-03-15 12:26:13 +0200 Sebastian Dröge <sebastian@centricular.com>
445 * gst-rtsp-server.doap:
448 2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
450 * gst/rtsp-server/rtsp-stream.c:
451 rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
452 Without this, RECORD pipelines are broken because
453 a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
454 added later. Previously it was there earlier and due to NO_PREROLL caused the
455 pipeline to preroll immediately
456 b) the udpsrc for the pipeline is added later and never set to PLAYING state,
457 as the corresponding code previously was only for PLAY pipelines.
458 https://bugzilla.gnome.org/show_bug.cgi?id=763281
460 2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
462 * gst/rtsp-server/rtsp-stream.c:
463 rtsp-stream: Fix typo in the docstring
464 gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
466 2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
468 * gst/rtsp-server/rtsp-stream.c:
469 rtsp-stream: Disable multicast loopback for all our sockets
470 On Windows this is a receiver-side setting, on Linux a sender-side setting. As
471 we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
472 loopback setting on the socket... while udpsink does which unfortunately has
473 no effect here on Windows but on Linux.
474 https://bugzilla.gnome.org/show_bug.cgi?id=757488
476 2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
478 * tests/check/gst/stream.c:
479 stream tests: added new tests
480 Test a case when the address pool only contains multicast addresses
481 and the client is requesting unicast udp.
482 Added tests for multicast ports allocation.
483 https://bugzilla.gnome.org/show_bug.cgi?id=757488
485 2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
487 * gst/rtsp-server/rtsp-stream.c:
488 rtsp-stream: Only bind multicast sockets to ANY on Windows
489 On Linux it is still needed to bind to the multicast address
490 to filter out random other packets, while on Windows binding
491 to multicast addresses just fails.
493 2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
495 * gst/rtsp-server/rtsp-stream.c:
496 rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
497 Otherwise we fail to allocate UDP ports if the pool only contains multicast
498 addresses, which is something that used to work before. For unicast addresses
499 if the pool contains none, we just allocate them as if there is no pool at
501 https://bugzilla.gnome.org/show_bug.cgi?id=757488
503 2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
505 * gst/rtsp-server/rtsp-client.c:
506 * gst/rtsp-server/rtsp-stream.c:
507 rtsp-server: Fix indentation
509 2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
511 * gst/rtsp-server/rtsp-stream.c:
512 rtsp-stream: Don't bind the sockets to multicast addresses
513 This works on Linux but fails completely on Windows. You're supposed
514 to bind to ANY and then join the multicast group.
515 https://bugzilla.gnome.org/show_bug.cgi?id=757488
517 === release 1.7.90 ===
519 2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
525 * gst-rtsp-server.doap:
528 2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
531 Automatic update of common submodule
532 From b64f03f to 6f2d209
534 2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
536 * gst/rtsp-sink/gstrtspclientsink.c:
537 * tests/check/gst/rtspclientsink.c:
538 rtspsink: Fix some leaks in rtspclientsink and the unit test.
539 https://bugzilla.gnome.org/show_bug.cgi?id=762525
541 2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
543 * tests/check/gst/media.c:
544 * tests/check/gst/rtspclientsink.c:
545 * tests/check/gst/rtspserver.c:
546 * tests/check/gst/stream.c:
547 tests: unit test fixes
548 Removed port allocation test from the media suite.
549 The port allocation failure is now in the stream suite.
551 Make sure that the media is suspended after the DESCRIBE request
552 before reconfiguring the UDP sinks.
554 In the RECORD case we have to set async property to false
555 for the appsink element in the test in order to make sure
556 that the media pipeline doesn't hang in start_preroll().
557 https://bugzilla.gnome.org/show_bug.cgi?id=757488
559 2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
561 * gst/rtsp-server/rtsp-client.c:
562 * gst/rtsp-server/rtsp-stream.c:
563 * gst/rtsp-server/rtsp-stream.h:
564 rtsp-stream: postpone UDP socket allocation until SETUP
565 Postpone the allocation of the UDP sockets until we know
566 what transport has been chosen by the client.
567 Both unicast and multicast UDP sources are created in one
569 https://bugzilla.gnome.org/show_bug.cgi?id=757488
571 2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
573 * gst/rtsp-server/rtsp-stream.c:
574 rtsp-stream: postpone the creation of the UDP sources
575 Code refactoring: allocate the UDP ports after the sender and
576 the reciver parts have been created.
577 We postpone the creation of the UDP sources until the UDP
578 ports have been allocated.
579 https://bugzilla.gnome.org/show_bug.cgi?id=757488
581 2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
583 * gst/rtsp-server/rtsp-stream.c:
584 rtsp-stream: added function for setting UDP sources to PLAYING state
585 Code refactoring: Introduced a function for setting UDP sources
587 https://bugzilla.gnome.org/show_bug.cgi?id=757488
589 2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
591 * gst/rtsp-server/rtsp-stream.c:
592 rtsp-stream: added function for creating and configuring UDP sources
593 Code refactoring: create and configure UDP sources in a separate function.
594 https://bugzilla.gnome.org/show_bug.cgi?id=757488
596 2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
598 * gst/rtsp-server/rtsp-stream.c:
599 rtsp-stream: added function for RTP/RTCP socket configuration
600 Code refactoring: configure RTP and RTCP sockets for UDP sinks
601 in a separate function.
602 https://bugzilla.gnome.org/show_bug.cgi?id=757488
604 2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
606 * gst/rtsp-server/rtsp-stream.c:
607 rtsp-stream: added function for creating and configuring UDP sinks
608 Code refactoring: create and configure UDP sinks in a separate function.
609 https://bugzilla.gnome.org/show_bug.cgi?id=757488
611 2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
613 * gst/rtsp-server/rtsp-stream.c:
614 rtsp-stream: added helper function for creating the sender/receiver parts
615 Code refactoring: introduced helper function for creating
616 the receiver and the sender parts of the streaming pipeline.
617 https://bugzilla.gnome.org/show_bug.cgi?id=757488
619 2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
624 === release 1.7.2 ===
626 2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
632 * gst-rtsp-server.doap:
635 2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
637 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
638 uninstalled.pc: add support for non libtool build systems
639 Currently the .la path is provided which requires to use libtool as
640 mentioned in the GStreamer manual section-helloworld-compilerun.html.
641 It is fine as long as the application is built using libtool.
642 So currently it is not possible to compile a GStreamer application
643 within gst-uninstalled with CMake or other build system different
645 This patch allows to do the following in gst-uninstalled env:
646 gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
647 gstreamer-rtsp-server-1.0)
648 Previously it required to prepend libtool --mode=link
649 https://bugzilla.gnome.org/show_bug.cgi?id=720778
651 2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
653 * gst/rtsp-sink/gstrtspclientsink.c:
654 rtspclientsink: remove check for impossible condition
655 Goto error label checks stream to see if it needs to be unreferenced before
656 returning, but this goto jumps happens before the stream is ever set, so it
657 will always be NULL in this error label.
660 2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
662 * gst/rtsp-sink/gstrtspclientsink.c:
663 rtspclientsink: clean switch statements
664 Coverity demands for fallthrough statements to be clearly commented,
665 to distinguish from accidental fall throughs. And it also needs all
666 cases to finish with a break, even if the break is never going to be
667 executed like in the case of a continue jump.
671 2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
673 * tests/check/Makefile.am:
674 tests: extend the AM_TESTS_ENVIRONMENT from check.mak
675 To get the CK_DEFAULT_TIMEOUT defined for all tests
676 Also removes a 120 seconds timeout that was set as default
677 explicitly in this module
678 https://bugzilla.gnome.org/show_bug.cgi?id=761472
680 2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
684 Automatic update of common submodule
685 From 86e4663 to b64f03f
687 2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
689 * gst/rtsp-server/rtsp-media.c:
690 rtsp-media: fix state_lock not locked again when preroll fails
691 https://bugzilla.gnome.org/show_bug.cgi?id=761399
693 2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
696 configure: Move plugin specific flags below all the others
697 They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
698 -no-undefined. And -no-undefined is required on Windows to build DLLs.
700 2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
702 * gst/rtsp-sink/gstrtspclientsink.c:
703 rtspclientsink: Simplify slightly using new -base API
704 Use the new Mikey and SDP API in the base plugins libs
705 to simplify some code.
706 https://bugzilla.gnome.org/show_bug.cgi?id=758180
708 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
713 * gst/rtsp-sink/Makefile.am:
714 * gst/rtsp-sink/gstrtspclientsink.c:
715 * gst/rtsp-sink/gstrtspclientsink.h:
716 * gst/rtsp-sink/plugin.c:
717 * tests/check/Makefile.am:
718 * tests/check/gst/rtspclientsink.c:
719 rtspsink: Add rtspclientsink element
720 Add an rtspclientsink element that accepts streams for which
721 there is a registered payloader and sends them to
722 an RTSP server using RECORD.
723 Sending is synchronised to the pipeline clock. Payload-types
724 are automatically selected. The 'new-payloader' signal is fired
725 for custom configuration of payloaders when they are created.
726 Can now stream a movie like this:
728 ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
729 decodebin name=depay1 ! audioconvert ! autoaudiosink )"
731 gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
732 queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
733 https://bugzilla.gnome.org/show_bug.cgi?id=758180
735 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
737 * gst/rtsp-server/rtsp-stream.c:
738 * gst/rtsp-server/rtsp-stream.h:
739 rtsp-stream: Add functions for using rtsp-stream from the client
740 Add a boolean to indicate that the rtsp-stream is running on the
741 'client' side of an RTSP connection, for sending streams via
742 RECORD. In that case, the roles of the client/server ports
743 in transport setup are swapped.
744 https://bugzilla.gnome.org/show_bug.cgi?id=758180
746 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
748 * gst/rtsp-server/rtsp-sdp.c:
749 * gst/rtsp-server/rtsp-sdp.h:
750 rtsp-sdp: Add gst_rtsp_sdp_from_stream()
751 A new function that adds info from a GstRTSPStream into an SDP message.
752 https://bugzilla.gnome.org/show_bug.cgi?id=758180
754 2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
756 * gst/rtsp-server/rtsp-media.c:
757 rtsp-media: Fix mutex beeing unlocked while they should be locked
758 https://bugzilla.gnome.org/show_bug.cgi?id=761226
760 2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
762 * gst/rtsp-server/rtsp-media-factory.c:
763 rtsp-media-factory: add missing break in "clock" property setter
766 2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
768 * gst/rtsp-server/rtsp-stream.c:
769 rtsp-stream: fixed assert during update transport
770 When RTSP server trying update transport during multicast, it throws an
771 assert. The assert is thrown because it is trying to get the parent of
772 an non-existing funnel element.
773 https://bugzilla.gnome.org/show_bug.cgi?id=760150
775 2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
777 * gst/rtsp-server/rtsp-permissions.h:
778 * gst/rtsp-server/rtsp-thread-pool.h:
779 * gst/rtsp-server/rtsp-token.h:
780 docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
781 gtk-doc can handle static inline functions just fine these days,
782 there's no need for this stuff any more.
784 2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
786 * gst/rtsp-server/rtsp-media.c:
787 * gst/rtsp-server/rtsp-sdp.c:
788 sdp: replace duplicated codes to call new base sdp apis
789 https://bugzilla.gnome.org/show_bug.cgi?id=745880
791 2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
793 * examples/test-netclock.c:
794 test-netclock: Use the new API to configure a clock directly
796 2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
798 * gst/rtsp-server/rtsp-media-factory.c:
799 * gst/rtsp-server/rtsp-media-factory.h:
800 * gst/rtsp-server/rtsp-media.c:
801 * gst/rtsp-server/rtsp-media.h:
802 rtsp-media: Add API to directly configure a clock on the media pipelines
804 2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
806 * gst/rtsp-server/rtsp-media.c:
807 rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
809 2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
811 * gst/rtsp-server/rtsp-media-factory.c:
812 rtsp-media-factory: Add FIXME for 2.0
814 2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
816 * gst/rtsp-server/rtsp-stream.c:
817 rtsp-stream: Fix indentation
819 2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
821 * gst/rtsp-server/rtsp-media.c:
822 rtsp-media: Do not prepare media after media times out
823 Deferred calls to start_prepare() can be deferred past the point until
824 which wait_preroll() and by proxy gst_rtsp_media_get_status() is
825 prepared to wait. Previously there was no lock and no check for this
826 situation. This meant that a media could be prepared and unprepared
827 simultaneously by two different threads. Now a lock is in place and a
828 suitable check is done.
829 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
831 2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
833 * gst/rtsp-server/rtsp-client.c:
834 * gst/rtsp-server/rtsp-media-factory.c:
835 * gst/rtsp-server/rtsp-media-factory.h:
836 * gst/rtsp-server/rtsp-media.c:
837 * gst/rtsp-server/rtsp-media.h:
838 rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
839 Without TEARDOWN it might be desireable to keep the media running and continue
840 sending data to the client, even if the RTSP connection itself is
842 Only do this for session medias that have only UDP transports. If there's at
843 least on TCP transport, it will stop working and cause problems when the
844 connection is disconnected.
845 https://bugzilla.gnome.org/show_bug.cgi?id=758999
847 2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
852 === release 1.7.1 ===
854 2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
860 * gst-rtsp-server.doap:
863 2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
866 configure: Make -Bsymbolic check work with clang.
867 Update the -Bsymbolic check with the version glib has. This version
869 https://bugzilla.gnome.org/show_bug.cgi?id=759713
871 2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
873 * gst/rtsp-server/rtsp-session-pool.c:
874 rtsp-session-pool: Avoid dollar sign ($) in session ids
875 Live555 in VLC strips off dollar signs and then gets very confused,
876 we don't loose too much entropy by just skipping it.
878 2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
880 * gst/rtsp-server/rtsp-address-pool.h:
881 * gst/rtsp-server/rtsp-auth.h:
882 * gst/rtsp-server/rtsp-client.h:
883 * gst/rtsp-server/rtsp-media-factory-uri.h:
884 * gst/rtsp-server/rtsp-media-factory.h:
885 * gst/rtsp-server/rtsp-media.h:
886 * gst/rtsp-server/rtsp-mount-points.h:
887 * gst/rtsp-server/rtsp-permissions.h:
888 * gst/rtsp-server/rtsp-server.h:
889 * gst/rtsp-server/rtsp-session-media.h:
890 * gst/rtsp-server/rtsp-session-pool.h:
891 * gst/rtsp-server/rtsp-session.h:
892 * gst/rtsp-server/rtsp-stream-transport.h:
893 * gst/rtsp-server/rtsp-stream.h:
894 * gst/rtsp-server/rtsp-thread-pool.h:
895 * gst/rtsp-server/rtsp-token.h:
896 rtsp-server: Add g_autoptr() support to all types
897 https://bugzilla.gnome.org/show_bug.cgi?id=754464
899 2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
901 * gst/rtsp-server/rtsp-stream.c:
902 rtsp-stream: fixed valgrind error
903 Fixed the valgrind error in unit test. The UDP source created during
904 gst_rtsp_stream_join_bin() was not released while destroying the rtp
906 https://bugzilla.gnome.org/show_bug.cgi?id=759010
908 2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
912 Automatic update of common submodule
913 From b319909 to 86e4663
915 2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
917 * gst/rtsp-server/rtsp-client.c:
918 rtsp-client: suspend media during setup request
919 SETUP request from clients needs to suspend the media to clear the
920 prerolled buffers. Otherwise it will not affect the prerolled buffer
921 and the prerolled buffers will be incorrect (for example block-size
922 from setup request will not affect the prerolled buffer unless the
924 https://bugzilla.gnome.org/show_bug.cgi?id=758268
926 2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
928 * gst/rtsp-server/rtsp-stream.c:
929 rtsp-stream: create stream pipeline based on transport
930 Based on the protocol, create the rtsp stream pipeline. If only TCP or
931 only UDP is set as the transport protocol, it will not add the extra tee
932 or queue element to the pipeline. Both these elements will be added, if
933 it supports both TCP and UDP protocols. This improves the pipeline
934 performance when one protocol is present.
935 https://bugzilla.gnome.org/show_bug.cgi?id=758179
937 2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
939 * gst/rtsp-server/rtsp-stream.c:
940 rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
941 Adding them when not needed will start some logic inside rtpbin that might be
942 problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
943 would start up a rtpjitterbuffer and behave in weird ways.
944 We still set up the UDP sources for RTP receiving for a sender media to be
945 able to receive any packets sent by the client for NAT traversal. They will
946 all go to a fakesink though.
947 Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
948 NO_PREROLL, which will cause deadlocks when seeking the media as it will never
949 receive ASYNC_DONE after a seek.
950 https://bugzilla.gnome.org/show_bug.cgi?id=758319
952 2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
954 * gst/rtsp-server/rtsp-stream.c:
955 rtsp-stream: Disable multicast loopback for the multicast udp sources too
956 On POSIX this setting is for sender sockets, on Windows for receiver sockets.
957 Previously we were only setting this for sender sockets, which caused looped
958 back packets to be received on Windows if a multicast transport was used.
960 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
962 * examples/test-record-auth.c:
963 * examples/test-record.c:
964 examples: Actually use the provided port in the record examples
966 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
968 * examples/test-record-auth.c:
969 test-record-auth: Add the option to build in TLS support
971 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
973 * examples/test-auth.c:
974 test-auth: Use an 'anonymous' user for unauthenticated default
975 There's a comment on one of the resources that 'user' and 'admin'
976 shouldn't even be able to see it, but they can if the default
977 token is 'admin2', since that gives them access anyway.
979 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
981 * examples/.gitignore:
982 * examples/Makefile.am:
983 * examples/test-record-auth.c:
984 Add test-record-auth example
986 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
988 * gst/rtsp-server/rtsp-client.c:
989 * tests/check/gst/client.c:
990 rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
992 2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
994 * gst/rtsp-server/rtsp-server.c:
995 rtsp-server: Change the logic so we don't pop a NULL context
996 When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
997 will sometimes fail. This call is made before any context is pushed
998 resulting in an attempt to pop a NULL context.
999 https://bugzilla.gnome.org/show_bug.cgi?id=757949
1001 2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
1003 * tests/check/gst/rtspserver.c:
1004 rtspserver: Add udp-mcast transport SETUP test
1005 Refactor utility functions in the test file so they can handle
1006 more than UDP and TCP as lower transport.
1007 https://bugzilla.gnome.org/show_bug.cgi?id=756969
1009 2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
1011 * gst/rtsp-server/rtsp-stream.c:
1012 rtsp-stream: Always unref return value of gst_object_get_parent()
1013 Fixes a leak of a GstBin in the udp-mcast case.
1014 https://bugzilla.gnome.org/show_bug.cgi?id=756968
1016 2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
1019 Automatic update of common submodule
1020 From b99800a to b319909
1022 2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
1025 Use new GST_ENABLE_EXTRA_CHECKS #define
1026 https://bugzilla.gnome.org/show_bug.cgi?id=756870
1028 2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
1031 Automatic update of common submodule
1032 From 6babecd to b99800a
1034 2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
1037 Update GLib dependency to 2.40.0
1039 2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1041 * examples/test-mp4.c:
1042 * gst/rtsp-server/rtsp-stream.c:
1043 stream: listen to sender ssrc signals
1044 https://bugzilla.gnome.org/show_bug.cgi?id=746747
1046 2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
1049 common: update for new suppression
1050 Makes check-valgrind pass with glib 2.46
1052 2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1054 * gst/rtsp-server/rtsp-media.c:
1055 rtsp-media: Take reference to media that will be prepared
1056 default_prepare() takes a transfer-none reference GstRTSPMedia object.
1057 Later on a g_idle_source_new() is created and a pointer to the media
1058 object is passed as user data. If the media is freed before the idle
1059 source is dispatched the media object pointer is invalid, but the idle
1060 source callback expects it to still be valid. To fix this a reference to
1061 the media object is taken when registering the source callback function
1062 and a corresponding release of the reference is done when the souce is
1064 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
1066 2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
1068 * examples/test-launch.c:
1069 * examples/test-mp4.c:
1070 * examples/test-ogg.c:
1071 * examples/test-record.c:
1072 * examples/test-uri.c:
1073 rtsp-server: Fix memory leaks when context parse fails
1074 When g_option_context_parse fails, context and error variables are not getting free'd
1075 which results in memory leaks. Free'ing the same.
1076 And replacing g_error_free with g_clear_error, which checks if the error being passed
1077 is not NULL and sets the variable to NULL on free'ing.
1078 https://bugzilla.gnome.org/show_bug.cgi?id=753863
1080 2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
1085 === release 1.6.0 ===
1087 2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
1093 * gst-rtsp-server.doap:
1096 === release 1.5.91 ===
1098 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
1104 * gst-rtsp-server.doap:
1107 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
1109 * docs/libs/gst-rtsp-server-sections.txt:
1110 * gst/rtsp-server/rtsp-stream.c:
1111 stream: fix docs for recently-added get/set_buffer_size API
1112 https://bugzilla.gnome.org/show_bug.cgi?id=749095
1114 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
1116 * gst/rtsp-server/rtsp-media.c:
1117 rtsp-media: Don't crash on encrypted RTX SDP
1118 In parse_keymgmt(), don't mutate the input string that's been passed
1119 as const, especially since we might need the original value again if
1120 the same key info applies to multiple streams (RTX, for example).
1121 https://bugzilla.gnome.org/show_bug.cgi?id=754753
1123 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
1125 * examples/test-mp4.c:
1126 test-mp4: Support filenames with spaces in them. Error out on too few arguments
1128 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
1130 * examples/test-record.c:
1131 test-record: Check parameter count and print out help
1132 If no launch pipeline was supplied, print out some help
1134 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
1136 * gst/rtsp-server/rtsp-media.c:
1137 * gst/rtsp-server/rtsp-stream.c:
1138 * gst/rtsp-server/rtsp-stream.h:
1139 rtsp-stream: Implement UDP buffer size setting.
1140 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
1142 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
1143 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
1145 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
1147 * gst/rtsp-server/rtsp-media.h:
1148 rtsp-media: Fix small typo causing gtk-doc to complain
1150 === release 1.5.90 ===
1152 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
1158 * gst-rtsp-server.doap:
1161 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1163 * gst/rtsp-server/rtsp-media-factory.c:
1164 media-factory: get port number through gst_rtsp_url_get_port
1165 https://bugzilla.gnome.org/show_bug.cgi?id=753473
1167 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
1169 * tests/check/gst/media.c:
1170 media-test: Removing unnecessary assertion
1171 https://bugzilla.gnome.org/show_bug.cgi?id=753385
1173 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1175 * gst/rtsp-server/rtsp-server.c:
1176 Document that source keeps a ref on server until it's destroyed
1177 https://bugzilla.gnome.org/show_bug.cgi?id=749227
1179 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1181 * tests/check/gst/media.c:
1182 media-test: Test for multiple dynamic payload
1183 https://bugzilla.gnome.org/show_bug.cgi?id=753385
1185 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1187 * gst/rtsp-server/rtsp-media.c:
1188 media: Only add fakesink once per pipeline
1189 The intention is to prevent going PLAYING state before pads are created.
1190 If there was mutilple dynamic payload, it would leak few fakesink and
1191 actually prevent from ever reaching playing state.
1192 https://bugzilla.gnome.org/show_bug.cgi?id=753385
1194 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1196 * gst/rtsp-server/rtsp-media.c:
1197 Revert "rtsp-media: Only add 1 fakesink per pipeline"
1198 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
1200 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1202 * gst/rtsp-server/rtsp-media.c:
1203 rtsp-media: Only add 1 fakesink per pipeline
1204 There should be only one fakesink per pipeline, not per dynpay. This
1205 would lead to element naming clash.
1207 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
1209 * gst/rtsp-server/rtsp-media.c:
1210 rtsp-media: assertion error due to wrong condition check
1211 In media to caps function, reserved_keys array is being used for variable i,
1212 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
1213 changed it to variable j
1214 https://bugzilla.gnome.org/show_bug.cgi?id=753009
1216 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
1218 * gst/rtsp-server/rtsp-media.c:
1219 rtsp-media: Strip keys from the fmtp that we use internally in our caps
1220 Skip keys from the fmtp, which we already use ourselves for the
1221 caps. Some software is adding random things like clock-rate into
1222 the fmtp, and we would otherwise here set a string-typed clock-rate
1223 in the caps... and thus fail to create valid RTP caps
1224 https://bugzilla.gnome.org/show_bug.cgi?id=753009
1226 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1228 * gst/rtsp-server/rtsp-thread-pool.c:
1229 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
1230 https://bugzilla.gnome.org/show_bug.cgi?id=752640
1232 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
1235 Automatic update of common submodule
1236 From f74b2df to 9aed1d7
1238 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
1243 === release 1.5.2 ===
1245 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
1251 * gst-rtsp-server.doap:
1254 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
1256 * gst/rtsp-server/rtsp-client.c:
1257 * gst/rtsp-server/rtsp-client.h:
1258 * tests/check/gst/client.c:
1259 rtsp-client: allow application to decide what requirements are supported
1260 Add "check-requirements" signal and vfunc to allow application
1261 (and subclasses) to check the requirements.
1262 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
1263 https://bugzilla.gnome.org/show_bug.cgi?id=749417
1265 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
1268 Automatic update of common submodule
1269 From 6015d26 to f74b2df
1271 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
1273 * gst/rtsp-server/rtsp-media.c:
1274 rtsp-media: Always use real payloader when creating streams
1275 A bin that contains the real payloader might be used as payloader. In this
1276 case we have to get the real payloader for the various properties it provides.
1277 Example use cases for this are bins that payload some media and then have
1278 additional elements that add metadata or RTP extension headers to the stream.
1279 https://bugzilla.gnome.org/show_bug.cgi?id=750800
1281 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
1283 * examples/test-netclock-client.c:
1284 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
1286 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
1288 * examples/test-netclock-client.c:
1289 * examples/test-netclock.c:
1290 test-netclock: Use new ntp-time-source property on rtpbin
1291 Select the clock time to be used as NTP time source. This allows proper
1292 synchronization between receivers, independent of sharing base times, and just
1293 requires them to use the same clock.
1295 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
1297 * examples/test-netclock-client.c:
1298 * examples/test-netclock.c:
1299 test-netclock: Setting the same base time on sender and receiver is not necessary
1300 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
1302 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1304 * gst/rtsp-server/rtsp-stream.c:
1305 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
1306 https://bugzilla.gnome.org/show_bug.cgi?id=750764
1308 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1310 * docs/libs/gst-rtsp-server.types:
1311 docs: add missing types
1312 https://bugzilla.gnome.org/show_bug.cgi?id=750764
1314 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1316 * docs/libs/gst-rtsp-server-sections.txt:
1317 docs: add missing apis
1318 https://bugzilla.gnome.org/show_bug.cgi?id=750764
1320 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
1322 * examples/test-netclock-client.c:
1323 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
1325 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1327 * docs/libs/gst-rtsp-server-sections.txt:
1328 * gst/rtsp-server/rtsp-auth.c:
1329 * gst/rtsp-server/rtsp-auth.h:
1330 GstRTSPAuth: Add client certificate authentication support
1331 https://bugzilla.gnome.org/show_bug.cgi?id=750471
1333 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
1335 * examples/test-netclock-client.c:
1336 test-netclock-client: Use new GstClock API to wait for clock synchronization
1338 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
1340 * examples/test-netclock-client.c:
1341 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
1342 A mainloop is needed to get glimagesink to display something on OSX, and
1343 the source-setup signal just makes things a little bit easier.
1345 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
1348 Automatic update of common submodule
1349 From d9a3353 to 6015d26
1351 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
1354 Automatic update of common submodule
1355 From d37af32 to d9a3353
1357 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
1360 Automatic update of common submodule
1361 From 21ba2e5 to d37af32
1363 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
1366 Automatic update of common submodule
1367 From c408583 to 21ba2e5
1369 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
1371 * docs/libs/Makefile.am:
1372 docs: remove variables that we define in the snippet from common
1373 This is syncing our Makefile.am with upstream gtkdoc.
1375 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
1378 Automatic update of common submodule
1379 From 44a3517 to c408583
1381 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
1386 === release 1.5.1 ===
1388 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
1394 * gst-rtsp-server.doap:
1397 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
1399 * gst/rtsp-server/rtsp-client.c:
1400 rtsp-client: No flush during Teardown.
1401 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
1402 backlog is empty it can happen that just a part of a message will be
1403 sent and rest is in backlog queue. If then flush during teardown
1404 just a part of message will be sent.This can lead to client miss
1405 teardown response since it expect to get the last part of message.
1406 The flushing during teardown was introduced to fix a deadlock that now
1407 is fixed more generally in handle_request by temporary setting backlog
1409 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
1411 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
1413 * tests/check/Makefile.am:
1414 tests: Use AM_TESTS_ENVIRONMENT
1415 Needed by the new automake test runner and the
1416 current version of the common submodule.
1418 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
1420 * gst/rtsp-server/rtsp-media.h:
1421 * gst/rtsp-server/rtsp-stream.h:
1422 rtsp-server: Use single-include rtsp header to make sure we get all definitions
1424 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
1426 * gst/rtsp-server/rtsp-media.c:
1427 rtsp-media: Mark some more functions static
1429 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
1431 * gst/rtsp-server/rtsp-media.c:
1432 rtsp-media: Only unblock the media in suspend() when actually changing the state
1433 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
1435 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
1437 * examples/test-video-rtx.c:
1438 examples: Use AVPF profile for the RTX example
1440 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
1442 * gst/rtsp-server/rtsp-sdp.c:
1443 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
1445 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1447 * gst/rtsp-server/rtsp-stream.c:
1448 rtsp-stream: get valid clock-rate from last-sample
1449 clock-rate in last-sample's caps is integer, not unsigned.
1450 To get this value properly, variable needs to be type-casted to int.
1451 https://bugzilla.gnome.org/show_bug.cgi?id=747614
1453 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
1457 autogen.sh: only run autopoint if gettext requested in configure.ac
1458 Not just because there happens to be a po directory.
1459 https://bugzilla.gnome.org/show_bug.cgi?id=748058
1461 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
1464 Revert "configure.ac: uncomment gettext version setup"
1465 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
1466 We don't need a gettext setup here and there's no po
1467 directory either, so no reason why autopoint would be
1468 run in the first place.
1469 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
1471 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
1473 * examples/test-multicast.c:
1474 * examples/test-multicast2.c:
1475 * examples/test-sdp.c:
1476 * examples/test-video-rtx.c:
1477 * examples/test-video.c:
1478 * tests/test-cleanup.c:
1479 * tests/test-reuse.c:
1480 Fix timeout function signatures across tests and examples
1482 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
1484 * tests/check/Makefile.am:
1485 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
1486 Make sure the test environment is set up.
1487 https://bugzilla.gnome.org//show_bug.cgi?id=747624
1489 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
1492 configure: bump automake requirement to 1.14 and autoconf to 2.69
1493 This is only required for builds from git, people can still
1494 build tarballs if they only have older autotools.
1495 https://bugzilla.gnome.org//show_bug.cgi?id=747624
1497 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
1500 configure.ac: uncomment gettext version setup
1501 Fixes autogen.sh. It would run autopoint, which would complain
1502 that it could not find the gettext version in configure.ac.
1503 https://bugzilla.gnome.org/show_bug.cgi?id=748058
1505 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1507 * examples/test-video-rtx.c:
1508 test-video-rtx: set exact payload type to PCMA payloader
1509 Setting wrong payload type causes failure to do retransmission through audio stream
1510 https://bugzilla.gnome.org/show_bug.cgi?id=747839
1512 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
1514 * gst/rtsp-server/rtsp-media.c:
1515 * gst/rtsp-server/rtsp-stream.c:
1516 * gst/rtsp-server/rtsp-stream.h:
1517 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
1518 Because of duplicated g_signal_connect for request-aux-sender signal,
1519 wrong stream pointer is passed to the signal handler.
1520 Instead of passing each stream, pass stream array and get the relevant stream.
1521 https://bugzilla.gnome.org/show_bug.cgi?id=747839
1523 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
1527 Update autogen.sh to latest version from common
1528 Fixes build after aclocal_check etc. helpers have been removed.
1530 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
1533 Automatic update of common submodule
1534 From bc76a8b to c8fb372
1536 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
1538 * gst/rtsp-server/rtsp-stream.c:
1539 rtsp-stream: Limit the queues to 1 buffer
1540 We only need them to be able to pre-roll, queueing up more data here
1541 is only going to harm latency and memory usage.
1543 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
1545 * gst/rtsp-server/rtsp-stream.c:
1546 rtsp-stream: Update comment and ASCII art to the latest code
1547 We have a queue in front of the udpsink too to prevent the pipeline from
1550 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
1552 * gst/rtsp-server/rtsp-stream.c:
1553 rtsp-media: Properly return first rtptime
1554 Instead we where returning first GstBuffer timestamp. This would result
1555 in clock skew and unwanted behaviour in RTSP playback.
1556 https://bugzilla.gnome.org/show_bug.cgi?id=746479
1558 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
1560 * gst/rtsp-server/rtsp-stream.c:
1561 rtsp-stream: Don't leave buffer mapped
1562 If the seq is NULL, the RTP buffer was left mapped. We should always
1565 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
1570 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
1572 * gst/rtsp-server/rtsp-media-factory.c:
1573 * tests/check/gst/client.c:
1574 Fix double semicolons
1576 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
1578 * gst/rtsp-server/rtsp-stream.c:
1579 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
1580 This gives more accurate values than asking the payloader. There might be
1581 queueing happening between the payloader and the sink.
1582 https://bugzilla.gnome.org/show_bug.cgi?id=745704
1584 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
1586 * gst/rtsp-server/rtsp-media.c:
1587 rtsp-media: Don't seek for PLAY if the position will not change
1588 https://bugzilla.gnome.org/show_bug.cgi?id=745704
1590 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
1592 * gst/rtsp-server/rtsp-media.c:
1593 rtsp-media: Don't include payload type in the caps for framesize
1594 When the sdp media attribute framesize are converted to caps
1595 the <payload> should not be included.
1596 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
1597 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
1599 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
1601 * gst/rtsp-server/rtsp-sdp.c:
1602 rtsp-sdp: add payload type to the sdp framesize attribute
1603 The sdp framesize attribute is desribed in RFC6064. It is specified
1604 for payloading of H263 and has the following form
1605 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
1606 should be added to the caps in a payloader and the <payload type> should
1607 be added by the rtsp-server.
1608 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
1610 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
1612 * examples/test-uri.c:
1613 examples: test-uri: fix tainted variable
1614 Insignificant but this keeps Coverity happy.
1617 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
1619 * examples/.gitignore:
1620 * examples/Makefile.am:
1621 * examples/test-netclock-client.c:
1622 * examples/test-netclock.c:
1623 examples: Add a simple example of network synch for live streams.
1624 An example server and client that works for synchronising live streams
1625 only - as it can't support pause/play.
1627 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
1629 * gst/rtsp-server/rtsp-media-factory.c:
1630 * gst/rtsp-server/rtsp-media-factory.h:
1631 rtsp-media-factory: Add functions to set/get the media gtype
1632 Allow specifying the GType of a GstRtspMedia subclass to create
1633 as a simpler way to get the factory to create a custom
1634 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
1636 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
1638 * gst/rtsp-server/rtsp-media.c:
1639 rtsp-media: fix double unlock in _get_buffer_size()
1640 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
1641 because of double g_mutex_unlock () usage.
1642 https://bugzilla.gnome.org/show_bug.cgi?id=745434
1644 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
1646 * gst/rtsp-server/rtsp-session-pool.c:
1647 * gst/rtsp-server/rtsp-session.c:
1648 * gst/rtsp-server/rtsp-session.h:
1649 rtsp-session: Use monotonic time for RTSP session timeout
1650 Changed RTSP session timeout handling to monotonic time
1651 and deprecating the API for current system time.
1652 This fixes timeouts when the system time changes.
1653 https://bugzilla.gnome.org/show_bug.cgi?id=743346
1655 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
1657 * gst/rtsp-server/rtsp-client.c:
1658 * gst/rtsp-server/rtsp-media.c:
1659 rtsp-client: Only error out in PLAY if seeking actually failed
1660 If the media was just not seekable, we continue from whatever position we are
1661 and let the client decide if that is what is wanted or not.
1662 Only if the actual seek failed, we can't really recover and should error out.
1664 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
1666 * gst/rtsp-server/rtsp-stream.c:
1667 rtsp-stream: Add necessary queues between tee and multiudpsink
1668 https://bugzilla.gnome.org/show_bug.cgi?id=744379
1670 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
1672 * gst/rtsp-server/rtsp-client.c:
1673 * gst/rtsp-server/rtsp-media.c:
1674 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
1675 Instead error out properly the same way as if the SEEKING query already
1678 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
1680 * gst/rtsp-server/rtsp-stream.h:
1681 rtsp-stream: minor code formatting fix
1683 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
1685 * gst/rtsp-server/rtsp-media.c:
1686 rtsp-media: fix logic for collect_streams
1687 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
1688 all streams it knows if it got any, and can check if the transport mode is OK.
1691 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
1693 * gst/rtsp-server/rtsp-media.c:
1694 rtsp-media: Don't set the transport mode based on what elements we find
1695 Just print a warning if the one that was set before disagrees with what
1696 elements we found. It must already be set to something before as this
1697 function is called after we received the SDP from ANNOUNCE in RECORD mode,
1698 and we would reject ANNOUNCE if the RECORD flag was not set.
1700 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
1702 * tests/check/gst/rtspserver.c:
1703 tests: rtspserver: rename shadowed variable
1704 We have two different 'sink' variables here,
1705 rename one of them for clarity.
1707 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
1709 * gst/rtsp-server/rtsp-client.c:
1710 rtsp-client: fix awkward if clause
1712 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
1714 * examples/test-uri.c:
1715 examples: test-uri: improve uri argument handling and accept file names
1716 Print an error if the argument passed is not a URI and can't
1717 be converted into one, or no arguments have been provided.
1719 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
1721 * examples/test-uri.c:
1722 examples: test-uri: don't remove mount point after 10 seconds
1723 It's very irritating when trying to test stuff repeatedly
1724 and serves no real purpose other than showing that it can
1727 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
1729 * examples/.gitignore:
1730 examples: add new test-record to .gitignore
1732 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
1734 * examples/test-record.c:
1735 * gst/rtsp-server/rtsp-client.c:
1736 * gst/rtsp-server/rtsp-media-factory.c:
1737 * gst/rtsp-server/rtsp-media-factory.h:
1738 * gst/rtsp-server/rtsp-media.c:
1739 * gst/rtsp-server/rtsp-media.h:
1740 * tests/check/gst/rtspserver.c:
1741 rtsp-media: Use flags to distinguish between PLAY and RECORD media
1743 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
1745 * examples/test-record.c:
1746 test-record: Set latency for playback-style example to 2s instead of 200ms
1748 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
1750 * tests/check/gst/rtspserver.c:
1751 tests: add some unit tests for ANNOUNCE and RECORD
1752 https://bugzilla.gnome.org/show_bug.cgi?id=743175
1754 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
1756 * gst/rtsp-server/rtsp-client.c:
1757 rtsp-client: fix a couple of leaks in handle_announce
1759 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
1761 * gst/rtsp-server/rtsp-media-factory.c:
1762 * gst/rtsp-server/rtsp-media-factory.h:
1763 * gst/rtsp-server/rtsp-media.c:
1764 * gst/rtsp-server/rtsp-media.h:
1765 rtsp-media: Expose latency setting for setting the rtpbin latency
1767 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
1769 * examples/test-record.c:
1770 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
1772 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
1774 * gst/rtsp-server/rtsp-stream.c:
1775 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
1777 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
1779 * examples/Makefile.am:
1780 * examples/test-record.c:
1781 * gst/rtsp-server/rtsp-client.c:
1782 * gst/rtsp-server/rtsp-client.h:
1783 * gst/rtsp-server/rtsp-media-factory.c:
1784 * gst/rtsp-server/rtsp-media-factory.h:
1785 * gst/rtsp-server/rtsp-media.c:
1786 * gst/rtsp-server/rtsp-media.h:
1787 * gst/rtsp-server/rtsp-session-media.c:
1788 * gst/rtsp-server/rtsp-stream.c:
1789 * gst/rtsp-server/rtsp-stream.h:
1790 Add initial support for RECORD
1791 We currently only support media that is RECORD or PLAY only, not both at once.
1792 https://bugzilla.gnome.org/show_bug.cgi?id=743175
1794 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
1796 * gst/rtsp-server/rtsp-stream.c:
1797 rtsp-stream: RTCP and RTP transport cache cookies seperated
1798 RTCP packets were not sent because the same tr_cache_cookie was used for
1799 both RTP and RTCP. So only one of the tr_cache lists were populated
1800 depending on which one was sent first. If the tr_cache list is not
1801 populated then no packets can be sent. Most often this happened to be
1802 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
1803 resulted in both the tr_cache_lists to be populated regardless of which
1805 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
1807 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
1809 * gst/rtsp-server/rtsp-stream.c:
1810 rtsp-stream: fix false compiler warning
1811 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
1813 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
1815 * gst/rtsp-server/rtsp-client.c:
1816 rtsp-client: log interleaved data received
1818 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
1820 * gst/rtsp-server/rtsp-client.c:
1821 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
1823 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
1825 * gst/rtsp-server/rtsp-client.c:
1826 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
1828 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
1830 * gst/rtsp-server/rtsp-client.c:
1831 rtsp-client: Use a random session ID in the SDP
1832 RFC4566 Section 5.2 says that it should make the username, session id,
1833 nettype, addrtype and unicast address tuple globally unique. Always using
1834 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
1835 Instead let's create a 64 bit random number, which at least brings us
1836 closer to the goal of global uniqueness.
1837 https://tools.ietf.org/html/rfc4566#section-5.2
1839 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
1841 * examples/test-launch.c:
1842 * examples/test-mp4.c:
1843 * examples/test-ogg.c:
1844 * examples/test-uri.c:
1845 examples: Don't call gst_init() and gst_get_option_group()
1846 The latter calls the former at the appropriate time.
1848 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
1850 * gst/rtsp-server/rtsp-client.c:
1851 rtsp-client: Drop trailing \0 of RTSP DATA messages
1852 We add a trailing \0 in GstRTSPConnection to make parsing of
1853 string message bodies easier (e.g. the SDP from DESCRIBE) but
1854 for actual data this means we have to drop it or otherwise
1855 create invalid data.
1857 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
1859 * gst/rtsp-server/rtsp-stream.c:
1860 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
1861 Fixes crash when two threads access handle_new_sample() at the same
1862 time, one for RTP, one for RTCP.
1863 Otherwise, when iterating over the transports cache, it might be modified by
1864 another thread at the same time if the transports cookie has changed.
1865 https://bugzilla.gnome.org/show_bug.cgi?id=742954
1867 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
1869 * gst/rtsp-server/rtsp-stream.c:
1870 rtsp-stream: Set format=TIME on our app sources for TCP
1872 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
1874 * gst/rtsp-server/rtsp-session-pool.c:
1875 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
1876 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
1877 RFC 2326 states that session IDs may consist of alphanumeric as well as
1878 the safe characters $-_.+ -- N.B. the percent character is not allowed.
1879 Previously the session ID was URI-escaped, this meant that any character
1880 which was not alphanumeric or any of the characters +-._~ would be
1881 percent encoded. While the RFC (surprisingly) mentions that linear white
1882 space in session IDs should be URI-escaped, it does not say anything
1883 about other characters. Moreover no white space is allowed in the
1884 session ID. Finally the percent character which is the result of
1885 URI-escaping is not allowed in a session ID.
1886 So there is no reason to do any URI-escaping, and now it is removed.
1887 https://bugzilla.gnome.org/show_bug.cgi?id=742869
1889 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
1892 Automatic update of common submodule
1893 From f2c6b95 to bc76a8b
1895 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
1898 Fix 'make check' from top-level directory
1900 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
1902 * examples/test-launch.c:
1903 * examples/test-mp4.c:
1904 * examples/test-ogg.c:
1905 * examples/test-uri.c:
1906 examples: Add command-line parsing and take a 'port' argument
1907 This allows users to run multiple servers on different ports for testing.
1908 Only done for examples that actually take arguments and hence are capable of
1909 outputting different streams for each instance on each port.
1910 https://bugzilla.gnome.org/show_bug.cgi?id=742115
1912 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
1914 * gst/rtsp-server/rtsp-client.c:
1915 * gst/rtsp-server/rtsp-client.h:
1916 rtsp-client: Add a send_message default signal handler
1917 This allows subclasses to easily hook into the response sending
1918 mechanism without doing everything from a signal, which seems
1919 awkward from subclasses.
1921 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
1924 Automatic update of common submodule
1925 From ef1ffdc to f2c6b95
1927 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1931 configure: add --disable-examples switch
1932 https://bugzilla.gnome.org/show_bug.cgi?id=741678
1934 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
1936 * examples/.gitignore:
1937 * examples/Makefile.am:
1938 * examples/test-video-rtx.c:
1939 examples: add a retransmisison example implementing RFC4588
1940 Currently only SSRC-multiplexed rtx streams are supported
1942 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
1944 * gst/rtsp-server/rtsp-stream.c:
1945 rtsp-stream: Fix some minor memory leaks
1947 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
1949 * gst/rtsp-server/rtsp-media.c:
1950 rtsp-media: Some minor cleanup
1952 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
1954 * gst/rtsp-server/rtsp-stream.c:
1955 rtsp-stream: Fix compiler warnings
1956 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
1957 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1959 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
1960 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1963 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
1965 * docs/libs/gst-rtsp-server-sections.txt:
1966 * gst/rtsp-server/rtsp-media-factory.c:
1967 * gst/rtsp-server/rtsp-media-factory.h:
1968 * gst/rtsp-server/rtsp-media.c:
1969 * gst/rtsp-server/rtsp-media.h:
1970 * gst/rtsp-server/rtsp-sdp.c:
1971 * gst/rtsp-server/rtsp-stream.c:
1972 * gst/rtsp-server/rtsp-stream.h:
1973 media: implement ssrc-multiplexed retransmission support
1974 based off RFC 4588 and the server-rtpaux example in -good
1976 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
1978 * gst/rtsp-server/rtsp-client.c:
1979 * gst/rtsp-server/rtsp-stream-transport.c:
1980 * gst/rtsp-server/rtsp-stream.c:
1981 rtsp: Ref transports in hash table.
1982 Also ref streams for transports.
1983 This solves a crash when reciving a rtcp after teardown but before
1985 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
1987 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
1990 Automatic update of common submodule
1991 From 7bb2bce to ef1ffdc
1993 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
1995 * gst/rtsp-server/rtsp-client.c:
1996 client: refactor cleanup of cached media
1998 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
2000 * tests/check/gst/client.c:
2002 The session leak is now fixed, lets remove those FIXME comments.
2004 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
2006 * tests/check/gst/rtspserver.c:
2007 tests: Test to setup two sessions on one connection
2008 https://bugzilla.gnome.org/show_bug.cgi?id=739112
2010 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
2012 * tests/check/gst/rtspserver.c:
2013 tests: Test setup with tcp transport
2014 https://bugzilla.gnome.org/show_bug.cgi?id=739112
2016 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
2018 * gst/rtsp-server/rtsp-client.c:
2019 client: Configure transport after creating session media
2020 The default implementation of configure_client_transport() in
2021 rtsp-client uses the session media when it chooses channels for
2022 interleaved traffic.
2023 https://bugzilla.gnome.org/show_bug.cgi?id=739112
2025 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
2027 * gst/rtsp-server/rtsp-client.c:
2028 * gst/rtsp-server/rtsp-session-media.c:
2029 client: Stop caching media in client when doing setup
2030 If the media has been managed by a session media, it should not be
2031 cached in the client any longer. The GstRTSPSessionMedia object is now
2032 responsible for unpreparing the GstRTSPMedia object using
2033 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
2035 https://bugzilla.gnome.org/show_bug.cgi?id=739112
2037 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2039 * gst/rtsp-server/rtsp-stream.c:
2040 rtsp-stream: unref srtp decoder when leaving bin
2041 https://bugzilla.gnome.org/show_bug.cgi?id=739481
2043 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2045 * gst/rtsp-server/rtsp-client.c:
2046 rtsp-client: mikey memory leaks
2047 https://bugzilla.gnome.org/show_bug.cgi?id=739383
2049 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
2052 Automatic update of common submodule
2053 From 84d06cd to 7bb2bce
2055 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
2058 Parallelise 'make check-valgrind'
2060 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
2063 Automatic update of common submodule
2064 From a8c8939 to 84d06cd
2066 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
2069 Automatic update of common submodule
2070 From 36388a1 to a8c8939
2072 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
2074 * gst/rtsp-server/rtsp-media.c:
2075 rtsp-media: deactivate media when shutting down from paused
2076 This was only done when going directly from playing.
2077 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
2079 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2081 * gst/rtsp-server/rtsp-client.c:
2082 * gst/rtsp-server/rtsp-context.h:
2083 rtsp-client: add stream transport to context
2084 We add the stream transport to the context so we can get the configured
2085 client stream transport in the setup request signal.
2086 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
2088 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2090 * gst/rtsp-server/rtsp-stream.c:
2091 stream: release lock even not all transports have been removed
2092 We don't want to keep the lock even we return FALSE because not all the
2093 transports have been removed. This could lead into a deadlock.
2094 https://bugzilla.gnome.org/show_bug.cgi?id=737797
2096 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
2098 * gst/rtsp-server/rtsp-sdp.c:
2099 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
2100 These were renamed in GstRTPBasePayload in 1.0
2102 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2104 * gst/rtsp-server/rtsp-client.c:
2105 client: set session media to NULL without the lock
2106 We need to set session medias to NULL without the client lock otherwise
2107 we can end up in a deadlock if another thread is waiting for the lock
2108 and media unprepare is also waiting for that thread to end.
2109 https://bugzilla.gnome.org/show_bug.cgi?id=737690
2111 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
2113 * gst/rtsp-server/rtsp-media.c:
2114 rtsp-media: Set state to UNPREPARING in all cases
2116 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
2118 * gst/rtsp-server/rtsp-media.c:
2119 media: set state to unpreparing when unprepare is initiated
2120 https://bugzilla.gnome.org/show_bug.cgi?id=737675
2122 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
2124 * gst/rtsp-server/rtsp-client.c:
2125 rtsp-client: Remove backlog limit while processings requests
2126 If the backlog limit is kept two cases of deadlocks may be
2127 encountered when streaming over TCP. Without the backlog
2128 limit this deadlocks can not happen, at the expence of
2130 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
2132 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
2134 * gst/rtsp-server/rtsp-client.c:
2135 rtsp-client: do not free main context before rtsp watch
2136 https://bugzilla.gnome.org/show_bug.cgi?id=737110
2138 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
2140 * tests/check/gst/rtspserver.c:
2141 tests: Extend unit test timeout to accomodate for valgrind
2142 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
2144 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
2146 * gst/rtsp-server/rtsp-client.c:
2147 * gst/rtsp-server/rtsp-session.c:
2148 * gst/rtsp-server/rtsp-stream-transport.c:
2149 rtsp-*: Treat sending packets to clients as keepalive
2150 As long as gst-rtsp-server can successfully send RTP/RTCP data to
2151 clients then the client must be reading. This change makes the server
2152 timeout the connection if the client stops reading.
2153 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
2155 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
2157 * gst/rtsp-server/rtsp-client.c:
2158 rtsp-client: Allow backlog to grow while expiring session
2159 Allow the send backlog in the RTSP watch to grow to unlimited size while
2160 attempting to bring the media pipeline to NULL due to a session
2161 expiring. Without this change the appsink element cannot change state
2162 because it is blocked while rendering data in the new_sample callback.
2163 This callback will block until it has successfully put the data into the
2164 send backlog. There is a chance that the send backlog is full at this
2165 point which means that the callback may block for a long time, possibly
2166 forever. Therefore the media pipeline may also be prevented from
2167 changing state for a long time.
2168 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
2170 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
2172 * gst/rtsp-server/rtsp-client.c:
2173 rtsp-client: Make old compilers happy
2174 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
2175 Just in case that guint8 doesn't fit in a pointer. Just in case ...
2177 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
2179 * gst/rtsp-server/rtsp-client.c:
2180 client: raise the backlog limits before pausing
2181 We need to raise the backlog limits before pausing the pipeline or else
2182 the appsink might be blocking in the render method in wait_backlog() and
2183 we would deadlock waiting for paused.
2184 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
2186 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
2188 * gst/rtsp-server/rtsp-client.c:
2189 client: make define for the WATCH_BACKLOG
2190 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
2192 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
2194 * gst/rtsp-server/rtsp-client.c:
2195 client: simplify session transport handling
2196 link/unlink of the transport in a session was done to keep track of all
2197 TCP transports and to send RTP/RTCP data to the streams. We can simplify
2198 that by putting all the TCP transports in a hashtable indexed with the
2200 We also don't need to link/unlink the transports when we pause/resume
2201 the streams. The same effect is already achieved when we pause/play the
2202 media. Indeed, when we pause the media, the transport is removed from
2203 the media and the callbacks will not be called anymore.
2204 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
2206 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
2208 * gst/rtsp-server/rtsp-stream-transport.c:
2209 * gst/rtsp-server/rtsp-stream-transport.h:
2210 stream-transport: make method to handle received data
2211 Make a method to handle the data received on a channel. It sends the
2212 data to the stream of the transport on the RTP or RTCP pads based on
2215 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
2217 * examples/test-mp4.c:
2218 test: add example of dumping RTCP reports
2220 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
2222 * gst/rtsp-server/rtsp-media.c:
2223 * gst/rtsp-server/rtsp-stream.c:
2224 * gst/rtsp-server/rtsp-stream.h:
2225 rtsp-media: Make sure that sequence numbers are monotonic after pause
2226 The sequence number is not monotonic for RTP packets after pause. The
2227 reason is basepayloader generates a randon sequence number when the
2228 pipeline goes from ready to pause. With this fix generation of sequence
2229 number will be monotonic when going from pause to play request.
2230 https://bugzilla.gnome.org/show_bug.cgi?id=736017
2232 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
2234 * gst/rtsp-server/rtsp-client.c:
2235 rtsp-client: Protect saved clients watch with a mutex
2236 Fixes a crash when close() is called while merging clients
2237 in handle_tunnel(). In that case close() would destroy the
2238 watch while it is still being used in handle_tunnel().
2239 https://bugzilla.gnome.org/show_bug.cgi?id=735570
2241 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
2243 * gst/rtsp-server/rtsp-stream.c:
2244 rtsp-stream: Remove the multicast group udp sources when removing from the bin
2246 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
2248 * gst/rtsp-server/rtsp-media.c:
2249 * gst/rtsp-server/rtsp-stream.c:
2250 * gst/rtsp-server/rtsp-stream.h:
2251 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
2252 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
2253 seeking and will always continue counting the time. This leads to
2254 the NPT after a backwards seek to be something completely different
2255 to the actual seek position.
2256 https://bugzilla.gnome.org/show_bug.cgi?id=732644
2258 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
2260 * examples/test-appsrc.c:
2261 examples: fix another reference leak
2262 gst_rtsp_media_get_element() returns a new ref.
2264 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2266 * examples/test-appsrc.c:
2267 examples: unref element after usage
2268 gst_bin_get_by_name_recurse_up() returns an element
2269 reference that must be unreffed after usage.
2270 https://bugzilla.gnome.org/show_bug.cgi?id=734546
2272 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
2274 * gst/rtsp-server/rtsp-media.c:
2275 signals: Fix copy-pasto in target-state signal offset
2277 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
2281 Makefile: Add usage of build-checks step
2282 Allows building checks without running them
2284 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
2286 * gst/rtsp-server/rtsp-stream.c:
2287 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
2288 When a UDP multicast transport is used it is expected that the server listens
2289 for RTP and RTCP packets on the multicast group with the corresponding port.
2290 Without this we will never get RTCP packets from clients in multicast mode.
2291 https://bugzilla.gnome.org/show_bug.cgi?id=732238
2293 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
2298 === release 1.4.0 ===
2300 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
2306 * gst-rtsp-server.doap:
2309 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
2311 * gst/rtsp-server/rtsp-media.h:
2312 media: correct misspelled words in description
2313 https://bugzilla.gnome.org/show_bug.cgi?id=733244
2315 === release 1.3.91 ===
2317 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
2323 * gst-rtsp-server.doap:
2326 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
2328 * docs/libs/gst-rtsp-server-sections.txt:
2331 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
2333 * gst/rtsp-server/rtsp-server.c:
2334 server: implement client REMOVE filter
2336 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
2338 * gst/rtsp-server/rtsp-client.c:
2339 * gst/rtsp-server/rtsp-client.h:
2340 client: expose _close() method
2341 Expose a previously internal close method to close the client
2344 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
2346 * gst/rtsp-server/rtsp-session-pool.c:
2347 session-pool: signal session-removed outside of the lock
2348 Release the lock before emiting the session-removed signal.
2350 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
2352 * gst/rtsp-server/rtsp-client.c:
2353 * gst/rtsp-server/rtsp-server.c:
2354 * gst/rtsp-server/rtsp-session-pool.c:
2355 * gst/rtsp-server/rtsp-session.c:
2356 * gst/rtsp-server/rtsp-stream.c:
2357 filter: Release lock in filter functions
2358 Release the object lock before calling the filter functions. We need to
2359 keep a cookie to detect when the list changed during the filter
2360 callback. We also keep a hashtable to make sure we only call the filter
2361 function once for each object in case of concurrent modification.
2362 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
2364 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
2366 * gst/rtsp-server/rtsp-client.c:
2367 client: check if watch is set in handle_teardown()
2368 The unit tests run without a watch
2370 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
2372 * tests/check/gst/client.c:
2373 client tests: send teardown to cleanup session
2375 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
2377 * tests/check/gst/rtspserver.c:
2378 server tests: send teardown to cleanup session
2380 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
2382 * gst/rtsp-server/rtsp-client.c:
2383 client: keep ref to client for the session removed handler
2384 This extra ref will be dropped when all client sessions have been
2385 removed. A session is removed when a client sends teardown, closes its
2386 endpoint of the TCP connection or the sessions expires.
2387 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
2389 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
2391 * gst/rtsp-server/rtsp-client.c:
2392 * gst/rtsp-server/rtsp-session.c:
2393 * tests/check/gst/client.c:
2394 client: manage media in session as a last step
2395 Once we manage a media in a session, we can't unmanage it anymore
2396 without destroying it. Therefore, first check everything before we
2397 manage the media, otherwise if something is wrong we have no way to
2399 If we created a new session and something went wrong, remove the session
2400 again. Fixes a leak in the unit test.
2402 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
2404 * examples/test-mp4.c:
2405 * examples/test-ogg.c:
2406 examples: print 'stream ready at url' for mp4 and ogg example
2408 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
2410 * gst/rtsp-server/rtsp-client.c:
2411 * gst/rtsp-server/rtsp-sdp.c:
2412 rtsp: fix for MIKEY api change
2414 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
2416 * gst/rtsp-server/rtsp-client.c:
2417 client: free watch context only once
2418 The watch context is freed when the source is destroyed. Avoids
2419 a CRITICAL when we try to unref the context twice.
2421 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
2423 * gst/rtsp-server/rtsp-client.c:
2426 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
2428 * gst/rtsp-server/rtsp-client.c:
2429 client: protect sessions with lock
2430 Protect the list of sessions with the lock.
2431 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
2433 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
2435 * gst/rtsp-server/rtsp-client.c:
2436 Client: keep a ref to the session
2437 Don't just keep a weak ref to the session objects but use a hard ref. We
2438 will be notified when a session is removed from the pool (expired) with
2439 the new session-removed signal.
2440 Don't automatically close the RTSP connection when all the sessions of
2441 a client are removed, a client can continue to operate and it can create
2442 a new session if it wants. If you want to remove the client from the
2443 server, you have to use gst_rtsp_server_client_filter() now.
2444 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
2445 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
2447 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
2449 * gst/rtsp-server/rtsp-session-pool.c:
2450 * gst/rtsp-server/rtsp-session-pool.h:
2451 session-pool: add session-removed signal
2452 Add a signal to be notified when a session is removed from the pool.
2454 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
2456 * gst/rtsp-server/Makefile.am:
2457 * gst/rtsp-server/rtsp-server.h:
2458 Make rtsp-server.h a single-include header, use it for G-I
2459 https://bugzilla.gnome.org/show_bug.cgi?id=732411
2461 === release 1.3.90 ===
2463 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
2469 * gst-rtsp-server.doap:
2472 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
2474 * gst/rtsp-server/rtsp-stream.c:
2475 stream: crypto can be NULL
2477 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
2479 * gst/rtsp-server/rtsp-client.c:
2480 * gst/rtsp-server/rtsp-media.c:
2481 * gst/rtsp-server/rtsp-mount-points.c:
2482 introspection: add missing allow-none annotations
2483 https://bugzilla.gnome.org/show_bug.cgi?id=730952
2485 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
2487 * gst/rtsp-server/rtsp-address-pool.c:
2488 * gst/rtsp-server/rtsp-media.c:
2489 * gst/rtsp-server/rtsp-session-media.c:
2490 * gst/rtsp-server/rtsp-session-pool.c:
2491 * gst/rtsp-server/rtsp-stream-transport.c:
2492 * gst/rtsp-server/rtsp-stream.c:
2493 * gst/rtsp-server/rtsp-token.c:
2494 introspection: add (nullable) annotations to return values
2495 https://bugzilla.gnome.org/show_bug.cgi?id=730952
2497 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
2499 * gst/rtsp-server/rtsp-client.c:
2500 * gst/rtsp-server/rtsp-stream.c:
2501 gi: improve annotations
2502 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
2504 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
2506 * gst/rtsp-server/rtsp-client.c:
2507 * gst/rtsp-server/rtsp-media-factory.c:
2508 * gst/rtsp-server/rtsp-media.c:
2509 * gst/rtsp-server/rtsp-server.c:
2510 signals: use generic marshal function
2511 Use the generic C marshal function.
2512 Use more explicit type instead of G_TYPE_POINTER
2514 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
2516 * gst/rtsp-server/rtsp-context.h:
2517 context: add type macro
2519 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
2521 * gst/rtsp-server/rtsp-client.c:
2522 * gst/rtsp-server/rtsp-sdp.c:
2523 * gst/rtsp-server/rtsp-sdp.h:
2524 sdp: hide key length defines
2525 They don't have a namespace.
2527 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
2532 === release 1.3.3 ===
2534 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
2540 * gst-rtsp-server.doap:
2543 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2545 * gst/rtsp-server/rtsp-client.c:
2546 * gst/rtsp-server/rtsp-sdp.c:
2547 * gst/rtsp-server/rtsp-sdp.h:
2548 mikey: add different key length parameters
2549 Add encryption and authentication key length parameters to MIKEY. For
2550 the encoders, the key lengths are obtained from the cipher and auth
2551 algorithms set in the caps. For the decoders, they are obtained while
2552 parsing the key management from the client.
2553 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
2555 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
2557 * tests/check/gst/stream.c:
2558 stream tests: Make sure we get right multicast address from stream
2559 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
2561 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
2563 * gst/rtsp-server/rtsp-client.c:
2564 client: ref the context until rtsp watch is alive
2565 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
2567 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
2569 * gst/rtsp-server/rtsp-client.c:
2570 client: Destroy the rtsp watch after connection close
2572 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
2574 * gst/rtsp-server/rtsp-media.c:
2575 media: fix confusing comment
2577 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
2579 * gst/rtsp-server/rtsp-session.c:
2580 rtsp-session: Timeout in header.
2581 Adding the possbilty to always have timout in header.
2582 This is configurabe with setting "timeout-always-visible".
2583 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
2585 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
2590 === release 1.3.2 ===
2592 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
2599 * gst-rtsp-server.doap:
2602 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
2605 Automatic update of common submodule
2606 From 211fa5f to 1f5d3c3
2608 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
2610 * gst/rtsp-server/rtsp-client.c:
2611 client: store TCP ports in transport
2612 Store the TCP ports in the transport when we are doing RTSP over TCP.
2613 This way, we can easily get to the ports from the transport.
2614 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
2616 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2618 * gst/rtsp-server/rtsp-stream.c:
2619 stream: add signals for new RTP/RTCP encoders
2620 New signals to allow the user to configure the dynamically created
2622 https://bugzilla.gnome.org/show_bug.cgi?id=730228
2624 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
2626 * gst/rtsp-server/rtsp-media.c:
2627 * gst/rtsp-server/rtsp-media.h:
2628 media: Make suspend()/unsuspend() virtual
2629 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
2631 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2633 * gst/rtsp-server/rtsp-client.c:
2634 client: fix send-message signal marshaller
2635 Use generic marshalling for the send-message signal. It has
2636 two POINTER arguments, not just one.
2637 https://bugzilla.gnome.org/show_bug.cgi?id=729900
2639 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
2641 * tests/check/gst/media.c:
2642 tests: add and remove pads only once
2643 In this test we simulate a dynamic pad by watching the caps event.
2644 Because of renegotiation in the base payloader now, this caps is sent
2645 multiple times but we can only deal with 1 invocation, use a variable to
2646 only 'add and remove' the pad once.
2648 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
2650 * tests/check/gst/rtspserver.c:
2651 tests: add unit test for correct handling of Require headers
2652 https://bugzilla.gnome.org/show_bug.cgi?id=729426
2654 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
2656 * gst/rtsp-server/rtsp-client.c:
2657 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
2658 Servers must handle Require headers and must report a failure
2659 if they don't handle any of the Required options, see RFC 2326,
2660 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
2661 https://bugzilla.gnome.org/show_bug.cgi?id=729426
2663 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
2668 === release 1.3.1 ===
2670 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
2676 * gst-rtsp-server.doap:
2679 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
2682 Automatic update of common submodule
2683 From bcb1518 to 211fa5f
2685 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
2690 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
2692 * tests/check/gst/sessionmedia.c:
2693 tests: fix memory leak in sessionmedia unit test
2695 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
2697 * gst/rtsp-server/rtsp-client.c:
2698 client: emit a signal before sending a message
2699 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
2701 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
2703 * gst/rtsp-server/rtsp-client.c:
2704 client: pass context to send_message
2705 Pass the current context to send_message, we will need it later.
2707 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
2709 * gst/rtsp-server/rtsp-client.c:
2710 client: fix typo in comment
2712 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
2714 * gst/rtsp-server/rtsp-media.c:
2715 media: Do not stop thread twice if default_prepare() fails
2717 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
2719 * gst/rtsp-server/rtsp-client.c:
2720 client: set the watch to flushing before going to NULL
2721 First set the watch to flushing so that we unblock any current and
2722 future attempt to send data on the watch, Then set the pipeline to
2724 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
2726 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
2728 * gst/rtsp-server/rtsp-session-pool.c:
2729 * tests/check/gst/sessionpool.c:
2730 rtsp-session-pool: Fixes annotation
2731 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
2732 in the sessionpool test.
2733 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
2735 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
2737 * gst/rtsp-server/rtsp-media.c:
2738 * gst/rtsp-server/rtsp-media.h:
2739 media: make media_prepare virtual
2740 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
2742 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
2744 * gst/rtsp-server/rtsp-media.c:
2745 * tests/check/gst/media.c:
2746 media: stop the thread in more error cases
2748 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
2750 * gst/rtsp-server/rtsp-media.c:
2751 * tests/check/gst/media.c:
2752 media: allow NULL as the thread
2753 Use the default context whan passing a NULL thread.
2755 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
2757 * gst/rtsp-server/rtsp-client.c:
2758 rtsp-client: indent cleanup
2759 Coverity was moaning about unreachable code, and I think it was just
2760 confused by { being before the label. We'll see if it pops up again.
2763 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
2765 * gst/rtsp-server/rtsp-client.c:
2766 * gst/rtsp-server/rtsp-media.c:
2767 client: Add drop-backlog property
2768 When we have too many messages queued for a client (currently hardcoded
2769 to 100) we overflow and drop the messages. Add a drop-backlog property
2770 to control this behaviour. Setting this property to FALSE will retry
2771 to send the messages to the client by waiting for more room in the
2773 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
2775 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
2777 * gst/rtsp-server/rtsp-client.c:
2778 client: support for POST before GET when setting up a tunnel
2780 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
2782 * gst/rtsp-server/rtsp-client.c:
2783 client: remove watch of the second client after http tunnel setup
2784 The second client will be freed after the HTTP tunnel has been set up.
2785 Make sure it's RTSP watch is never dispatched again.
2786 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
2788 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
2790 * gst/rtsp-server/rtsp-media.c:
2791 * tests/check/gst/media.c:
2792 media: Make media_prepare() fail if port allocation fails
2793 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
2795 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
2797 * tests/check/gst/media.c:
2798 media test: cleanup the thread pool in tests
2800 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
2802 * gst/rtsp-server/rtsp-media.c:
2803 * tests/check/gst/media.c:
2804 rtsp-media: Unblock blocked streams in unprepare
2805 The streams will be blocked when a live media is prepared.
2806 The streams should be unblocked in gst_rtsp_media_unprepare.
2807 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
2809 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
2811 * gst/rtsp-server/rtsp-media.c:
2812 media: release the state lock when going to NULL
2813 Set our state to UNPREPARING and release the state-lock before
2814 setting the pipeline to the NULL state. This way, any pad-added
2815 callback will be able to take the state-lock and check that we are now
2816 unpreparing instead of deadlocking.
2817 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
2819 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
2821 * gst/rtsp-server/rtsp-media.c:
2822 media: protect status with lock
2823 Make sure we only update the status with the lock.
2825 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
2827 * gst/rtsp-server/rtsp-client.c:
2828 * gst/rtsp-server/rtsp-sdp.c:
2829 rtsp: update for MIKEY API changes
2831 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
2833 * gst/rtsp-server/rtsp-client.c:
2834 client: parse the mikey response from the client
2835 Parse the mikey response from the client and update the policy for
2838 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
2840 * gst/rtsp-server/rtsp-stream.c:
2841 * gst/rtsp-server/rtsp-stream.h:
2842 stream: add method to set crypto info
2843 Make a method to configure the crypto information of a stream.
2844 Set udpsrc in READY instead of PAUSED so that we can configure caps
2847 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
2849 * gst/rtsp-server/rtsp-client.c:
2850 client: cleanup error paths
2852 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
2854 * gst/rtsp-server/rtsp-media.c:
2857 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
2859 * examples/test-video.c:
2860 test: enable SRTP only on RTSPS
2861 We only want to enable SRTP when doing rtsp over TLS so that we can
2862 exchange the keys in a secure way.
2864 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
2866 * examples/test-video.c:
2867 test: print an error on failure
2869 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
2872 * examples/test-video.c:
2873 * gst/rtsp-server/rtsp-sdp.c:
2874 * gst/rtsp-server/rtsp-stream.c:
2875 * tests/check/Makefile.am:
2876 stream: add SRTP support
2877 Install srtp encoder and decoder elements in rtpbin
2880 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2882 * tests/check/Makefile.am:
2883 * tests/check/gst/sessionpool.c:
2884 tests: Add unit tests for sessionpool
2885 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
2887 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2889 * tests/check/gst/threadpool.c:
2890 tests: Improve code coverage of rtsp-threadpool tests
2891 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
2893 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2895 * tests/check/gst/sessionmedia.c:
2896 tests: Improve code coverage for rtsp-session-media
2897 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
2899 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2901 gobject-introspection: Add annotations to support language bindings
2902 In addition a few cosmetic changes:
2903 * Adjust the order of arguments
2904 * Fix typo: occured -> occurred
2905 * Fix indentation after Return:-clauses
2906 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
2908 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2910 * gst/rtsp-server/rtsp-stream.c:
2911 rtsp-stream: Don't mix IPv4 and IPv6 addresses
2912 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
2914 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
2916 * gst/rtsp-server/rtsp-stream.c:
2917 stream: take caps after the session manager
2918 Take the caps for the SDP after they leave the rtpbin so that we can
2919 also get the properties added by rtpbin elements.
2921 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
2923 * gst/rtsp-server/rtsp-stream.c:
2924 stream: release lock while pushing out packets
2925 Keep a cache of the transports and use this to iterate the transport
2926 while pushing packets. This allows us to release the lock early.
2927 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
2929 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
2931 * gst/rtsp-server/rtsp-client.c:
2932 * gst/rtsp-server/rtsp-client.h:
2933 rtsp-client: vmethod for modifying tunnel GET response
2934 Add a vmethod tunnel_http_response where the response to the HTTP GET
2935 for tunneled connections can be modified.
2936 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
2938 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
2940 * gst/rtsp-server/rtsp-sdp.c:
2941 sdp: make 1 media line per profile
2942 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
2943 line in the SDP for each profile. The client is then supposed to pick
2944 one of the profiles in the SETUP request. Because the m= lines have the
2945 same pt, the client also knows that only 1 option is possible.
2947 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
2949 * gst/rtsp-server/rtsp-media-factory.c:
2950 * gst/rtsp-server/rtsp-media-factory.h:
2951 * gst/rtsp-server/rtsp-media.c:
2952 factory: add profile property and pass to media and streams
2954 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
2956 * examples/test-multicast.c:
2957 * gst/rtsp-server/rtsp-sdp.c:
2958 sdp: pass multicast connection for multicast-only stream
2959 Pass the multicast address of the stream in the connection info in the
2960 SDP so that clients try a multicast connection first.
2961 Only allow multicast connections in the test-multicast example. Also
2962 increase the TTL a little.
2964 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2967 .gitignore: Ignore gcov intermediate files
2968 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
2970 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
2972 * gst/rtsp-server/rtsp-stream.c:
2973 stream: release some locks in error cases
2975 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2977 docs: Enable and fix gtk-doc warnings
2978 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
2979 * addresspool/mediafactory: Add missing annotation colon
2980 * stream: Annotate return value
2981 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
2983 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
2986 Automatic update of common submodule
2987 From fe1672e to bcb1518
2989 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
2992 Automatic update of common submodule
2993 From 1a07da9 to fe1672e
2995 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
2997 * examples/Makefile.am:
2998 examples: use LDADD for libs instead of LDFLAGS
3000 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
3003 configure: make sure releases are in .doap file
3005 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
3007 * examples/test-cgroups.c:
3008 examples: test-cgroups: don't put code with side effects into g_assert()
3009 The g_assert() might get compiled out with the right
3010 compiler/preprocessor flags.
3012 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
3014 * examples/.gitignore:
3015 examples: add cgroup test binary to .gitignore
3017 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
3019 * examples/test-cgroups.c:
3020 examples: fix cgroup test build
3021 Fixes build failure caused by compiler warning:
3022 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
3024 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
3027 .gitignore: ignore temp files created in the course of 'make check'
3029 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
3031 * gst/rtsp-server/rtsp-media.c:
3032 rtsp-media: don't loose frames handling new PLAY request
3033 If client supplied a range check if the range specifies the start point.
3034 If not, then do an accurate seek to the current position. If a start
3035 point was specified do do a key unit seek to make sure the streaming
3036 starts with decodeable frames.
3037 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
3039 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
3041 * gst/rtsp-server/rtsp-media.c:
3042 Revert "media: only flush when setting a new start position"
3043 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
3044 We need to do the flush in all cases, demuxer block currently for
3047 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
3049 * gst/rtsp-server/rtsp-media.c:
3050 media: only flush when setting a new start position
3051 Only flush the pipeline when we change the start position with
3053 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
3055 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
3057 * gst/rtsp-server/rtsp-stream.c:
3058 stream: set ttl-mc before adding the socket
3059 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
3060 never be set on socket.
3061 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
3063 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
3065 * gst/rtsp-server/rtsp-media.c:
3066 media: stop thread if media is already prepared
3067 in gst_rtsp_media_prepare() the thread is not used if media is already
3068 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
3070 https://bugzilla.gnome.org/show_bug.cgi?id=724182
3072 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
3075 build: Ship gst-rtsp-server.doap file
3077 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
3079 * tests/check/gst/rtspserver.c:
3080 tests: Fix another compiler warning with gcc
3082 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
3084 * gst/rtsp-server/rtsp-client.c:
3085 * gst/rtsp-server/rtsp-mount-points.c:
3086 * gst/rtsp-server/rtsp-stream.c:
3087 * tests/check/gst/client.c:
3088 rtsp-server: Fix lots of compiler warnings with clang
3090 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
3093 * gst-rtsp-server.doap:
3094 * tests/Makefile.am:
3095 configure: Synchronise with the configure scripts of the other modules
3097 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
3100 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
3102 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
3104 * gst/rtsp-server/rtsp-media.c:
3105 * gst/rtsp-server/rtsp-stream.c:
3106 Revert "rtsp-server: support build against last stable release"
3107 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
3108 Let us require 1.2.3 now, which is going to be released in a few
3111 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
3113 * gst/rtsp-server/rtsp-session-media.c:
3114 * gst/rtsp-server/rtsp-stream-transport.c:
3115 session: improve RTP-Info
3116 Ignore streams that can't generate RTP-Info instead of failing.
3117 Don't return the empty string when all streams are unconfigured but
3118 return NULL so that we don't generate and empty RTP-Info header.
3119 Improve docs a little.
3121 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
3123 * gst/rtsp-server/rtsp-session-media.c:
3124 Don't free rtpinfo GString when it is NULL
3125 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
3127 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
3129 * gst/rtsp-server/rtsp-media.c:
3130 media: only set keyframe flag when modifying start
3131 Only set the keyframe flag when we modify the start position. The
3132 keyframe flag should probably be ignored when no change is requested but
3133 until we can claim this is all documented properly and all demuxer
3134 implement this, avoid setting the flag.
3135 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
3137 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
3139 * gst/rtsp-server/rtsp-thread-pool.c:
3140 thread-pool: Unref source after mainloop has quit to avoid races in GLib
3141 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
3143 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
3145 * gst/rtsp-server/rtsp-stream.c:
3146 stream: handle NULL seqnum and rtptime arguments
3148 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
3150 * gst/rtsp-server/rtsp-thread-pool.c:
3151 * tests/check/gst/threadpool.c:
3152 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
3153 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
3155 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
3157 * gst/rtsp-server/rtsp-stream.c:
3158 stream: add fallback for missing stats property
3159 Use a fallback when the payloader does not have a stats property
3160 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
3162 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
3165 Automatic update of common submodule
3166 From f7bc1c3 to 1a07da9
3168 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
3170 * gst/rtsp-server/rtsp-stream.c:
3171 stream: don't leak stats structure
3172 Don't leak the stats structure and deal with NULL stats.
3174 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
3176 * gst/rtsp-server/rtsp-stream.c:
3177 stream: Get rtpinfo properties atomically from payloader
3178 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
3180 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
3182 * gst/rtsp-server/rtsp-media.c:
3183 media: refactor state change functions and signals
3184 Make functions to set the target state and the pipeline state and emit
3185 the signals from those functions.
3187 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
3189 * gst/rtsp-server/rtsp-media.c:
3190 * gst/rtsp-server/rtsp-media.h:
3191 media: add signal to notify of pending state changes
3193 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
3195 * gst/rtsp-server/rtsp-media.c:
3196 * gst/rtsp-server/rtsp-stream.c:
3197 rtsp-server: support build against last stable release
3198 Until 1.2.3 is out with the new get_type function and we
3201 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
3203 * gst/rtsp-server/rtsp-stream.c:
3204 stream: fix compilation
3206 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
3208 * gst/rtsp-server/rtsp-media.c:
3209 * gst/rtsp-server/rtsp-media.h:
3210 * gst/rtsp-server/rtsp-stream.c:
3211 * gst/rtsp-server/rtsp-stream.h:
3212 stream: add property to configure profiles
3214 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
3216 * gst/rtsp-server/rtsp-client.c:
3217 client: let stream check supported transport
3218 Delegate the check if a transport is allowed to the stream.
3219 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
3221 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
3223 * gst/rtsp-server/rtsp-stream.c:
3224 * gst/rtsp-server/rtsp-stream.h:
3225 stream: add method to check supported transport
3226 Add a method to check if a transport is supported
3228 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
3231 configure.ac: Only check for gstreamer-check, not check
3232 We include check in gstreamer-check since quite some time now.
3234 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
3236 * gst/rtsp-server/rtsp-session-media.c:
3237 * gst/rtsp-server/rtsp-stream-transport.c:
3238 * gst/rtsp-server/rtsp-stream.c:
3239 * gst/rtsp-server/rtsp-stream.h:
3240 stream: return clock-rate from get_rtpinfo
3241 And use it to correct the rtptime to the requested start-time.
3242 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
3244 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
3246 * gst/rtsp-server/rtsp-session-media.c:
3247 * gst/rtsp-server/rtsp-stream-transport.c:
3248 * gst/rtsp-server/rtsp-stream-transport.h:
3249 session-media: calculate start-time
3251 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
3253 * gst/rtsp-server/rtsp-stream-transport.c:
3254 * gst/rtsp-server/rtsp-stream.c:
3255 * gst/rtsp-server/rtsp-stream.h:
3256 stream: also return the running-time
3257 Return the running-time in the rtpinfo as well.
3259 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
3261 * gst/rtsp-server/rtsp-client.c:
3262 * gst/rtsp-server/rtsp-session-media.c:
3263 * gst/rtsp-server/rtsp-session-media.h:
3264 * gst/rtsp-server/rtsp-stream-transport.c:
3265 * gst/rtsp-server/rtsp-stream-transport.h:
3266 session-media: let the session-media make the RTPInfo
3267 Add method to create the RTPInfo for a stream-transport.
3268 Add method to create the RTPInfo for all stream-transports in a
3270 Use the session-media RTPInfo code in client. This allows us to refactor
3271 another method to link the TCP callbacks.
3273 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
3275 mount-points: sort sequence before g_sequence_lookup
3276 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
3277 sort sequence if dirty, otherwise lookup will fail.
3278 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
3280 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
3283 configure: rename package from gst-rtsp to gst-rtsp-server
3284 To match git module name and avoid confusion with the
3285 rtsp lib in gst-plugins-base and rtsp plugin in -good.
3287 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
3290 configure: bump core/base/good requirement to 1.2.0
3291 Bump to released stable version and make implicit
3292 requirements explicit.
3294 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
3299 Fix broken gettext setup which is not used anyway
3301 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
3304 Automatic update of common submodule
3305 From dbedaa0 to d48bed3
3307 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
3309 * gst/rtsp-server/rtsp-client.c:
3310 * gst/rtsp-server/rtsp-media.c:
3311 * gst/rtsp-server/rtsp-media.h:
3312 media: add setup_sdp vmethod
3313 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
3314 gst_rtsp_media_setup_sdp.
3315 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
3317 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
3319 * gst/rtsp-server/rtsp-stream.c:
3320 rtsp-stream: Check return value of sscanf
3321 streamid is only valid if sscanf matched something.
3323 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
3325 * gst/rtsp-server/rtsp-client.c:
3326 rtsp-client: Fix iteration
3327 Wouldn't even enter the code block otherwise (i++ was used as the check
3328 and not the postfix).
3330 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
3332 * gst/rtsp-server/rtsp-client.c:
3333 * gst/rtsp-server/rtsp-client.h:
3334 client: add vmethod to configure media and streams
3335 Implement a vmethod that can be used to configure the media and the
3336 streams based on the current context. Handle the blocksize handling in
3337 the default handler.
3338 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
3340 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
3343 Make git ignore more unit test binaries
3345 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
3347 * gst/rtsp-server/rtsp-address-pool.h:
3348 * gst/rtsp-server/rtsp-auth.h:
3349 * gst/rtsp-server/rtsp-client.h:
3350 * gst/rtsp-server/rtsp-context.h:
3351 * gst/rtsp-server/rtsp-media-factory-uri.h:
3352 * gst/rtsp-server/rtsp-media-factory.h:
3353 * gst/rtsp-server/rtsp-media.h:
3354 * gst/rtsp-server/rtsp-mount-points.h:
3355 * gst/rtsp-server/rtsp-server.h:
3356 * gst/rtsp-server/rtsp-session-media.h:
3357 * gst/rtsp-server/rtsp-session-pool.h:
3358 * gst/rtsp-server/rtsp-session.h:
3359 * gst/rtsp-server/rtsp-stream-transport.h:
3360 * gst/rtsp-server/rtsp-stream.h:
3361 * gst/rtsp-server/rtsp-thread-pool.h:
3362 * gst/rtsp-server/rtsp-token.h:
3363 rtsp-server: add padding to many public structures
3364 Not mini objects though, since they are not subclassable
3365 anyway, nor kept on the stack or inlined in a structure.
3367 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
3369 media: add new create_rtpbin vmethod
3370 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
3371 https://bugzilla.gnome.org/show_bug.cgi?id=719734
3373 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
3375 * tests/check/gst/media.c:
3376 tests: fix memory leak, free test's thread pool
3377 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
3379 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
3381 * gst/rtsp-server/rtsp-stream-transport.c:
3382 stream-transport: free url in finalize
3384 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
3386 * gst/rtsp-server/rtsp-media.c:
3387 media: also do state change in suspended state
3389 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
3391 * gst/rtsp-server/rtsp-client.c:
3392 * gst/rtsp-server/rtsp-media.c:
3393 media: also handle prepare and range in suspended state
3394 When we are suspended, we are already prepared.
3395 We can get the range in the suspended state.
3397 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
3399 * tests/check/Makefile.am:
3400 * tests/check/gst/sessionmedia.c:
3401 check: add test for uri in setup
3402 Added unit tests for the new functionality in GstRTSPStreamTransport.
3403 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
3405 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
3407 * gst/rtsp-server/rtsp-client.c:
3408 client: store setup uri and use in PLAY response
3409 Store the uri used when doing the setup and use that in the PLAY
3411 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
3413 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
3415 * gst/rtsp-server/rtsp-stream-transport.c:
3416 * gst/rtsp-server/rtsp-stream-transport.h:
3417 stream-transport: add method to get/set url
3419 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
3421 * gst/rtsp-server/rtsp-client.c:
3422 client: suspend after SDP and unsuspend before PLAYING
3423 Based on patches by Ognyan Tonchev <ognyan@axis.com>
3424 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
3426 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
3428 * gst/rtsp-server/rtsp-media-factory.c:
3429 * gst/rtsp-server/rtsp-media-factory.h:
3430 * gst/rtsp-server/rtsp-media.c:
3431 * gst/rtsp-server/rtsp-media.h:
3432 * gst/rtsp-server/rtsp-session-media.c:
3433 * gst/rtsp-server/rtsp-session.c:
3434 * tests/check/gst/media.c:
3435 * tests/check/gst/mediafactory.c:
3436 media: add suspend modes
3437 Add support for different suspend modes. The stream is suspended right after
3438 producing the SDP and after PAUSE. Different suspend modes are available that
3439 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
3440 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
3441 state and RESET will bring the pipeline to the NULL state.
3442 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
3443 this means that the pipeline needs to be prerolled again.
3444 Base on patches by Ognyan Tonchev <ognyan@axis.com>
3445 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3447 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
3449 * gst/rtsp-server/rtsp-media.c:
3450 media: start live streams in blocked state
3451 Start live streams in the blocked state and make them preroll using the
3452 messages. This ensure that no data is played by the sink until we explicitly
3453 unblock the stream right before going to PLAYING.
3454 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3456 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
3458 * gst/rtsp-server/rtsp-media.c:
3459 media: refactor starting and waiting for preroll
3460 Based on patches from Ognyan Tonchev <ognyan@axis.com>
3461 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3463 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
3465 * gst/rtsp-server/rtsp-stream.c:
3466 * gst/rtsp-server/rtsp-stream.h:
3467 stream: add API to block streams
3468 Add an API to block on the streams and make it post a message.
3469 Based on patch by Ognyan Tonchev <ognyan@axis.com>
3470 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
3472 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
3474 * docs/libs/Makefile.am:
3475 docs: Specify the override file
3476 Even if it's empty (for now) it avoids make distcheck complaining
3478 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
3480 * gst/rtsp-server/rtsp-media.c:
3481 media: move default implementations to where they are used
3483 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
3485 * gst/rtsp-server/rtsp-media.c:
3486 media: take the right lock in gst_rtsp_media_set_pipeline_state()
3487 We need to take the state_lock when calling this method.
3489 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
3491 * gst/rtsp-server/rtsp-media.c:
3492 media: handle add-added on non-bins too
3493 Handle dynamic payloaders that are not bins, as used in the unit-test.
3495 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3497 * gst/rtsp-server/rtsp-media-factory.c:
3498 * gst/rtsp-server/rtsp-media-factory.h:
3499 * gst/rtsp-server/rtsp-media.c:
3500 rtsp-media/-factory: Fix request pad name comments
3501 These must be escaped for gtk-doc to parse the comments without warnings.
3503 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
3505 rtsp-media: remove transports if media is in error status
3506 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
3507 trying to change to GST_STATE_NULL and media is in error status, we
3508 remove all transports.
3509 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
3511 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
3513 * gst/rtsp-server/rtsp-media.c:
3514 rtsp-media: use element metadata to find payloader
3515 Use the element metadata to find the payloader instead of checking
3517 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
3519 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
3521 rtsp-stream: add getter for payload type
3522 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
3523 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
3524 element and create the stream with this one instead of the dynpay%d
3526 https://bugzilla.gnome.org/show_bug.cgi?id=712396
3528 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3530 * gst/rtsp-server/rtsp-client.c:
3531 * gst/rtsp-server/rtsp-context.h:
3532 * gst/rtsp-server/rtsp-media.c:
3533 * gst/rtsp-server/rtsp-mount-points.c:
3534 * gst/rtsp-server/rtsp-server.c:
3535 * gst/rtsp-server/rtsp-token.c:
3536 rtsp-*: Refer to NULL as a constant in comments
3538 https://bugzilla.gnome.org/show_bug.cgi?id=714988
3540 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3542 rtsp-*: Fix type name typos in comments
3543 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
3544 * rtsp-auth: Refer to part of constant name as text
3545 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
3546 * rtsp-session-media: Fix GstRTSPSessionMedia typo
3547 * rtsp-stream: Fix typo when refering to GstBin
3548 https://bugzilla.gnome.org/show_bug.cgi?id=714988
3550 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3553 * docs/libs/gst-rtsp-server-docs.sgml:
3554 * docs/libs/gst-rtsp-server-sections.txt:
3555 docs: Improve documentation
3556 * Include annotation-glossary to quiet gtk-doc
3557 * Rename remaining ClientState -> Context
3558 * Rename object hierarchy file
3559 * Remove stale chapter references
3560 * Add missing function and object references
3561 * Include missing GstRTSPAddressPoolResult
3562 https://bugzilla.gnome.org/show_bug.cgi?id=714988
3564 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
3566 * gst/rtsp-server/rtsp-client.c:
3567 * gst/rtsp-server/rtsp-server.c:
3568 * gst/rtsp-server/rtsp-session-pool.c:
3569 * gst/rtsp-server/rtsp-session.c:
3570 * gst/rtsp-server/rtsp-stream.c:
3571 rtsp-server: sprinkle some allow-none annotations for g-i
3573 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
3575 * gst/rtsp-server/rtsp-stream.c:
3576 * gst/rtsp-server/rtsp-stream.h:
3577 stream: add method to filter transports
3578 Add a method to safely iterate and collect the stream transports
3579 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
3581 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
3583 * gst/rtsp-server/rtsp-client.c:
3584 * gst/rtsp-server/rtsp-server.c:
3585 * gst/rtsp-server/rtsp-session-pool.c:
3586 * gst/rtsp-server/rtsp-session.c:
3587 rtsp: allow NULL func in filters
3588 Passing a null function make the filters return a list of
3591 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
3593 * gst/rtsp-server/rtsp-address-pool.c:
3594 * tests/check/gst/addresspool.c:
3595 address-pool: fix address increment
3596 Use a guint instead of guint8 to increment the address. It's still not
3597 completely correct because a guint might not be able to hold the complete
3598 address range, but that's an enhacement for later.
3599 Add unit test to test improved behaviour.
3600 https://bugzilla.gnome.org/show_bug.cgi?id=708237
3602 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
3604 * gst/rtsp-server/rtsp-client.c:
3605 * tests/check/gst/client.c:
3606 client: allow absolute path in requests
3607 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
3609 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
3611 * gst/rtsp-server/rtsp-client.c:
3612 * gst/rtsp-server/rtsp-client.h:
3613 client: make make_path_from_uri a vmethod
3615 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
3617 * docs/libs/gst-rtsp-server-sections.txt:
3618 * gst/rtsp-server/rtsp-stream.c:
3619 * gst/rtsp-server/rtsp-stream.h:
3620 * tests/check/Makefile.am:
3621 * tests/check/gst/stream.c:
3622 stream: Add functions to get rtp and rtcp sockets
3623 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
3625 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
3627 * gst/rtsp-server/rtsp-context.c:
3628 * gst/rtsp-server/rtsp-context.h:
3629 context: defing a GType for the context
3630 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
3632 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
3634 * gst/rtsp-server/Makefile.am:
3635 * gst/rtsp-server/rtsp-auth.c:
3636 * gst/rtsp-server/rtsp-context.c:
3637 * gst/rtsp-server/rtsp-media.c:
3638 * gst/rtsp-server/rtsp-mount-points.c:
3639 * gst/rtsp-server/rtsp-server.h:
3640 * gst/rtsp-server/rtsp-session-media.c:
3641 * gst/rtsp-server/rtsp-session.c:
3642 * gst/rtsp-server/rtsp-stream.c:
3643 Fixed several GIR warnings
3645 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
3647 * gst/rtsp-server/rtsp-auth.c:
3650 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3652 * tests/check/Makefile.am:
3653 * tests/check/gst/token.c:
3654 tests: Add unit tests for token
3655 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
3657 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3659 * gst/rtsp-server/rtsp-token.c:
3660 token: Validate args for gst_rtsp_token_is_allowed
3661 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
3663 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3665 * gst/rtsp-server/rtsp-token.c:
3666 token: Fix bug when creating empty token
3667 We always want to have a valid GstStructure in the token.
3668 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
3670 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
3672 * gst/rtsp-server/rtsp-thread-pool.c:
3673 thread-pool: avoid race in shutdown
3674 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
3675 don't actually stop the mainloop ever. Solve this race by adding an idle source
3676 to the mainloop that calls the _quit. This way we immediately exit the mainloop
3677 if quit was called before we started it.
3679 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3681 * tests/check/Makefile.am:
3682 * tests/check/gst/permissions.c:
3683 tests: Add unit tests for permissions
3684 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
3686 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3688 * tests/check/gst/mediafactory.c:
3689 tests: Test mediafactory permissions
3690 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
3692 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3694 * gst/rtsp-server/rtsp-permissions.c:
3695 permissions: Fix refcounting when adding/removing roles
3696 Previously a role that was removed was unreffed twice, and when
3697 replacing an existing role the replaced role was freed while still being
3698 referenced. Both bugs are now fixed.
3699 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
3701 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3703 * tests/check/gst/media.c:
3704 * tests/check/gst/mediafactory.c:
3705 * tests/check/gst/rtspserver.c:
3706 tests: Check gst_rtsp_url_parse return value
3707 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
3709 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
3712 Automatic update of common submodule
3713 From 865aa20 to dbedaa0
3715 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
3717 * gst/rtsp-server/rtsp-server.c:
3718 rtsp-server: Fix socket leak
3719 https://bugzilla.gnome.org/show_bug.cgi?id=710088
3721 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
3723 * gst/rtsp-server/rtsp-session-pool.c:
3724 rtsp-session-pool: Make sure session IDs are properly URI-escaped
3725 https://bugzilla.gnome.org/show_bug.cgi?id=643812
3727 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
3729 * examples/.gitignore:
3730 * examples/test-video.c:
3731 examples: fix compilation when WITH_AUTH is defined
3732 https://bugzilla.gnome.org/show_bug.cgi?id=710228
3734 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
3737 gitignore: Add new test binary
3739 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
3741 * tests/check/Makefile.am:
3742 * tests/check/gst/threadpool.c:
3743 thread-pool: Add unit test for the thread pools
3744 https://bugzilla.gnome.org/show_bug.cgi?id=710228
3746 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
3748 * gst/rtsp-server/rtsp-thread-pool.c:
3749 thread-pool: Fix thread leak when reusing threads
3750 https://bugzilla.gnome.org/show_bug.cgi?id=709730
3752 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
3754 * gst/rtsp-server/rtsp-server.c:
3755 * tests/check/gst/rtspserver.c:
3756 tests: fixed racy behavior in rtspserver tests
3757 https://bugzilla.gnome.org/show_bug.cgi?id=710078
3759 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3761 * tests/check/gst/addresspool.c:
3762 tests: Improve address pool unit tests
3763 Add a range with mixed IPV4 and IPV6 addresses to pool.
3764 Get an IPV4 address from an IPV6-only pool.
3765 Get an IPV6 address from an IPV4-only pool.
3766 Reserve a IPV6 address from an IPV4-only pool.
3767 Check for unicast addresses in multicast-only pool.
3768 Check for unicast addresses in uni-/multicast-mixed pool.
3769 https://bugzilla.gnome.org/show_bug.cgi?id=710128
3771 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3773 * gst/rtsp-server/rtsp-client.c:
3774 client: append query string in PAUSE/PLAY/TEARDOWN as well
3776 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
3778 * gst/rtsp-server/rtsp-client.c:
3779 client: Add query to control path
3780 If the SETUP url contains a query it must be appended to the control
3781 path so that it matches any already created stream in the media. The
3782 query will also be appended to the session media path.
3784 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3786 * gst/rtsp-server/rtsp-media.c:
3787 rtsp-media: remove old line
3789 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
3791 * gst/rtsp-server/rtsp-stream.c:
3792 stream: Correct control comparison
3793 https://bugzilla.gnome.org/show_bug.cgi?id=709176
3795 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3797 * gst/rtsp-server/rtsp-media.c:
3798 media: Check dynamically if the pipeline supports seeking
3799 We should not depend on whether or not the pipeline state change
3800 returned NO_PREROLL or not. A media could dynamically change its
3801 element and switch from seekable to non seekable so it's best to test
3802 the seekable nature of the pipeline dynamically when we try to do a seek.
3804 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3806 * gst/rtsp-server/rtsp-media.c:
3807 media: Return FALSE if seeking is not supported
3809 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3811 * gst/rtsp-server/rtsp-media.c:
3812 rtsp-media: don't seek accurate by default
3813 Accurate seeking is perhaps a little overkill in the most common situation and
3814 causes some formats (mp3) over slow media to seek extremely slowly.
3816 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
3818 * tests/check/gst/rtspserver.c:
3819 tests: fix unit test
3820 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
3822 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
3824 * gst/rtsp-server/rtsp-client.c:
3825 client: Reply 400 if media cannot be constructed
3826 Reply 400 Bad Request instead of 503 Service Unavailable if media
3827 cannot be constructed in SETUP.
3828 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
3830 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
3832 * gst/rtsp-server/rtsp-client.c:
3833 client: Send setup reply once only
3834 If find_media() failed in handle_setup_request() two replies was sent.
3835 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
3837 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
3840 Automatic update of common submodule
3841 From 6b03ba7 to 865aa20
3843 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
3845 * gst/rtsp-server/rtsp-server.c:
3846 server: Emit client-connected signal earlier
3847 Emit client-connected before the client ref is given to a GSource,
3848 otherwise client-connected can be emitted after the client object has
3851 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
3853 * gst/rtsp-server/rtsp-address-pool.c:
3854 * gst/rtsp-server/rtsp-address-pool.h:
3855 * gst/rtsp-server/rtsp-stream.c:
3856 * tests/check/gst/addresspool.c:
3857 addresspool: return reason of failure
3858 Let gst_rtsp_address_pool_reserve_address() return the reason why
3859 the address could not be reserved.
3860 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
3862 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
3865 autogen.sh: Sync behaviour with other GStreamer modules
3866 Allows building from outside of tree amongst other things
3868 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
3871 Automatic update of common submodule
3872 From b613661 to 6b03ba7
3874 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
3877 Automatic update of common submodule
3878 From 74a6857 to b613661
3880 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
3883 Automatic update of common submodule
3884 From 01a7a46 to 74a6857
3886 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
3888 * gst/rtsp-server/rtsp-client.c:
3889 client: Do not read beyond end of path string
3890 If the setup was done without a control url, make sure we don't try to read the
3891 non-existing control string and crash.
3893 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3895 * gst/rtsp-server/rtsp-client.c:
3896 client: Fix RTPInfo header
3897 Refactor the method to make the content_base.
3898 Use the content-base and the control url to construct the RTPInfo
3901 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3903 * gst/rtsp-server/rtsp-client.c:
3904 client: map url to path only in describe
3905 Only map the request url to a path in the DESCRIBE method. The SDP then
3906 contains the base and control urls that should be used to SETUP/PAUSE/
3907 PLAY/TEARDOWN the media.
3909 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3911 * gst/rtsp-server/rtsp-client.c:
3912 Revert "client: map URL to path in requests"
3913 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
3914 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
3915 contains the base and control urls which are used in the SETUP, PLAY,
3916 PAUSE and TEARDOWN requests.
3918 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3920 * gst/rtsp-server/rtsp-client.c:
3921 client: map URL to path in requests
3923 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3925 * gst/rtsp-server/rtsp-client.c:
3926 * gst/rtsp-server/rtsp-mount-points.c:
3927 * gst/rtsp-server/rtsp-mount-points.h:
3928 mount-points: make vmethod to make path from uri
3929 Make a vmethod to transform an url into a path. The path is then used to lookup
3930 the factory. This makes it possible to also use other bits of the url, such as
3931 the query parameters, to locate the factory.
3933 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
3935 * gst/rtsp-server/rtsp-thread-pool.c:
3936 * gst/rtsp-server/rtsp-thread-pool.h:
3937 thread-pool: Add cleanup to wait for the threadpool to finish
3938 Also fix race condition if two threads are asking for the first
3939 thread from the thread pool at once. This would case two internal
3940 GThreadPools to be created.
3941 https://bugzilla.gnome.org/show_bug.cgi?id=707753
3943 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
3945 * gst/rtsp-server/rtsp-client.c:
3946 * tests/check/gst/client.c:
3947 client: free threadpool
3948 https://bugzilla.gnome.org/show_bug.cgi?id=707638
3950 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
3952 * tests/check/gst/mountpoints.c:
3953 mountpoints tests: unref matched factories
3954 https://bugzilla.gnome.org/show_bug.cgi?id=707638
3956 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
3958 * tests/check/gst/media.c:
3959 media tests: unref thread pool and caps
3960 https://bugzilla.gnome.org/show_bug.cgi?id=707638
3962 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
3964 * gst/rtsp-server/rtsp-auth.c:
3965 * gst/rtsp-server/rtsp-media-factory.c:
3966 * gst/rtsp-server/rtsp-media.c:
3967 auth, media, media-factory: unref permissions
3968 https://bugzilla.gnome.org/show_bug.cgi?id=707638
3970 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3972 * examples/Makefile.am:
3973 Makefile: add rule for appsrc example
3975 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3977 * examples/test-appsrc.c:
3978 tests: add appsrc example
3979 Add an example on how to use appsrc to feed the server pipeline with data.
3981 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
3983 * gst/rtsp-server/rtsp-client.c:
3984 rtsp-client: remove query part from content-base string
3985 Make sure that after the control url has been resolved, it's
3986 not a part of the query-string.
3987 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
3989 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3991 * gst/rtsp-server/rtsp-client.c:
3992 client: don't check url in response
3993 There is no url or method in the response to check
3995 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3997 * gst/rtsp-server/rtsp-client.c:
3998 * gst/rtsp-server/rtsp-client.h:
3999 Add handle-response signal for when we receive a GET_PARAMETER response
4001 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4003 * gst/rtsp-server/rtsp-server.c:
4004 Fix gst_rtsp_server_client_filter, using wrong variable type
4006 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
4008 * gst/rtsp-server/rtsp-media-factory-uri.c:
4009 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
4010 For AAC we need to check for framed=true instead of parsed=true.
4011 https://bugzilla.gnome.org/show_bug.cgi?id=701384
4013 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4015 * gst/rtsp-server/rtsp-stream.c:
4016 stream: optimize pipeline for protocols
4017 When TCP is not an allowed protocol for the stream, avoid creating the
4018 appsrc/appsink/queue and tee elements.
4020 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4022 * gst/rtsp-server/rtsp-media.c:
4023 media: set protocols on streams
4025 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4027 * gst/rtsp-server/rtsp-client.c:
4028 client: use protocols supported by stream
4030 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4032 * gst/rtsp-server/rtsp-media-factory.c:
4033 * gst/rtsp-server/rtsp-media.c:
4034 * gst/rtsp-server/rtsp-stream.c:
4035 media-factory: allow all protocols
4037 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4039 * gst/rtsp-server/rtsp-media.c:
4040 media: configure protocols in new streams
4042 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4044 * gst/rtsp-server/rtsp-stream.c:
4045 * gst/rtsp-server/rtsp-stream.h:
4046 stream: add protocols property
4048 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4050 * gst/rtsp-server/rtsp-media.c:
4051 rtsp-media: send state in "new-state" signal
4052 https://bugzilla.gnome.org/show_bug.cgi?id=705110
4054 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
4057 build: add subdir-objects to AM_INIT_AUTOMAKE
4058 Fixes warnings with automake 1.14
4059 https://bugzilla.gnome.org/show_bug.cgi?id=705350
4061 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4063 * docs/libs/gst-rtsp-server-sections.txt:
4064 * gst/rtsp-server/rtsp-client.c:
4065 * gst/rtsp-server/rtsp-server.c:
4066 * gst/rtsp-server/rtsp-server.h:
4067 server: add method to iterate clients of server
4069 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4071 * gst/rtsp-server/rtsp-media.c:
4072 * gst/rtsp-server/rtsp-media.h:
4073 Add vmethod for rtsp-media subclass to access rtpbin
4075 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4077 * gst/rtsp-server/rtsp-client.h:
4078 small documentation fix
4080 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4082 * gst/rtsp-server/rtsp-client.c:
4083 Do not take range header if range is invalid
4085 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4087 * docs/libs/gst-rtsp-server-sections.txt:
4088 * gst/rtsp-server/rtsp-media.c:
4089 media: add docs for new method
4091 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4093 * gst/rtsp-server/rtsp-media.c:
4094 * gst/rtsp-server/rtsp-media.h:
4095 Add API to rtsp-media set the pipeline's state
4097 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4099 * gst/rtsp-server/rtsp-media.c:
4100 Update current position/duration when gst_rtsp_media_get_range_string is called
4102 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4104 * examples/test-cgroups.c:
4105 tests: add some more docs
4107 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4109 * examples/test-cgroups.c:
4110 * gst/rtsp-server/Makefile.am:
4111 * gst/rtsp-server/rtsp-auth.c:
4112 * gst/rtsp-server/rtsp-auth.h:
4113 * gst/rtsp-server/rtsp-client.c:
4114 * gst/rtsp-server/rtsp-client.h:
4115 * gst/rtsp-server/rtsp-context.c:
4116 * gst/rtsp-server/rtsp-context.h:
4117 * gst/rtsp-server/rtsp-params.c:
4118 * gst/rtsp-server/rtsp-params.h:
4119 * gst/rtsp-server/rtsp-server.c:
4120 * gst/rtsp-server/rtsp-thread-pool.c:
4121 * gst/rtsp-server/rtsp-thread-pool.h:
4122 * tests/check/gst/client.c:
4123 ClientState -> Context
4124 Rename the clientstate to context and put the code in a separate file.
4126 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4128 * examples/test-auth.c:
4129 * gst/rtsp-server/rtsp-auth.c:
4130 * gst/rtsp-server/rtsp-auth.h:
4131 auth: add support for default token
4132 The default token is used when the user is not authenticated and can be used to
4133 give minimal permissions.
4135 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4137 * examples/test-auth.c:
4138 * gst/rtsp-server/rtsp-auth.c:
4139 auth: use defines when possible
4141 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4143 * gst/rtsp-server/rtsp-address-pool.c:
4144 address-pool: improve docs
4146 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4148 * gst/rtsp-server/rtsp-permissions.c:
4149 permissions: add the role to the copy
4151 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
4153 * gst/rtsp-server/rtsp-permissions.c:
4154 permissions: Also copy the roles
4156 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
4158 * gst/rtsp-server/rtsp-permissions.c:
4159 permissions: Make it build
4161 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4163 * gst/rtsp-server/rtsp-address-pool.h:
4166 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4168 * docs/libs/gst-rtsp-server-sections.txt:
4169 * gst/rtsp-server/rtsp-auth.c:
4170 * gst/rtsp-server/rtsp-auth.h:
4171 * gst/rtsp-server/rtsp-media.c:
4172 * gst/rtsp-server/rtsp-session-media.c:
4173 * gst/rtsp-server/rtsp-stream-transport.c:
4174 * gst/rtsp-server/rtsp-stream-transport.h:
4175 * gst/rtsp-server/rtsp-stream.c:
4176 * tests/check/gst/client.c:
4179 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4181 * docs/libs/gst-rtsp-server-sections.txt:
4182 * gst/rtsp-server/rtsp-address-pool.c:
4183 * gst/rtsp-server/rtsp-address-pool.h:
4184 * tests/check/gst/addresspool.c:
4185 * tests/check/gst/rtspserver.c:
4186 address-pool: cleanups
4187 Remove redundant method, improve docs.
4189 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4191 * docs/libs/gst-rtsp-server-sections.txt:
4192 * gst/rtsp-server/rtsp-auth.h:
4193 * gst/rtsp-server/rtsp-permissions.c:
4194 * gst/rtsp-server/rtsp-permissions.h:
4195 * gst/rtsp-server/rtsp-token.c:
4198 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4200 * gst/rtsp-server/rtsp-permissions.c:
4201 permissions: implement _remove_role
4203 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4205 * gst/rtsp-server/rtsp-permissions.c:
4206 permissions: update docs
4208 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4210 * tests/check/gst/client.c:
4211 tests: simplify tests
4212 Client settings are now disabled by default so we don't need an auth
4213 module to disable them.
4215 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4217 * gst/rtsp-server/rtsp-auth.c:
4218 auth: add default authorizations
4219 When no auth module is specified, use our table of defaults to look up the
4220 default value of the check instead of always allowing everything. This was
4221 we can disallow client settings by default.
4223 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4226 README: update readme
4228 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4230 * gst/rtsp-server/rtsp-thread-pool.c:
4231 * gst/rtsp-server/rtsp-thread-pool.h:
4232 thread-pool: add more docs
4234 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4236 * gst/rtsp-server/rtsp-thread-pool.c:
4237 * gst/rtsp-server/rtsp-thread-pool.h:
4238 thread-pool: fix race in thread reuse
4239 If we try to reuse a thread right after we made it stop, we end up using a
4240 stopped thread. Catch this case and only reuse threads that are not stopping.
4242 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4244 * gst/rtsp-server/rtsp-server.c:
4245 server: add small debug
4247 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4249 * tests/check/gst/client.c:
4251 Add some permissions to media so we can use the auth and enable
4254 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4256 * gst/rtsp-server/rtsp-client.c:
4257 client: support pushed context in handle_request
4258 If we already have a pushed state, reuse it and add our own things. This makes
4259 it easier to write tests.
4261 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4263 * gst/rtsp-server/rtsp-auth.c:
4264 auth: don't auth on methods
4265 Don't authorize on methods anymore but on the resources that we
4266 try to access, this is more flexible.
4267 Move the authorization checks to where they are needed and let the
4268 check return the response on error.
4270 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4272 * gst/rtsp-server/rtsp-mount-points.c:
4273 mount-points: add some debug
4275 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4277 * tests/check/gst/client.c:
4278 tests: almost fix test
4280 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4282 * gst/rtsp-server/rtsp-auth.c:
4283 * gst/rtsp-server/rtsp-auth.h:
4284 * gst/rtsp-server/rtsp-client.c:
4285 * gst/rtsp-server/rtsp-client.h:
4286 * gst/rtsp-server/rtsp-server.c:
4287 * gst/rtsp-server/rtsp-server.h:
4288 auth: let the auth module check client_settings
4289 Let the auth module decide if client settings are allowed for the
4292 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4294 * gst/rtsp-server/rtsp-token.c:
4295 * gst/rtsp-server/rtsp-token.h:
4296 token: add method to check boolean permission
4298 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4300 * examples/test-auth.c:
4301 * examples/test-cgroups.c:
4302 * gst/rtsp-server/rtsp-token.c:
4303 * gst/rtsp-server/rtsp-token.h:
4304 token: simplify token constructor
4305 Use variable arguments to make easier API.
4307 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4309 * examples/test-auth.c:
4310 * examples/test-cgroups.c:
4311 * gst/rtsp-server/rtsp-media-factory.c:
4312 * gst/rtsp-server/rtsp-media-factory.h:
4313 media-factory: add convenience API for factory
4315 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4317 * examples/test-auth.c:
4318 * examples/test-cgroups.c:
4319 * gst/rtsp-server/rtsp-permissions.c:
4320 * gst/rtsp-server/rtsp-permissions.h:
4321 permissions: simplify API a little
4322 Avoid passing GstStructure in the add_role method, use varargs instead
4323 to construct the structure behind the scenes. We can then also use the
4324 structure name as the role and simplify some more logic.
4326 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4328 * gst/rtsp-server/rtsp-auth.c:
4331 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4333 * gst/rtsp-server/rtsp-auth.c:
4334 * gst/rtsp-server/rtsp-auth.h:
4335 * gst/rtsp-server/rtsp-client.c:
4336 auth: handle unauthorized response
4337 Move handling of the unauthorized response to the auth module, it can add
4338 the appropriate headers to request authorization for the required method
4339 much better than the client.
4341 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4343 * gst/rtsp-server/rtsp-client.c:
4344 * gst/rtsp-server/rtsp-client.h:
4345 client: allow for sending any message, not only requests
4346 Change the _send_request() method to _send_message() so that we
4347 can both send requests and replies.
4349 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4351 * docs/libs/gst-rtsp-server-sections.txt:
4352 * gst/rtsp-server/rtsp-server.h:
4355 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4357 * examples/test-video.c:
4358 * gst/rtsp-server/rtsp-auth.c:
4359 * gst/rtsp-server/rtsp-auth.h:
4360 * gst/rtsp-server/rtsp-server.c:
4361 * gst/rtsp-server/rtsp-server.h:
4362 auth: move TLS handling to auth module
4363 Remove the TLS settings on the server and move it to the auth module because
4364 that is where security related bits go.
4366 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4368 * gst/rtsp-server/rtsp-client.c:
4369 * gst/rtsp-server/rtsp-client.h:
4370 client: add state push/pop
4372 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4374 * gst/rtsp-server/rtsp-client.c:
4375 * gst/rtsp-server/rtsp-client.h:
4376 client: add connection to state
4378 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4380 * gst/rtsp-server/rtsp-mount-points.c:
4381 mount-points: fix debug
4383 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4385 * tests/check/gst/media.c:
4386 tests: fix media test
4388 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4390 * gst/rtsp-server/rtsp-thread-pool.c:
4391 thread-pool: we don't require a state
4393 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4395 * gst/rtsp-server/rtsp-server.c:
4396 server: let context ref the server
4397 So that we don't risk losing the server object early anc crash.
4399 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4401 * tests/check/gst/client.c:
4402 tests: fix client test
4404 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4407 * docs/libs/gst-rtsp-server-docs.sgml:
4408 * docs/libs/gst-rtsp-server-sections.txt:
4409 * gst/rtsp-server/rtsp-address-pool.c:
4410 * gst/rtsp-server/rtsp-auth.c:
4411 * gst/rtsp-server/rtsp-client.c:
4412 * gst/rtsp-server/rtsp-client.h:
4413 * gst/rtsp-server/rtsp-media-factory-uri.c:
4414 * gst/rtsp-server/rtsp-media-factory.c:
4415 * gst/rtsp-server/rtsp-media-factory.h:
4416 * gst/rtsp-server/rtsp-media.c:
4417 * gst/rtsp-server/rtsp-mount-points.c:
4418 * gst/rtsp-server/rtsp-params.c:
4419 * gst/rtsp-server/rtsp-permissions.c:
4420 * gst/rtsp-server/rtsp-sdp.c:
4421 * gst/rtsp-server/rtsp-server.c:
4422 * gst/rtsp-server/rtsp-server.h:
4423 * gst/rtsp-server/rtsp-session-media.c:
4424 * gst/rtsp-server/rtsp-session-pool.c:
4425 * gst/rtsp-server/rtsp-session.c:
4426 * gst/rtsp-server/rtsp-stream-transport.c:
4427 * gst/rtsp-server/rtsp-stream.c:
4428 * gst/rtsp-server/rtsp-thread-pool.c:
4429 * gst/rtsp-server/rtsp-token.c:
4432 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4434 * gst/rtsp-server/rtsp-session-pool.c:
4435 * gst/rtsp-server/rtsp-session-pool.h:
4436 session-pool: make vmethod to create a session
4437 Make a vmethod to create a sessions so that subclasses can create
4438 custom session objects
4440 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4442 * gst/rtsp-server/rtsp-auth.c:
4443 * gst/rtsp-server/rtsp-media-factory.h:
4444 * gst/rtsp-server/rtsp-media.h:
4445 * gst/rtsp-server/rtsp-mount-points.h:
4446 * gst/rtsp-server/rtsp-session-pool.h:
4447 * gst/rtsp-server/rtsp-stream.h:
4450 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4452 * docs/libs/gst-rtsp-server-docs.sgml:
4453 * docs/libs/gst-rtsp-server-sections.txt:
4454 * gst/rtsp-server/rtsp-address-pool.c:
4455 * gst/rtsp-server/rtsp-address-pool.h:
4456 * gst/rtsp-server/rtsp-auth.c:
4457 * gst/rtsp-server/rtsp-client.h:
4458 * gst/rtsp-server/rtsp-media-factory.h:
4459 * gst/rtsp-server/rtsp-media.c:
4460 * gst/rtsp-server/rtsp-media.h:
4461 * gst/rtsp-server/rtsp-permissions.c:
4462 * gst/rtsp-server/rtsp-permissions.h:
4463 * gst/rtsp-server/rtsp-server.h:
4464 * gst/rtsp-server/rtsp-session-media.c:
4465 * gst/rtsp-server/rtsp-session-media.h:
4466 * gst/rtsp-server/rtsp-session-pool.h:
4467 * gst/rtsp-server/rtsp-session.h:
4468 * gst/rtsp-server/rtsp-stream-transport.h:
4469 * gst/rtsp-server/rtsp-stream.c:
4470 * gst/rtsp-server/rtsp-thread-pool.h:
4473 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4476 * examples/Makefile.am:
4477 configure: compile cgroup example conditionally
4478 Only compile the cgroup example when we have libcgroup
4480 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4483 * examples/Makefile.am:
4484 * examples/test-cgroups.c:
4485 examples: add cgroups example
4487 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4489 * tests/check/gst/rtspserver.c:
4490 tests: fix compilation
4492 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4494 * gst/rtsp-server/rtsp-thread-pool.c:
4495 thread-pool: fix vmethod invocation
4497 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4499 * gst/rtsp-server/rtsp-thread-pool.c:
4500 * gst/rtsp-server/rtsp-thread-pool.h:
4501 thread-pool: store thread type in thread
4503 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4505 * gst/rtsp-server/rtsp-client.c:
4506 client: pass thread from pool to media _prepare
4507 Get a thread from the configured threadpool and pass it to the prepare method of
4510 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4512 * gst/rtsp-server/rtsp-media.c:
4513 * gst/rtsp-server/rtsp-media.h:
4514 media: Accept a thread in _prepare
4515 Remove out own threadpool handling and use the provided thread and
4516 maincontext for the bus messages and the state changes.
4518 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4520 * gst/rtsp-server/rtsp-server.c:
4521 server: configure client thread pool
4523 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4525 * gst/rtsp-server/rtsp-client.c:
4526 * gst/rtsp-server/rtsp-client.h:
4527 client: add method to configure thread pool
4529 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4531 * gst/rtsp-server/rtsp-client.h:
4532 * gst/rtsp-server/rtsp-server.c:
4533 * gst/rtsp-server/rtsp-server.h:
4534 server: use thread pool
4535 Use the thread pool instead of doing our own thing.
4537 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4539 * gst/rtsp-server/Makefile.am:
4540 * gst/rtsp-server/rtsp-thread-pool.c:
4541 * gst/rtsp-server/rtsp-thread-pool.h:
4542 thread-pool: add object to manage threads
4543 Add an object to manage the client and media threads.
4545 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4547 * gst/rtsp-server/rtsp-auth.c:
4548 auth: debug authorization check
4550 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4552 * gst/rtsp-server/rtsp-media.c:
4553 media: start media pipeline in context
4554 Start the media pipeline in the provided context (or our default one
4555 when NULL). This makes sure that we run the bus thread in this context and that
4556 all media threads are children of this context.
4558 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4560 * gst/rtsp-server/rtsp-media-factory.c:
4561 factory: pass permissions to media by default
4563 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4565 * examples/test-auth.c:
4566 test: add permissions to auth test
4567 Ass some permissions to the media factory in the test.
4569 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4571 * gst/rtsp-server/rtsp-auth.c:
4572 * gst/rtsp-server/rtsp-auth.h:
4573 * gst/rtsp-server/rtsp-client.c:
4574 auth: simplify auth checks
4575 Remove client from methods, it's now in the state
4576 Perform the check specified by the string, use the information from the
4577 thread local context.
4579 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4581 * gst/rtsp-server/rtsp-client.c:
4582 * gst/rtsp-server/rtsp-client.h:
4583 client: add state to current thread
4584 Add the client to the ClientState object.
4585 Place the ClientState on the current thread.
4587 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4589 * gst/rtsp-server/rtsp-media-factory.c:
4590 * gst/rtsp-server/rtsp-media-factory.h:
4591 * gst/rtsp-server/rtsp-media.c:
4592 * gst/rtsp-server/rtsp-media.h:
4593 media: make it possible to set permissions
4594 Make it possible to set permissions on media and media factory objects
4596 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4598 * gst/rtsp-server/Makefile.am:
4599 * gst/rtsp-server/rtsp-permissions.c:
4600 * gst/rtsp-server/rtsp-permissions.h:
4601 permissions: add permissions object
4602 Add a mini object to store permissions based on a role.
4604 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4606 * examples/test-auth.c:
4607 * gst/rtsp-server/rtsp-auth.c:
4608 * gst/rtsp-server/rtsp-auth.h:
4609 * gst/rtsp-server/rtsp-client.c:
4610 auth: add auth checks
4611 Add an enum with auth checks and implement the checks in the auth object.
4612 Perform the checks from the client.
4614 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4616 * examples/test-auth.c:
4617 * gst/rtsp-server/rtsp-auth.c:
4618 * gst/rtsp-server/rtsp-auth.h:
4619 * gst/rtsp-server/rtsp-client.h:
4620 auth: use the token after authentication
4621 After we authenticated a user, keep the Token around in the state.
4623 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4625 * gst/rtsp-server/rtsp-client.c:
4626 * gst/rtsp-server/rtsp-media.c:
4627 * gst/rtsp-server/rtsp-media.h:
4628 * tests/check/gst/media.c:
4629 media: add optional context for bus messages
4630 Add an optional mainloop to _prepare that will handle the bus messages instead
4631 of always using the shared mainloop.
4633 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4635 * gst/rtsp-server/Makefile.am:
4636 * gst/rtsp-server/rtsp-token.c:
4637 * gst/rtsp-server/rtsp-token.h:
4638 token: add authorization token
4639 Add a simply miniobject that contains the authorizations. The object contains a
4640 GstStructure that hold all authorization fields. When a user is authenticated,
4641 the auth module will create a Token for the user. The token is then used to
4642 check what operations the user is allowed to do and various other configuration
4645 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4647 * examples/test-auth.c:
4648 * gst/rtsp-server/rtsp-auth.c:
4649 * gst/rtsp-server/rtsp-auth.h:
4650 * gst/rtsp-server/rtsp-client.c:
4651 * gst/rtsp-server/rtsp-client.h:
4652 * gst/rtsp-server/rtsp-media-factory.c:
4653 * gst/rtsp-server/rtsp-media-factory.h:
4654 * gst/rtsp-server/rtsp-media.c:
4655 * gst/rtsp-server/rtsp-media.h:
4656 auth: remove auth from media and factory
4657 Remove the auth object from media and factory. We want to have the RTSPClient
4658 authenticate and authorize resources, there is no need to place another auth
4659 manager on the media/factory.
4661 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4663 * examples/test-auth.c:
4664 * gst/rtsp-server/rtsp-auth.c:
4665 * gst/rtsp-server/rtsp-auth.h:
4666 * gst/rtsp-server/rtsp-client.h:
4667 auth: add support for multiple basic auth tokens
4668 Make it possible to add multiple basic authorisation tokens to one authorization
4669 object. Associate with each token an authorization group that will define what
4670 capabilities are allowed.
4672 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4674 * gst/rtsp-server/rtsp-client.c:
4675 client: error out on non-aggregate control
4676 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
4678 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4680 * gst/rtsp-server/rtsp-client.c:
4681 client: rework setup request a little
4682 Cache the media in DESCRIBE based on the longest matching path with the uri
4683 that we can find in the mount points.
4684 Rework the setup request a little to get the media from the session or from
4685 the longest matching path, this way we can derive the control string as
4686 everything after the path instead of hardcoding it.
4687 Find the stream based on the control string and only open a session when all
4690 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4692 * gst/rtsp-server/rtsp-media.c:
4693 * gst/rtsp-server/rtsp-media.h:
4694 media: add method to find a stream by control url
4696 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4698 * gst/rtsp-server/rtsp-stream.c:
4699 * gst/rtsp-server/rtsp-stream.h:
4700 stream: add method to check control url of stream
4702 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4704 * gst/rtsp-server/rtsp-client.c:
4705 * gst/rtsp-server/rtsp-session-media.c:
4706 * gst/rtsp-server/rtsp-session-media.h:
4707 * gst/rtsp-server/rtsp-session.c:
4708 * gst/rtsp-server/rtsp-session.h:
4709 session: use path matching for session media
4710 Use a path string instead of a uri to lookup session media in the sessions. Also
4711 use path matching to find the largest possible path that matches.
4713 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4715 * gst/rtsp-server/rtsp-client.c:
4716 * gst/rtsp-server/rtsp-mount-points.c:
4717 * gst/rtsp-server/rtsp-mount-points.h:
4718 * tests/check/gst/mountpoints.c:
4719 mount-points: remove useless vmethod
4720 Making lookups in the mount points should not be done with a URL, if there is a
4721 mapping to be done from URL to mount points, we'll need to do it somewhere
4724 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4726 * gst/rtsp-server/rtsp-mount-points.c:
4727 * gst/rtsp-server/rtsp-mount-points.h:
4728 * tests/check/gst/mountpoints.c:
4729 mount-points: improve mount point searching
4730 Use a GSequence to keep track of the mount points.
4731 Match a URL to the longest matching registered mount point. This should be the
4732 URL to perform aggreagate control and the remainder is the stream specific
4734 Add some unit tests for this.
4736 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
4738 * gst/rtsp-server/Makefile.am:
4739 rtsp-server: Allow building of static library
4741 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4743 * tests/check/gst/mediafactory.c:
4744 tests: fix compilation
4746 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4748 * gst/rtsp-server/rtsp-sdp.c:
4749 sdp: get control string from stream
4750 Use the control string as configured in the stream.
4752 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4754 * gst/rtsp-server/rtsp-stream.c:
4755 * gst/rtsp-server/rtsp-stream.h:
4756 stream: add methods and property to set control string
4758 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4760 * gst/rtsp-server/rtsp-client.c:
4762 Rename variables for clarity
4763 Keep media in state when we can
4765 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4767 * gst/rtsp-server/rtsp-client.c:
4768 * gst/rtsp-server/rtsp-stream.c:
4769 * gst/rtsp-server/rtsp-stream.h:
4770 stream: add more support for IPv6
4771 Rename _get_address to _get_multicast_address in GstRTSPStream to
4772 make it clear that this function only deals with multicast.
4773 Make it possible to have both an IPv4 and IPv6 multicast address on
4774 a stream. Give the client an IPv4 or IPv6 address depending on the
4775 address it used to connect to the server.
4776 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
4778 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4780 * gst/rtsp-server/rtsp-client.c:
4783 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4785 * gst/rtsp-server/rtsp-stream.c:
4786 stream: handle failed port allocation
4787 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
4788 can't allocate any family at all. Also keep track of what port families we
4790 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
4792 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4794 * gst/rtsp-server/rtsp-stream.c:
4795 stream: improve docs
4797 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4799 * gst/rtsp-server/rtsp-stream-transport.c:
4800 stream-transport: remove old if 0 block
4802 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
4804 * tests/check/gst/client.c:
4806 gst_rtsp_client_get_uri() has been removed
4807 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
4809 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4811 * gst/rtsp-server/rtsp-client.c:
4812 * gst/rtsp-server/rtsp-client.h:
4813 client: add method to filter managed sessions
4814 Add a method to filter the sessions managed by this client connection.
4815 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
4817 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4819 * gst/rtsp-server/rtsp-client.c:
4820 * gst/rtsp-server/rtsp-client.h:
4821 client: remove _get_uri() method
4822 Remove the get_uri() method on the client. A client has no uri, the uri
4823 property is an internal property to manage the last cached media for
4826 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4828 * gst/rtsp-server/rtsp-media-factory.h:
4829 media-factory: fix typo
4831 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
4833 * gst/rtsp-server/rtsp-media.c:
4834 rtsp-media: Do not leak the query in default_query_stop
4835 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
4837 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4839 * gst/rtsp-server/rtsp-media.c:
4840 media: don't unlock when conversion fails
4841 Don't unlock the state lock when conversion fails because it was not locked.
4843 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4845 * gst/rtsp-server/rtsp-media.c:
4846 * gst/rtsp-server/rtsp-media.h:
4847 Add query_position and query_stop vmethods to rtsp-media
4849 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4851 * gst/rtsp-server/rtsp-media.c:
4852 Fix typo in property install for rtsp-media's time-provider
4854 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4856 * gst/rtsp-server/rtsp-client.c:
4857 * gst/rtsp-server/rtsp-client.h:
4858 client: clean some variables
4859 Clean some variables and add some guards to _send_request()
4861 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
4863 * gst/rtsp-server/rtsp-client.c:
4864 * gst/rtsp-server/rtsp-client.h:
4865 Add gst_rtsp_client_send_request API
4866 This makes it possible to send arbitrary messages to a client, such as
4867 SET_PARAMETER or GET_PARAMETER
4869 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4871 * gst/rtsp-server/rtsp-media.c:
4872 * gst/rtsp-server/rtsp-media.h:
4873 media: add _get_element() method
4874 Add method to get the element used when creating the media.
4875 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
4877 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4879 * gst/rtsp-server/rtsp-media.c:
4882 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4884 * gst/rtsp-server/rtsp-stream.c:
4885 * gst/rtsp-server/rtsp-stream.h:
4886 stream: allow access to the rtp session
4887 https://bugzilla.gnome.org/show_bug.cgi?id=703004
4889 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
4891 * gst/rtsp-server/rtsp-stream.c:
4892 * gst/rtsp-server/rtsp-stream.h:
4893 dscp qos support in gst-rtsp-stream
4894 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
4896 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4898 * tests/check/gst/rtspserver.c:
4900 Actually do what the comment says. Also keep the old code around, not sure what
4901 should happen when you get a 454 from a TEARDOWN, does it close the connection?
4902 it currently doesn't.
4904 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4906 * gst/rtsp-server/rtsp-client.c:
4907 client: also watch newly created session
4908 When we newly created a session, start watching it immediately instead of
4909 on the next request.
4911 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
4913 * tests/check/gst/client.c:
4914 tests: add unit test for new-session
4915 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
4917 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4919 * gst/rtsp-server/rtsp-client.c:
4920 client: emit new-session when new session is created
4921 Only emit new-session when we created a new session for a client, not when a
4922 client picked up a previous session.
4923 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
4925 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
4927 * gst/rtsp-server/rtsp-client.c:
4928 client: handle asterisk as path in requests
4929 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
4931 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4933 * gst/rtsp-server/rtsp-media.c:
4934 media: handle segment query format mismatch
4935 It's possible that the segment query returns with a different format than what
4936 we asked for, handle this case also.
4938 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
4940 * gst/rtsp-server/rtsp-media.c:
4941 media: use segment stop in collect_media_stats
4942 Use segment stop instead of duration as range end point.
4943 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
4945 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
4947 * gst/rtsp-server/rtsp-media.c:
4948 * tests/check/gst/media.c:
4949 rtsp-media: Do not leak the element in take_pipeline
4950 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
4952 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
4954 * gst/rtsp-server/rtsp-client.c:
4955 * gst/rtsp-server/rtsp-client.h:
4956 rtsp-client: Make configure_client_transport virtual
4957 This patch makes configure_client_transport virtual. The functionality is
4958 needed to handle some weird clients sending multicast transport settings as url
4960 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
4962 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
4964 * gst/rtsp-server/rtsp-client.c:
4965 * gst/rtsp-server/rtsp-client.h:
4966 rtsp-client: Make param_set and param_get virtual
4967 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
4969 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
4971 * gst/rtsp-server/rtsp-client.c:
4972 * gst/rtsp-server/rtsp-media.c:
4973 * gst/rtsp-server/rtsp-media.h:
4974 media: convert_range replaces get_range_times
4975 get_range_times worked for handling UTC ranges for seeks, but we also
4976 need to convert back from NPT to the requested unit in
4977 get_range_string. convert_range is now used for both.
4978 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
4980 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4982 * gst/rtsp-server/rtsp-client.c:
4983 * gst/rtsp-server/rtsp-sdp.c:
4984 * gst/rtsp-server/rtsp-sdp.h:
4985 sdp: cleanup sdp info
4986 We don't need to pass the proto, we can more easily check a boolean.
4987 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
4989 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
4991 * gst/rtsp-server/rtsp-sdp.c:
4992 use 0.0.0.0 or :: for c= line instead of server address
4994 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
4996 * gst/rtsp-server/rtsp-client.c:
4997 use local address, not remote, in SDP
4998 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
5000 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5003 Automatic update of common submodule
5004 From 098c0d7 to 01a7a46
5006 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
5008 * gst/rtsp-server/rtsp-media.c:
5009 * gst/rtsp-server/rtsp-media.h:
5010 media: possibility to override range time conversion
5011 Make it possible to override the conversion from GstRTSPTimeRange to
5012 GstClockTimes, that is done before seeking on the media
5013 pipeline. Overriding can be useful for UTC ranges, where the default
5014 conversion gives nanoseconds since 1900.
5015 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
5017 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
5019 * gst/rtsp-server/rtsp-server.c:
5020 * gst/rtsp-server/rtsp-server.h:
5021 rtsp-server: Expose the use_client_settings API
5022 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
5024 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
5026 * gst/rtsp-server/rtsp-client.c:
5027 * gst/rtsp-server/rtsp-stream.c:
5028 * gst/rtsp-server/rtsp-stream.h:
5029 rtspstream: handle both ipv4 and ipv6 clients
5030 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
5032 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5034 * gst/rtsp-server/rtsp-sdp.c:
5035 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
5036 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
5037 We already have a way to place extra attributes in the SDP by using a string
5038 property with prefix x- or a- in the caps.
5040 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5042 * gst/rtsp-server/rtsp-sdp.c:
5043 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
5044 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
5045 We already have a way to place extra attributes in the SDP, just make a string
5046 property in the payloader with a- or x- prefix.
5048 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5050 * gst/rtsp-server/rtsp-sdp.c:
5051 rtsp: place a- and x- properties as attributes
5052 application/x-rtp has properties with a- and x- prefixes that should be
5053 placed as attributes in the SDP for the media instead of being added to the
5056 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5058 * examples/Makefile.am:
5059 * examples/test-video.c:
5060 example: add TLS example
5062 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5064 * gst/rtsp-server/rtsp-server.c:
5065 * gst/rtsp-server/rtsp-server.h:
5066 server: add support for TLS
5067 Add methods to set and get a TLS certificate.
5068 Add vmethod to configure a new connection. By default, configure the TLS
5069 certificate in a new connection if needed.
5071 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5073 * gst/rtsp-server/rtsp-server.c:
5074 * gst/rtsp-server/rtsp-server.h:
5075 server: remove accept_client vmethod
5076 This vmethod is not very useful so remove it.
5078 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5080 * gst/rtsp-server/rtsp-server.c:
5081 server: don't crash on NULL GError
5083 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
5085 * gst/rtsp-server/rtsp-session-pool.c:
5086 rtsp-session-pool: corrected session timeout detection
5087 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
5089 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5091 * gst/rtsp-server/rtsp-client.c:
5092 client: improve debug
5094 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5096 * gst/rtsp-server/rtsp-client.c:
5097 * gst/rtsp-server/rtsp-client.h:
5098 * gst/rtsp-server/rtsp-server.c:
5099 server: refactor connection setup
5100 Let the server accept the socket connection and construct a GstRTSPConnection
5101 from it. Remove the code from the client and let the client only deal with
5102 a fully configure GstRTSPConnection object.
5103 We will need this later when the server will configure the connection for
5106 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5108 * gst/rtsp-server/rtsp-stream.c:
5109 stream: keep the transport object alive
5110 Keep the transport object alive while we have it as qdata on the
5113 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
5115 * gst/rtsp-server/rtsp-client.c:
5116 * gst/rtsp-server/rtsp-server.c:
5117 rtsp-server: Do not crash on nmapping of server
5118 * generate error when gst_rtsp_connection_accept fails
5119 * do not stop accepting incoming connections because
5120 accepting a client fails
5121 https://bugzilla.gnome.org/show_bug.cgi?id=701072
5123 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
5125 * gst/rtsp-server/rtsp-client.c:
5126 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
5127 https://bugzilla.gnome.org/show_bug.cgi?id=700953
5129 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
5131 * gst/rtsp-server/rtsp-sdp.c:
5132 rtsp-sdp: Parse framerate caps field and set SDP attribute
5133 The SDP attribute and its format is described in RFC4566.
5134 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
5136 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
5138 * gst/rtsp-server/rtsp-sdp.c:
5139 rtsp-sdp: Parse width/height from caps and set SDP attribute
5140 The SDP attribute and its format is described in RFC6064.
5141 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
5143 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
5145 * gst/rtsp-server/rtsp-sdp.c:
5146 * tests/check/gst/client.c:
5147 rtsp-sdp: add bandwidth line
5148 https://bugzilla.gnome.org/show_bug.cgi?id=699220
5150 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5153 Automatic update of common submodule
5154 From 5edcd85 to 098c0d7
5156 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
5158 * tests/check/gst/media.c:
5159 tests: add dynamic payloader prepare/unprepare check
5161 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5163 * gst/rtsp-server/rtsp-media.c:
5164 media: release lock when removing fakesink
5166 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5168 * gst/rtsp-server/rtsp-stream.c:
5169 stream: set elements to NULL before removing
5170 When removing a stream, set the elements to NULL first. This avoids
5171 element-is-not-in-NULL-state errors when we dispose the elements.
5173 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
5176 Automatic update of common submodule
5177 From 3cb3d3c to 5edcd85
5179 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5181 * gst/rtsp-server/rtsp-media.c:
5182 * gst/rtsp-server/rtsp-media.h:
5183 media: listen to pad-removed signals
5184 Listen to the pad-removed signal and remove the stream associated with the
5186 Add signal to be notified of the removed pad.
5187 Remove the fakesink in unprepare()
5188 Fix signatures of the signal methods
5190 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5192 * examples/test-sdp.c:
5193 tests: add example of reusable pipelines
5195 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
5197 * gst/rtsp-server/rtsp-stream.c:
5198 * gst/rtsp-server/rtsp-stream.h:
5199 stream: add method to get the srcpad
5201 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
5203 * tests/check/gst/media.c:
5204 check: add media prepare/unprepare test
5205 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
5207 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
5209 * gst/rtsp-server/rtsp-media.c:
5210 media: disconnect from signal handlers in unprepare()
5211 We connected to the pad-added and no-more-pads signals in prepare() so
5212 we need to disconnect from them in unprepare().
5213 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
5215 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
5217 * gst/rtsp-server/rtsp-media.c:
5218 media: don't free streams array
5219 Don't free the streams array in the unprepare() method, they were not
5221 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
5223 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
5225 * gst/rtsp-server/rtsp-media.c:
5226 media: don't unref the pipeline in unprepare
5227 Unprepare() should undo what prepare() does. Because the pipeline is
5228 not created in prepare(), we should not unref it in unprepare()
5230 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
5232 * gst/rtsp-server/rtsp-stream.c:
5233 stream: clear session and caps for reuse
5234 Set the session and caps to NULL after unref otherwise we might unref
5236 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
5238 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
5240 * gst/rtsp-server/rtsp-client.c:
5241 client: send out teardown signal before tearing down
5242 The advantage is that in the signal handler you get direct access to
5243 information about what streams are about to get torn down (in the
5244 GstRTSPClientState).
5245 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
5247 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
5249 * gst/rtsp-server/rtsp-client.c:
5250 * gst/rtsp-server/rtsp-client.h:
5251 client: expose connection
5252 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
5254 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
5257 Automatic update of common submodule
5258 From aed87ae to 3cb3d3c
5260 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5262 * gst/rtsp-server/rtsp-media.c:
5263 * gst/rtsp-server/rtsp-media.h:
5264 * gst/rtsp-server/rtsp-session-media.c:
5265 * gst/rtsp-server/rtsp-session-media.h:
5266 media: add method to get the base_time of the pipeline
5267 Together with a shared clock, this base-time could eventually be sent to
5268 the client so that it can reconstruct the exact running-time of the clock
5271 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5273 * gst/rtsp-server/Makefile.am:
5274 * gst/rtsp-server/rtsp-media.c:
5275 * gst/rtsp-server/rtsp-media.h:
5276 * gst/rtsp-server/rtsp-sdp.c:
5277 media: add GstNetTimeProvider support
5278 Add a property to let the media provide a GstNetTimeProvider for its clock.
5279 Make methods to get the clock and nettimeprovider
5280 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
5281 provider and also the current time of the clock. This should make it possible
5282 for (GStreamer) clients to slave their clock to the server clock.
5284 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
5287 Automatic update of common submodule
5288 From 04c7a1e to aed87ae
5290 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5292 * gst/rtsp-server/rtsp-media.c:
5293 media: wait for buffering to complete
5294 Wait for buffering to complete before changing the state to the target state.
5296 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5298 * gst/rtsp-server/rtsp-media.c:
5299 media: small cleanup
5301 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
5303 * tests/check/gst/rtspserver.c:
5304 tests: remove extra unref in test_setup_non_existing_stream
5305 The unref is not needed anymore, teardown runs without it.
5306 https://bugzilla.gnome.org/show_bug.cgi?id=696542
5308 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
5310 * tests/check/gst/rtspserver.c:
5311 tests: GSocketService cleanup in test_bind_already_in_use
5312 Use g_socket_service_stop so the rtspserver test stops listening for
5313 incoming connections in test_bind_already_in_use.
5314 https://bugzilla.gnome.org/show_bug.cgi?id=696541
5316 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
5318 * gst/rtsp-server/rtsp-media-factory.c:
5319 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
5320 Instead use a GWeakRef which is safe to use
5321 This is a known GLib bug, see:
5322 https://bugzilla.gnome.org/show_bug.cgi?id=667145
5324 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
5326 * gst/rtsp-server/rtsp-client.c:
5327 * gst/rtsp-server/rtsp-media.c:
5328 * gst/rtsp-server/rtsp-media.h:
5329 * gst/rtsp-server/rtsp-sdp.c:
5330 * tests/check/gst/media.c:
5331 * tests/check/gst/rtspserver.c:
5332 rtsp-media/client: Reply to PLAY request with same type of Range
5333 Remember the type of Range from the PLAY request and use the same type for
5336 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
5338 * gst/rtsp-server/rtsp-client.c:
5339 * gst/rtsp-server/rtsp-client.h:
5340 * tests/check/gst/client.c:
5341 rtsp-client: expose uri
5343 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
5345 * tests/check/gst/mediafactory.c:
5346 tests: Hold ref while creating second media
5347 To test if the media aren't shared, make sure we keep the first one while creating a second
5348 otherwise the same memory address may be reused.
5350 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
5353 configure: remove out-of-date comment
5355 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
5358 .gitignore: ignore more build files
5360 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
5362 * tests/check/Makefile.am:
5363 tests: use right _LIBS variable for gst-plugins-base libs
5365 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5367 * tests/check/Makefile.am:
5368 check: add librtp to libs
5370 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
5372 * tests/check/gst/rtspserver.c:
5373 tests: Add test to check selecting a port the server will send from
5375 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
5377 * tests/check/gst/rtspserver.c:
5378 tests: Make sure packets are actually received
5380 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
5382 * gst/rtsp-server/rtsp-stream.c:
5383 stream: Select unicast address from pool if appropriate
5385 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
5387 * gst/rtsp-server/rtsp-stream.c:
5388 stream: Properties are always there in Gst 1.0
5390 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
5392 * tests/check/gst/addresspool.c:
5393 tests: Add tests for unicast addresses in pool
5395 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
5397 * gst/rtsp-server/rtsp-address-pool.c:
5398 * tests/check/gst/addresspool.c:
5399 address-pool: Verify that multicast addresses are used for multicast and vice-versa
5401 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
5403 * docs/libs/gst-rtsp-server-sections.txt:
5404 * gst/rtsp-server/rtsp-address-pool.c:
5405 * gst/rtsp-server/rtsp-address-pool.h:
5406 * gst/rtsp-server/rtsp-stream.c:
5407 * tests/check/gst/addresspool.c:
5408 address-pool: Add unicast addresses
5410 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
5413 * gst/rtsp-server/rtsp-server.c:
5414 * tests/check/gst/rtspserver.c:
5415 rtsp-server: Limit the number of threads per server instance
5416 If we exceed the maximum, just round robin the clients over the existing
5419 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
5421 * gst/rtsp-server/rtsp-server.c:
5422 rtsp-server: No need to store the GMainContext in the client context
5424 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
5426 * tests/check/gst/rtspserver.c:
5427 tests: Add test for client disconnection
5429 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
5431 * tests/check/gst/rtspserver.c:
5432 tests: Test client and session timeouts with multiple threads
5434 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
5436 * gst/rtsp-server/rtsp-address-pool.c:
5437 * gst/rtsp-server/rtsp-auth.c:
5438 * gst/rtsp-server/rtsp-client.c:
5439 * gst/rtsp-server/rtsp-media-factory-uri.c:
5440 * gst/rtsp-server/rtsp-media-factory.c:
5441 * gst/rtsp-server/rtsp-media.c:
5442 * gst/rtsp-server/rtsp-mount-points.c:
5443 * gst/rtsp-server/rtsp-server.c:
5444 * gst/rtsp-server/rtsp-session-media.c:
5445 * gst/rtsp-server/rtsp-session-pool.c:
5446 * gst/rtsp-server/rtsp-session.c:
5447 Document locking and its order
5449 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
5451 * tests/check/gst/rtspserver.c:
5452 tests: Test that slow DESCRIBE don't block other clients
5454 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
5456 * tests/check/gst/client.c:
5457 tests: Add tests for client-requested multicast address
5459 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
5461 * docs/libs/gst-rtsp-server-sections.txt:
5462 docs: Put the various functions in the right sections
5464 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
5466 * docs/libs/gst-rtsp-server-docs.sgml:
5467 * docs/libs/gst-rtsp-server-sections.txt:
5468 * gst/rtsp-server/rtsp-address-pool.c:
5469 * gst/rtsp-server/rtsp-address-pool.h:
5470 docs: Generate docs for GstRTSPAddressPool
5472 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
5474 * gst/rtsp-server/rtsp-client.c:
5475 * gst/rtsp-server/rtsp-stream.c:
5476 * gst/rtsp-server/rtsp-stream.h:
5477 client: Check client provided addresses against the address pool
5479 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
5481 * gst/rtsp-server/rtsp-address-pool.c:
5482 * gst/rtsp-server/rtsp-address-pool.h:
5483 * tests/check/gst/addresspool.c:
5484 address-pool: Add API to request a specific address from the pool
5485 Also add relevant unit tests.
5487 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
5489 * tests/check/gst/mediafactory.c:
5490 tests: Check the passing around of a RTSPAddressPool
5491 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
5492 way down to the stream.
5494 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
5496 * tests/check/gst/addresspool.c:
5497 tests: Add more tests for the address pool
5499 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
5501 * gst/rtsp-server/rtsp-address-pool.c:
5502 address-pool: Fix off by one error
5503 When splitting a port range, the port after a skip is not part of range.
5505 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
5508 Automatic update of common submodule
5509 From 2de221c to 04c7a1e
5511 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
5514 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
5515 AM_CONFIG_HEADER was removed in automake 1.13
5516 https://bugzilla.gnome.org/show_bug.cgi?id=693368
5518 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
5521 Automatic update of common submodule
5522 From a942293 to 2de221c
5524 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5526 * gst/rtsp-server/rtsp-client.c:
5527 client: make sure the watch exists while sending data
5528 Protect the send_func with a lock. This allows us to wait for sending
5529 to complete before changing the send_func and user_data. We add an
5530 extra ref to the watch to make sure that it remains valid during
5532 When closing the connection, set the send_func to NULL
5533 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
5535 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5537 * tests/check/Makefile.am:
5538 tests: use GST_*_1_0 environment variables everywhere
5539 The _1_0 suffixed environment variables override the
5540 non-suffixed ones, so if we're in an environment that
5541 sets the _1_0 suffixed ones, such as jhbuild, we need
5542 to set those to make sure ours actually always get
5545 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5548 Automatic update of common submodule
5549 From acb04d9 to a942293
5551 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5553 * gst/rtsp-server/rtsp-client.c:
5554 rtsp-client: set the client backlog
5555 Set the client backlog to a reasonable default
5557 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
5559 * gst/rtsp-server/rtsp-media.c:
5560 rtsp-media: Make the element a constructor parameter
5561 https://bugzilla.gnome.org/show_bug.cgi?id=689594
5563 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5565 * docs/libs/Makefile.am:
5566 docs: Link with gcov library when gcov is enabled
5567 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
5569 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5571 * gst/rtsp-server/rtsp-media.c:
5572 media: match prepare with unprepare
5573 Really unprepare when there were an equal amount of prepare calls.
5575 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5577 * gst/rtsp-server/rtsp-media.c:
5578 media: media has to be unprepared in finalize
5579 Because unprepare takes away the last ref on the media.
5581 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5583 * gst/rtsp-server/rtsp-client.c:
5584 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
5585 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
5586 We can't use the refcount to trigger unprepare because it is the unprepare call
5587 that removes the last refcount after all messages are consumed. What we should
5588 probably do is make a prepared refcount and only unprepare when the refcount
5591 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5593 * gst/rtsp-server/rtsp-media.c:
5594 media: let the source unref the last media ref
5595 the last ref to the media is held by the source so we don't need to add more ref
5596 and unrefs, we simply destroy the media when the source is gone.
5598 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5600 * gst/rtsp-server/rtsp-media.c:
5601 media: improve debug
5603 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5605 * gst/rtsp-server/rtsp-media.c:
5607 Make sure we are in the right state when collecting the position and duration.
5608 Only make ourselves PREPARED when we were previously PREPARING.
5610 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5612 * gst/rtsp-server/rtsp-media.c:
5613 media: use g_object_ref/unref for GObjects
5615 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
5617 * gst/rtsp-server/rtsp-client.c:
5618 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
5619 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
5620 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
5621 isn't being used anymore.
5623 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
5625 * gst/rtsp-server/rtsp-media.c:
5626 Fix compiler warning
5628 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
5630 * gst/rtsp-server/rtsp-media-factory-uri.c:
5631 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
5633 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5635 * gst/rtsp-server/rtsp-session-media.h:
5638 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5640 * gst/rtsp-server/rtsp-media.c:
5641 * tests/check/gst/media.c:
5642 media: avoid element leak
5644 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5646 * gst/rtsp-server/rtsp-media.c:
5647 media: require an element in media constructor
5649 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5651 * gst/rtsp-server/rtsp-client.c:
5652 Revert "client: TEARDOWN brings that state to Init again"
5653 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
5654 The object is already disposed, there is no point in setting the state.
5656 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5658 * gst/rtsp-server/rtsp-client.c:
5659 client: TEARDOWN brings that state to Init again
5661 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5663 * docs/libs/gst-rtsp-server-sections.txt:
5664 * examples/test-auth.c:
5665 * gst/rtsp-server/rtsp-auth.c:
5666 * gst/rtsp-server/rtsp-auth.h:
5667 * gst/rtsp-server/rtsp-client.c:
5668 * gst/rtsp-server/rtsp-client.h:
5669 * gst/rtsp-server/rtsp-media-factory-uri.c:
5670 * gst/rtsp-server/rtsp-media-factory-uri.h:
5671 * gst/rtsp-server/rtsp-media-factory.c:
5672 * gst/rtsp-server/rtsp-media-factory.h:
5673 * gst/rtsp-server/rtsp-media.c:
5674 * gst/rtsp-server/rtsp-media.h:
5675 * gst/rtsp-server/rtsp-mount-points.c:
5676 * gst/rtsp-server/rtsp-mount-points.h:
5677 * gst/rtsp-server/rtsp-sdp.c:
5678 * gst/rtsp-server/rtsp-server.c:
5679 * gst/rtsp-server/rtsp-server.h:
5680 * gst/rtsp-server/rtsp-session-media.c:
5681 * gst/rtsp-server/rtsp-session-media.h:
5682 * gst/rtsp-server/rtsp-session-pool.c:
5683 * gst/rtsp-server/rtsp-session-pool.h:
5684 * gst/rtsp-server/rtsp-session.c:
5685 * gst/rtsp-server/rtsp-session.h:
5686 * gst/rtsp-server/rtsp-stream-transport.c:
5687 * gst/rtsp-server/rtsp-stream-transport.h:
5688 * gst/rtsp-server/rtsp-stream.c:
5689 * gst/rtsp-server/rtsp-stream.h:
5690 * tests/check/gst/media.c:
5691 rtsp: make object details private
5692 Make all object details private
5693 Add methods to access private bits
5695 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5697 * tests/check/Makefile.am:
5698 * tests/check/gst/media.c:
5699 tests: add media tests
5701 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5703 * gst/rtsp-server/rtsp-media.c:
5704 media: check if prepared for some methods
5705 Check that the media object is prepared before doing seek and getting the
5706 current position etc.
5707 Add some g_return checks.
5709 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5711 * tests/check/Makefile.am:
5712 * tests/check/gst/mediafactory.c:
5713 tests: add mediafactory test
5715 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5717 * gst/rtsp-server/rtsp-stream.c:
5718 stream: improve debug
5720 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5722 * gst/rtsp-server/rtsp-media.c:
5723 * gst/rtsp-server/rtsp-media.h:
5724 media: unref pipeline in finalize to avoid leaking it
5726 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5728 * gst/rtsp-server/rtsp-media-factory-uri.c:
5729 * gst/rtsp-server/rtsp-media.c:
5730 rtsp: use gst_object_unref on GstObjects
5732 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5734 * gst/rtsp-server/rtsp-media-factory.c:
5735 media-factory: require an url
5737 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5739 * examples/test-uri.c:
5740 examples: fix include
5742 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5744 * gst/rtsp-server/rtsp-server.h:
5745 server: remove unused include
5747 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5749 * tests/check/Makefile.am:
5750 * tests/check/gst/mountpoints.c:
5751 tests: add test for mountpoints
5753 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5755 * gst/rtsp-server/rtsp-client.c:
5756 client: fix factory leak
5757 Keep the factory in the state object only for authorization checks and make
5758 sure we unref it on failure. Also don't keep invalid objects in the state
5761 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5763 * gst/rtsp-server/rtsp-mount-points.c:
5764 mounts: add g_return_if guards
5766 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5768 * tests/check/gst/client.c:
5769 tests: add more tests
5771 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5773 * gst/rtsp-server/rtsp-client.c:
5774 client: improve debug
5776 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5778 * gst/rtsp-server/rtsp-client.c:
5779 client: improve debug and fix leaks
5780 Cleanup the uri and session when there is a bad request.
5782 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5787 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5789 * tests/check/gst/client.c:
5790 test: add test for session in options request
5792 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5794 * gst/rtsp-server/rtsp-client.c:
5795 client: use 454 when session can't be found
5796 We should use 454 when a session can't be found because there was no session
5797 pool configured in the server. This is not a server configuration problem
5798 because the server on which the request is done might not be the same one that
5799 will keep the sessions for us and so it does not need to support sessions.
5801 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5803 * gst/rtsp-server/rtsp-client.c:
5804 client: only free connection when there is one
5805 It's possible that the client doesn't have a connection when we try to free it.
5807 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5809 * tests/check/Makefile.am:
5810 * tests/check/gst/client.c:
5811 tests: add unit test for the client object
5813 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5815 * gst/rtsp-server/rtsp-client.c:
5816 client: small cleanup
5818 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5820 * gst/rtsp-server/rtsp-client.h:
5821 client: remove unused include
5823 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5825 * gst/rtsp-server/rtsp-client.c:
5826 client: fix compilation
5828 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5830 * gst/rtsp-server/rtsp-client.c:
5831 client: call destroy without the lock
5833 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5835 * gst/rtsp-server/rtsp-client.c:
5836 * gst/rtsp-server/rtsp-client.h:
5837 client: make the client usable without a socket
5838 Make a method to let the client handle a message and a callback when the client
5839 wants us to send a response message back. This makes it possible to also use the
5840 client object without the sockets, which should make it easier to test.
5842 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5844 * gst/rtsp-server/rtsp-client.c:
5845 * gst/rtsp-server/rtsp-client.h:
5846 client: small cleanup
5848 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5850 * docs/libs/gst-rtsp-server-sections.txt:
5851 * gst/rtsp-server/rtsp-client.c:
5852 * gst/rtsp-server/rtsp-client.h:
5853 * gst/rtsp-server/rtsp-server.c:
5854 client: remove reference to server
5855 We don't need to keep a ref to the server
5857 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5859 * gst/rtsp-server/rtsp-client.c:
5860 * gst/rtsp-server/rtsp-client.h:
5862 Also add some g_return_if()
5864 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5866 * gst/rtsp-server/rtsp-client.c:
5867 client: log more errors
5869 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5871 * gst/rtsp-server/rtsp-client.c:
5872 client: fix compilation
5874 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5876 * gst/rtsp-server/rtsp-client.c:
5877 * gst/rtsp-server/rtsp-client.h:
5878 client: add generic close-after-send support
5879 Add a property to send_response() to close the connection after the response has
5880 been sent to the client.
5882 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5885 * docs/libs/gst-rtsp-server-docs.sgml:
5886 * docs/libs/gst-rtsp-server-sections.txt:
5887 * docs/libs/gst-rtsp-server.types:
5888 * examples/test-auth.c:
5889 * examples/test-launch.c:
5890 * examples/test-mp4.c:
5891 * examples/test-multicast.c:
5892 * examples/test-multicast2.c:
5893 * examples/test-ogg.c:
5894 * examples/test-readme.c:
5895 * examples/test-sdp.c:
5896 * examples/test-uri.c:
5897 * examples/test-video.c:
5898 * gst/rtsp-server/Makefile.am:
5899 * gst/rtsp-server/rtsp-auth.h:
5900 * gst/rtsp-server/rtsp-client.c:
5901 * gst/rtsp-server/rtsp-client.h:
5902 * gst/rtsp-server/rtsp-media-mapping.c:
5903 * gst/rtsp-server/rtsp-media-mapping.h:
5904 * gst/rtsp-server/rtsp-mount-points.c:
5905 * gst/rtsp-server/rtsp-mount-points.h:
5906 * gst/rtsp-server/rtsp-server.c:
5907 * gst/rtsp-server/rtsp-server.h:
5908 * gst/rtsp-server/rtsp-session-media.c:
5909 * gst/rtsp-server/rtsp-session-pool.c:
5910 * gst/rtsp-server/rtsp-session-pool.h:
5911 * tests/check/gst/rtspserver.c:
5912 MediaMapping -> MountPoints
5913 Describes better what the object manages.
5915 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5918 configure: bump required version of -base
5920 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5922 * gst/rtsp-server/rtsp-media.c:
5925 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5927 * gst/rtsp-server/rtsp-media.c:
5928 * gst/rtsp-server/rtsp-media.h:
5929 media: support more Range formats
5930 Use the new -base methods to convert the Range string into a seek start and stop
5933 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5935 * examples/test-launch.c:
5936 examples: fix whitespace
5938 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5940 * examples/test-auth.c:
5941 test-auth: add example of how to remove sessions
5942 Add an example of the session filter api.
5944 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5946 * examples/test-uri.c:
5947 test-uri: remove mapping example
5949 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5951 * examples/test-uri.c:
5952 test-uri: fix callback signature
5954 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5956 * gst/rtsp-server/rtsp-media-factory.c:
5957 factory: keep ref to factory while media active
5958 While the media from a factory is alive, keep a ref to the factory.
5959 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
5961 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5963 * gst/rtsp-server/rtsp-media-factory-uri.c:
5964 factory-uri: add some debug
5966 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5968 * gst/rtsp-server/rtsp-stream.c:
5969 stream: set udp sources to PLAYING
5970 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
5971 so that it doesn't cause our pipeline to produce ASYNC-DONE.
5973 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5975 * gst/rtsp-server/rtsp-media-factory-uri.c:
5976 factory-uri: take ref to factory
5977 Take a ref to the factory that we place in our list.
5979 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5981 * tests/Makefile.am:
5982 * tests/test-reuse.c:
5983 test: add test for server reuse
5984 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
5986 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
5988 * gst/rtsp-server/rtsp-server.c:
5989 server: start and stop multiple times
5990 Stop listening on the RTSP port when the GSource is removed, so clients
5991 can't connect and the server can be started again.
5992 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
5994 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5996 * gst/rtsp-server/rtsp-server.c:
5997 server: fix small leak
5999 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6001 * gst/rtsp-server/rtsp-media.c:
6002 media: unref source in finish_unprepare
6003 The source is created in prepare, unref it in finish_unprepare.
6004 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
6006 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
6008 * gst/rtsp-server/rtsp-client.c:
6009 * gst/rtsp-server/rtsp-media.c:
6010 rtsp-media: remove bus watch before finalizing
6011 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
6012 * An extra media ref is added for the bus watch. This extra ref is unreffed by
6013 the GDestroyNotify function.
6014 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
6015 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
6016 gst_rtsp_media_unprepare before unreffing the media.
6017 This way, the bus watch will be removed before the media is finalized.
6018 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
6020 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
6022 * gst/rtsp-server/rtsp-client.c:
6023 * gst/rtsp-server/rtsp-client.h:
6024 client: wait until the TEARDOWN response is sent to close the connection
6025 Responses can be sent async so we need to wait until the TEARDOWN response has
6026 been written before we close the connection to the client. This avoids the risk
6027 of writing/polling closed sockets.
6028 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
6030 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
6032 * gst/rtsp-server/rtsp-stream.c:
6033 rtsp-stream: plug socket leak
6034 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
6036 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
6039 Automatic update of common submodule
6040 From 6bb6951 to a72faea
6042 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
6044 * gst/rtsp-server/rtsp-media-factory-uri.c:
6045 rtsp-server: don't use deprecated API
6047 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
6049 * gst/rtsp-server/rtsp-client.c:
6050 rtsp-client: fix unused-but-set-variable compiler warning
6051 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
6053 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6056 * docs/libs/gst-rtsp-server-sections.txt:
6057 * gst/rtsp-server/rtsp-client.c:
6060 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6062 * examples/Makefile.am:
6063 * examples/test-multicast2.c:
6064 examples: add another multicast example
6065 Add an example for how to configure separate multicast ranges for each media
6068 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6070 * examples/test-multicast.c:
6073 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6075 * gst/rtsp-server/rtsp-client.c:
6076 * gst/rtsp-server/rtsp-media.c:
6077 * gst/rtsp-server/rtsp-session-media.c:
6078 * gst/rtsp-server/rtsp-session-media.h:
6079 * gst/rtsp-server/rtsp-stream-transport.c:
6080 * gst/rtsp-server/rtsp-stream-transport.h:
6081 stream: use the address managed by the stream
6082 Use the address managed by the stream for multicast. This allows us to have 1
6083 multicast address for each stream.
6084 Because the address is now managed by the stream we don't have to pass it around
6086 Set the address pool on the streams.
6088 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6090 * gst/rtsp-server/rtsp-client.c:
6091 * gst/rtsp-server/rtsp-media.c:
6092 * gst/rtsp-server/rtsp-stream.c:
6095 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6097 * gst/rtsp-server/rtsp-media.c:
6098 * gst/rtsp-server/rtsp-media.h:
6099 media: add signal for new streams
6100 This allows applications to listen for new streams and configure properties on
6101 them, like the address pool.
6103 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6105 * gst/rtsp-server/rtsp-media.c:
6106 media: configure address pool in new streams
6108 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6110 * gst/rtsp-server/rtsp-stream.c:
6111 * gst/rtsp-server/rtsp-stream.h:
6112 stream: add methods to deal with address pool
6113 Add methods to get and set the address pool for the stream
6114 Add method to allocate and get the multicast addresses for this stream.
6116 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6118 * docs/libs/gst-rtsp-server-sections.txt:
6119 * gst/rtsp-server/rtsp-media.c:
6120 * gst/rtsp-server/rtsp-media.h:
6121 media: remove MTU property
6122 It is a stream property
6124 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6126 * gst/rtsp-server/rtsp-client.c:
6127 client: set blocksize only on stream
6128 Set the blocksize only on the current stream.
6130 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6132 * gst/rtsp-server/rtsp-stream.c:
6133 stream: share src and sink sockets
6134 the allocated socket is in the used-socket property, not socket.
6136 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6138 * gst/rtsp-server/rtsp-address-pool.c:
6139 * gst/rtsp-server/rtsp-address-pool.h:
6140 * gst/rtsp-server/rtsp-client.c:
6141 * gst/rtsp-server/rtsp-session-media.c:
6142 * gst/rtsp-server/rtsp-session-media.h:
6143 * gst/rtsp-server/rtsp-stream-transport.c:
6144 * gst/rtsp-server/rtsp-stream-transport.h:
6145 * tests/check/gst/addresspool.c:
6146 rtsp: make address-pool return an address object
6147 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
6148 store more info in the structure and allows us to more easily return the address
6149 to the right pool when no longer needed.
6150 Pass the address to the StreamTransport so that we can return it to the pool
6151 when the stream transport is freed or changed.
6153 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6155 * examples/Makefile.am:
6156 * examples/test-multicast.c:
6157 examples: add multicast example
6158 Show how to set up the multicast address pool so that media can be
6159 server with multicast.
6161 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6163 * gst/rtsp-server/rtsp-client.c:
6164 * gst/rtsp-server/rtsp-media-factory.c:
6165 * gst/rtsp-server/rtsp-media-factory.h:
6166 * gst/rtsp-server/rtsp-media.c:
6167 * gst/rtsp-server/rtsp-media.h:
6168 rtsp: use AddressPool
6169 Remove the multicast_group property.
6170 Use the configured addresspool to allocate multicast addresses.
6172 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6174 * gst/rtsp-server/rtsp-address-pool.c:
6175 * gst/rtsp-server/rtsp-address-pool.h:
6176 address-pool: add clear method
6178 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6180 * gst/rtsp-server/rtsp-address-pool.c:
6181 address-pool: small cleanups
6183 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6185 * tests/check/Makefile.am:
6186 * tests/check/gst/addresspool.c:
6187 tests: add addresspool unit test
6189 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6191 * gst/rtsp-server/Makefile.am:
6192 * gst/rtsp-server/rtsp-address-pool.c:
6193 * gst/rtsp-server/rtsp-address-pool.h:
6194 address-pool: add object to manage multicast addresses
6195 Make an object that can manage a rage of multicast addresses and ports.
6197 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6199 * gst/rtsp-server/rtsp-server.c:
6200 server: set default max-threads property
6202 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6204 * gst/rtsp-server/rtsp-media.c:
6205 media: wait for concurrent _prepare
6206 If a prepare is busy, wait for the result.
6208 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6210 * gst/rtsp-server/rtsp-media.c:
6211 media: add lock around message handler
6212 We don't want to dispatch messages while we are still processing the result of
6215 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6217 * gst/rtsp-server/rtsp-media.c:
6218 * gst/rtsp-server/rtsp-media.h:
6219 media: add lock to protect state changes
6221 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6223 * gst/rtsp-server/rtsp-stream.c:
6224 * gst/rtsp-server/rtsp-stream.h:
6227 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6229 * gst/rtsp-server/rtsp-stream-transport.c:
6230 * gst/rtsp-server/rtsp-stream-transport.h:
6231 * gst/rtsp-server/rtsp-stream.c:
6232 stream-transport: add keep-alive method
6234 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6236 * gst/rtsp-server/rtsp-stream-transport.c:
6237 * gst/rtsp-server/rtsp-stream-transport.h:
6238 * gst/rtsp-server/rtsp-stream.c:
6239 stream-transport: add method to handle RTP/RTCP
6240 Call new methods instead of poking into the structures directly.
6242 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6244 * gst/rtsp-server/rtsp-session-media.c:
6245 * gst/rtsp-server/rtsp-session-media.h:
6246 session-media: add locking
6248 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6250 * gst/rtsp-server/rtsp-session.c:
6251 * gst/rtsp-server/rtsp-session.h:
6252 session: add locking
6254 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6256 * gst/rtsp-server/rtsp-server.c:
6257 server: free old socket
6259 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6261 * gst/rtsp-server/rtsp-media-mapping.c:
6262 * gst/rtsp-server/rtsp-media-mapping.h:
6263 mapping: add locking
6265 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6267 * gst/rtsp-server/rtsp-media-factory.c:
6268 media-factory: add locking
6270 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6272 * gst/rtsp-server/rtsp-auth.c:
6273 * gst/rtsp-server/rtsp-auth.h:
6276 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6278 * gst/rtsp-server/rtsp-server.c:
6279 * gst/rtsp-server/rtsp-server.h:
6280 server: add max-thread property
6282 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6284 * gst/rtsp-server/rtsp-server.c:
6285 * gst/rtsp-server/rtsp-server.h:
6286 server: use a threadpool for the mainloops
6288 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6290 * gst/rtsp-server/rtsp-client.c:
6291 * gst/rtsp-server/rtsp-client.h:
6292 client: rename method
6293 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
6294 don't really create the client from the socket, we use the socket for the
6297 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6299 * gst/rtsp-server/rtsp-client.c:
6300 * gst/rtsp-server/rtsp-client.h:
6301 * gst/rtsp-server/rtsp-server.c:
6302 server: rework maincontext handling in clients
6303 Make a separate method to attach a client to a MainContext.
6304 Let the server decide in what GMainContext the client will operate and give this
6305 context to the client in attach. Then the server can later decide to use a
6306 separate thread for each client or just use the mainthread.
6308 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6310 * gst/rtsp-server/rtsp-client.c:
6311 * gst/rtsp-server/rtsp-session.c:
6312 * gst/rtsp-server/rtsp-session.h:
6313 session: move session header code in session object
6315 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
6319 * examples/test-auth.c:
6320 * examples/test-launch.c:
6321 * examples/test-mp4.c:
6322 * examples/test-ogg.c:
6323 * examples/test-readme.c:
6324 * examples/test-sdp.c:
6325 * examples/test-uri.c:
6326 * examples/test-video.c:
6327 * gst/rtsp-server/rtsp-auth.c:
6328 * gst/rtsp-server/rtsp-auth.h:
6329 * gst/rtsp-server/rtsp-client.c:
6330 * gst/rtsp-server/rtsp-client.h:
6331 * gst/rtsp-server/rtsp-media-factory-uri.c:
6332 * gst/rtsp-server/rtsp-media-factory-uri.h:
6333 * gst/rtsp-server/rtsp-media-factory.c:
6334 * gst/rtsp-server/rtsp-media-factory.h:
6335 * gst/rtsp-server/rtsp-media-mapping.c:
6336 * gst/rtsp-server/rtsp-media-mapping.h:
6337 * gst/rtsp-server/rtsp-media.c:
6338 * gst/rtsp-server/rtsp-media.h:
6339 * gst/rtsp-server/rtsp-params.c:
6340 * gst/rtsp-server/rtsp-params.h:
6341 * gst/rtsp-server/rtsp-sdp.c:
6342 * gst/rtsp-server/rtsp-sdp.h:
6343 * gst/rtsp-server/rtsp-server.c:
6344 * gst/rtsp-server/rtsp-server.h:
6345 * gst/rtsp-server/rtsp-session-media.c:
6346 * gst/rtsp-server/rtsp-session-media.h:
6347 * gst/rtsp-server/rtsp-session-pool.c:
6348 * gst/rtsp-server/rtsp-session-pool.h:
6349 * gst/rtsp-server/rtsp-session.c:
6350 * gst/rtsp-server/rtsp-session.h:
6351 * gst/rtsp-server/rtsp-stream-transport.c:
6352 * gst/rtsp-server/rtsp-stream-transport.h:
6353 * gst/rtsp-server/rtsp-stream.c:
6354 * gst/rtsp-server/rtsp-stream.h:
6355 * tests/check/gst/rtspserver.c:
6356 * tests/test-cleanup.c:
6359 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6361 * gst/rtsp-server/rtsp-media.c:
6362 * gst/rtsp-server/rtsp-session-media.c:
6363 * gst/rtsp-server/rtsp-session.c:
6364 rtsp-server: added annotations to indicate type of ownership transfer of return values
6365 https://bugzilla.gnome.org/show_bug.cgi?id=680777
6367 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
6370 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
6372 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
6375 * bindings/Makefile.am:
6376 * bindings/vala/Makefile.am:
6377 * bindings/vala/gst-rtsp-server-0.10.deps:
6378 * bindings/vala/gst-rtsp-server-0.10.vapi:
6379 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
6380 * bindings/vala/packages/gst-rtsp-server-0.10.files:
6381 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6382 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6383 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
6385 bindings: remove vala bindings
6386 They'll be reunited with the other GStreamer bindings
6387 https://bugzilla.gnome.org/show_bug.cgi?id=680777
6389 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6391 * gst/rtsp-server/rtsp-client.c:
6392 * gst/rtsp-server/rtsp-session-media.c:
6393 * gst/rtsp-server/rtsp-session-media.h:
6394 * gst/rtsp-server/rtsp-stream-transport.c:
6395 * gst/rtsp-server/rtsp-stream-transport.h:
6396 rtsp: only create transport when needed
6397 Only create the StreamTransport when configured.
6399 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6401 * gst/rtsp-server/rtsp-client.c:
6402 client: small cleanup
6404 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6406 * gst/rtsp-server/rtsp-client.c:
6407 * gst/rtsp-server/rtsp-client.h:
6408 * gst/rtsp-server/rtsp-stream-transport.c:
6409 * gst/rtsp-server/rtsp-stream-transport.h:
6410 rtsp: refactor configuration of transport
6411 Move the configuration of the transport to a place where it makes
6414 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6416 * gst/rtsp-server/rtsp-client.c:
6417 client: refactor transport parsing
6419 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6421 * gst/rtsp-server/rtsp-client.c:
6422 client: refuse to change the MTU on shared media
6423 If we change the MTU of chared media, it changes for all clients.
6424 We don't want to set the MTU to something large for clients that
6427 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6429 * examples/test-mp4.c:
6430 * gst/rtsp-server/rtsp-media.c:
6431 small fixes to docs and debug
6433 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6435 * gst/rtsp-server/rtsp-stream.c:
6436 stream: transports must already have been removed
6438 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6440 * gst/rtsp-server/rtsp-media.c:
6441 * gst/rtsp-server/rtsp-stream.c:
6442 * gst/rtsp-server/rtsp-stream.h:
6443 stream: improve join and leave of the pipeline
6445 Do the cleanup properly
6448 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6450 * gst/rtsp-server/rtsp-media.c:
6451 media: move unprepare below default implementation
6452 Makes it easier to find the default implementation
6454 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6456 * gst/rtsp-server/rtsp-media.c:
6457 media: signal unprepared when we actually finish
6459 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6461 * gst/rtsp-server/rtsp-media.c:
6462 media: no need to unlock, unprepare does that when needed
6464 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6466 * docs/libs/gst-rtsp-server-sections.txt:
6467 * gst/rtsp-server/rtsp-media-factory.h:
6468 * gst/rtsp-server/rtsp-media-mapping.c:
6469 * gst/rtsp-server/rtsp-media.h:
6470 * gst/rtsp-server/rtsp-params.c:
6471 * gst/rtsp-server/rtsp-server.c:
6472 * gst/rtsp-server/rtsp-session-pool.h:
6473 * gst/rtsp-server/rtsp-session.c:
6474 * gst/rtsp-server/rtsp-session.h:
6475 * gst/rtsp-server/rtsp-stream-transport.h:
6476 * gst/rtsp-server/rtsp-stream.h:
6479 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6481 * gst/rtsp-server/rtsp-client.c:
6482 * gst/rtsp-server/rtsp-media-mapping.h:
6483 * gst/rtsp-server/rtsp-media.c:
6484 * gst/rtsp-server/rtsp-media.h:
6485 * gst/rtsp-server/rtsp-server.h:
6486 * gst/rtsp-server/rtsp-stream.c:
6487 * gst/rtsp-server/rtsp-stream.h:
6488 rtsp: fix MTU setting
6489 Fix setting of the MTU. There is no need for a vmethod.
6491 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6496 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6499 configure: bump version number after refactoring
6501 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6503 * gst/rtsp-server/Makefile.am:
6504 * gst/rtsp-server/rtsp-client.c:
6505 * gst/rtsp-server/rtsp-client.h:
6506 * gst/rtsp-server/rtsp-media-factory-uri.c:
6507 * gst/rtsp-server/rtsp-media-factory.c:
6508 * gst/rtsp-server/rtsp-media-factory.h:
6509 * gst/rtsp-server/rtsp-media.c:
6510 * gst/rtsp-server/rtsp-media.h:
6511 * gst/rtsp-server/rtsp-sdp.c:
6512 * gst/rtsp-server/rtsp-session-media.c:
6513 * gst/rtsp-server/rtsp-session-media.h:
6514 * gst/rtsp-server/rtsp-session.c:
6515 * gst/rtsp-server/rtsp-session.h:
6516 * gst/rtsp-server/rtsp-stream-transport.c:
6517 * gst/rtsp-server/rtsp-stream-transport.h:
6518 * gst/rtsp-server/rtsp-stream.c:
6519 * gst/rtsp-server/rtsp-stream.h:
6520 rtsp: massive refactoring
6521 Make GObjects from the remaining simple structures.
6522 Remove GstRTSPSessionStream, it's not needed.
6523 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
6524 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
6525 a GstRTSPStream should be transported to a client.
6526 Rename GstRTSPMediaFactory::get_element -> create_element because that
6527 more accurately describes what it does.
6528 Make nice methods instead of poking in the structures.
6529 Move some methods inside the relevant object source code.
6530 Use GPtrArray to store objects instead of plain arrays, it is more
6531 natural and allows us to more easily clean up.
6532 Move the allocation of udp ports to the Stream object. The Stream object
6533 contains the elements needed to stream the media to a client.
6534 Improve the prepare and unprepare methods. Unprepare should now undo
6535 everything prepare did. Improve also async unprepare when doing EOS on
6536 shutdown. Make sure we always unprepare correctly.
6538 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
6540 * gst/rtsp-server/rtsp-client.c:
6541 rtsp-client: Unref server address clients connected to
6542 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
6544 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
6546 * gst/rtsp-server/rtsp-server.c:
6547 rtsp-server: don't ref server socket if it is NULL
6548 Fixes test_bind_already_in_use unit test again after commit 6a497440.
6549 https://bugzilla.gnome.org/show_bug.cgi?id=686644
6551 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
6553 * tests/check/Makefile.am:
6554 tests: Add libgio link dependency
6555 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
6557 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6559 * gst/rtsp-server/rtsp-media-mapping.c:
6560 * gst/rtsp-server/rtsp-media-mapping.h:
6561 rtsp-media-mapping: rename find_media vfunc to find_factory
6562 The virtual method and class method should have the same name
6563 so it is correctly represented in GIR file
6564 https://bugzilla.gnome.org/show_bug.cgi?id=680777
6566 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6568 * gst/rtsp-server/rtsp-auth.c:
6569 * gst/rtsp-server/rtsp-client.c:
6570 * gst/rtsp-server/rtsp-media-factory-uri.c:
6571 * gst/rtsp-server/rtsp-media-factory.c:
6572 * gst/rtsp-server/rtsp-media-mapping.c:
6573 * gst/rtsp-server/rtsp-media.c:
6574 * gst/rtsp-server/rtsp-server.c:
6575 * gst/rtsp-server/rtsp-session-pool.c:
6576 * gst/rtsp-server/rtsp-session.c:
6577 rtsp-server: fixed comments and GIR annotations
6578 https://bugzilla.gnome.org/show_bug.cgi?id=680777
6580 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
6582 * gst/rtsp-server/rtsp-media-mapping.c:
6583 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
6585 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
6587 * gst/rtsp-server/rtsp-server.c:
6588 rtsp-server: allow binding on port 0 (binds on a random port)
6590 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
6592 * gst/rtsp-server/rtsp-server.c:
6593 * gst/rtsp-server/rtsp-server.h:
6594 rtsp-server: add bound-port property
6595 bound-port can be used to retrieve the port number when the server is bound on
6596 port 0, which binds on a random port.
6598 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
6600 * gst/rtsp-server/rtsp-media-factory.c:
6601 * gst/rtsp-server/rtsp-media-factory.h:
6602 rtsp-media-factory: make ::get_element overridable by GI bindings
6603 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
6604 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
6605 as the invoker for ::get_element(), making it overridable by GI generated
6608 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
6610 * gst/rtsp-server/rtsp-media-factory-uri.c:
6611 rtsp-media-factory-uri: don't autoplug parsers in a loop
6612 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
6615 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
6617 * gst/rtsp-server/Makefile.am:
6618 Explicitly link against gio. Fix link error on mac.
6620 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
6622 * gst/rtsp-server/rtsp-session.c:
6623 session: add ttl to the transport header in SETUP
6624 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
6626 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
6628 * gst/rtsp-server/rtsp-client.c:
6629 * gst/rtsp-server/rtsp-client.h:
6630 * gst/rtsp-server/rtsp-media.c:
6631 client: Use client transport settings for multicast if allowed.
6632 This patch makes it possible for the client to send transport settings for
6633 multicast (destination && ttl). Client settings must be explicitly allowed or
6634 the server will use its own settings.
6635 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
6637 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
6640 Automatic update of common submodule
6641 From 6c0b52c to 6bb6951
6643 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
6645 * gst/rtsp-server/rtsp-client.c:
6646 rtsp-client: do not destroy the rtsp watch
6647 Don't destroy the client watch while dispatching. The rtsp watch is
6648 automatically destroyed after the rtsp watch function closed() has
6650 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
6652 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
6655 Automatic update of common submodule
6656 From 4f962f7 to 6c0b52c
6658 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
6660 * gst/rtsp-server/rtsp-media.c:
6661 media: fix check for seekability
6663 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6665 * gst/rtsp-server/rtsp-client.c:
6666 client: use more GIO
6667 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
6669 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6671 * gst/rtsp-server/rtsp-server.c:
6672 server: remove obsolete includes
6674 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
6676 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
6677 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
6678 be available in "on_new_ssrc". The transports are added in
6679 gst_rtsp_media_set_state when going to PLAYING state. However,
6680 "on_new_ssrc" might be called before this happens.
6681 https://bugzilla.gnome.org/show_bug.cgi?id=683304
6683 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
6685 * gst/rtsp-server/rtsp-client.c:
6686 * gst/rtsp-server/rtsp-client.h:
6687 rtsp-client: add signals for rtsp requests (fixes #683287)
6689 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
6691 * gst/rtsp-server/rtsp-client.c:
6692 * gst/rtsp-server/rtsp-client.h:
6693 add new-session signal to rtsp-client (fixes #683058)
6695 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
6698 Automatic update of common submodule
6699 From 668acee to 4f962f7
6701 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
6703 * gst/rtsp-server/rtsp-server.c:
6704 * tests/check/gst/rtspserver.c:
6705 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
6706 Do not assume that *error is set in g_socket_address_enumerator_next.
6707 Added test_bind_already_in_use unit-test.
6708 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
6710 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
6713 Automatic update of common submodule
6714 From 94ccf4c to 668acee
6716 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
6718 * gst/rtsp-server/rtsp-client.c:
6719 * gst/rtsp-server/rtsp-client.h:
6720 rtsp-client: make create_sdp virtual method
6721 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
6723 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6726 Automatic update of common submodule
6727 From 98e386f to 94ccf4c
6729 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6731 * gst/rtsp-server/rtsp-client.c:
6734 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
6736 * gst/rtsp-server/rtsp-client.c:
6737 * gst/rtsp-server/rtsp-client.h:
6738 * gst/rtsp-server/rtsp-server.c:
6739 * gst/rtsp-server/rtsp-server.h:
6740 rtsp-server: use an existing socket to establish HTTP tunnel
6741 Make it possible to transfer a socket from an HTTP server to be used as
6742 an RTSP over HTTP tunnel.
6744 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
6746 * gst/rtsp-server/rtsp-client.c:
6747 * gst/rtsp-server/rtsp-media.c:
6748 * gst/rtsp-server/rtsp-media.h:
6749 rtsp: Handle the blocksize parameter
6750 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
6752 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
6754 * tests/check/Makefile.am:
6755 * tests/check/gst/rtspserver.c:
6756 Have unit test get header from source dir, not installed dir
6757 This makes compilation of unit tests work in a build directory other
6758 than the source directory.
6759 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
6761 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
6763 * gst/rtsp-server/rtsp-media.c:
6764 rtsp-media: update for gst_element_make_from_uri() changes
6766 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
6769 * tests/Makefile.am:
6770 * tests/check/Makefile.am:
6771 * tests/check/gst/rtspserver.c:
6773 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
6775 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
6777 * gst/rtsp-server/rtsp-media.c:
6778 rtsp-media: don't collect media stats when going to NULL
6779 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
6781 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6783 * gst/rtsp-server/rtsp-client.c:
6784 client: don't leak transports
6786 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
6788 * gst/rtsp-server/rtsp-client.c:
6789 rtsp-client: free transport on no_stream in SETUP handler
6791 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
6793 * gst/rtsp-server/rtsp-client.c:
6794 rtsp-client: changed session media iteration
6795 In client_unlink_session: now don't iterate in session->medias
6796 list where items are removed by gst_rtsp_session_release_media.
6797 Instead, repeatedly remove the first item.
6799 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
6801 * gst/rtsp-server/rtsp-client.c:
6802 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
6803 GstRTSPSessionMedia is not a GObject type. When the
6804 GstRTSPSession is freed, it will free the media.
6806 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
6808 * gst/rtsp-server/rtsp-media-factory.c:
6809 factory: plug pad leak in collect_streams
6810 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
6811 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
6812 will take one reference, and the other reference will otherwise
6815 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
6818 configure: suppress some warnings when debug is disabled
6819 Warnings about unused variables should be suppressed if core has the
6820 debug system disabled.
6821 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
6823 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6825 * docs/libs/Makefile.am:
6826 docs: fix build in uninstalled setup
6827 Include gst-plugins-base libs properly.
6829 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
6831 * docs/libs/gst-rtsp-server.types:
6832 docs: include headers defining rtsp-server object types
6833 Fixes compiler warnings during docs build.
6834 https://bugzilla.gnome.org/show_bug.cgi?id=676824
6836 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
6839 configure: Add warning flags for compiler when configuring
6840 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
6842 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6845 Automatic update of common submodule
6846 From 03a0e57 to 98e386f
6848 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6851 Automatic update of common submodule
6852 From 1fab359 to 03a0e57
6854 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
6856 * gst/rtsp-server/rtsp-client.c:
6857 client: fix GSocketAddress leak in gst_rtsp_client_accept
6858 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
6860 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6863 Automatic update of common submodule
6864 From f1b5a96 to 1fab359
6866 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6869 Automatic update of common submodule
6870 From 92b7266 to f1b5a96
6872 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6875 Automatic update of common submodule
6876 From ec1c4a8 to 92b7266
6878 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6881 Automatic update of common submodule
6882 From 3429ba6 to ec1c4a8
6884 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
6886 * gst/rtsp-server/rtsp-auth.c:
6887 * gst/rtsp-server/rtsp-client.c:
6888 * gst/rtsp-server/rtsp-media-factory-uri.c:
6889 * gst/rtsp-server/rtsp-server.c:
6890 rtsp: fix compiler warnings
6891 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
6893 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6896 Automatic update of common submodule
6897 From dc70203 to 3429ba6
6899 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6901 * gst/rtsp-server/rtsp-client.c:
6902 * gst/rtsp-server/rtsp-media-factory.c:
6903 * gst/rtsp-server/rtsp-media-factory.h:
6904 * gst/rtsp-server/rtsp-media.c:
6905 * gst/rtsp-server/rtsp-media.h:
6906 * gst/rtsp-server/rtsp-server.c:
6907 * gst/rtsp-server/rtsp-server.h:
6908 * gst/rtsp-server/rtsp-session-pool.c:
6909 * gst/rtsp-server/rtsp-session-pool.h:
6910 rtsp-server: port to new thread API
6912 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6915 Automatic update of common submodule
6916 From 6db25be to dc70203
6918 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6920 * gst/rtsp-server/rtsp-auth.c:
6921 * gst/rtsp-server/rtsp-auth.h:
6922 * gst/rtsp-server/rtsp-client.c:
6923 rtsp-server: Fix compilation and compiler warnings
6925 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6929 * gst/rtsp-server/Makefile.am:
6930 configure: Modernize autotools setup a bit
6931 Also we now only create tar.bz2 and tar.xz tarballs.
6933 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6936 Automatic update of common submodule
6937 From 464fe15 to 6db25be
6939 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6942 Automatic update of common submodule
6943 From 7fda524 to 464fe15
6945 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6948 * docs/libs/Makefile.am:
6949 * docs/version.entities.in:
6951 * gst/rtsp-server/Makefile.am:
6952 * pkgconfig/Makefile.am:
6953 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
6954 * pkgconfig/gstreamer-rtsp-server.pc.in:
6955 * tests/Makefile.am:
6956 rtsp-server: Update versioning
6958 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6960 Merge remote-tracking branch 'origin/0.10'
6962 gst/rtsp-server/rtsp-session-pool.c
6964 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6966 * gst/rtsp-server/rtsp-session-pool.c:
6967 rtsp-server: Don't use deprecated GLib API
6969 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6971 Replace master with 0.11
6973 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6975 Merge branch 'master' into 0.11
6977 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6979 Merge branch 'master' into 0.11
6981 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
6984 A couple minor typo fixes
6986 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6988 * gst/rtsp-server/rtsp-media.c:
6989 media: fix state of the appqueue
6991 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6993 * gst/rtsp-server/rtsp-media-factory-uri.c:
6994 factory: use videoconvert
6996 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6998 * gst/rtsp-server/rtsp-media-factory-uri.c:
6999 factory: change to new style caps
7001 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7003 * gst/rtsp-server/rtsp-client.c:
7004 * gst/rtsp-server/rtsp-client.h:
7005 * gst/rtsp-server/rtsp-media-factory-uri.c:
7006 * gst/rtsp-server/rtsp-media.c:
7007 * gst/rtsp-server/rtsp-server.c:
7008 * gst/rtsp-server/rtsp-server.h:
7009 * gst/rtsp-server/rtsp-session-pool.c:
7010 rtsp-server: port to GIO
7013 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7016 configure: fix build
7018 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7021 docs: fix for gst_rtsp_server_set_port() -> _set_service()
7022 https://bugzilla.gnome.org/show_bug.cgi?id=666548
7024 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7027 * examples/Makefile.am:
7028 First rule of gst-rtsp-server club: don't talk about gst-phonon
7030 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7033 * pkgconfig/Makefile.am:
7034 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
7035 * pkgconfig/gstreamer-rtsp-server.pc.in:
7036 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
7037 For consistency with all other modules.
7039 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7041 * gst/rtsp-server/rtsp-client.c:
7042 rtsp-client: update for new map API
7044 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7047 * bindings/Makefile.am:
7048 * bindings/python/Makefile.am:
7049 * bindings/python/arg-types.py:
7050 * bindings/python/codegen/Makefile.am:
7051 * bindings/python/codegen/__init__.py:
7052 * bindings/python/codegen/argtypes.py:
7053 * bindings/python/codegen/code-coverage.py:
7054 * bindings/python/codegen/codegen.py:
7055 * bindings/python/codegen/definitions.py:
7056 * bindings/python/codegen/defsparser.py:
7057 * bindings/python/codegen/docextract.py:
7058 * bindings/python/codegen/docgen.py:
7059 * bindings/python/codegen/fileprefix.override:
7060 * bindings/python/codegen/fileprefixmodule.c:
7061 * bindings/python/codegen/h2def.py:
7062 * bindings/python/codegen/mergedefs.py:
7063 * bindings/python/codegen/mkskel.py:
7064 * bindings/python/codegen/override.py:
7065 * bindings/python/codegen/reversewrapper.py:
7066 * bindings/python/codegen/scmexpr.py:
7067 * bindings/python/rtspserver-types.defs:
7068 * bindings/python/rtspserver.defs:
7069 * bindings/python/rtspserver.override:
7070 * bindings/python/rtspservermodule.c:
7071 * bindings/python/test.py:
7073 python: remove pygst-based python bindings
7074 pygi is the future, apparently.
7076 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
7079 Automatic update of common submodule
7080 From c463bc0 to 7fda524
7082 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7085 Automatic update of common submodule
7086 From 2a59016 to c463bc0
7088 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7091 Automatic update of common submodule
7092 From 0807187 to 2a59016
7094 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7097 Automatic update of common submodule
7098 From 11f0cd5 to 0807187
7100 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7102 * examples/test-auth.c:
7103 example: update for new caps
7105 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7107 * examples/test-video.c:
7108 * gst/rtsp-server/rtsp-client.c:
7109 * gst/rtsp-server/rtsp-media-factory-uri.c:
7110 * gst/rtsp-server/rtsp-media.c:
7111 * gst/rtsp-server/rtsp-media.h:
7112 * gst/rtsp-server/rtsp-session.c:
7113 * gst/rtsp-server/rtsp-session.h:
7114 rtsp-server: port some more to 0.11
7116 Remove bufferlist stuff
7118 Add queue before appsink now that preroll-queue-len is gone.
7119 Update for request pad changes.
7121 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7123 Merge branch 'master' into 0.11
7125 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
7127 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7128 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
7129 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
7131 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
7133 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7134 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
7135 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
7137 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7139 Merge branch 'master' into 0.11
7141 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7143 * gst/rtsp-server/rtsp-media.c:
7144 * gst/rtsp-server/rtsp-media.h:
7145 media: add a seekable boolean
7146 Maintain the seekable state with a new variable instead of reusing the
7149 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
7151 * gst/rtsp-server/rtsp-media.c:
7152 Disallow seek in live media
7154 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7156 Merge branch 'master' into 0.11
7158 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
7160 * gst/rtsp-server/rtsp-server.c:
7161 #ifdef statements for windows socket creation were missing
7163 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
7166 Automatic update of common submodule
7167 From a39eb83 to 11f0cd5
7169 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
7172 Automatic update of common submodule
7173 From 605cd9a to a39eb83
7175 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7177 Merge branch 'master' into 0.11
7179 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7181 * gst/rtsp-server/rtsp-client.c:
7182 client: use method to access property
7184 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7186 * gst/rtsp-server/rtsp-media-factory.c:
7187 * gst/rtsp-server/rtsp-media-factory.h:
7188 media-factory: add protocols property
7189 Add a property to configure the allowed protocols in the media created from the
7192 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7194 * gst/rtsp-server/rtsp-media-factory.c:
7195 * gst/rtsp-server/rtsp-media-factory.h:
7196 media-factory: add media-configure signal
7197 Add signal to allow the application to configure the media after it was created
7200 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7202 * gst/rtsp-server/rtsp-client.c:
7203 client: use method to access property
7205 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7207 * gst/rtsp-server/rtsp-media-factory.c:
7208 * gst/rtsp-server/rtsp-media-factory.h:
7209 media-factory: add protocols property
7210 Add a property to configure the allowed protocols in the media created from the
7213 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7215 * gst/rtsp-server/rtsp-media-factory.c:
7216 * gst/rtsp-server/rtsp-media-factory.h:
7217 media-factory: add media-configure signal
7218 Add signal to allow the application to configure the media after it was created
7221 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7223 Merge branch 'master' into 0.11
7225 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7227 * gst/rtsp-server/rtsp-client.c:
7228 client: use media multicast group
7230 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7232 * gst/rtsp-server/rtsp-media-factory.h:
7233 * gst/rtsp-server/rtsp-server.h:
7234 * gst/rtsp-server/rtsp-session-pool.h:
7235 * gst/rtsp-server/rtsp-session.h:
7238 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
7240 * gst/rtsp-server/rtsp-client.c:
7241 * gst/rtsp-server/rtsp-sdp.h:
7242 sdp: copy and free the server ip address
7243 Copy and free the server ip address to make memory management easier later.
7245 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7247 * gst/rtsp-server/rtsp-media-factory.c:
7248 media-factory: configure multicast in media
7250 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7252 * gst/rtsp-server/rtsp-media.c:
7253 * gst/rtsp-server/rtsp-media.h:
7254 media: add property for multicast group
7255 Add a property to configure the multicast group in the media.
7256 Based on patches from Marc Leeman and Robert Krakora.
7258 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7260 * gst/rtsp-server/rtsp-media-factory.c:
7261 * gst/rtsp-server/rtsp-media-factory.h:
7262 media-factory: add property for multicast group
7263 Add a property to configure the multicast group in the media factory.
7264 Based on patches from Marc Leeman and Robert Krakora.
7266 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7268 * gst/rtsp-server/rtsp-client.c:
7269 client: do configuration of transport in one place
7270 Move the configuration of the transport destination address to where we also
7271 configure the other bits.
7273 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7275 * gst/rtsp-server/rtsp-client.c:
7276 client: use media multicast group
7278 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7280 * gst/rtsp-server/rtsp-media-factory.h:
7281 * gst/rtsp-server/rtsp-server.h:
7282 * gst/rtsp-server/rtsp-session-pool.h:
7283 * gst/rtsp-server/rtsp-session.h:
7286 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
7288 * gst/rtsp-server/rtsp-client.c:
7289 * gst/rtsp-server/rtsp-sdp.h:
7290 sdp: copy and free the server ip address
7291 Copy and free the server ip address to make memory management easier later.
7293 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7295 * gst/rtsp-server/rtsp-media-factory.c:
7296 media-factory: configure multicast in media
7298 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7300 * gst/rtsp-server/rtsp-media.c:
7301 * gst/rtsp-server/rtsp-media.h:
7302 media: add property for multicast group
7303 Add a property to configure the multicast group in the media.
7304 Based on patches from Marc Leeman and Robert Krakora.
7306 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7308 * gst/rtsp-server/rtsp-media-factory.c:
7309 * gst/rtsp-server/rtsp-media-factory.h:
7310 media-factory: add property for multicast group
7311 Add a property to configure the multicast group in the media factory.
7312 Based on patches from Marc Leeman and Robert Krakora.
7314 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7316 * gst/rtsp-server/rtsp-client.c:
7317 client: do configuration of transport in one place
7318 Move the configuration of the transport destination address to where we also
7319 configure the other bits.
7321 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7323 Merge branch 'master' into 0.11
7325 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
7327 * gst/rtsp-server/rtsp-client.c:
7328 client: destroy pipeline on client disconnect with no prior TEARDOWN.
7329 The problem occurs when the client abruptly closes the connection without
7330 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
7331 server is where the pipeline gets torn down. Since this handler is not called,
7332 the pipeline remains and is up and running. Subsequent clients get their own
7333 pipelines and if the do not issue TEARDOWNs then those pipelines will also
7334 remain up and running. This is a resource leak.
7336 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7338 Merge branch 'master' into 0.11
7340 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
7342 * gst/rtsp-server/rtsp-media-factory.c:
7343 * gst/rtsp-server/rtsp-media-factory.h:
7344 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
7345 For example, it can be used to retrieve source elements like appsrc, in a more
7346 convenient way than subclassing get_element.
7348 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7350 Merge branch 'master' into 0.11
7352 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
7354 * gst/rtsp-server/rtsp-server.c:
7355 rtsp-server: hold on to reference while using object
7357 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7359 * gst/rtsp-server/rtsp-media.c:
7362 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7365 configure: use unstable api
7367 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
7369 * gst/rtsp-server/rtsp-client.c:
7370 client: fix reference counting
7372 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
7374 * gst/rtsp-server/rtsp-client.c:
7375 * gst/rtsp-server/rtsp-media.c:
7376 fix compiler warnings about unused variables
7378 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
7380 * examples/test-launch.c:
7381 * examples/test-readme.c:
7382 * examples/test-uri.c:
7383 * examples/test-video.c:
7384 examples: tell rtsp uri when ready
7386 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
7389 Automatic update of common submodule
7390 From 69b981f to 605cd9a
7392 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7394 * gst/rtsp-server/rtsp-client.c:
7395 client: update for buffer API change
7397 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7399 * gst/rtsp-server/Makefile.am:
7400 Makefile.am: 0.10 => @GST_MAJORMINOR@
7402 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7404 * gst/rtsp-server/rtsp-media-factory-uri.c:
7405 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
7407 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7409 * gst/rtsp-server/.gitignore:
7410 .gitignore: 0.10 => 0.11
7412 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
7414 * gst/rtsp-server/Makefile.am:
7415 Makefile.am: 0.10 => @GST_MAJORMINOR@
7417 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7419 Merge branch 'master' into 0.11
7421 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
7424 Automatic update of common submodule
7425 From 9e5bbd5 to 69b981f
7427 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
7430 Automatic update of common submodule
7431 From fd35073 to 9e5bbd5
7433 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
7436 Automatic update of common submodule
7437 From 46dfcea to fd35073
7439 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7441 * gst/rtsp-server/rtsp-media-factory-uri.c:
7442 * gst/rtsp-server/rtsp-media.c:
7443 media: port to new caps API
7445 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7447 Merge branch 'master' into 0.11
7449 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
7451 * bindings/vala/gst-rtsp-server-0.10.vapi:
7452 Updated Vala bindings.
7453 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
7455 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
7457 * gst/rtsp-server/rtsp-server.c:
7458 * gst/rtsp-server/rtsp-server.h:
7459 Add a signal for newly connected clients.
7460 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
7462 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
7464 * bindings/python/rtspserver.override:
7465 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
7467 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7469 * gst/rtsp-server/Makefile.am:
7470 * gst/rtsp-server/rtsp-client.c:
7471 * gst/rtsp-server/rtsp-funnel.c:
7472 * gst/rtsp-server/rtsp-funnel.h:
7473 * gst/rtsp-server/rtsp-media.c:
7474 rtsp-server: port to 0.11
7476 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7481 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7483 Merge branch 'master' into 0.11
7488 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7491 Automatic update of common submodule
7492 From c3cafe1 to 46dfcea
7494 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
7496 * bindings/python/Makefile.am:
7497 * bindings/python/rtspserver.defs:
7498 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
7500 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
7502 * bindings/python/arg-types.py:
7503 python bindings: add GstRTSPUrlParam
7504 Needed to implement MediaFactory virtual proxies
7506 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
7508 * bindings/python/arg-types.py:
7509 python bindings: fix returning GstRTSPUrl types
7511 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
7513 * bindings/python/arg-types.py:
7514 python bindings: add arg type for GstRTSPUrl
7516 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
7518 * bindings/python/rtspserver.defs:
7519 python bindings: fix the definition of MediaFactory.collect_stream
7521 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
7524 Automatic update of common submodule
7525 From 1ccbe09 to c3cafe1
7527 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7530 Automatic update of common submodule
7531 From 193b717 to 1ccbe09
7533 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
7536 Automatic update of common submodule
7537 From b77e2bf to 193b717
7539 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7542 build: Include lcov.mak to allow test coverage report generation
7544 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7547 Automatic update of common submodule
7548 From d8814b6 to b77e2bf
7550 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7553 Automatic update of common submodule
7554 From 6aaa286 to d8814b6
7556 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
7559 Automatic update of common submodule
7560 From 6aec6b9 to 6aaa286
7562 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
7565 autogen: wingo signed comment
7567 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
7569 * gst/rtsp-server/rtsp-session-pool.c:
7570 session: use full charset for RTSP session ID
7571 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
7572 session ID more difficult.
7573 https://bugzilla.gnome.org/show_bug.cgi?id=643812
7575 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7577 * gst/rtsp-server/Makefile.am:
7578 rtsp-server: Don't install the funnel header
7580 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
7583 Automatic update of common submodule
7584 From 1de7f6a to 6aec6b9
7586 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7589 configure: require core/base 0.10.31
7590 Needed at least for gst_plugin_feature_rank_compare_func().
7592 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
7595 Automatic update of common submodule
7596 From f94d739 to 1de7f6a
7598 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7600 * gst/rtsp-server/rtsp-media.c:
7601 media: remove more unused code
7603 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7605 * gst/rtsp-server/rtsp-media.c:
7606 * gst/rtsp-server/rtsp-media.h:
7607 media: remove duplicate filtering
7608 Remove the duplicate filtering code now that we have a released -good version.
7609 Give a warning instead.
7611 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7613 * gst/rtsp-server/rtsp-media-factory.c:
7614 * gst/rtsp-server/rtsp-media.c:
7615 media: fix default buffer size
7617 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7619 * gst/rtsp-server/rtsp-media-factory.c:
7620 * gst/rtsp-server/rtsp-media-factory.h:
7621 media-factory: add property to configure the buffer-size
7622 Add a property to configure the kernel UDP buffer size.
7624 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7626 * gst/rtsp-server/rtsp-media.c:
7627 * gst/rtsp-server/rtsp-media.h:
7628 media: add property to configure kernel buffer sizes
7629 Add a property to configure the kernel UDP buffer size.
7631 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7634 configure: set PYGOBJECT_REQ before using it
7635 https://bugzilla.gnome.org/show_bug.cgi?id=640641
7637 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7640 docs: recursive into sub-directories on 'make upload'
7642 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7644 * docs/libs/gst-rtsp-server-docs.sgml:
7645 * docs/version.entities.in:
7646 docs: mention full version these docs are for, not just major-minor
7648 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7653 === release 0.10.8 ===
7655 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7660 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7662 * gst/rtsp-server/rtsp-server.c:
7663 rtsp-server: clarify docs a little
7665 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7667 * gst/rtsp-server/rtsp-media.c:
7668 media: init debug category before starting thread
7670 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7672 * gst/rtsp-server/rtsp-auth.c:
7673 auth: add realm to make it more spec compliant
7675 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7677 * gst/rtsp-server/rtsp-server.c:
7678 * gst/rtsp-server/rtsp-server.h:
7681 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7683 * examples/test-video.c:
7684 example: improve example docs a little
7686 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7688 * gst/rtsp-server/rtsp-server.c:
7689 server: ensure the watch has a ref to the server
7691 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7693 * gst/rtsp-server/rtsp-server.c:
7694 server: simpify channel function
7696 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7698 * gst/rtsp-server/rtsp-server.c:
7699 * gst/rtsp-server/rtsp-server.h:
7700 server: simplify management of channel and source
7701 We don't need to keep around the channel and source objects. Let the mainloop
7702 and the source manage the source and channel respectively.
7704 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7710 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7713 * tests/Makefile.am:
7714 * tests/test-cleanup.c:
7715 tests: add tests directory and cleanup test
7717 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7719 * gst/rtsp-server/rtsp-media-factory-uri.c:
7720 * gst/rtsp-server/rtsp-media-factory.c:
7721 * gst/rtsp-server/rtsp-media-mapping.c:
7722 * gst/rtsp-server/rtsp-media.c:
7723 * gst/rtsp-server/rtsp-session-pool.c:
7724 * gst/rtsp-server/rtsp-session.c:
7725 server: improve debugging in various objects
7727 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7729 * gst/rtsp-server/rtsp-server.c:
7730 server: chain up to the parent finalize
7732 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
7734 * bindings/python/rtspserver-types.defs:
7735 * bindings/python/rtspserver.defs:
7736 * bindings/python/rtspserver.override:
7737 * bindings/python/test.py:
7738 gst-rtsp-server: update python bindings
7740 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7742 * gst/rtsp-server/rtsp-client.c:
7743 client: use the response from the clientstate
7744 Create the response object only once and store in the client state.
7745 Make all methods use the state response,
7747 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7749 * gst/rtsp-server/rtsp-server.c:
7750 server: use signal to keep track of clients
7751 Keep track of all the clients that the server creates and remove them when they
7752 fire the 'closed' signal.
7754 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7756 * gst/rtsp-server/rtsp-client.c:
7757 * gst/rtsp-server/rtsp-client.h:
7758 client: emit signal when closing
7760 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7762 * examples/.gitignore:
7763 * examples/Makefile.am:
7764 * examples/test-auth.c:
7765 * examples/test-video.c:
7766 * gst/rtsp-server/rtsp-auth.c:
7767 * gst/rtsp-server/rtsp-auth.h:
7768 * gst/rtsp-server/rtsp-client.c:
7769 * gst/rtsp-server/rtsp-media-factory.c:
7770 * gst/rtsp-server/rtsp-media.c:
7771 * gst/rtsp-server/rtsp-media.h:
7772 * gst/rtsp-server/rtsp-session-pool.h:
7773 * gst/rtsp-server/rtsp-session.h:
7774 media: enable per factory authorisations
7775 Allow for adding a GstRTSPAuth on the factory and media level and check
7776 permissions when accessing the factory.
7777 Add hints to the auth methods for future more fine grained authorisation.
7778 Add example application for per factory authentication.
7780 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7782 * gst/rtsp-server/rtsp-auth.c:
7783 * gst/rtsp-server/rtsp-auth.h:
7784 * gst/rtsp-server/rtsp-client.c:
7785 * gst/rtsp-server/rtsp-client.h:
7786 * gst/rtsp-server/rtsp-params.c:
7787 * gst/rtsp-server/rtsp-params.h:
7788 rtsp-server: Pass ClientState structure arround
7789 Pass the collected information for the ongoing request in a GstRTSPClientState
7790 structure that we can then pass around to simplify the method arguments. This
7791 will also be handy when we implement logging functionality.
7793 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7795 * gst/rtsp-server/rtsp-media-factory.c:
7796 * gst/rtsp-server/rtsp-media-factory.h:
7797 media-factory: add methods to configure authorisation
7799 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7801 * gst/rtsp-server/rtsp-client.c:
7802 client: unref auth in finalize
7804 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7806 * gst/rtsp-server/rtsp-server.c:
7807 server: unref auth in finalize
7809 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7811 * docs/libs/gst-rtsp-server-docs.sgml:
7812 * docs/libs/gst-rtsp-server-sections.txt:
7813 * docs/libs/gst-rtsp-server.types:
7816 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7818 * gst/rtsp-server/rtsp-server.c:
7819 * gst/rtsp-server/rtsp-server.h:
7820 server: separate create and accept
7821 Create separate create and accept methods so that subclasses can create custom
7823 Configure the server in the client object and prepare for keeping track of
7826 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7828 * gst/rtsp-server/rtsp-client.c:
7829 * gst/rtsp-server/rtsp-client.h:
7830 client: add support for setting the server.
7831 Add support for keeping a ref to the server that started this client
7834 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7836 * gst/rtsp-server/rtsp-auth.c:
7837 auth: fix memleak and add some docs
7838 Fix a memleak of the basic auth token.
7839 Add docs for the helper function
7841 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7843 * gst/rtsp-server/rtsp-auth.c:
7844 * gst/rtsp-server/rtsp-auth.h:
7845 * gst/rtsp-server/rtsp-client.c:
7846 client: delegate setup of auth to the manager
7847 Delegate the configuration of the authentication tokens to the manager object
7850 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7852 * examples/test-video.c:
7853 * gst/rtsp-server/Makefile.am:
7854 * gst/rtsp-server/rtsp-auth.c:
7855 * gst/rtsp-server/rtsp-auth.h:
7856 * gst/rtsp-server/rtsp-client.c:
7857 * gst/rtsp-server/rtsp-client.h:
7858 * gst/rtsp-server/rtsp-server.c:
7859 * gst/rtsp-server/rtsp-server.h:
7860 auth: add authentication object
7861 Add an object that can check the authorization of requests.
7862 Implement basic authentication.
7863 Add example authentication to test-video
7865 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7867 * gst/rtsp-server/rtsp-server.c:
7868 * gst/rtsp-server/rtsp-server.h:
7869 server: move includes back
7870 the includes are needed for sockaddr_in.
7872 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7874 * gst/rtsp-server/rtsp-client.c:
7875 * gst/rtsp-server/rtsp-client.h:
7876 * gst/rtsp-server/rtsp-server.c:
7877 * gst/rtsp-server/rtsp-server.h:
7878 rtsp: move network includes where they are needed
7880 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
7882 * gst/rtsp-server/rtsp-media.h:
7883 rtsp-media.h: Minor corrections in comments.
7886 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
7889 Automatic update of common submodule
7890 From e572c87 to f94d739
7892 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7896 * docs/libs/.gitignore:
7897 * examples/.gitignore:
7898 * gst/rtsp-server/.gitignore:
7901 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7903 * docs/libs/Makefile.am:
7904 docs: We don't build ps/pdf for API reference docs
7906 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7909 Automatic update of common submodule
7910 From ccbaa85 to e572c87
7912 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7915 Automatic update of common submodule
7916 From 46445ad to ccbaa85
7918 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7920 * gst/rtsp-server/Makefile.am:
7921 * gst/rtsp-server/rtsp-funnel.c:
7922 * gst/rtsp-server/rtsp-funnel.h:
7923 * gst/rtsp-server/rtsp-media.c:
7924 funnel: rename fsfunnel to rtspfunnel
7925 Rename the funnel to avoid conflicts with the farsight one.
7927 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7929 * gst/rtsp-server/Makefile.am:
7930 * gst/rtsp-server/fs-funnel.c:
7931 * gst/rtsp-server/fs-funnel.h:
7932 * gst/rtsp-server/rtsp-media.c:
7933 rtsp-media: add and use fsfunnel
7934 Add a copy of fsfunnel to the build because input-selector removed the (broken)
7935 select-all property that we need.
7937 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7939 * gst/rtsp-server/Makefile.am:
7940 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
7941 Use PKG_CONFIG_PATH specified at configure time (if any) as well
7942 for the g-ir-compiler, rather than just assuming the env var has
7945 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7952 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
7954 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7957 * gst/rtsp-server/Makefile.am:
7958 gobject-introspection: fix g-i build for uninstalled setup
7959 Requires gst-plugins-base git (> 0.10.31.2).
7961 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7963 * examples/test-uri.c:
7964 examples: add some more options and comments
7966 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7968 * gst/rtsp-server/rtsp-media-factory-uri.c:
7969 factory-uri: use right property type
7971 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7973 * gst/rtsp-server/rtsp-media-factory-uri.c:
7974 factory-uri: attempt to configure buffer-lists
7975 Attempt to configure buffer lists in the payloader for improved performance.
7977 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7979 * gst/rtsp-server/rtsp-media.c:
7980 media: attempt to configure bigger UDP buffers
7981 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
7982 send buffers with high bitrate streams.
7984 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
7986 * gst/rtsp-server/rtsp-client.c:
7987 client: use the socket length from getsockname
7988 Use the length returned by getsockname to perform the getnameinfo call because
7989 the size can depend on the socket type and platform.
7992 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7994 * docs/libs/gst-rtsp-server-docs.sgml:
7995 * docs/libs/gst-rtsp-server-sections.txt:
7996 docs: add uri factory to the docs
7998 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8000 * gst/rtsp-server/rtsp-client.c:
8001 * gst/rtsp-server/rtsp-media.h:
8004 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8006 * gst/rtsp-server/rtsp-client.c:
8007 * gst/rtsp-server/rtsp-media.c:
8008 * gst/rtsp-server/rtsp-media.h:
8009 * gst/rtsp-server/rtsp-session.c:
8010 * gst/rtsp-server/rtsp-session.h:
8011 rtsp-server: add support for buffer lists
8012 Add support for sending bufferlists received from appsink.
8015 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8017 * gst/rtsp-server/rtsp-client.c:
8018 * gst/rtsp-server/rtsp-media.c:
8019 * gst/rtsp-server/rtsp-media.h:
8020 * gst/rtsp-server/rtsp-sdp.c:
8021 media: make method to retrieve the play range
8022 Make a method to retrieve the playback range so that we can conditionally create
8023 a different range for the SDP and the PLAY requests.
8025 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8027 * gst/rtsp-server/rtsp-media.c:
8028 * gst/rtsp-server/rtsp-media.h:
8029 media: add signal to notify of state changes
8031 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8033 * gst/rtsp-server/rtsp-client.h:
8034 client: cleanup headers
8036 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8038 * gst/rtsp-server/rtsp-client.c:
8041 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8043 * gst/rtsp-server/rtsp-media-factory-uri.c:
8044 * gst/rtsp-server/rtsp-media-factory-uri.h:
8045 factory-uri: add support for gstpay
8046 Add an option to prefer gstpay over decoder + raw payloader.
8048 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8050 * gst/rtsp-server/rtsp-media-factory-uri.c:
8051 * gst/rtsp-server/rtsp-media-factory-uri.h:
8052 factory-uri: rework the autoplugger.
8053 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
8056 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8058 * gst/rtsp-server/rtsp-media-factory-uri.c:
8059 factory-uri: use better factory filter
8060 Make better payloader filter based on autoplug rank and RTP use case.
8062 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8065 Automatic update of common submodule
8066 From 169462a to 46445ad
8068 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8070 * gst/rtsp-server/rtsp-server.c:
8071 server: set SO_REUSEADDR before bind
8072 Set the SO_REUSEADDR _before_ bind() to make it actually work.
8074 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8076 * gst/rtsp-server/rtsp-media.c:
8077 * gst/rtsp-server/rtsp-media.h:
8078 media: emit prepared signal when prepared
8079 Make a 'prepared' signal and emit it when we successfully prepared the element.
8080 This signal can be used to configure the media object after it has been prepared
8083 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
8086 Automatic update of common submodule
8087 From 011bcc8 to 169462a
8089 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
8091 python an optional dependency
8092 * configure.ac: Move up valgrind and g-i checks. Make the python
8093 dependency optional, as it was before.
8095 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8097 Merge branch 'master' into 0.11
8102 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8104 * gst/rtsp-server/rtsp-media.c:
8105 media: update range when active clients changed
8106 When we changed the number of active clients, update the current range
8107 information because we want the second client connecting to a shared resource
8108 continue from where the stream currently.
8110 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8112 * gst/rtsp-server/rtsp-media-factory-uri.c:
8113 * gst/rtsp-server/rtsp-media-factory-uri.h:
8114 factory-uri: add colorspace and fix pt
8115 Rework the way we pass data to the autoplugger.
8116 When we have raw caps, plug a converter element to make pluggin to raw
8117 payloaders more successful.
8118 Make sure all dynamically plugged payloaders have a unique payload types.
8120 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8122 * examples/Makefile.am:
8123 * examples/test-uri.c:
8124 example: add example of the uri factory
8126 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8128 * gst/rtsp-server/Makefile.am:
8129 * gst/rtsp-server/rtsp-media-factory-uri.c:
8130 * gst/rtsp-server/rtsp-media-factory-uri.h:
8131 * gst/rtsp-server/rtsp-server.h:
8132 factory-uri: add a factory to stream any URI
8133 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
8136 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8138 * gst/rtsp-server/rtsp-media.c:
8139 * gst/rtsp-server/rtsp-media.h:
8140 media: ignore spurious ASYNC_DONE messages
8141 When we are dynamically adding pads, the addition of the udpsrc elements will
8142 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
8143 the real ASYNC_DONE when everything is prerolled.
8145 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8147 * gst/rtsp-server/rtsp-media-factory.c:
8148 * gst/rtsp-server/rtsp-media-factory.h:
8149 media-factory: make lock macro
8151 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
8153 * gst/rtsp-server/rtsp-client.c:
8154 rtsp-server: Remove unused variable and dead assignment
8156 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
8158 * examples/test-launch.c:
8159 * examples/test-mp4.c:
8160 * examples/test-ogg.c:
8161 * examples/test-readme.c:
8162 * examples/test-sdp.c:
8163 * examples/test-video.c:
8164 examples: Run gst-indent
8166 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
8168 * gst/rtsp-server/rtsp-client.c:
8169 * gst/rtsp-server/rtsp-media-factory.c:
8170 * gst/rtsp-server/rtsp-media-mapping.c:
8171 * gst/rtsp-server/rtsp-media.c:
8172 * gst/rtsp-server/rtsp-params.c:
8173 * gst/rtsp-server/rtsp-sdp.c:
8174 * gst/rtsp-server/rtsp-server.c:
8175 * gst/rtsp-server/rtsp-session-pool.c:
8176 * gst/rtsp-server/rtsp-session.c:
8177 rtsp-server: Run gst-indent
8178 Since it wasn't using the upstream common previously, there was no
8179 indentation check before commiting.
8181 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
8183 * gst/rtsp-server/rtsp-media-mapping.h:
8184 * gst/rtsp-server/rtsp-media.c:
8185 * gst/rtsp-server/rtsp-media.h:
8186 * gst/rtsp-server/rtsp-sdp.c:
8187 * gst/rtsp-server/rtsp-session-pool.h:
8188 * gst/rtsp-server/rtsp-session.c:
8189 * gst/rtsp-server/rtsp-session.h:
8190 rtsp-server: Some more doc fixups
8192 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8195 Makefile: Add cruft-cleaning support
8197 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8202 * docs/libs/Makefile.am:
8203 * docs/libs/gst-rtsp-server-docs.sgml:
8204 * docs/libs/gst-rtsp-server-sections.txt:
8205 * docs/libs/gst-rtsp-server.types:
8206 * docs/version.entities.in:
8207 docs: Add gtk-doc build system
8209 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8211 * gst/rtsp-server/Makefile.am:
8212 Makefile.am: Use standard GIR make behaviour
8214 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
8218 autogen/configure: Bring more in sync to standard gst module behaviour
8220 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8222 * gst/rtsp-server/rtsp-media.c:
8223 media: warn and fail when gstrtpbin is not found
8225 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8228 configure: open 0.11 branch
8230 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
8234 Add common submodule
8236 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
8239 * common/Makefile.am:
8240 * common/c-to-xml.py:
8242 * common/coverage/coverage-report-entry.pl:
8243 * common/coverage/coverage-report.pl:
8244 * common/coverage/coverage-report.xsl:
8245 * common/coverage/lcov.mak:
8246 * common/gettext.patch:
8247 * common/glib-gen.mak:
8248 * common/gst-autogen.sh:
8249 * common/gst-xmlinspect.py:
8251 * common/gstdoc-scangobj:
8252 * common/gtk-doc-plugins.mak:
8253 * common/gtk-doc.mak:
8254 * common/m4/.gitignore:
8255 * common/m4/Makefile.am:
8257 * common/m4/as-ac-expand.m4:
8258 * common/m4/as-auto-alt.m4:
8259 * common/m4/as-compiler-flag.m4:
8260 * common/m4/as-compiler.m4:
8261 * common/m4/as-docbook.m4:
8262 * common/m4/as-libtool-tags.m4:
8263 * common/m4/as-libtool.m4:
8264 * common/m4/as-python.m4:
8265 * common/m4/as-scrub-include.m4:
8266 * common/m4/as-version.m4:
8267 * common/m4/ax_create_stdint_h.m4:
8268 * common/m4/check.m4:
8269 * common/m4/glib-gettext.m4:
8270 * common/m4/gst-arch.m4:
8271 * common/m4/gst-args.m4:
8272 * common/m4/gst-check.m4:
8273 * common/m4/gst-debuginfo.m4:
8274 * common/m4/gst-default.m4:
8275 * common/m4/gst-doc.m4:
8276 * common/m4/gst-error.m4:
8277 * common/m4/gst-feature.m4:
8278 * common/m4/gst-function.m4:
8279 * common/m4/gst-gettext.m4:
8280 * common/m4/gst-glib2.m4:
8281 * common/m4/gst-libxml2.m4:
8282 * common/m4/gst-plugindir.m4:
8283 * common/m4/gst-valgrind.m4:
8284 * common/m4/gtk-doc.m4:
8285 * common/m4/introspection.m4:
8287 * common/mangle-tmpl.py:
8288 * common/plugins.xsl:
8290 * common/release.mak:
8291 * common/scangobj-merge.py:
8292 * common/upload.mak:
8293 common: Remove static version
8295 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
8297 * common/m4/introspection.m4:
8298 Update introspection.m4 to match usage
8300 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8304 Remove old stuff from the README
8306 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8311 === release 0.10.7 ===
8313 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8318 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8320 * examples/test-ogg.c:
8321 test-ogg: remove parsers
8322 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
8323 buffers with timestamps. Using the parsers also seems to break things.
8325 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8327 * bindings/vala/gst-rtsp-server-0.10.vapi:
8328 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8329 Updated Vala bindings
8331 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8333 * common/m4/introspection.m4:
8335 * gst/rtsp-server/Makefile.am:
8336 Added initial gobject-introspection support
8338 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8340 * gst/rtsp-server/rtsp-media-factory.c:
8341 media-factory: don't use host for shared hash key
8342 When we generate the key to share made between connections, don't include the
8343 host used to connect so that we can share media even if between clients that
8344 connected with localhost and ones with the ip address.
8346 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8348 * bindings/vala/Makefile.am:
8349 build: fix distcheck
8351 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8353 * bindings/vala/gst-rtsp-server-0.10.vapi:
8354 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8355 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8356 Update Vala bindings
8358 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8360 * bindings/vala/Makefile.am:
8362 Fix configure checks and installation location for Vala bindings
8365 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8370 === release 0.10.6 ===
8372 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8375 configure: release 0.10.6
8377 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8379 * gst/rtsp-server/rtsp-media.c:
8380 media: help the compiler a little
8382 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8384 * gst/rtsp-server/rtsp-media.c:
8385 * gst/rtsp-server/rtsp-media.h:
8386 * gst/rtsp-server/rtsp-session.c:
8387 media: cleanup media transport before freeing
8388 Cleanup the media transport data before freeing. In particular, remove the qdata
8389 from the rtpsource object.
8391 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8393 * gst/rtsp-server/rtsp-media-factory.c:
8394 * gst/rtsp-server/rtsp-media-factory.h:
8395 * gst/rtsp-server/rtsp-media.c:
8396 * gst/rtsp-server/rtsp-media.h:
8397 media-factory: add eos-shutdown property
8398 Add an eos-shutdown property that will send an EOS to the pipeline before
8399 shutting it down. This allows for nice cleanup in case of a muxer.
8402 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8404 * gst/rtsp-server/rtsp-media.c:
8405 * gst/rtsp-server/rtsp-media.h:
8406 media: use multiudpsink send-duplicates when we can
8407 If we have a new enough multiudpsink with the send-duplicates property, use this
8408 instead of doing our own filtering. Our custom filtering code should eventually
8409 be removed when we can depend on a released -good.
8411 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8413 * gst/rtsp-server/rtsp-media.c:
8414 media: don't leak destinations
8415 Refactor and cleanup the destinations array when the stream is destroyed.
8417 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8419 * gst/rtsp-server/rtsp-media.c:
8420 * gst/rtsp-server/rtsp-media.h:
8421 media: don't add udp addresses multiple times
8422 Keep track of the udp addresses we added to udpsink and never add the same udp
8423 destination twice. This avoids duplicate packets when using multicast.
8425 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8427 * gst/rtsp-server/rtsp-server.c:
8428 server: disable use of SO_LINGER
8429 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
8430 server close()s the connection.
8432 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8434 * gst/rtsp-server/rtsp-server.c:
8435 server: use 5 second linger period in SO_LINGER
8436 Wait 5 seconds before clearing the send buffers and reseting the connection with
8437 the client when we do a close. This should be enough time to get the message to
8441 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
8443 * gst/rtsp-server/rtsp-server.c:
8444 server: use SO_LINGER
8445 SO_LINGER on the socket will make sure that any pending data on the socket is
8446 flushed ASAP and that the socket connection is reset. This makes sure that the
8447 socket can be reused immediately.
8450 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8453 README: add blurb about shared media factories
8455 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
8457 * gst/rtsp-server/rtsp-media.c:
8458 Add stdlib.h for atoi()
8460 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8462 * bindings/python/Makefile.am:
8463 * bindings/vala/Makefile.am:
8464 build: distcheck fixes
8465 Fix 'make distcheck', somewhat (it still fails because it tries to
8466 install files into /usr/share/vala/vapi/ irrespective of the
8469 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8472 configure: bump core/base requirements to released version
8473 Makes things less confusing for people.
8475 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8478 configure: fail if GStreamer core/base requirements are not met
8480 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8482 * gst/rtsp-server/rtsp-client.c:
8483 client: improve client cleanups
8484 Make sure the session does not timeout when using TCP. We need to do this
8485 because quicktime player does not send RTCP for some reason in tunneled
8487 Refactor some cleanup code.
8490 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8492 * gst/rtsp-server/rtsp-session.c:
8493 * gst/rtsp-server/rtsp-session.h:
8494 session: add support for prevent session timeouts
8495 Add an atomix counter to prevent session timeouts when we are, for example,
8498 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8500 * gst/rtsp-server/rtsp-client.c:
8501 client: fix unlink on session timeouts
8502 When our session times out, make sure we unlink all streams in this
8504 Remove the tunnelid when closing the connection.
8506 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8508 * gst/rtsp-server/rtsp-session.c:
8509 session: small cleanups
8511 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8513 * gst/rtsp-server/rtsp-client.c:
8514 client: handle lost_tunnel callbacks
8515 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
8516 hashtable so that we can reuse it for when the client reopens the POST
8518 Close the connection after a TEARDOWN.
8519 Make sure or watchid is cleared when the watch is removed.
8522 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8524 * gst/rtsp-server/rtsp-client.c:
8525 * gst/rtsp-server/rtsp-media.c:
8526 * gst/rtsp-server/rtsp-sdp.c:
8527 rtsp-server: add more support for multicast
8529 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8532 * gst/rtsp-server/rtsp-media.c:
8533 * gst/rtsp-server/rtsp-media.h:
8534 media: allow configuration of allowed lower transport
8536 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8538 * gst/rtsp-server/rtsp-client.h:
8539 * gst/rtsp-server/rtsp-media.c:
8540 * gst/rtsp-server/rtsp-media.h:
8541 * gst/rtsp-server/rtsp-sdp.c:
8542 * gst/rtsp-server/rtsp-sdp.h:
8543 * gst/rtsp-server/rtsp-server.c:
8544 rtsp: keep track of server ip and ipv6
8545 Keep track of how the client connected to the server and setup the udp ports
8546 with the same protocol.
8547 Copy the server ip address in the SDP so that clients can send RTCP back to
8550 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8552 * gst/rtsp-server/rtsp-session.c:
8555 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8557 * gst/rtsp-server/rtsp-client.c:
8558 client: use right size for malloc
8560 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8562 * gst/rtsp-server/rtsp-server.c:
8563 server: comment ipv6 server listening address
8565 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8567 * gst/rtsp-server/rtsp-media.c:
8568 media: allow for ipv6 sockets
8570 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8572 * gst/rtsp-server/rtsp-server.c:
8573 * gst/rtsp-server/rtsp-server.h:
8574 server: rework server part
8575 Allow setting a bind address, make sure we can deal with ipv6.
8576 Remove the port property and change with the service property.
8578 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8580 * gst/rtsp-server/rtsp-media.h:
8581 media: update comments a little
8583 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8585 * gst/rtsp-server/rtsp-client.c:
8586 client: make content-base better
8587 Use the URI formatting functions to make a content-base. Also make sure that
8588 there is a trailing / at the end.
8590 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8592 * gst/rtsp-server/rtsp-client.c:
8593 client: guard against invalid paths
8595 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8597 * examples/test-video.c:
8598 test: catch server bind errors
8600 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
8602 * gst/rtsp-server/rtsp-media.c:
8603 rtspmedia: emit "unprepared" if _prepare fails.
8604 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
8605 media object is removed from its factory's cache.
8607 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8609 * gst/rtsp-server/rtsp-media.c:
8610 media: collect media position when seek completes
8612 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
8614 * gst/rtsp-server/rtsp-client.c:
8615 client: call unlink_streams in client finalize
8618 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8620 * gst/rtsp-server/rtsp-media.c:
8621 media: limit the time to wait to something huge
8622 Avoid waiting forever but limit the timeout to 20 seconds.
8624 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8626 * gst/rtsp-server/rtsp-sdp.c:
8627 sdp: reindent and check for prepared status
8629 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8631 * gst/rtsp-server/rtsp-media.c:
8632 * gst/rtsp-server/rtsp-media.h:
8633 * gst/rtsp-server/rtsp-session.c:
8634 media: avoid doing _get_state() for state changes
8635 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
8636 until the media is prerolled or in error. This avoids doing a blocking call of
8637 gst_element_get_state() that can cause lockups when there is an error.
8640 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8642 * gst/rtsp-server/rtsp-media.c:
8645 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8647 * gst/rtsp-server/rtsp-media-factory.c:
8648 media-factory: better error handling
8649 Improve the error handling a bit.
8651 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8653 * gst/rtsp-server/rtsp-client.c:
8654 client: rework transport parsing
8655 Rework the transport parsing code so that we can ignore transports we don't
8656 support instead of just picking the first one we can parse.
8657 Configure a (for now hardcoded) destination for multicast transports.
8659 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8661 * gst/rtsp-server/rtsp-media.c:
8662 media: set multicast sink parameters
8663 Disable loop and automatic multicast join on the udpsink elements.
8664 Add some more debug info.
8665 Reset some state variables in the right place.
8666 Use the right port numbers for multicast.
8668 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8670 * gst/rtsp-server/rtsp-session.c:
8671 session: handle transport setup correctly
8672 Handle UDP, MCAST and TCP transport negotiation more correctly.
8673 Store the server session SSRC in the transport.
8675 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8677 * gst/rtsp-server/rtsp-client.c:
8678 rtsp-client: implement error_full
8679 Implement error_full to avoid some segfaults when the rtspconnection calls it.
8682 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8685 * gst/rtsp-server/rtsp-client.c:
8686 * gst/rtsp-server/rtsp-server.c:
8687 docs: update docs and comments
8689 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
8691 * gst/rtsp-server/rtsp-sdp.c:
8692 sdp: make server work better when behind a proxy
8694 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8696 * gst/rtsp-server/rtsp-client.c:
8697 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
8699 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8701 * gst/rtsp-server/rtsp-client.c:
8702 * gst/rtsp-server/rtsp-media-factory.c:
8703 * gst/rtsp-server/rtsp-media-mapping.c:
8704 * gst/rtsp-server/rtsp-media.c:
8705 * gst/rtsp-server/rtsp-server.c:
8706 * gst/rtsp-server/rtsp-session-pool.c:
8707 * gst/rtsp-server/rtsp-session.c:
8708 Use GStreamer's debugging subsystem
8710 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8712 * gst/rtsp-server/rtsp-media-factory.c:
8713 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
8715 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8720 === release 0.10.5 ===
8722 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8727 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8730 configure: bump required versions
8732 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
8734 * gst/rtsp-server/rtsp-client.c:
8735 client: call weak-unref on client->sessions from finalize
8738 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8740 * gst/rtsp-server/rtsp-media.c:
8741 media: Fixed crasher where caps got unref'ed too often
8743 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8746 * pkgconfig/.gitignore:
8747 * pkgconfig/Makefile.am:
8748 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
8749 Added pkg-config file to use gst-rtsp-server uninstalled
8751 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8753 * gst/rtsp-server/rtsp-media.c:
8754 media: add some docs
8756 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
8758 * gst/rtsp-server/rtsp-client.c:
8759 rtsp: Use gst_rtsp_watch_send_message().
8760 Use gst_rtsp_watch_send_message() since the old API which used
8761 gst_rtsp_watch_queue_message() has been deprecated.
8763 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8768 === release 0.10.4 ===
8770 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8775 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8777 * gst/rtsp-server/rtsp-client.c:
8778 * gst/rtsp-server/rtsp-session.c:
8779 * gst/rtsp-server/rtsp-session.h:
8780 rtsp: allocate channels in TCP mode
8781 When the client does not provide us with channels in TCP mode, allocate channels
8784 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8786 * gst/rtsp-server/rtsp-client.c:
8787 client: don't crash when tunnelid is missing
8788 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
8789 don't crash but return an error response to the client.
8792 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8794 * bindings/vala/gst-rtsp-server-0.10.vapi:
8795 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8796 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8797 bindings: update vala bindings with new method
8799 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8801 * gst/rtsp-server/rtsp-session-pool.c:
8802 * gst/rtsp-server/rtsp-session-pool.h:
8803 sessionpool: add function to filter sessions
8804 Add generic function to retrieve/remove sessions.
8806 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8809 configure: bump core/base requirements to release
8811 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8813 * gst/rtsp-server/rtsp-media.c:
8814 media: fix indentation
8816 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8818 * gst/rtsp-server/rtsp-media.c:
8819 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
8821 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8823 * gst/rtsp-server/rtsp-media.c:
8824 set state and remove elements of media in for loop
8826 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
8828 * bindings/vala/gst-rtsp-server-0.10.vapi:
8829 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8830 Added gst_rtsp_media_remove_elements function to Vala bindings
8832 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
8834 * gst/rtsp-server/rtsp-media.c:
8835 * gst/rtsp-server/rtsp-media.h:
8836 Added gst_rtsp_media_remove_elements function
8838 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
8840 * gst/rtsp-server/rtsp-media.c:
8841 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
8843 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8845 * bindings/vala/gst-rtsp-server-0.10.vapi:
8846 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8847 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8848 Updated Vala bindings
8850 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8852 * gst/rtsp-server/rtsp-media.c:
8853 * gst/rtsp-server/rtsp-media.h:
8854 Added vmethod unprepare to GstRTSPMedia
8855 The default implementation sets the state of the pipeline to GST_STATE_NULL
8857 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8859 * gst/rtsp-server/rtsp-media-factory.c:
8860 * gst/rtsp-server/rtsp-media-factory.h:
8861 Made collect_streams function public
8863 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8865 * gst/rtsp-server/rtsp-media-factory.c:
8866 * gst/rtsp-server/rtsp-media-factory.h:
8867 * gst/rtsp-server/rtsp-media.c:
8868 Added vmethod create_pipeline to GstRTSPMediaFactory
8869 The pipeline is created in this method and the GstRTSPMedia's element is added to it
8871 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8873 * gst/rtsp-server/rtsp-client.c:
8874 client: use g_source_destroy()
8875 We need to use g_source_destroy() because we might have added the source to a
8876 different main context than the default one.
8878 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8880 * gst/rtsp-server/Makefile.am:
8881 * gst/rtsp-server/rtsp-client.c:
8882 * gst/rtsp-server/rtsp-params.c:
8883 * gst/rtsp-server/rtsp-params.h:
8884 rtsp: prepare for handling GET/SET_PARAMETER
8885 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
8887 Fix return codes of handlers.
8889 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8891 * gst/rtsp-server/rtsp-media.c:
8892 media: don't leak session pads
8894 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8896 * gst/rtsp-server/rtsp-media.c:
8897 media: clean up the messages a bit
8899 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8901 * gst/rtsp-server/rtsp-sdp.c:
8902 sdp: warn and skip streams without media
8904 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8906 * bindings/vala/gst-rtsp-server-0.10.vapi:
8907 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8908 vala: Fixed typo in header file of RTSPMediaStream
8910 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8912 * gst/rtsp-server/rtsp-media.c:
8915 Make dumping RTCP stats configurable
8917 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8919 * gst/rtsp-server/rtsp-media.c:
8920 media: be less verbose and leak less
8922 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8924 * gst/rtsp-server/rtsp-media.c:
8925 media: don't leak the destination address
8927 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8929 * gst/rtsp-server/rtsp-client.c:
8930 * gst/rtsp-server/rtsp-media.c:
8931 * gst/rtsp-server/rtsp-media.h:
8932 * gst/rtsp-server/rtsp-session.c:
8933 * gst/rtsp-server/rtsp-session.h:
8934 rtsp: use RTCP to keep the session alive
8935 Use the RTCP rtcp-from stats field to find the associated session and use this
8936 to keep the session alive.
8938 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8940 * gst/rtsp-server/rtsp-session.c:
8941 session: add 5sec to the real session timeout
8942 Allow the session to live 5sec longer before really timing out. This should give
8943 clients some extra time to keep the session active.
8945 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8947 * gst/rtsp-server/rtsp-client.c:
8948 client: replay OK to GET/SET_PARAMETER
8949 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
8950 so that we return OK for those requests.
8952 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8954 * gst/rtsp-server/rtsp-media.c:
8955 * gst/rtsp-server/rtsp-media.h:
8956 media: keep track of active transports
8957 Keep track of which transport is active to avoid closing the connection too
8959 Remove the destination transport also when going to NULL.
8960 Print some stats about the SDES and other RTCP messages we receive from the
8963 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8965 * examples/.gitignore:
8966 * examples/Makefile.am:
8967 * examples/test-sdp.c:
8968 example: add SDP relay example
8970 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8972 * gst/rtsp-server/rtsp-media.c:
8973 media: also count active TCP connections
8975 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8977 * gst/rtsp-server/rtsp-media-factory.c:
8978 * gst/rtsp-server/rtsp-media.c:
8979 * gst/rtsp-server/rtsp-media.h:
8980 rtsp: add support for dynamic elements
8981 Add support for dynamic elements.
8982 Don't set live pipelines back to paused.
8984 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8986 * gst/rtsp-server/rtsp-sdp.c:
8987 sdp: don't add encoding name when absent in caps
8989 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8991 * gst/rtsp-server/rtsp-client.c:
8992 client: warn when we can't do RTP-Info
8994 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8996 * gst/rtsp-server/rtsp-media-factory.c:
8997 factory: factor out the stream construction
8999 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9001 * gst/rtsp-server/rtsp-client.c:
9002 client: only add RTP-Info when we have the info
9003 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
9006 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9011 === release 0.10.3 ===
9013 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9017 - Fixes a bug where it put the wrong verion in pkgconfig
9018 - Link RTP and RTCP sources
9020 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9022 * gst/rtsp-server/rtsp-media.c:
9023 * gst/rtsp-server/rtsp-media.h:
9024 media: link the RTP udpsrc to the session manager
9025 Link the RTP udpsrc and the appsrc to the session manager so that they don't
9026 shut down when the client sends a packet to open firewalls.
9028 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9030 * pkgconfig/gst-rtsp-server.pc.in:
9031 Don't use hard-coded version number in pkg-config file
9033 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9038 === release 0.10.2 ===
9040 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9045 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9048 * common/m4/.gitignore:
9049 * examples/.gitignore:
9050 * pkgconfig/.gitignore:
9051 add some .gitignore files
9053 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9055 * gst/rtsp-server/rtsp-media.c:
9056 media: seek to key frames
9058 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9060 * gst/rtsp-server/rtsp-media.c:
9061 media: emit the unprepared signal by id
9062 Emit the unprepared signal by id instead of name and set the media as
9065 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9067 * gst/rtsp-server/rtsp-media.c:
9068 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
9070 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9072 * gst/rtsp-server/rtsp-server.c:
9073 Added finalize function to GstRTPSPServer to unref session pool and media mapping
9075 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9077 * bindings/vala/gst-rtsp-server-0.10.vapi:
9078 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9079 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9080 Updated vala bindings
9082 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9084 * gst/rtsp-server/Makefile.am:
9085 * gst/rtsp-server/rtsp-client.c:
9086 * gst/rtsp-server/rtsp-media.c:
9087 server: use appsink and appsrc with the API
9088 Use the appsink/appsrc API instead of the signals for higher
9091 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9093 * examples/test-ogg.c:
9094 tests: set the payload type correctly
9096 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9098 * gst/rtsp-server/rtsp-media-factory.c:
9099 factory: connect to the unprepare signal
9100 Connect to the unprepare signal for non-reusable media so that we can remove
9101 them from the cache.
9103 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9105 * gst/rtsp-server/rtsp-media.c:
9106 * gst/rtsp-server/rtsp-media.h:
9107 media: add signal to notify of unprepare
9109 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9111 * gst/rtsp-server/rtsp-media.c:
9112 * gst/rtsp-server/rtsp-media.h:
9113 media: more work on making the media shared
9114 Add a reusable flag to medias, indicating that they can be reused after a state
9118 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9120 * examples/test-readme.c:
9121 examples: mark the example as shared for testing
9123 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9125 * gst/rtsp-server/rtsp-media.c:
9126 * gst/rtsp-server/rtsp-media.h:
9127 client: support shared media
9128 Always perform the state actions even if the target state of the pipeline is
9129 already correct, we still want to add/remove the transports when we are dealing
9131 Keep a counter of the number of active transports for a media so that we can use
9132 this to perform a state change when needed.
9133 Perform a state change of the pipeline only when the first transport was added
9134 or when there are no active transports.
9136 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9138 * gst/rtsp-server/rtsp-client.c:
9139 client: fix refcounting crasher
9140 Don't need to remove the weak refs in the finalize methods, they are already
9141 removed in the dispose.
9142 Don't register the callback with a DestroyNofity.
9144 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9146 * gst/rtsp-server/rtsp-client.c:
9147 Fix rtsp client refcount management in TCP mode.
9148 Don't unref a client ref we never had. Fixes an unref
9149 of an already-free client object after a client
9150 teardown request for me.
9152 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9154 * gst/rtsp-server/rtsp-session.c:
9155 docs: fix typo in API docs
9157 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9159 * gst/rtsp-server/rtsp-media.c:
9161 Keep the udp sources in playing even if we go to paused. unlock the sources when
9163 Add some more debug info.
9164 Only seek when we need to.
9165 Keep track of the position when we go to paused.
9167 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9169 * gst/rtsp-server/rtsp-client.c:
9170 * gst/rtsp-server/rtsp-media.c:
9171 * gst/rtsp-server/rtsp-media.h:
9172 Add beginnings of seeking.
9173 Parse the Range header and perform a seek on the pipeline for the requested
9174 position. It's disabled currently until I figure out what's going wrong.
9176 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9178 * gst/rtsp-server/rtsp-client.c:
9179 allow pause requests for now.
9182 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9184 * gst/rtsp-server/rtsp-client.c:
9185 Remove weak ref on the session in teardown
9186 We need to remove our weakref from the session when we do a teardown because
9187 else we close the TCP connection prematurely.
9189 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9191 * gst/rtsp-server/rtsp-client.c:
9192 * gst/rtsp-server/rtsp-client.h:
9193 * gst/rtsp-server/rtsp-session-pool.c:
9194 Do some more session cleanup
9195 Make session timeout kill the TCP connection that currently watches the
9197 Remove the client timeout property.
9199 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9201 * gst/rtsp-server/rtsp-client.c:
9202 * gst/rtsp-server/rtsp-client.h:
9203 * gst/rtsp-server/rtsp-media.c:
9204 * gst/rtsp-server/rtsp-media.h:
9205 * gst/rtsp-server/rtsp-server.c:
9206 * gst/rtsp-server/rtsp-session.c:
9207 * gst/rtsp-server/rtsp-session.h:
9209 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
9212 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9214 * examples/Makefile.am:
9215 * examples/test-launch.c:
9216 Add example server that takes launch lines
9217 Add an example server that streams any -launch line.
9219 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9221 * examples/test-readme.c:
9222 * gst/rtsp-server/rtsp-client.c:
9223 * gst/rtsp-server/rtsp-media.c:
9224 * gst/rtsp-server/rtsp-media.h:
9225 Add support for live streams
9226 Add support for live streams and ranges
9227 Start on handling TCP data transfer.
9229 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9231 * gst/rtsp-server/rtsp-media.c:
9232 Free the pipeline before other things
9235 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9237 * gst/rtsp-server/rtsp-client.c:
9238 Only free the pending tunnel if there is one
9241 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9243 * gst/rtsp-server/rtsp-client.c:
9244 * gst/rtsp-server/rtsp-client.h:
9245 * gst/rtsp-server/rtsp-media.c:
9246 rtsp-server: Add support for tunneling
9247 Add support for tunneling over HTTP.
9248 Use new connection methods to retrieve the url.
9249 Dispatch messages based on the message type instead of blindly
9250 assuming it's always a request.
9251 Keep track of the watch id so that we can remove it later.
9252 Set the media pipeline to NULL before unreffing the pipeline.
9254 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9256 * gst/rtsp-server/rtsp-client.c:
9257 * gst/rtsp-server/rtsp-client.h:
9258 Fix for channel -> watch rename in gstreamer
9259 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
9261 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9263 * gst/rtsp-server/rtsp-client.c:
9264 * gst/rtsp-server/rtsp-client.h:
9266 Use the async RTSP channels instead of spawning a new thread for each client.
9267 If a sessionid is specified in a request, fail if we don't have the session.
9269 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9271 * gst/rtsp-server/rtsp-media.c:
9272 Add better debug info
9273 Add some better debug info.
9275 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9277 * examples/test-video.c:
9279 Add support for session timeouts in the example.
9281 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9283 * gst/rtsp-server/rtsp-session-pool.c:
9284 * gst/rtsp-server/rtsp-session-pool.h:
9285 Pass GTimeVal around for performance reasons
9286 Get the current time only once and pass it around so that sessions don't have to
9287 get the current time anymore.
9288 Add experimental support for a GSource that dispatches when the session needs to
9291 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9293 * gst/rtsp-server/rtsp-session.c:
9294 * gst/rtsp-server/rtsp-session.h:
9295 Add better support for session timeouts
9296 Add a method to request the number of milliseconds when a session will timeout.
9298 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9300 * gst/rtsp-server/rtsp-media.c:
9301 * gst/rtsp-server/rtsp-media.h:
9302 Add suport for RTP manager monitoring
9303 Add the first stage in monitoring the rtp manager.
9304 Make sure we don't update the state to something we don't want.
9306 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9308 * gst/rtsp-server/rtsp-client.c:
9309 Add support for session keepalive
9310 Get and update the session timeout for all requests. get the session as early as
9313 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9315 * gst/rtsp-server/rtsp-media-factory.h:
9316 * gst/rtsp-server/rtsp-media.c:
9317 * gst/rtsp-server/rtsp-media.h:
9318 Handle media bus messages
9319 Handle media bus messages in a custom mainloop and dispatch them to the
9320 RTSPMedia objects. Let the default implementation handle some common messages.
9322 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9324 * gst/rtsp-server/rtsp-client.c:
9325 * gst/rtsp-server/rtsp-session-pool.c:
9326 * gst/rtsp-server/rtsp-session.c:
9327 Some more session timeout handling
9328 Move the session header setting code to a central place so that we always add
9329 the timeout parameter too.
9330 Handle timeouts by running the session cleanup code.
9331 Stop media before cleaning up.
9333 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9335 * gst/rtsp-server/rtsp-client.c:
9336 * gst/rtsp-server/rtsp-client.h:
9337 Add timeout property
9338 Add a timeout property ot the client and make the other properties into GObject
9341 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9343 * gst/rtsp-server/rtsp-session-pool.c:
9344 Use getters and setters in property code
9345 Use the getters and setters for the timeout property instead of locking
9348 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9350 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
9352 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9354 * gst/rtsp-server/rtsp-session-pool.c:
9355 * gst/rtsp-server/rtsp-session-pool.h:
9356 * gst/rtsp-server/rtsp-session.c:
9357 * gst/rtsp-server/rtsp-session.h:
9358 Add more timeout stuff
9359 Add method to check if a session is expired.
9360 Add method to perform cleanup on a session pool.
9362 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9364 * gst/rtsp-server/rtsp-client.c:
9365 * gst/rtsp-server/rtsp-session-pool.c:
9366 * gst/rtsp-server/rtsp-session-pool.h:
9367 * gst/rtsp-server/rtsp-session.c:
9368 * gst/rtsp-server/rtsp-session.h:
9369 Add beginnings of session timeouts and limits
9370 Add the timeout value to the Session header for unusual timeout values.
9371 Allow us to configure a limit to the amount of active sessions in a pool. Set a
9372 limit on the amount of retry we do after a sessionid collision.
9373 Add properties to the sessionid and the timeout of a session. Keep track of
9374 creation time and last access time for sessions.
9376 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9378 * gst/rtsp-server/rtsp-client.c:
9379 * gst/rtsp-server/rtsp-media.c:
9380 * gst/rtsp-server/rtsp-media.h:
9381 * gst/rtsp-server/rtsp-sdp.c:
9382 * gst/rtsp-server/rtsp-session-pool.c:
9383 * gst/rtsp-server/rtsp-session.c:
9384 * gst/rtsp-server/rtsp-session.h:
9385 Cleanup of sessions and more
9386 Fix the refcounting of media and sessions in the client. Properly clean up the
9387 session data when the client performs a teardown.
9388 Add Server header to responses.
9389 Allow for multiple uri setups in one session.
9390 Add Range header to the PLAY response and add the range attribute to the SDP
9392 Fix the session pool remove method, it used the wrong key in the hashtable. Also
9393 give the ownership of the sessionid to the session object.
9395 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9397 * gst/rtsp-server/rtsp-server.c:
9398 * gst/rtsp-server/rtsp-server.h:
9400 Rename the 'server_port' variable to simply 'port'.
9402 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9405 * gst/rtsp-server/rtsp-client.c:
9406 * gst/rtsp-server/rtsp-media.c:
9407 * gst/rtsp-server/rtsp-media.h:
9408 * gst/rtsp-server/rtsp-session.c:
9409 * gst/rtsp-server/rtsp-session.h:
9410 Rework the way we handle transports for streams
9411 Make the media accept an array of transports for the streams that we have
9412 configured for the play/pause requests.
9413 Implement server states for a client and its media.
9414 Require 0.10.22.1 (git HEAD) of gstreamer.
9416 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9418 * gst/rtsp-server/rtsp-client.c:
9419 * gst/rtsp-server/rtsp-media-factory.c:
9420 Drop const from functions dealing with urls
9421 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
9422 have the right const in them.
9424 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9426 * gst/rtsp-server/rtsp-client.c:
9427 * gst/rtsp-server/rtsp-media.c:
9428 * gst/rtsp-server/rtsp-sdp.c:
9432 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9434 * gst/rtsp-server/rtsp-client.c:
9435 * gst/rtsp-server/rtsp-media-factory.c:
9436 * gst/rtsp-server/rtsp-media.c:
9437 * gst/rtsp-server/rtsp-media.h:
9439 Don't keep a reference to the GstRTSPMedia in the stream.
9440 Free more things when freeing the GstRTSPMedia.
9442 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9445 * gst/rtsp-server/rtsp-media-factory.c:
9446 * gst/rtsp-server/rtsp-media-factory.h:
9447 * gst/rtsp-server/rtsp-media.c:
9448 * gst/rtsp-server/rtsp-media.h:
9449 * gst/rtsp-server/rtsp-server.c:
9450 * gst/rtsp-server/rtsp-server.h:
9451 More docs and small cleanups
9452 Add some more docs and update the README
9453 Cleanup some method names.
9454 Remove an unneeded idx field in the GstRTSPMediaStream
9456 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9459 * examples/Makefile.am:
9460 * examples/test-readme.c:
9461 Add a README and more example code
9462 Add a README file that contains a small introduction on how to use the server
9463 along with the example code explained in the readme.
9465 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9467 * gst/rtsp-server/rtsp-media.c:
9468 * gst/rtsp-server/rtsp-server.c:
9469 Fix some leaks and change default port
9470 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
9471 we finished the initial preroll. If we keep them locked, setting the pipeline to
9472 NULL will not stop and clean up the sources correctly.
9473 Change the default RTSP port to 8554 aka the official alternative RTSP port.
9475 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9477 * gst/rtsp-server/rtsp-session.c:
9478 * gst/rtsp-server/rtsp-session.h:
9479 Cleanups to the session object
9480 Remove some unneeded variables in the session state of a stream such as the
9481 owner media and the server transport.
9482 Get the configuration of a media stream in a session based on the media_stream
9483 in the original object instead of our cached index.
9484 Free more data in the finalize method.
9486 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9488 * gst/rtsp-server/rtsp-client.c:
9489 * gst/rtsp-server/rtsp-client.h:
9490 Cleanups and reuse media from DESCRIBE
9491 Handle thread create errors.
9492 Rename some internal methods to better match what they actually do.
9493 Handle misconfiguration of session_pool and media_mapping gracefully.
9494 Cache the DESCRIBE media and uri in the client connection and reuse them when
9495 we receive a SETUP request in the same connection for the same uri.
9496 Cleanup the client connection object.
9498 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9500 * gst/rtsp-server/rtsp-media-factory.c:
9501 * gst/rtsp-server/rtsp-media-factory.h:
9502 * gst/rtsp-server/rtsp-media.c:
9503 * gst/rtsp-server/rtsp-media.h:
9504 Add shared properties to media and factory
9505 Add the shared property to media.
9506 Implement some simple caching in the factory depending on if the media is shared
9509 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9511 * gst/rtsp-server/rtsp-client.c:
9512 Add a little comment
9513 Add some comment about the content-base header.
9515 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9517 * examples/Makefile.am:
9518 * examples/test-mp4.c:
9519 * examples/test-ogg.c:
9520 * examples/test-video.c:
9521 * gst/rtsp-server/Makefile.am:
9522 * gst/rtsp-server/rtsp-client.c:
9523 * gst/rtsp-server/rtsp-client.h:
9524 * gst/rtsp-server/rtsp-media-factory.c:
9525 * gst/rtsp-server/rtsp-media-factory.h:
9526 * gst/rtsp-server/rtsp-media.c:
9527 * gst/rtsp-server/rtsp-media.h:
9528 * gst/rtsp-server/rtsp-sdp.c:
9529 * gst/rtsp-server/rtsp-sdp.h:
9530 * gst/rtsp-server/rtsp-server.c:
9531 * gst/rtsp-server/rtsp-server.h:
9532 * gst/rtsp-server/rtsp-session.c:
9533 * gst/rtsp-server/rtsp-session.h:
9534 Reorganize things, prepare for media sharing
9535 Added various other test server examples
9536 Move the SDP message generation to a separate helper.
9537 Refactor common code for finding the session.
9538 Add content-base for realplayer compatibility
9539 Clean up request uris before processing for better vlc compatibility.
9540 Move prerolling and pipeline construction to the RTSPMedia object.
9541 Use multiudpsink for future pipeline reuse.
9543 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9549 === release 0.10.1 ===
9551 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9557 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9559 * bindings/vala/Makefile.am:
9561 Add more directories and files to the dist.
9563 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9565 * bindings/python/Makefile.am:
9566 * bindings/python/rtspserver.override:
9567 Fixed compile error of python bindings
9569 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9571 * bindings/vala/gst-rtsp-server-0.10.vapi:
9572 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9573 Marked values as nullable accordingly
9575 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9577 * bindings/vala/gst-rtsp-server-0.10.vapi:
9578 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
9579 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9580 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9581 Updated Vala bindings
9583 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9585 * gst/rtsp-server/rtsp-client.c:
9586 * gst/rtsp-server/rtsp-media-mapping.c:
9587 * gst/rtsp-server/rtsp-media-mapping.h:
9588 * gst/rtsp-server/rtsp-media.h:
9589 * gst/rtsp-server/rtsp-session-pool.h:
9590 Cleanups and doc updates
9591 Add some more documentation and do some minor cleanups here and there.
9593 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9595 * gst/rtsp-server/rtsp-client.c:
9596 * gst/rtsp-server/rtsp-media-factory.c:
9597 * gst/rtsp-server/rtsp-media-factory.h:
9598 * gst/rtsp-server/rtsp-media.c:
9599 * gst/rtsp-server/rtsp-media.h:
9600 * gst/rtsp-server/rtsp-session.c:
9601 * gst/rtsp-server/rtsp-session.h:
9603 Rename GstRTSPMediaBin to GstRTSPMedia
9604 Parse the request url into a GstRTSPUri object and pass this object to the
9605 various handlers and methods that require the uri.
9607 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9611 Add some more docs and remove some old code from the example.
9613 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9615 * gst/rtsp-server/rtsp-client.c:
9616 Handle state change failures better
9617 Handle state change failures better when changing the state of the pipeline to
9620 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9622 * gst/rtsp-server/rtsp-media-factory.c:
9623 * gst/rtsp-server/rtsp-media-factory.h:
9624 Make element creation more extendible
9625 Add get_element vmethod to the default MediaFactory so that subclasses can just
9626 override that method and still use the default logic for making a MediaBin from
9629 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9632 * gst/rtsp-server/Makefile.am:
9633 * gst/rtsp-server/rtsp-client.c:
9634 * gst/rtsp-server/rtsp-client.h:
9635 * gst/rtsp-server/rtsp-media-factory.c:
9636 * gst/rtsp-server/rtsp-media-factory.h:
9637 * gst/rtsp-server/rtsp-media-mapping.c:
9638 * gst/rtsp-server/rtsp-media-mapping.h:
9639 * gst/rtsp-server/rtsp-media.c:
9640 * gst/rtsp-server/rtsp-media.h:
9641 * gst/rtsp-server/rtsp-server.c:
9642 * gst/rtsp-server/rtsp-server.h:
9643 * gst/rtsp-server/rtsp-session.c:
9644 * gst/rtsp-server/rtsp-session.h:
9645 Make the server handle arbitrary pipelines
9646 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
9647 The GstMediaBin object has a handle to a bin with elements and to a list of
9648 GstMediaStream objects that this bin produces.
9649 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
9650 with methods to register and remove those mappings.
9651 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
9652 used by the server instance.
9653 Modify the example application so that it shows how to create custom pipelines
9654 attached to a specific mount point.
9655 Various misc cleanps.
9657 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9659 * gst/rtsp-server/rtsp-server.c:
9660 * gst/rtsp-server/rtsp-server.h:
9661 Allow setting a custom media factory for a server
9663 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9665 * gst/rtsp-server/rtsp-client.c:
9666 * gst/rtsp-server/rtsp-client.h:
9667 Allow setting a custom media factory for a client.
9669 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9671 * gst/rtsp-server/Makefile.am:
9672 Add Makefile entry for the media factory
9674 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9676 * gst/rtsp-server/rtsp-media-factory.c:
9677 * gst/rtsp-server/rtsp-media-factory.h:
9678 Add media factory to map urls to media pipeline objects.
9680 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9682 * gst/rtsp-server/rtsp-media.c:
9683 * gst/rtsp-server/rtsp-media.h:
9684 Add comments. Remove unused field
9686 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9688 * gst/rtsp-server/rtsp-session-pool.c:
9689 * gst/rtsp-server/rtsp-session-pool.h:
9690 Allow custom session pools to override the session id allocation algorithms Add some comments.
9692 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9694 * gst/rtsp-server/rtsp-session.h:
9697 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9699 * gst/rtsp-server/rtsp-client.c:
9700 * gst/rtsp-server/rtsp-client.h:
9701 Move the connection code in one place Add some comments
9703 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9705 * gst/rtsp-server/rtsp-server.c:
9706 * gst/rtsp-server/rtsp-server.h:
9707 Make vmethod to create and accept new clients. Add some docs.
9709 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9711 * gst/rtsp-server/rtsp-server.c:
9712 * gst/rtsp-server/rtsp-server.h:
9713 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
9715 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9717 * gst/rtsp-server/rtsp-client.c:
9718 * gst/rtsp-server/rtsp-client.h:
9719 Name the parameters more appropriately.
9721 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9723 * gst/rtsp-server/rtsp-session-pool.c:
9724 Do some more cleanup of the session pool.
9726 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9728 * gst/rtsp-server/Makefile.am:
9729 * gst/rtsp-server/rtsp-client.c:
9730 Check if return value of gst_rtsp_session_get_media is not NULL
9732 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9734 * gst/rtsp-server/Makefile.am:
9735 Install rtsp-session and rtsp-session-pool headers
9737 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9742 * bindings/python/Makefile.am:
9743 * bindings/python/arg-types.py:
9744 * bindings/python/codegen/Makefile.am:
9745 * bindings/python/codegen/__init__.py:
9746 * bindings/python/codegen/argtypes.py:
9747 * bindings/python/codegen/code-coverage.py:
9748 * bindings/python/codegen/codegen.py:
9749 * bindings/python/codegen/definitions.py:
9750 * bindings/python/codegen/defsparser.py:
9751 * bindings/python/codegen/docextract.py:
9752 * bindings/python/codegen/docgen.py:
9753 * bindings/python/codegen/fileprefix.override:
9754 * bindings/python/codegen/fileprefixmodule.c:
9755 * bindings/python/codegen/h2def.py:
9756 * bindings/python/codegen/mergedefs.py:
9757 * bindings/python/codegen/mkskel.py:
9758 * bindings/python/codegen/override.py:
9759 * bindings/python/codegen/reversewrapper.py:
9760 * bindings/python/codegen/scmexpr.py:
9761 * bindings/python/rtspserver-types.defs:
9762 * bindings/python/rtspserver.defs:
9763 * bindings/python/rtspserver.override:
9764 * bindings/python/rtspservermodule.c:
9766 Add python bindings.
9768 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9770 * bindings/Makefile.am:
9772 Don't go into python dir when requirements for python bindings are missing
9774 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9776 * bindings/Makefile.am:
9777 * bindings/vala/Makefile.am:
9779 Install Vala bindings if vala is available
9781 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9783 * bindings/vala/gst-rtsp-server-0.10.deps:
9784 * bindings/vala/gst-rtsp-server-0.10.vapi:
9785 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
9786 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
9787 * bindings/vala/packages/gst-rtsp-server-0.10.files:
9788 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9789 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9790 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
9791 Regenerated Vala bindings
9793 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9795 * bindings/vala/gst-rtsp-server.vapi:
9796 * bindings/vala/packages/gst-rtsp-server.metadata:
9797 Fixed typo in included headers for vala bindings
9799 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9803 * pkgconfig/Makefile.am:
9804 * pkgconfig/gst-rtsp-server.pc.in:
9805 Added pkgconfig file
9807 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
9809 * bindings/vala/gst-rtsp-server.vapi:
9810 * bindings/vala/packages/gst-rtsp-server.excludes:
9811 * bindings/vala/packages/gst-rtsp-server.gi:
9812 * bindings/vala/packages/gst-rtsp-server.metadata:
9813 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
9815 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
9817 * bindings/vala/gst-rtsp-server.vapi:
9818 * bindings/vala/packages/gst-rtsp-server.deps:
9819 * bindings/vala/packages/gst-rtsp-server.files:
9820 * bindings/vala/packages/gst-rtsp-server.gi:
9821 * bindings/vala/packages/gst-rtsp-server.metadata:
9822 * bindings/vala/packages/gst-rtsp-server.namespace:
9825 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
9827 * gst/rtsp-server/rtsp-session.c:
9828 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
9830 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
9832 * examples/Makefile.am:
9833 * gst/rtsp-server/Makefile.am:
9834 Put GStreamer version in library name
9836 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9838 * examples/Makefile.am:
9839 * gst/rtsp-server/Makefile.am:
9840 Fix some issues to pass distcheck
9842 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9844 * gst/rtsp-server/rtsp-server.c:
9845 Added port property to GstRTSPServer class.
9847 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9852 * examples/Makefile.am:
9855 * gst/rtsp-server/Makefile.am:
9856 * gst/rtsp-server/rtsp-client.c:
9857 * gst/rtsp-server/rtsp-client.h:
9858 * gst/rtsp-server/rtsp-media.c:
9859 * gst/rtsp-server/rtsp-media.h:
9860 * gst/rtsp-server/rtsp-server.c:
9861 * gst/rtsp-server/rtsp-server.h:
9862 * gst/rtsp-server/rtsp-session-pool.c:
9863 * gst/rtsp-server/rtsp-session-pool.h:
9864 * gst/rtsp-server/rtsp-session.c:
9865 * gst/rtsp-server/rtsp-session.h:
9867 Split in library and example program
9869 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
9871 * src/rtsp-client.h:
9872 Removed obsolete variable
9874 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
9876 * src/rtsp-client.c:
9877 * src/rtsp-client.h:
9878 Removed pipeline variable GstRTSPClient, because it's only used in one function
9880 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9883 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
9885 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
9887 * src/rtsp-session.c:
9888 Initialize some more vars.
9890 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
9892 * src/rtsp-session.c:
9893 Initialize variable to avoid compiler warning.
9895 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
9898 Add a reasonable generic .gitignore
projects / platform / upstream / gstreamer.git / blob