1 === release 1.13.90 ===
3 2018-03-03 22:49:34 +0000 Tim-Philipp Müller <tim@centricular.com>
8 * gst-rtsp-server.doap:
12 2018-03-02 16:24:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
14 * gst/rtsp-server/rtsp-media-factory.c:
15 * gst/rtsp-server/rtsp-permissions.c:
16 permissions: add Since tags and example for new API
18 2018-03-02 01:36:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
20 * docs/libs/gst-rtsp-server-sections.txt:
21 * gst/rtsp-server/rtsp-media-factory.c:
22 * gst/rtsp-server/rtsp-media-factory.h:
23 * gst/rtsp-server/rtsp-permissions.c:
24 * gst/rtsp-server/rtsp-permissions.h:
25 * tests/check/gst/permissions.c:
26 permissions: more bindings-friendly API
27 https://bugzilla.gnome.org/show_bug.cgi?id=793975
29 2018-03-01 19:28:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
32 meson: enable more warnings
34 2018-02-28 21:12:43 +0200 Sebastian Dröge <sebastian@centricular.com>
36 * gst/rtsp-server/rtsp-client.c:
37 rtsp-client: Place netaddress meta on packets received via TCP
38 This allows us to later map signals from rtpbin/rtpsource back to the
39 corresponding stream transport, and allows to do keep-alive based on
40 RTCP packets in case of TCP media transport.
41 https://bugzilla.gnome.org/show_bug.cgi?id=789646
43 2018-02-27 20:34:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
45 * gst/rtsp-sink/gstrtspclientsink.c:
46 rtspclientsink: if OPEN failed, unqueue next command
47 As READY_TO_PAUSED can no longer return async, the RECORD
48 command will be queued before the OPEN command fails
49 (for example in case the server could not be connected),
50 and record then waits for ever.
51 https://bugzilla.gnome.org/show_bug.cgi?id=793896
53 2018-02-26 22:59:17 +0100 Mathieu Duponchelle <mathieu@centricular.com>
55 * gst/rtsp-sink/gstrtspclientsink.c:
56 rtspclientsink: fix retrieval of custom payloader caps
57 If a bin is passed as the custom payloader, the caps of
58 its factory will be empty, the correct way to obtain the caps
59 is to query its sinkpad.
61 2018-02-26 22:59:00 +0100 Mathieu Duponchelle <mathieu@centricular.com>
63 * gst/rtsp-sink/gstrtspclientsink.c:
64 rtspclientsink: fix extra unref of custom payloader
66 2018-02-26 22:57:39 +0100 Mathieu Duponchelle <mathieu@centricular.com>
68 * gst/rtsp-sink/gstrtspclientsink.c:
69 rspclientsink: fix recent code indentation
71 2018-02-26 20:27:57 +0100 Mathieu Duponchelle <mathieu@centricular.com>
73 * gst/rtsp-sink/gstrtspclientsink.c:
74 rtspclientsink: add missing get_type prototype
76 2018-02-24 03:52:15 +0100 Mathieu Duponchelle <mathieu@centricular.com>
78 * gst/rtsp-sink/gstrtspclientsink.c:
79 rtspclientsink: allow setting payloader as pad property
80 This was a FIXME item, and can be quite useful, also
81 allowing to specify payloader properties from the command
82 line, which is always nice.
83 https://bugzilla.gnome.org/show_bug.cgi?id=793776
85 2018-02-26 14:16:54 +0100 Carlos Rafael Giani <dv@pseudoterminal.org>
87 * gst/rtsp-server/rtsp-media.c:
88 rtsp-media: Replace g_print() log line
89 https://bugzilla.gnome.org/show_bug.cgi?id=793838
91 2018-02-22 20:17:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
93 * gst/rtsp-server/rtsp-media.c:
94 * tests/check/gst/rtspclientsink.c:
95 rtsp-media: fix RECORD getting stuck
96 The test_record case was working because async=false had
97 been added in https://bugzilla.gnome.org/show_bug.cgi?id=757488
98 but that was incorrect, as it should not be needed.
99 Removing async=false made the test fail as expected, this is
100 fixed by not trying to preroll when preparing the media for
101 RECORD, as start_prepare is called upon receiving ANNOUNCE,
102 and our peer will not start sending media until it has received
103 a response to that request, and sent and received a response
104 to RECORD as well, thus obviously preventing preroll.
105 https://bugzilla.gnome.org/show_bug.cgi?id=793738
107 2018-02-23 03:26:21 +0100 Mathieu Duponchelle <mathieu@centricular.com>
109 * gst/rtsp-server/rtsp-auth.c:
110 rtsp-auth: fix set_tls_authentication_mode annotation
112 2018-02-19 11:57:29 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
114 * gst/rtsp-server/rtsp-onvif-media.c:
115 rtp-server: remove redefined variable
116 res is a boolean variable which is defined in the function scope and
117 redefined, with no reason, in the loop scope. This patch removes the
119 https://bugzilla.gnome.org/show_bug.cgi?id=793592
121 2018-02-05 11:49:07 +0100 Ognyan Tonchev <ognyan@axis.com>
123 * gst/rtsp-server/rtsp-media.c:
124 * gst/rtsp-server/rtsp-stream.c:
125 * gst/rtsp-server/rtsp-stream.h:
126 stream: Add functions for checking if stream is receiver or sender
127 ...and replace all checks for RECORD in GstRTSPMedia which are really
128 for "sender-only". This way the code becomes more generic and introducing
129 support for onvif-backchannel later on will require no changes in
132 2017-10-21 14:06:30 +0200 Ognyan Tonchev <ognyan@axis.com>
134 * gst/rtsp-server/rtsp-onvif-media-factory.c:
135 * gst/rtsp-server/rtsp-onvif-media-factory.h:
136 onvif: Make requires_backchannel() public
137 ...in order to let subclasses building the onvif part of the pipeline
138 check whether backchannel shall be included or not.
140 2018-01-22 12:46:34 +0200 Sebastian Dröge <sebastian@centricular.com>
142 * gst/rtsp-server/rtsp-onvif-media.c:
143 rtsp-server: Switch around sendonly/recvonly attributes
144 They are wrong in the ONVIF streaming spec. The backchannel should be
145 recvonly and the normal media should be sendonly: direction is always
146 from the point of view of the SDP offerer (the server) according to
149 2017-09-25 19:41:05 +0300 Sebastian Dröge <sebastian@centricular.com>
151 * docs/libs/gst-rtsp-server-docs.sgml:
152 * docs/libs/gst-rtsp-server-sections.txt:
153 * examples/.gitignore:
154 * examples/Makefile.am:
155 * examples/test-onvif-backchannel.c:
156 * gst/rtsp-server/Makefile.am:
157 * gst/rtsp-server/rtsp-media.h:
158 * gst/rtsp-server/rtsp-onvif-client.c:
159 * gst/rtsp-server/rtsp-onvif-client.h:
160 * gst/rtsp-server/rtsp-onvif-media-factory.c:
161 * gst/rtsp-server/rtsp-onvif-media-factory.h:
162 * gst/rtsp-server/rtsp-onvif-media.c:
163 * gst/rtsp-server/rtsp-onvif-media.h:
164 * gst/rtsp-server/rtsp-onvif-server.c:
165 * gst/rtsp-server/rtsp-onvif-server.h:
166 * gst/rtsp-server/rtsp-sdp.c:
167 * gst/rtsp-server/rtsp-sdp.h:
168 rtsp: Add support for ONVIF backchannel
169 This adds a new RTSP server, client, media-factory and media subclass
170 for handling the specifics of the backchannel. Ideally this later can be
171 extended with other ONVIF specific features.
173 2017-10-12 21:00:16 +0300 Sebastian Dröge <sebastian@centricular.com>
175 * gst/rtsp-server/rtsp-media.c:
176 rtsp-media: Add support for sending+receiving medias
177 We need to add an appsrc/appsink in that case because otherwise the
178 media bin will be a sink and a source for rtpbin, causing a pipeline
180 https://bugzilla.gnome.org/show_bug.cgi?id=788950
182 2018-02-15 19:44:28 +0000 Tim-Philipp Müller <tim@centricular.com>
188 === release 1.13.1 ===
190 2018-02-15 17:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
194 * gst-rtsp-server.doap:
198 2018-02-14 17:11:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
200 * gst/rtsp-server/rtsp-session-pool.c:
201 session-pool: remove nullable return annotation
202 create_watch can only return NULL from the API guards, no
205 2018-02-13 18:59:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
207 * gst/rtsp-server/rtsp-media-factory.c:
208 * gst/rtsp-server/rtsp-media.c:
209 set_clock functions: Add nullable annotations
211 2018-02-10 00:07:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
213 * gst/rtsp-server/rtsp-auth.c:
214 * gst/rtsp-server/rtsp-client.c:
215 * gst/rtsp-server/rtsp-media-factory.c:
216 * gst/rtsp-server/rtsp-media.c:
217 * gst/rtsp-server/rtsp-mount-points.c:
218 * gst/rtsp-server/rtsp-server.c:
219 * gst/rtsp-server/rtsp-session-media.c:
220 * gst/rtsp-server/rtsp-session-pool.c:
221 * gst/rtsp-server/rtsp-session.c:
222 * gst/rtsp-server/rtsp-stream-transport.c:
223 * gst/rtsp-server/rtsp-stream.c:
224 * gst/rtsp-server/rtsp-thread-pool.c:
225 All around: add annotations and API guards
227 2018-02-12 19:12:35 +0100 Mathieu Duponchelle <mathieu@centricular.com>
229 * tests/test-cleanup.c:
230 test-cleanup: bind any port
231 The meson test suite runs tests in parallel, trying to bind
232 a single port made the test fail.
234 2018-02-08 19:15:10 +0000 Tim-Philipp Müller <tim@centricular.com>
237 meson: make version numbers ints and fix int/string comparison
238 WARNING: Trying to compare values of different types (str, int).
239 The result of this is undefined and will become a hard error
240 in a future Meson release.
242 2018-02-06 18:00:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
244 * gst/rtsp-server/rtsp-context.c:
245 gst_rtsp_context_get_current: add (skip) annotation
246 The return value type is defined with G_DEFINE_POINTER_TYPE,
247 and gi emits the following warning:
248 Invalid non-constant return of bare structure or union; register as
251 2018-02-06 17:58:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
253 * gst/rtsp-server/rtsp-client.c:
254 rtsp-client: add type annotations
255 gi doesn't seem to be able to figure out the type of the
256 signal parameters when defined with G_DEFINE_POINTER_TYPE
258 2018-02-04 12:24:09 +0100 Tim-Philipp Müller <tim@centricular.com>
261 autotools: use -fno-strict-aliasing where supported
262 https://bugzilla.gnome.org/show_bug.cgi?id=769183
264 2018-01-30 20:35:21 +0000 Tim-Philipp Müller <tim@centricular.com>
267 meson: use -fno-strict-aliasing where supported
268 https://bugzilla.gnome.org/show_bug.cgi?id=769183
270 2018-01-25 12:09:03 +0000 Tim-Philipp Müller <tim@centricular.com>
272 * gst/rtsp-server/rtsp-mount-points.c:
273 mount-points: bail out of loop again when matching mount points
274 Previous patch led to us iterating the entire sequence. Bail out
275 of the loop again if we have a match but are moving away from it.
276 https://bugzilla.gnome.org/show_bug.cgi?id=771555
278 2018-01-25 12:06:57 +0000 Tim-Philipp Müller <tim@centricular.com>
280 * tests/check/gst/mountpoints.c:
281 tests: mountpoints: add more checks for mount point path matching
282 https://bugzilla.gnome.org/show_bug.cgi?id=771555
284 2016-09-16 20:41:19 +0000 Andrew Bott <andrew.bott@blackmoth.com>
286 * gst/rtsp-server/rtsp-mount-points.c:
287 mount-points: fix matching of paths where there's also an entry with a common prefix
288 e.g. with the following mount points
292 _match() would not match /raw/video and /raw/snapshot correctly.
293 https://bugzilla.gnome.org/show_bug.cgi?id=771555
295 2018-01-18 23:53:20 +0000 Tim-Philipp Müller <tim@centricular.com>
297 * docs/libs/gst-rtsp-server-sections.txt:
298 * gst/rtsp-server/rtsp-permissions.c:
299 * gst/rtsp-server/rtsp-permissions.h:
300 * tests/check/gst/permissions.c:
301 permissions: add some new API to make this usable from bindings
302 https://bugzilla.gnome.org/show_bug.cgi?id=787073
304 2018-01-18 11:32:32 +0000 Tim-Philipp Müller <tim@centricular.com>
306 * gst/rtsp-server/rtsp-token.c:
307 rtsp-token: annotate constructors for bindings
308 This maps _new_empty() to _new(), which also makes RTSPToken()
309 work properly now. Since this API wasn't usable from bindings
310 before, this should hopefully be fine.
311 https://bugzilla.gnome.org/show_bug.cgi?id=787073
313 2018-01-18 11:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
315 * docs/libs/gst-rtsp-server-sections.txt:
316 * gst/rtsp-server/rtsp-token.c:
317 * gst/rtsp-server/rtsp-token.h:
318 * tests/check/gst/token.c:
319 rtsp-token: add some API to set fields from bindings
320 The existing functions are all vararg-based and as such
321 not usable from bindings.
322 https://bugzilla.gnome.org/show_bug.cgi?id=787073
324 2018-01-13 15:02:28 +0000 Tim-Philipp Müller <tim@centricular.com>
326 * tests/check/gst/rtspclientsink.c:
327 * tests/check/gst/rtspserver.c:
328 * tests/check/gst/sessionpool.c:
329 * tests/check/gst/stream.c:
330 tests: fix indentation
333 2018-01-13 14:58:55 +0000 Tim-Philipp Müller <tim@centricular.com>
335 * tests/check/gst/rtspserver.c:
336 tests: rtspserver: fix another ref leak
337 Even if this didn't show up in valgrind.
339 2018-01-13 14:58:00 +0000 Tim-Philipp Müller <tim@centricular.com>
341 * tests/check/gst/rtspclientsink.c:
342 tests: rtspclientsink: fix leak
344 2018-01-02 14:19:31 +0100 Branko Subasic <branko@axis.com>
346 * tests/check/gst/rtspserver.c:
347 test: rtspserver: plug memory leak in test_no_session_timeout
348 In test_no_session_timeout, unref the rtsp session object when the
350 https://bugzilla.gnome.org/show_bug.cgi?id=792127
352 2017-12-20 14:17:02 +0100 Edward Hervey <edward@centricular.com>
354 * gst/rtsp-sink/gstrtspclientsink.c:
355 rtpsclientsink: Initialize and clear newly added mutex and cond
356 While it *did* work, glib would automatically create new mutex and cond
357 ... which never got freed
359 2017-12-19 11:34:37 +0200 Sebastian Dröge <sebastian@centricular.com>
361 * gst/rtsp-server/rtsp-stream.c:
362 rtsp-stream: Set multicast TTL on the multicast sockets
363 And not if we do unicast UDP.
364 https://bugzilla.gnome.org/show_bug.cgi?id=791743
366 2017-12-19 11:14:48 +0200 Sebastian Dröge <sebastian@centricular.com>
368 * gst/rtsp-server/rtsp-stream.c:
369 rtsp-stream: Decide based on the sockets, not the addresses if we already allocated a socket
370 In the multicast case (as in test-multicast, not test-multicast2), the
371 address could be allocated/reserved (and thus set) already without
372 allocating the actual socket. We need to allocate the socket here still
373 instead of just claiming that it was already allocated.
374 See https://bugzilla.gnome.org/show_bug.cgi?id=791743#c2
376 2017-12-16 21:46:53 +0100 Patricia Muscalu <patricia@dovakhiin.com>
378 * gst/rtsp-sink/gstrtspclientsink.c:
379 * gst/rtsp-sink/gstrtspclientsink.h:
380 rtspclientsink: Use the new rtsp-stream API
381 https://bugzilla.gnome.org/show_bug.cgi?id=790412
383 2017-12-16 21:01:43 +0100 Patricia Muscalu <patricia@dovakhiin.com>
385 * gst/rtsp-sink/gstrtspclientsink.c:
386 * gst/rtsp-sink/gstrtspclientsink.h:
387 rtspclientsink: Wait until OPEN has been scheduled
388 Make sure that the sink thread has started opening connection
389 to the server before continuing.
390 https://bugzilla.gnome.org/show_bug.cgi?id=790412
392 2017-12-14 14:53:35 +1100 Matthew Waters <matthew@centricular.com>
395 Automatic update of common submodule
396 From e8c7a71 to 3fa2c9e
398 2017-12-07 16:08:29 +0100 Edward Hervey <edward@centricular.com>
400 * gst/rtsp-server/rtsp-media.c:
401 * gst/rtsp-server/rtsp-session-media.c:
402 * gst/rtsp-server/rtsp-stream.c:
403 rtsp-server: Minor doc fixes
406 2017-12-06 20:47:22 +0000 Tim-Philipp Müller <tim@centricular.com>
410 tests: disable all tests when --disable-tests is used
411 Move conditional subdir include into top level.
412 Based on patch by: Joel Holdsworth
413 https://bugzilla.gnome.org/show_bug.cgi?id=757703
415 2017-12-06 20:42:39 +0000 Tim-Philipp Müller <tim@centricular.com>
420 meson: build more tests and add options to disable tests and examples
422 2017-11-26 13:26:39 -0300 Thibault Saunier <tsaunier@gnome.org>
424 * gst/rtsp-server/rtsp-session.c:
425 Fix build when -Werror=deprecated-declarations is on
426 As gst_rtsp_session_next_timeout is deprecated.
428 ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-session.c:760:3: error: ‘gst_rtsp_session_next_timeout’ is deprecated: Use 'gst_rtsp_session_next_timeout_usec' instead [-Werror=deprecated-declarations]
429 res = (gst_rtsp_session_next_timeout (session, now) == 0);
431 ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-session.c:685:1: note: declared here
432 gst_rtsp_session_next_timeout (GstRTSPSession * session, GTimeVal * now)
433 ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~
436 2017-11-27 20:18:24 +1100 Matthew Waters <matthew@centricular.com>
439 Automatic update of common submodule
440 From 3f4aa96 to e8c7a71
442 2017-11-25 20:34:16 +0100 Patricia Muscalu <patricia@dovakhiin.com>
444 * tests/check/gst/media.c:
445 check/media: Add seekability test case: not all streams are active
446 Media contains two streams but only one is complete and prepared
448 https://bugzilla.gnome.org/show_bug.cgi?id=790674
450 2017-11-25 20:32:02 +0100 Patricia Muscalu <patricia@dovakhiin.com>
452 * gst/rtsp-server/rtsp-stream.c:
453 rtsp-stream: Do not reset 'blocking' if stream is already blocked
454 https://bugzilla.gnome.org/show_bug.cgi?id=790674
456 2017-11-25 20:45:44 +0100 Patricia Muscalu <patricia@dovakhiin.com>
458 * gst/rtsp-server/rtsp-media.c:
459 rtsp-media: Fix missing lock in gst_rtsp_media_seekable()
460 https://bugzilla.gnome.org/show_bug.cgi?id=790674
462 2017-11-26 16:29:49 +0000 Tim-Philipp Müller <tim@centricular.com>
465 meson: remove vs_module_defs_dir variable which is no longer needed
467 2017-11-26 14:46:05 +0000 Tim-Philipp Müller <tim@centricular.com>
469 * gst/rtsp-server/rtsp-session.h:
472 2017-11-26 12:53:42 +0000 Tim-Philipp Müller <tim@centricular.com>
475 * gst/rtsp-server/meson.build:
477 * win32/common/libgstrtspserver.def:
478 win32: remove .def file with exports
479 They're no longer needed, symbol exporting is now explicit
480 via GST_EXPORT in all cases (autotools, meson, incl. MSVC).
482 2017-11-26 12:28:40 +0000 Tim-Philipp Müller <tim@centricular.com>
485 autotools: stop controlling symbol visibility with -export-symbols-regex
486 Instead, use -fvisibility=hidden and explicit exports via GST_EXPORT.
487 This should result in consistent behaviour for the autotools and
490 2017-11-26 12:47:08 +0000 Tim-Philipp Müller <tim@centricular.com>
492 * gst/rtsp-server/rtsp-media.h:
493 * gst/rtsp-server/rtsp-server.h:
494 * gst/rtsp-server/rtsp-session.c:
495 * gst/rtsp-server/rtsp-session.h:
496 rtsp-server: add missing GST_EXPORT and export deprecated funcs
498 2017-11-25 07:53:30 +0100 Edward Hervey <edward@centricular.com>
500 * tests/check/gst/media.c:
501 check: Add seekability testing on medias
502 Make sure that once GstRTSPMedia are prepared they returned
503 the expected seekability results
504 https://bugzilla.gnome.org/show_bug.cgi?id=790674
506 2017-11-24 17:34:31 +0100 Edward Hervey <edward@centricular.com>
508 * docs/libs/gst-rtsp-server-sections.txt:
509 * gst/rtsp-server/rtsp-media.c:
510 * gst/rtsp-server/rtsp-stream.c:
511 * gst/rtsp-server/rtsp-stream.h:
512 * win32/common/libgstrtspserver.def:
513 rtsp-media: Enable seeking query before pipeline is complete
514 SDP are now provided *before* the pipeline is fully complete. In order
515 to know whether a media is seekable or not therefore requires asking
516 the invididual streams.
517 API: gst_rtsp_stream_seekable
518 https://bugzilla.gnome.org/show_bug.cgi?id=790674
520 2017-11-23 20:34:03 +0100 Patricia Muscalu <patricia@axis.com>
522 * gst/rtsp-server/rtsp-media.c:
523 rtsp-media: Fix handling in default_unsuspend()
524 Handle the case when streams are not blocked and media
525 is suspended from PAUSED.
526 Change-Id: I2f3d222ea7b9b20a0732ea5dc81a32d17ab75040
527 https://bugzilla.gnome.org/show_bug.cgi?id=790674
529 2017-11-23 18:51:21 +0100 Patricia Muscalu <patricia@axis.com>
531 * tests/check/gst/media.c:
532 check/media: Fix thread pool leak.
533 Change-Id: I0f92b1caca0ee518ae64a7dacfbd28a214c3eea1
534 https://bugzilla.gnome.org/show_bug.cgi?id=790674
536 2017-11-23 18:39:44 +0100 Patricia Muscalu <patricia@axis.com>
538 * gst/rtsp-server/rtsp-media.c:
539 rtsp-media: Removed fakesink elements
540 There is not need of adding fakesink elements to the media
541 pipeline in the dynamic-payloader case.
542 The media pipeline itself is dynamically updated with
543 the receiver and sender parts that are based on the client
544 transport information known after SETUP has been received.
545 Change-Id: I4e88c9b500c04030669822f0d03b1842913f6cb9
546 https://bugzilla.gnome.org/show_bug.cgi?id=790674
548 2017-11-23 09:10:54 +0100 Patricia Muscalu <patricia@axis.com>
550 * gst/rtsp-server/rtsp-media.c:
551 rtsp-media: Corrected ASYNC_DONE handling
552 Media is complete when all the transport based parts are
553 added to the media pipeline. At this point ASYNC_DONE is
554 posted by the media pipeline and media is ready to enter
556 Change-Id: I50fb8dfed88ebaf057d9a35fca2d7f0a70e9d1fa
557 https://bugzilla.gnome.org/show_bug.cgi?id=790674
559 2017-11-22 12:24:38 +0100 Edward Hervey <bilboed@bilboed.com>
561 * tests/check/gst/media.c:
562 check/media: Check that prepared media can provide a SDP
563 Whenever a RTSPMedia is prepared, it should be able to provide a SDP
565 2017-11-21 09:53:19 +0100 Edward Hervey <edward@centricular.com>
567 * gst/rtsp-server/rtsp-client.c:
568 rtsp-client: Don't leak addr
571 2017-11-21 09:53:08 +0100 Edward Hervey <bilboed@bilboed.com>
573 * gst/rtsp-server/rtsp-client.c:
574 * gst/rtsp-server/rtsp-session-media.c:
575 * gst/rtsp-server/rtsp-stream.c:
578 2017-11-20 18:30:19 +0100 Edward Hervey <bilboed@bilboed.com>
580 * gst/rtsp-server/rtsp-media.c:
581 rtsp-media: Don't unblock with remaining dynamic payloaders
582 If we still have some dynamic paylaoders which haven't posted
583 no-more-pads yet, don't go to PREPARED if one of the streams
585 The risk was that we would end up not exposing/using all specified
587 The downside is that if you have _multiple_ _live_ _dynamic_ payloaders
588 then it will take a bit more time to start. But only if those 3
589 conditions are present.
590 https://bugzilla.gnome.org/show_bug.cgi?id=769521
592 2017-11-20 16:49:29 +0100 Edward Hervey <edward@centricular.com>
594 * gst/rtsp-server/rtsp-media.c:
597 2017-11-20 16:48:55 +0100 Edward Hervey <edward@centricular.com>
599 * gst/rtsp-server/rtsp-media.c:
600 rtsp-media: Don't set float on a gint64 variable
601 Just use 0. Fixes 'undefined' behaviour from clang
603 2017-11-20 18:29:02 +0100 Edward Hervey <edward@centricular.com>
605 * gst/rtsp-server/rtsp-media.c:
606 rtsp-media: Fix previous commit
607 We only want to count dynamic payloaders
609 2017-11-20 09:32:07 +0100 Edward Hervey <edward@centricular.com>
611 * gst/rtsp-server/rtsp-media.c:
612 * tests/check/gst/media.c:
613 rtsp-media: Handle multiple dynamic elements
614 If we have more than one dynamic payloader in the pipeline, we need
615 to wait until the *last* one emits 'no-more-pads' before switching
617 Failure to do so would result in a race where some of the streams
618 wouldn't properly be prepared
619 https://bugzilla.gnome.org/show_bug.cgi?id=769521
621 2017-11-16 12:18:20 +0200 Sebastian Dröge <sebastian@centricular.com>
623 * win32/common/libgstrtspserver.def:
624 win32: Fix exported symbols list
626 2017-11-15 19:52:29 +0200 Sebastian Dröge <sebastian@centricular.com>
628 * gst/rtsp-server/rtsp-stream.c:
629 rtsp-stream: Only update the RTP udpsink if it actually exists
630 For send-only streams it does not exist, but the RTCP udpsink might.
632 2017-11-15 18:15:53 +0200 Sebastian Dröge <sebastian@centricular.com>
634 * win32/common/libgstrtspserver.def:
635 win32: Update exports
637 2017-10-23 09:49:09 +0200 Patricia Muscalu <patricia@axis.com>
639 * gst/rtsp-server/rtsp-media.c:
640 * gst/rtsp-server/rtsp-stream.c:
641 * gst/rtsp-server/rtsp-stream.h:
642 rtsp-media: seek on media pipelines that are complete
643 Make sure that a seek is performed on pipelines that
644 contain at least one sink element.
645 Change-Id: Icf398e10add3191d104b1289de612412da326819
646 https://bugzilla.gnome.org/show_bug.cgi?id=788340
648 2017-10-17 10:44:33 +0200 Patricia Muscalu <patricia@axis.com>
650 * gst/rtsp-server/rtsp-client.c:
651 * gst/rtsp-server/rtsp-media.c:
652 * gst/rtsp-server/rtsp-media.h:
653 * gst/rtsp-server/rtsp-stream.c:
654 * gst/rtsp-server/rtsp-stream.h:
655 * tests/check/gst/client.c:
656 * tests/check/gst/media.c:
657 * tests/check/gst/rtspserver.c:
658 * tests/check/gst/stream.c:
659 Dynamically reconfigure pipeline in PLAY based on transports
660 The initial pipeline does not contain specific transport
661 elements. The receiver and the sender parts are added
663 If the media is shared, the streams are dynamically
664 reconfigured after each PLAY.
665 https://bugzilla.gnome.org/show_bug.cgi?id=788340
667 2017-10-16 12:40:57 +0200 Patricia Muscalu <patricia@axis.com>
669 * gst/rtsp-server/rtsp-stream.c:
670 rtsp-stream: obtain stream position from pad
671 If no sinks have been added yet, obtain the current and
672 the stop position of the stream from the send_src pad.
673 Change-Id: Iacd4ab4bdc69f6b49370d06012880ce48a7d595a
674 https://bugzilla.gnome.org/show_bug.cgi?id=788340
676 2017-10-16 11:35:10 +0200 Patricia Muscalu <patricia@axis.com>
678 * gst/rtsp-server/rtsp-session-media.c:
679 * gst/rtsp-server/rtsp-session-media.h:
680 rtsp-session-media: add function to get a list of transports
681 Change-Id: I817e10624da0f3200f24d1b232cff481099278e3
682 https://bugzilla.gnome.org/show_bug.cgi?id=788340
684 2017-10-16 11:15:55 +0200 Patricia Muscalu <patricia@axis.com>
686 * gst/rtsp-server/rtsp-stream.c:
687 * gst/rtsp-server/rtsp-stream.h:
688 rtsp-stream: add functions to get rtp and rtcp multicast sockets
689 Change-Id: Iddfe6e0bd250cb0159096d5eba9e4202d22b56db
690 https://bugzilla.gnome.org/show_bug.cgi?id=788340
692 2017-10-20 12:21:48 +0200 Patricia Muscalu <patricia@axis.com>
694 * gst/rtsp-server/rtsp-stream.c:
695 stream: set async=sync=false only for RTCP appsink
696 Change-Id: I929a218a9adf4759f61322b6f2063aacc5595f90
697 https://bugzilla.gnome.org/show_bug.cgi?id=788340
699 2017-10-16 10:10:17 +0200 Patricia Muscalu <patricia@axis.com>
701 * gst/rtsp-server/rtsp-media.c:
702 rtsp-media: return minimum value in query position case
703 The minimum position should be returned as we are interested
704 in the whole interval.
705 Change-Id: I30e297fc040c995ae40c25dee8ff56321612fe2b
706 https://bugzilla.gnome.org/show_bug.cgi?id=788340
708 2017-08-09 11:52:38 +0200 Jonathan Karlsson <jonakn@axis.com>
710 * gst/rtsp-server/rtsp-session.c:
711 * tests/check/gst/rtspserver.c:
712 rtsp-session: Handle the case when timeout=0
713 According to the documentation, a timeout of value 0 means
714 that the session never timeouts. This adds handling of that.
715 If timeout=0 we just return with a -1 from
716 gst_rtsp_session_next_timeout_usec ().
717 https://bugzilla.gnome.org/show_bug.cgi?id=785058
719 2017-07-17 17:15:22 +0300 Sebastian Dröge <sebastian@centricular.com>
721 * gst/rtsp-sink/gstrtspclientsink.c:
722 rtspclientsink: Add "accept-certificate" signal for manually checking a TLS certificate for validity
723 https://bugzilla.gnome.org/show_bug.cgi?id=785024
725 2017-10-26 14:43:19 +0200 Mathieu Duponchelle <mathieu@centricular.com>
727 * docs/libs/gst-rtsp-server-sections.txt:
728 * gst/rtsp-server/rtsp-media-factory.c:
729 docs: add media factory transport mode accessors
730 and fix the documentation for the return value of the getter
732 2017-10-09 12:43:01 +0200 Branko Subasic <branko@axis.com>
734 * gst/rtsp-server/rtsp-client.c:
735 rtsp-client: unref 'pipelined_requests' in finalize
736 The hash table priv->pipelined_requests is not unref:ed in the
737 finalize funktion. Make sure it is.
738 https://bugzilla.gnome.org/show_bug.cgi?id=788704
740 2017-10-09 14:44:40 +0200 Thibault Saunier <thibault.saunier@osg.samsung.com>
742 * gst/rtsp-server/rtsp-media.c:
743 rtsp-media: Initialize scalar variable
746 2017-10-06 10:27:34 +0200 Edward Hervey <edward@centricular.com>
748 * win32/common/libgstrtspserver.def:
749 win32: Update export file
751 2017-04-22 09:26:07 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
753 * gst/rtsp-server/rtsp-client.c:
754 * gst/rtsp-server/rtsp-media.c:
755 * gst/rtsp-server/rtsp-media.h:
756 Start support for RTSP 2.0
757 This adds basic support for new 2.0 features, though the protocol is
758 subposdely backward incompatible, most semantics are the sames.
761 * version negotiation
762 * pipelined requests support
763 * Media-Properties support
764 * Accept-Ranges support
766 * gst_rtsp_media_seekable
767 The RTSP methods that have been removed when using 2.0 now return
769 https://bugzilla.gnome.org/show_bug.cgi?id=781446
771 2017-06-02 15:37:54 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
773 * gst/rtsp-server/rtsp-stream.c:
774 stream: Use stream duration as stream-stop if segment was not configured with a stop
775 Allowing client to know stream duration when no seeking happened.
776 https://bugzilla.gnome.org/show_bug.cgi?id=783435
778 2017-09-25 19:40:17 +0300 Sebastian Dröge <sebastian@centricular.com>
780 * gst/rtsp-server/rtsp-media-factory.c:
781 rtsp-media-factory: Don't cache any media if NULL was returned as key
782 The docs already mentioned this, but we actually stored it in the hash
783 table with key==NULL and leaked its reference forever.
785 2017-09-18 19:31:31 +0200 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
787 * gst/rtsp-sink/gstrtspclientsink.c:
788 * gst/rtsp-sink/gstrtspclientsink.h:
789 rtspclientsink: Use a mutex for protecting against concurrent send/receives
790 This is a simple port of:
791 * a722f6e8329032c6eda4865d6a07f4ba5981d7ea
792 * c438545dc9e2f14f657bc0ef261fff726449867b
793 * cd17c71dcea5c9310d21f1347c7520983e5869ac
796 2017-08-31 13:24:15 +0530 Satya Prakash Gupta <sp.gupta@samsung.com>
798 * gst/rtsp-server/rtsp-sdp.c:
799 sdp: fix Memory leak in error case
800 https://bugzilla.gnome.org/show_bug.cgi?id=787059
802 2017-08-18 17:37:01 +0100 Tim-Philipp Müller <tim@centricular.com>
804 * pkgconfig/meson.build:
805 meson: don't install -uninstalled.pc file
806 https://bugzilla.gnome.org/show_bug.cgi?id=786457
808 2017-08-17 12:26:17 +0100 Tim-Philipp Müller <tim@centricular.com>
811 Automatic update of common submodule
812 From 48a5d85 to 3f4aa96
814 2017-08-14 21:04:23 +0300 Sebastian Dröge <sebastian@centricular.com>
816 * gst/rtsp-server/rtsp-client.c:
817 rtsp-client: Fix typo in debug message
819 2017-08-11 14:14:32 +0100 Tim-Philipp Müller <tim@centricular.com>
822 meson: hide symbols by default unless explicitly exported
824 2017-08-10 14:20:12 +0100 Tim-Philipp Müller <tim@centricular.com>
826 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
827 pkgconfig: remove -I@srcdir@/.. which duplicates abs_top_srcdir
828 Fixes meson warning about undefined @srcdir@.
830 2017-07-21 13:36:00 +0100 Tim-Philipp Müller <tim@centricular.com>
833 meson: skip tests on windows for now
834 As we do in the other modules. As libgstcheck is currently not
835 built on windows. Fixes "Fallback variable 'gst_check_dep' in
836 the subproject 'gstreamer' does not exist"" Meson error.
838 2017-06-22 07:25:07 -0700 Julien Isorce <julien.isorce@gmail.com>
840 * gst/rtsp-server/rtsp-stream.c:
841 rtsp-stream: fix connection delay due to wrong assumption on last-sample
842 Commit 852cc09f542af5cadd79ffd7fe79d6475cf57e14 assumed that
843 multiudpsink's last-sample always comes from the payloader. Which
844 is wrong if auxiliary streams are multiplexed in the same stream.
845 So check the buffer's ssrc against the caps'ssrc before to use its
846 seqnum. If not the same ssrc just use the payloader as done prior
847 the commit above or when there is no last-sample yet.
848 https://bugzilla.gnome.org/show_bug.cgi?id=784094
850 2017-06-23 16:19:04 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
853 meson: Allow using glib as a subproject
855 2017-06-26 09:55:49 +0100 Tim-Philipp Müller <tim@centricular.com>
858 meson: fix with-package-name option
859 https://bugzilla.gnome.org/show_bug.cgi?id=784082
861 2017-06-09 20:16:28 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
864 Distribute meson_options.txt
866 2017-06-09 20:11:47 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
869 And config.h.meson is no longer dist either
871 2017-06-09 21:27:09 +0100 Tim-Philipp Müller <tim@centricular.com>
875 meson: config.h.meson is no longer needed
877 2017-06-07 13:04:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
879 * tests/check/meson.build:
881 meson: Fix building tests and activate them again
883 2017-06-07 12:55:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
885 * tests/check/meson.build:
886 meson: Do not use path separator in test names
887 Avoiding warnings like:
888 WARNING: Target "elements/audioamplify" has a path separator in its name.
890 2017-05-20 15:07:31 +0100 Tim-Philipp Müller <tim@centricular.com>
894 meson: add options to set package name and origin
895 https://bugzilla.gnome.org/show_bug.cgi?id=782172
897 2017-05-18 10:35:18 +0100 Tim-Philipp Müller <tim@centricular.com>
899 * gst/rtsp-server/rtsp-address-pool.h:
900 * gst/rtsp-server/rtsp-auth.h:
901 * gst/rtsp-server/rtsp-client.h:
902 * gst/rtsp-server/rtsp-context.h:
903 * gst/rtsp-server/rtsp-media-factory-uri.h:
904 * gst/rtsp-server/rtsp-media-factory.h:
905 * gst/rtsp-server/rtsp-media.h:
906 * gst/rtsp-server/rtsp-mount-points.h:
907 * gst/rtsp-server/rtsp-params.h:
908 * gst/rtsp-server/rtsp-permissions.h:
909 * gst/rtsp-server/rtsp-sdp.h:
910 * gst/rtsp-server/rtsp-server.h:
911 * gst/rtsp-server/rtsp-session-media.h:
912 * gst/rtsp-server/rtsp-session-pool.h:
913 * gst/rtsp-server/rtsp-session.h:
914 * gst/rtsp-server/rtsp-stream-transport.h:
915 * gst/rtsp-server/rtsp-stream.h:
916 * gst/rtsp-server/rtsp-thread-pool.h:
917 * gst/rtsp-server/rtsp-token.h:
918 Mark symbols explicitly for export with GST_EXPORT
920 2017-05-16 14:44:43 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
923 * gst/rtsp-sink/Makefile.am:
924 Remove plugin specific static build option
925 Static and dynamic plugins now have the same interface. The standard
926 --enable-static/--enable-shared toggle are sufficient.
928 2017-05-04 18:59:14 +0300 Sebastian Dröge <sebastian@centricular.com>
934 === release 1.12.0 ===
936 2017-05-04 15:40:46 +0300 Sebastian Dröge <sebastian@centricular.com>
942 * gst-rtsp-server.doap:
946 === release 1.11.91 ===
948 2017-04-27 17:42:02 +0300 Sebastian Dröge <sebastian@centricular.com>
954 * gst-rtsp-server.doap:
958 2017-04-24 20:30:37 +0100 Tim-Philipp Müller <tim@centricular.com>
961 Automatic update of common submodule
962 From 60aeef6 to 48a5d85
964 2017-04-13 14:20:10 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
966 * gst/rtsp-server/rtsp-media-factory.c:
967 * gst/rtsp-server/rtsp-media.c:
968 * gst/rtsp-server/rtsp-session.c:
969 * gst/rtsp-server/rtsp-stream.c:
970 gi: Fix some annotations and docstrings
972 2017-04-13 13:52:26 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
974 * gst/rtsp-server/meson.build:
979 2017-04-10 23:51:12 +0100 Tim-Philipp Müller <tim@centricular.com>
983 Automatic update of common submodule
984 From 39ac2f5 to 60aeef6
986 === release 1.11.90 ===
988 2017-04-07 16:35:03 +0300 Sebastian Dröge <sebastian@centricular.com>
994 * gst-rtsp-server.doap:
998 2017-03-27 18:19:33 +0100 Tim-Philipp Müller <tim@centricular.com>
1000 * examples/test-launch.c:
1001 examples: make test-launch pipeline shared by default as well
1003 2017-02-27 19:10:44 +0200 Sebastian Dröge <sebastian@centricular.com>
1005 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
1006 gstreamer-rtsp-server: Add both srcdir and builddir to the include path
1007 Just the build dir is not going to work for srcdir!=builddir.
1009 2017-02-24 15:59:54 +0200 Sebastian Dröge <sebastian@centricular.com>
1012 meson: Update version
1014 2017-02-24 15:37:49 +0200 Sebastian Dröge <sebastian@centricular.com>
1019 === release 1.11.2 ===
1021 2017-02-24 15:10:07 +0200 Sebastian Dröge <sebastian@centricular.com>
1027 * gst-rtsp-server.doap:
1030 2017-02-14 20:40:26 +0000 Tim-Philipp Müller <tim@centricular.com>
1033 meson: dist meson build files
1034 Ship meson build files in tarballs, so people who use tarballs
1035 in their builds can start playing with meson already.
1037 2017-02-07 23:39:37 +1100 Jan Schmidt <jan@centricular.com>
1039 * examples/test-record.c:
1040 examples/test-record: Add extra line to initial printout
1041 Add an example line of how to deliver a stream to the
1044 2017-01-19 14:57:19 +0200 Sebastian Dröge <sebastian@centricular.com>
1046 * gst/rtsp-server/rtsp-client.c:
1047 rtsp-client: Also handle the (S|G)ET_PARAMETER case of size==0 || !data as keep-alive
1048 If there is no Content-Length header, no body would be allocated and the
1049 '\0' would also not be appended to the body.
1051 2017-01-19 14:24:07 +0200 Sebastian Dröge <sebastian@centricular.com>
1053 * gst/rtsp-server/rtsp-client.c:
1054 rtsp-client: Fix handling of keep-alive GET_PARAMETER/SET_PARAMETER
1055 While they logically have 0 bytes length, GstRTSPConnection is appending
1056 a '\0' to everything making the size be 1 instead.
1058 2017-01-13 12:39:36 +0000 Tim-Philipp Müller <tim@centricular.com>
1063 2017-01-12 19:04:23 +0200 Sebastian Dröge <sebastian@centricular.com>
1065 * gst/rtsp-server/rtsp-session.c:
1066 rtsp-session: Only remove deprecated API if requested to do so, not just when disabling
1067 gst_rtsp_session_is_expired() and gst_rtsp_session_next_timeout() were
1070 2017-01-12 16:32:59 +0200 Sebastian Dröge <sebastian@centricular.com>
1075 === release 1.11.1 ===
1077 2017-01-12 16:14:46 +0200 Sebastian Dröge <sebastian@centricular.com>
1083 * gst-rtsp-server.doap:
1084 * win32/common/libgstrtspserver.def:
1087 2017-01-10 08:34:50 +0100 Patricia Muscalu <patricia@axis.com>
1089 * gst/rtsp-server/rtsp-stream.c:
1090 rtsp-stream: corrected if-statement in _get_server_port()
1091 This bug was accidentally introduced while fixing a segfault
1092 in _get_server_port() function.
1093 https://bugzilla.gnome.org/show_bug.cgi?id=776345
1095 2017-01-09 14:12:05 +0100 Patricia Muscalu <patricia@axis.com>
1097 * gst/rtsp-server/rtsp-stream.c:
1098 * tests/check/gst/stream.c:
1099 rtsp-stream: fixed segmenation fault in _get_server_port()
1100 Calling function gst_rtsp_stream_get_server_port() results in
1101 segmenation fault in the RTP/RTSP/TCP case.
1102 Port that the server will use to receive RTCP makes only
1103 sense in the UDP case, however the function should handle
1104 the TCP case in a nicer way.
1105 https://bugzilla.gnome.org/show_bug.cgi?id=776345
1107 2017-01-09 12:22:40 +0300 Aleksandr Slobodeniuk <alenuke@yandex.ru>
1109 * gst/rtsp-server/rtsp-media-factory.c:
1110 dosc: Fix a little typo
1111 https://bugzilla.gnome.org/show_bug.cgi?id=777037
1113 2017-01-04 16:20:54 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
1115 * pkgconfig/Makefile.am:
1116 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
1117 * pkgconfig/meson.build:
1118 meson: generate pkg-config -uninstalled pc files
1119 Generating those files is useful for users building the GStreamer stack
1120 using meson and having to link it to another project which is still
1121 using the autotools.
1122 https://bugzilla.gnome.org/show_bug.cgi?id=776810
1124 2017-01-04 16:11:08 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
1126 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
1127 pkgconfig: fix -uninstalled pc file
1128 pcfiledir was never defined so the paths were wrong.
1129 https://bugzilla.gnome.org/show_bug.cgi?id=776867
1131 2016-12-21 13:41:50 +0100 Patricia Muscalu <patricia@axis.com>
1133 * gst/rtsp-server/rtsp-stream.c:
1134 * tests/check/gst/rtspserver.c:
1135 rtsp-stream: Fixed TCP transport case
1136 Make sure that the appsink element is actually added to
1137 the bin before trying to link it with the elements in it.
1138 https://bugzilla.gnome.org/show_bug.cgi?id=776343
1140 2016-12-16 17:26:04 +0000 Tim-Philipp Müller <tim@centricular.com>
1146 Remove generated .spec file
1147 Likely extremely bitrotten, and we should not ship this anyway.
1149 2016-12-03 08:21:02 +0100 Edward Hervey <bilboed@bilboed.com>
1152 Automatic update of common submodule
1153 From f980fd9 to 39ac2f5
1155 2016-12-02 15:40:09 +0100 Edward Hervey <edward@centricular.com>
1157 * gst/rtsp-server/rtsp-media.c:
1158 media: Fix pt map caps
1159 Since decryption is handled within rtpbin, all outcoming stream
1160 caps will be application/x-rtp (i.e. regular rtp)
1161 Fixes RECORD with SRTP streams
1163 2016-12-02 15:38:04 +0100 Edward Hervey <edward@centricular.com>
1165 * gst/rtsp-server/rtsp-media-factory.c:
1166 media-factory: Create media objects with the proper transport mode
1167 The function called immediately afterwards (collect_streams()) will
1168 need it to work properly
1170 2016-12-02 14:36:50 +0200 Sebastian Dröge <sebastian@centricular.com>
1172 * gst/rtsp-server/rtsp-auth.c:
1173 rtsp-auth: Don't remove digest-auth nonces that already/still have a client connected
1175 2016-12-01 18:04:34 +0200 Sebastian Dröge <sebastian@centricular.com>
1177 * gst/rtsp-server/rtsp-media-factory.c:
1178 rtsp-media-factory: Don't create a pipeline for the media pipeline string
1179 We're going to put a pipeline into a pipeline otherwise, which is not
1182 2016-10-25 15:41:28 +0300 Kseniia Vasilchuk <vasilchukkseniia@gmail.com>
1184 * gst/rtsp-server/rtsp-media.c:
1185 media: Fix race condition around finish_unprepare() if called multiple time
1186 https://bugzilla.gnome.org/show_bug.cgi?id=755329
1188 2016-11-30 14:06:36 +1100 Jan Schmidt <jan@centricular.com>
1190 * gst/rtsp-sink/gstrtspclientsink.c:
1191 rtspclientsink: Don't leave stale pointer after unref
1192 Fix a warning on shutdown - don't keep a pointer to an
1193 alread-unreffed object.
1195 2016-11-26 11:24:50 +0000 Tim-Philipp Müller <tim@centricular.com>
1198 common: use https protocol for common submodule
1199 https://bugzilla.gnome.org/show_bug.cgi?id=775110
1201 2016-11-21 23:29:56 +1100 Matthew Waters <matthew@centricular.com>
1203 * gst/rtsp-server/rtsp-stream.c:
1204 stream: block the output of rtpbin instead of the source pipeline
1205 85c52e194bcb81928b96614be0ae47d59eccb1ce introduced a more correct
1206 detection of the srtp rollover counter to add to the SDP.
1207 Unfortunately, it was incomplete for live pipelines where the logic
1208 blocks the source bin before creating the SDP and thus would never have
1209 the necessary informaiton to create a correct SDP with srtp encryption.
1210 Move the pad blocks to rtpbin's output pads instead so that the
1211 necessary information can be created before we need the information for
1213 https://bugzilla.gnome.org/show_bug.cgi?id=770239
1215 2016-11-21 16:02:39 +0100 Dag Gullberg <dagg@axis.com>
1217 * gst/rtsp-server/rtsp-client.c:
1218 rtsp-client: add IDLE timeout, before session exists
1219 The RTSP server will not timeout an idle RTSP connection
1220 (note this is different from doing timeout on a RTSP
1222 At least for Apache this is a problem when running RTSP over
1223 HTTPS since it uses one of the threads (there is a rather
1224 limited number) that are available for handling requests.
1225 https://bugzilla.gnome.org/show_bug.cgi?id=771830
1227 2016-11-23 09:45:08 +0000 Tim-Philipp Müller <tim@centricular.com>
1232 2016-11-21 13:05:50 +0100 Göran Jönsson <goranjn@axis.com>
1234 * gst/rtsp-server/rtsp-stream.c:
1235 rtsp-stream: Set close-socket FALSE on UDP src:es
1236 With this RTSP server can use the sockets independent on the udpsrc
1238 When the udp src is finalized it will unref socket and when g_socket
1239 is finalized the socket will be closed.
1240 https://bugzilla.gnome.org/show_bug.cgi?id=765673
1242 2016-11-18 17:47:13 +0200 Sebastian Dröge <sebastian@centricular.com>
1244 * gst/rtsp-sink/gstrtspclientsink.c:
1245 rtspclientsink: Move to new helper function to parse authentication responses
1246 https://bugzilla.gnome.org/show_bug.cgi?id=774416
1248 2016-11-16 08:42:24 +0200 Sebastian Dröge <sebastian@centricular.com>
1250 * examples/Makefile.am:
1251 * examples/test-auth-digest.c:
1252 * gst/rtsp-server/rtsp-auth.c:
1253 * gst/rtsp-server/rtsp-auth.h:
1254 * win32/common/libgstrtspserver.def:
1255 rtsp-auth: Add support for Digest authentication
1256 https://bugzilla.gnome.org/show_bug.cgi?id=774416
1258 2016-11-17 09:41:53 -0800 Scott D Phillips <scott.d.phillips@intel.com>
1261 * gst/rtsp-server/meson.build:
1263 * tests/check/meson.build:
1265 * win32/common/libgstrtspserver.def:
1266 Enable building with MSVC
1267 https://bugzilla.gnome.org/show_bug.cgi?id=774640
1269 2016-11-18 20:23:14 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
1272 meson: gstreamer gst_check_dep does not exist on windows
1274 2016-11-17 09:43:37 -0800 Scott D Phillips <scott.d.phillips@intel.com>
1276 * gst/rtsp-server/rtsp-client.c:
1277 client: update do_send_message to match type GstRTSPClientSendFunc
1278 This type mismatch fails building with MSVC
1279 https://bugzilla.gnome.org/show_bug.cgi?id=774640
1281 2016-11-11 14:42:08 +0200 Sebastian Dröge <sebastian@centricular.com>
1283 * gst/rtsp-server/rtsp-sdp.c:
1284 rtsp-sdp: Fix indentation
1286 2016-11-10 05:16:00 +0000 Neha Arora <arora.neha@samsung.com>
1288 * gst/rtsp-server/rtsp-media.c:
1289 rtsp-media: Only signal "new-state" if the state has actually changed
1290 https://bugzilla.gnome.org/show_bug.cgi?id=774173
1292 2016-08-24 11:39:13 +0200 Branko Subasic <branko@axis.com>
1294 * gst/rtsp-server/rtsp-client.c:
1295 * gst/rtsp-server/rtsp-client.h:
1296 client: emit signal in the beginning of each rtsp request
1297 These signals let the application validate the requests, configure the
1298 media/stream in a certain way and also generate error status code in
1299 case of error or bad request.
1300 https://bugzilla.gnome.org/show_bug.cgi?id=758062
1302 2016-11-01 18:10:35 +0000 Tim-Philipp Müller <tim@centricular.com>
1305 meson: update version
1307 === release 1.11.0 ===
1309 2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com>
1314 === release 1.10.0 ===
1316 2016-11-01 18:06:46 +0200 Sebastian Dröge <sebastian@centricular.com>
1322 * gst-rtsp-server.doap:
1325 2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com>
1327 * tests/check/gst/rtspserver.c:
1328 * tests/check/gst/stream.c:
1329 tests: try to avoid using the same ports in different tests
1330 Causes problems with client multicast tests otherwise if
1331 tests are run in parallel.
1332 https://bugzilla.gnome.org/show_bug.cgi?id=773640
1334 2016-10-28 17:50:59 +0100 Tim-Philipp Müller <tim@centricular.com>
1336 * tests/check/gst/client.c:
1337 tests: client: use fail_unless_equals_foo() for better failure reporting
1339 2016-09-26 11:16:04 +0200 Göran Jönsson <goranjn@axis.com>
1341 * gst/rtsp-server/rtsp-client.c:
1342 rtsp-client: Session filter in unwatch session
1343 Call session filter with filter_session_media as paramer in
1344 client_unwatch_session if using drop_backlog = FALSE.
1345 In client_unwatch_session its allowed to grow the watchs backlog.
1346 If using drop_backlog = FALSE and the backlog is full it will cause
1347 a deadlock when setting session media state to NULL
1348 if the backlog is not allowed to grow.
1349 https://bugzilla.gnome.org/show_bug.cgi?id=771983
1351 2016-10-20 21:40:18 +0100 Tim-Philipp Müller <tim@centricular.com>
1354 meson: add fallbacks for gst modules
1357 2016-09-14 17:48:39 +0300 Nikita Bobkov <NikitaDBobkov@gmail.com>
1359 * gst/rtsp-server/rtsp-client.c:
1360 rtsp-client: Fix factory leaking in find_media() in error cases
1361 https://bugzilla.gnome.org/show_bug.cgi?id=771488
1363 2016-10-06 11:47:50 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1365 * gst/rtsp-server/rtsp-stream.c:
1366 stream: Fix randomly missing streams from SDP with dynamic elements
1367 When using dynamic elements, gst_rtsp_stream_join_bin() is called from
1368 "pad-added" signal. In that case priv->srcpad could already have its caps,
1369 and they'll be sent to priv->send_src[0] pad. That means that when it
1370 connects "notify::caps" signal, that pad could already have received its
1371 caps and the signal won't be emitted anymore.
1372 In that case priv->caps stay to NULL and when building the SDP that stream
1373 gets ignored. Leading to missing video or audio when playing in client side.
1374 https://bugzilla.gnome.org/show_bug.cgi?id=772478
1376 2016-09-30 11:42:08 +0100 Tim-Philipp Müller <tim@centricular.com>
1379 meson: update version
1381 === release 1.9.90 ===
1383 2016-09-30 13:04:12 +0300 Sebastian Dröge <sebastian@centricular.com>
1389 * gst-rtsp-server.doap:
1392 2016-09-17 13:17:19 +0100 Ian Jamison <ian.dev@arkver.com>
1394 * gst/rtsp-server/rtsp-media-factory.c:
1395 * gst/rtsp-server/rtsp-media.c:
1396 * gst/rtsp-server/rtsp-stream.c:
1397 rtsp-server: Hint that set_multicast_iface expects the name of the interface
1398 To prevent any possibly confusion with IPs or anything else.
1399 https://bugzilla.gnome.org/show_bug.cgi?id=771530
1401 2016-09-18 09:58:55 -0400 Sebastian Dröge <sebastian@centricular.com>
1403 * gst/rtsp-server/rtsp-media-factory.c:
1404 * gst/rtsp-server/rtsp-media.c:
1405 rtsp-media: Call g_free() instead of g_object_unref() on multicast-iface strings
1406 https://bugzilla.gnome.org/show_bug.cgi?id=763000#c5
1408 2016-09-14 11:31:15 +0200 Sebastian Dröge <sebastian@centricular.com>
1411 configure: Depend on gstreamer 1.9.2.1
1413 2016-09-10 20:52:31 +1000 Jan Schmidt <jan@centricular.com>
1417 Automatic update of common submodule
1418 From b18d820 to f980fd9
1420 2016-09-10 09:58:31 +1000 Jan Schmidt <jan@centricular.com>
1424 Automatic update of common submodule
1425 From 6f2d209 to b18d820
1427 2016-09-07 18:44:34 +0300 Sebastian Dröge <sebastian@centricular.com>
1429 * gst/rtsp-server/rtsp-stream.c:
1430 rtsp-stream: Remove unused _locked() variant of a function
1431 It was added during refactoring.
1433 2016-09-07 10:21:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1435 * gst/rtsp-server/rtsp-stream.c:
1436 stream: cosmetic cleanup
1437 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1439 2016-09-07 10:16:19 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1441 * gst/rtsp-server/rtsp-stream.c:
1442 stream: Compare IP addresses case insensitive in more places
1443 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1445 2016-09-07 10:12:18 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1448 * gst/rtsp-server/rtsp-stream.c:
1449 stream: Fix leaked joined_bin
1450 There is no need to keep a strong ref on it, and _leave_bin() was
1451 setting it to NULL before calling g_clear_object() so it was leaked.
1452 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1454 2016-09-06 19:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
1456 * gst/rtsp-server/rtsp-stream.c:
1457 rtsp-stream: Compare IP address strings case insensitive
1458 Otherwise IPv6 addresses might fail this comparision.
1460 2016-09-06 19:10:21 +0300 Sebastian Dröge <sebastian@centricular.com>
1462 * gst/rtsp-server/rtsp-stream.c:
1463 rtsp-stream: Bind multicast sockets to ANY as before
1464 https://bugzilla.gnome.org/show_bug.cgi?id=766612#c48
1466 2016-09-05 18:31:36 +0300 Kseniia <vasilchukkseniia@gmail.com>
1468 * gst/rtsp-server/rtsp-session.c:
1469 rtsp-session: Fix segfault when doing keep-alive after removing the session
1470 If keep-alive happens after removing the session but before finalizing the
1471 stream transport, we would segfault.
1472 https://bugzilla.gnome.org/show_bug.cgi?id=750544
1474 2016-09-05 18:04:50 +0300 Sebastian Dröge <sebastian@centricular.com>
1476 * gst/rtsp-server/rtsp-stream.c:
1477 rtsp-stream: Always create multicast UDP elements if the protocol flag is set
1478 Adding them later will cause deadlocks due to
1479 1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
1480 2) adding the multicast sink
1481 3) waiting for it to get data to preroll again
1482 3) never happens because the queues after the tee are full.
1484 2016-09-05 16:32:57 +0300 Sebastian Dröge <sebastian@centricular.com>
1486 * gst/rtsp-server/rtsp-stream.c:
1487 rtsp-stream: Fix up various multicast related issues
1489 2016-09-05 13:40:59 +0300 Sebastian Dröge <sebastian@centricular.com>
1491 * tests/check/gst/stream.c:
1492 tests: Fix compilation
1494 2016-07-28 15:33:05 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1496 * gst/rtsp-server/rtsp-client.c:
1497 * gst/rtsp-server/rtsp-stream.c:
1498 * tests/check/gst/stream.c:
1499 stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast
1500 This is basically reverting changes introduced in commit f62a9a7,
1501 because it was introducing various regressions:
1502 - It introduces a leak of udpsrc elements that got wrongly fixed by adding
1503 an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
1504 ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
1505 - If a mcast client connects, it creates a new socket in SETUP to try to respect
1506 the destination/port given by the client in the transport, and overrides the
1507 socket already set on the udpsink element. That means that if we already had a
1508 client connected, the source address on the udp packets it receives suddenly
1510 - If a 2nd mcast client connects, the destination/port in its transport is
1511 ignored but its transport wasn't updated.
1512 What this patch does:
1513 - Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
1514 - Always have a tee+queue when udp is enabled. This could be optimized
1515 again in a later patch, but is more complicated. If no unicast clients
1516 connects then those elements are useless, this could be also optimized
1518 - When mcast transport is added, it creates a new set of udpsrc/udpsink,
1519 seperated from those for unicast clients. Since we already support only
1520 one mcast address, we also create only one set of elements.
1521 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1523 2016-07-28 15:20:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1525 * gst/rtsp-server/rtsp-stream.c:
1526 stream: factor our plug_src function
1527 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1529 2016-07-21 21:46:16 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1531 * gst/rtsp-server/rtsp-stream.c:
1532 stream: factor out plug_sink function
1533 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1535 2016-07-20 23:05:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1537 * gst/rtsp-server/rtsp-stream.c:
1538 stream: small documentation clarification
1539 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1541 2016-07-20 15:35:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1543 * gst/rtsp-server/rtsp-stream.c:
1544 stream: rename addr_v4/6 to mcast_addr_v4/6 for clarity
1545 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1547 2016-07-14 11:10:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1549 * gst/rtsp-server/rtsp-stream.c:
1550 stream: Keep a ref on joined bin
1551 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1553 2016-07-20 15:11:32 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1555 * gst/rtsp-server/rtsp-stream.c:
1556 stream: code cleanup
1557 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1559 2016-07-20 23:18:23 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1561 * gst/rtsp-server/rtsp-stream.c:
1562 stream: small fix in error code path
1563 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1565 2016-07-20 20:09:57 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1567 * gst/rtsp-server/rtsp-stream.c:
1568 Revert "rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc"
1569 This partly reverts commit cba045e1b19fad6e689e10206f57903e15f1229a,
1570 but keeps unit tests.
1571 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1573 2016-09-01 12:33:00 +0300 Sebastian Dröge <sebastian@centricular.com>
1578 === release 1.9.2 ===
1580 2016-09-01 12:32:51 +0300 Sebastian Dröge <sebastian@centricular.com>
1586 * gst-rtsp-server.doap:
1589 2016-01-27 01:03:52 +0000 Tim-Philipp Müller <tim@centricular.com>
1592 * examples/meson.build:
1594 * gst/rtsp-server/meson.build:
1595 * gst/rtsp-sink/meson.build:
1597 * pkgconfig/meson.build:
1598 * tests/check/meson.build:
1599 * tests/meson.build:
1600 Add support for Meson as alternative/parallel build system
1601 https://github.com/mesonbuild/meson
1603 2016-08-26 21:56:13 +0200 Josep Torra <n770galaxy@gmail.com>
1606 * tests/check/Makefile.am:
1607 build: silence error about pthread for 'make check' in osx
1608 Fixes "clang: error: argument unused during compilation: '-pthread'"
1610 2015-09-25 15:04:00 +0000 Nikita Bobkov <NikitaDBobkov@gmail.com>
1612 * gst/rtsp-server/rtsp-client.c:
1613 rtsp-client: Fix leaking of media in error cases
1614 With additional fixes by Kseniya Vasilchuk <vasilchukkseniia@gmail.com>
1615 and myself to make the media refcounting a bit easier to follow.
1616 https://bugzilla.gnome.org/show_bug.cgi?id=755632
1618 2016-08-02 15:08:22 +0300 Sebastian Dröge <sebastian@centricular.com>
1620 * gst/rtsp-server/rtsp-client.c:
1621 rtsp-client: Fix leaking of session in error cases
1622 https://bugzilla.gnome.org/show_bug.cgi?id=755632
1624 2016-07-11 21:16:04 +0200 Stefan Sauer <ensonic@users.sf.net>
1627 Automatic update of common submodule
1628 From f363b32 to f49c55e
1630 2016-07-06 13:51:15 +0300 Sebastian Dröge <sebastian@centricular.com>
1635 === release 1.9.1 ===
1637 2016-07-06 13:28:12 +0300 Sebastian Dröge <sebastian@centricular.com>
1643 * gst-rtsp-server.doap:
1646 2016-06-24 02:02:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
1649 configure: Need to add -DGST_STATIC_COMPILATION when building only statically
1650 https://bugzilla.gnome.org/show_bug.cgi?id=767463
1652 2016-06-21 11:49:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
1655 Automatic update of common submodule
1656 From ac2f647 to f363b32
1658 2016-04-14 22:56:11 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1660 * gst/rtsp-server/rtsp-sdp.c:
1661 * gst/rtsp-server/rtsp-sdp.h:
1662 * gst/rtsp-server/rtsp-stream.c:
1663 * gst/rtsp-server/rtsp-stream.h:
1664 sdp: add rollover counters for all sender SSRC
1665 We add different crypto sessions in MIKEY, one for each sender
1666 SSRC. Currently, all of them will have the same security policy, 0.
1667 The rollover counters are obtained from the srtpenc element using the
1669 https://bugzilla.gnome.org/show_bug.cgi?id=730539
1671 2016-06-07 20:44:42 +0100 Tim-Philipp Müller <tim@centricular.com>
1673 * gst/rtsp-server/rtsp-media-factory.h:
1674 * gst/rtsp-server/rtsp-server.h:
1675 docs: fix some typos
1677 2016-05-25 10:28:43 +0100 Tim-Philipp Müller <tim@centricular.com>
1679 * gst/rtsp-server/Makefile.am:
1680 g-i: pass compiler env to g-ir-scanner
1681 It's what introspection.mak does as well. Should
1682 fix spurious build failures on gnome-continuous
1683 (caused by g-ir-scanner getting compiler details
1684 via python which is broken in some environments
1685 so passing the compiler details bypasses that).
1687 2016-05-18 16:48:44 +0100 Ian <ian.arkver.dev@gmail.com>
1689 * gst/rtsp-server/rtsp-session.c:
1690 rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header
1691 This works with rtspsrc and live555, but fails with e.g. ffmpeg.
1692 https://bugzilla.gnome.org/show_bug.cgi?id=766619
1694 2016-03-07 14:48:38 +0100 Edward Hervey <bilboed@bilboed.com>
1696 * gst/rtsp-sink/gstrtspclientsink.c:
1697 rtspclientsink: Check return value of sscanf
1698 And just make sure we always have 0/0 if we have an error
1701 2016-04-25 08:55:25 -0400 Jake Foytik <jake.foytik@ipconfigure.com>
1703 * gst/rtsp-server/rtsp-stream.c:
1704 * tests/check/gst/rtspserver.c:
1705 * tests/check/gst/stream.c:
1706 rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
1707 - Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
1708 - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
1709 - Create unit test for shared media.
1710 https://bugzilla.gnome.org/show_bug.cgi?id=764744
1712 2016-04-11 10:55:23 +0300 Sebastian Dröge <sebastian@centricular.com>
1714 * gst/rtsp-server/rtsp-stream.c:
1715 rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
1716 For IPv6 addresses, binding to a multicast group does not work on Linux
1717 either. Always bind to ANY and then later join the multicast group.
1718 https://bugzilla.gnome.org/show_bug.cgi?id=764679
1720 2016-04-14 10:05:02 +0100 Julien Isorce <j.isorce@samsung.com>
1723 Automatic update of common submodule
1724 From 6f2d209 to ac2f647
1726 2016-04-06 10:09:46 +0200 Patricia Muscalu <patricia@axis.com>
1728 * gst/rtsp-server/rtsp-thread-pool.c:
1729 rtsp-thread-pool: explained why GSource is a part of ThreadImpl
1730 Clarified why it is necessary to add source information to
1731 GstRTSPThreadImpl. See the reported bug in GLib:
1732 https://bugzilla.gnome.org/show_bug.cgi?id=720186
1733 for more information.
1734 https://bugzilla.gnome.org/show_bug.cgi?id=761702
1736 2016-04-04 12:58:38 +0300 Sebastian Dröge <sebastian@centricular.com>
1738 * examples/Makefile.am:
1739 examples: Clean up CFLAGS/LDADD even more
1740 The internal .la should come first and is part of LDADD, as is
1743 2016-04-04 12:39:39 +0300 Sebastian Dröge <sebastian@centricular.com>
1745 * examples/Makefile.am:
1746 examples: Clean up CFLAGS/LDADD to link with the correct versions of all libraries
1748 2016-04-03 12:06:29 +0300 Sebastian Dröge <sebastian@centricular.com>
1750 * gst/rtsp-server/Makefile.am:
1751 rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS)
1753 2015-12-30 18:39:05 +0200 Sebastian Dröge <sebastian@centricular.com>
1755 * gst/rtsp-server/rtsp-client.c:
1756 * gst/rtsp-server/rtsp-media-factory.c:
1757 * gst/rtsp-server/rtsp-media-factory.h:
1758 * gst/rtsp-server/rtsp-media.c:
1759 * gst/rtsp-server/rtsp-media.h:
1760 * gst/rtsp-server/rtsp-sdp.c:
1761 * gst/rtsp-server/rtsp-stream.c:
1762 * gst/rtsp-server/rtsp-stream.h:
1763 rtsp-server: Implement clock signalling according to RFC7273
1764 For NTP and PTP clocks we signal the actual clock that is used and signal
1765 the direct media clock offset.
1766 For all other clocks we at least signal that it's the local sender clock.
1767 This allows receivers to know which clock was used to generate the media and
1768 its RTP timestamps. Receivers can then implement network synchronization,
1769 either absolute or at least relative by getting the sender clock rate directly
1770 via NTP/PTP instead of estimating it from RTP timestamps and packet receive
1772 https://bugzilla.gnome.org/show_bug.cgi?id=760005
1774 2016-03-02 19:42:58 +0200 Sebastian Dröge <sebastian@centricular.com>
1776 * gst/rtsp-sink/gstrtspclientsink.c:
1777 rtspclientsink: Add support for setting the multicast interface
1778 https://bugzilla.gnome.org/show_bug.cgi?id=763000
1780 2016-03-02 19:42:13 +0200 Sebastian Dröge <sebastian@centricular.com>
1782 * gst/rtsp-server/rtsp-media-factory.c:
1783 * gst/rtsp-server/rtsp-media-factory.h:
1784 * gst/rtsp-server/rtsp-media.c:
1785 * gst/rtsp-server/rtsp-media.h:
1786 * gst/rtsp-server/rtsp-stream.c:
1787 * gst/rtsp-server/rtsp-stream.h:
1788 rtsp-media: Add support for setting the multicast interface
1789 https://bugzilla.gnome.org/show_bug.cgi?id=763000
1791 2016-03-07 08:50:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
1793 * gst/rtsp-sink/gstrtspclientsink.c:
1794 rtspclientsink: use new gst_element_class_add_static_pad_template()
1795 https://bugzilla.gnome.org/show_bug.cgi?id=763196
1797 2016-03-24 13:33:43 +0200 Sebastian Dröge <sebastian@centricular.com>
1802 === release 1.8.0 ===
1804 2016-03-24 13:00:35 +0200 Sebastian Dröge <sebastian@centricular.com>
1810 * gst-rtsp-server.doap:
1813 2016-03-16 23:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
1815 * gst/rtsp-server/rtsp-stream.c:
1816 rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
1817 This would get us NO_PREROLL in the bin again and break seeking.
1818 Thanks to Carlos Rafael Giani for helping to debug this!
1819 https://bugzilla.gnome.org/show_bug.cgi?id=740509
1821 === release 1.7.91 ===
1823 2016-03-15 12:26:13 +0200 Sebastian Dröge <sebastian@centricular.com>
1829 * gst-rtsp-server.doap:
1832 2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
1834 * gst/rtsp-server/rtsp-stream.c:
1835 rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
1836 Without this, RECORD pipelines are broken because
1837 a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
1838 added later. Previously it was there earlier and due to NO_PREROLL caused the
1839 pipeline to preroll immediately
1840 b) the udpsrc for the pipeline is added later and never set to PLAYING state,
1841 as the corresponding code previously was only for PLAY pipelines.
1842 https://bugzilla.gnome.org/show_bug.cgi?id=763281
1844 2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
1846 * gst/rtsp-server/rtsp-stream.c:
1847 rtsp-stream: Fix typo in the docstring
1848 gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
1850 2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
1852 * gst/rtsp-server/rtsp-stream.c:
1853 rtsp-stream: Disable multicast loopback for all our sockets
1854 On Windows this is a receiver-side setting, on Linux a sender-side setting. As
1855 we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
1856 loopback setting on the socket... while udpsink does which unfortunately has
1857 no effect here on Windows but on Linux.
1858 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1860 2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
1862 * tests/check/gst/stream.c:
1863 stream tests: added new tests
1864 Test a case when the address pool only contains multicast addresses
1865 and the client is requesting unicast udp.
1866 Added tests for multicast ports allocation.
1867 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1869 2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
1871 * gst/rtsp-server/rtsp-stream.c:
1872 rtsp-stream: Only bind multicast sockets to ANY on Windows
1873 On Linux it is still needed to bind to the multicast address
1874 to filter out random other packets, while on Windows binding
1875 to multicast addresses just fails.
1877 2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
1879 * gst/rtsp-server/rtsp-stream.c:
1880 rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
1881 Otherwise we fail to allocate UDP ports if the pool only contains multicast
1882 addresses, which is something that used to work before. For unicast addresses
1883 if the pool contains none, we just allocate them as if there is no pool at
1885 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1887 2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
1889 * gst/rtsp-server/rtsp-client.c:
1890 * gst/rtsp-server/rtsp-stream.c:
1891 rtsp-server: Fix indentation
1893 2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
1895 * gst/rtsp-server/rtsp-stream.c:
1896 rtsp-stream: Don't bind the sockets to multicast addresses
1897 This works on Linux but fails completely on Windows. You're supposed
1898 to bind to ANY and then join the multicast group.
1899 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1901 === release 1.7.90 ===
1903 2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
1909 * gst-rtsp-server.doap:
1912 2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
1915 Automatic update of common submodule
1916 From b64f03f to 6f2d209
1918 2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
1920 * gst/rtsp-sink/gstrtspclientsink.c:
1921 * tests/check/gst/rtspclientsink.c:
1922 rtspsink: Fix some leaks in rtspclientsink and the unit test.
1923 https://bugzilla.gnome.org/show_bug.cgi?id=762525
1925 2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
1927 * tests/check/gst/media.c:
1928 * tests/check/gst/rtspclientsink.c:
1929 * tests/check/gst/rtspserver.c:
1930 * tests/check/gst/stream.c:
1931 tests: unit test fixes
1932 Removed port allocation test from the media suite.
1933 The port allocation failure is now in the stream suite.
1935 Make sure that the media is suspended after the DESCRIBE request
1936 before reconfiguring the UDP sinks.
1938 In the RECORD case we have to set async property to false
1939 for the appsink element in the test in order to make sure
1940 that the media pipeline doesn't hang in start_preroll().
1941 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1943 2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
1945 * gst/rtsp-server/rtsp-client.c:
1946 * gst/rtsp-server/rtsp-stream.c:
1947 * gst/rtsp-server/rtsp-stream.h:
1948 rtsp-stream: postpone UDP socket allocation until SETUP
1949 Postpone the allocation of the UDP sockets until we know
1950 what transport has been chosen by the client.
1951 Both unicast and multicast UDP sources are created in one
1953 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1955 2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
1957 * gst/rtsp-server/rtsp-stream.c:
1958 rtsp-stream: postpone the creation of the UDP sources
1959 Code refactoring: allocate the UDP ports after the sender and
1960 the reciver parts have been created.
1961 We postpone the creation of the UDP sources until the UDP
1962 ports have been allocated.
1963 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1965 2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
1967 * gst/rtsp-server/rtsp-stream.c:
1968 rtsp-stream: added function for setting UDP sources to PLAYING state
1969 Code refactoring: Introduced a function for setting UDP sources
1971 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1973 2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
1975 * gst/rtsp-server/rtsp-stream.c:
1976 rtsp-stream: added function for creating and configuring UDP sources
1977 Code refactoring: create and configure UDP sources in a separate function.
1978 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1980 2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
1982 * gst/rtsp-server/rtsp-stream.c:
1983 rtsp-stream: added function for RTP/RTCP socket configuration
1984 Code refactoring: configure RTP and RTCP sockets for UDP sinks
1985 in a separate function.
1986 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1988 2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
1990 * gst/rtsp-server/rtsp-stream.c:
1991 rtsp-stream: added function for creating and configuring UDP sinks
1992 Code refactoring: create and configure UDP sinks in a separate function.
1993 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1995 2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
1997 * gst/rtsp-server/rtsp-stream.c:
1998 rtsp-stream: added helper function for creating the sender/receiver parts
1999 Code refactoring: introduced helper function for creating
2000 the receiver and the sender parts of the streaming pipeline.
2001 https://bugzilla.gnome.org/show_bug.cgi?id=757488
2003 2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
2008 === release 1.7.2 ===
2010 2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
2016 * gst-rtsp-server.doap:
2019 2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
2021 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
2022 uninstalled.pc: add support for non libtool build systems
2023 Currently the .la path is provided which requires to use libtool as
2024 mentioned in the GStreamer manual section-helloworld-compilerun.html.
2025 It is fine as long as the application is built using libtool.
2026 So currently it is not possible to compile a GStreamer application
2027 within gst-uninstalled with CMake or other build system different
2029 This patch allows to do the following in gst-uninstalled env:
2030 gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
2031 gstreamer-rtsp-server-1.0)
2032 Previously it required to prepend libtool --mode=link
2033 https://bugzilla.gnome.org/show_bug.cgi?id=720778
2035 2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
2037 * gst/rtsp-sink/gstrtspclientsink.c:
2038 rtspclientsink: remove check for impossible condition
2039 Goto error label checks stream to see if it needs to be unreferenced before
2040 returning, but this goto jumps happens before the stream is ever set, so it
2041 will always be NULL in this error label.
2044 2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
2046 * gst/rtsp-sink/gstrtspclientsink.c:
2047 rtspclientsink: clean switch statements
2048 Coverity demands for fallthrough statements to be clearly commented,
2049 to distinguish from accidental fall throughs. And it also needs all
2050 cases to finish with a break, even if the break is never going to be
2051 executed like in the case of a continue jump.
2055 2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
2057 * tests/check/Makefile.am:
2058 tests: extend the AM_TESTS_ENVIRONMENT from check.mak
2059 To get the CK_DEFAULT_TIMEOUT defined for all tests
2060 Also removes a 120 seconds timeout that was set as default
2061 explicitly in this module
2062 https://bugzilla.gnome.org/show_bug.cgi?id=761472
2064 2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
2068 Automatic update of common submodule
2069 From 86e4663 to b64f03f
2071 2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
2073 * gst/rtsp-server/rtsp-media.c:
2074 rtsp-media: fix state_lock not locked again when preroll fails
2075 https://bugzilla.gnome.org/show_bug.cgi?id=761399
2077 2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
2080 configure: Move plugin specific flags below all the others
2081 They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
2082 -no-undefined. And -no-undefined is required on Windows to build DLLs.
2084 2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
2086 * gst/rtsp-sink/gstrtspclientsink.c:
2087 rtspclientsink: Simplify slightly using new -base API
2088 Use the new Mikey and SDP API in the base plugins libs
2089 to simplify some code.
2090 https://bugzilla.gnome.org/show_bug.cgi?id=758180
2092 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
2097 * gst/rtsp-sink/Makefile.am:
2098 * gst/rtsp-sink/gstrtspclientsink.c:
2099 * gst/rtsp-sink/gstrtspclientsink.h:
2100 * gst/rtsp-sink/plugin.c:
2101 * tests/check/Makefile.am:
2102 * tests/check/gst/rtspclientsink.c:
2103 rtspsink: Add rtspclientsink element
2104 Add an rtspclientsink element that accepts streams for which
2105 there is a registered payloader and sends them to
2106 an RTSP server using RECORD.
2107 Sending is synchronised to the pipeline clock. Payload-types
2108 are automatically selected. The 'new-payloader' signal is fired
2109 for custom configuration of payloaders when they are created.
2110 Can now stream a movie like this:
2112 ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
2113 decodebin name=depay1 ! audioconvert ! autoaudiosink )"
2115 gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
2116 queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
2117 https://bugzilla.gnome.org/show_bug.cgi?id=758180
2119 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
2121 * gst/rtsp-server/rtsp-stream.c:
2122 * gst/rtsp-server/rtsp-stream.h:
2123 rtsp-stream: Add functions for using rtsp-stream from the client
2124 Add a boolean to indicate that the rtsp-stream is running on the
2125 'client' side of an RTSP connection, for sending streams via
2126 RECORD. In that case, the roles of the client/server ports
2127 in transport setup are swapped.
2128 https://bugzilla.gnome.org/show_bug.cgi?id=758180
2130 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
2132 * gst/rtsp-server/rtsp-sdp.c:
2133 * gst/rtsp-server/rtsp-sdp.h:
2134 rtsp-sdp: Add gst_rtsp_sdp_from_stream()
2135 A new function that adds info from a GstRTSPStream into an SDP message.
2136 https://bugzilla.gnome.org/show_bug.cgi?id=758180
2138 2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
2140 * gst/rtsp-server/rtsp-media.c:
2141 rtsp-media: Fix mutex beeing unlocked while they should be locked
2142 https://bugzilla.gnome.org/show_bug.cgi?id=761226
2144 2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
2146 * gst/rtsp-server/rtsp-media-factory.c:
2147 rtsp-media-factory: add missing break in "clock" property setter
2150 2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
2152 * gst/rtsp-server/rtsp-stream.c:
2153 rtsp-stream: fixed assert during update transport
2154 When RTSP server trying update transport during multicast, it throws an
2155 assert. The assert is thrown because it is trying to get the parent of
2156 an non-existing funnel element.
2157 https://bugzilla.gnome.org/show_bug.cgi?id=760150
2159 2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
2161 * gst/rtsp-server/rtsp-permissions.h:
2162 * gst/rtsp-server/rtsp-thread-pool.h:
2163 * gst/rtsp-server/rtsp-token.h:
2164 docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
2165 gtk-doc can handle static inline functions just fine these days,
2166 there's no need for this stuff any more.
2168 2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
2170 * gst/rtsp-server/rtsp-media.c:
2171 * gst/rtsp-server/rtsp-sdp.c:
2172 sdp: replace duplicated codes to call new base sdp apis
2173 https://bugzilla.gnome.org/show_bug.cgi?id=745880
2175 2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
2177 * examples/test-netclock.c:
2178 test-netclock: Use the new API to configure a clock directly
2180 2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
2182 * gst/rtsp-server/rtsp-media-factory.c:
2183 * gst/rtsp-server/rtsp-media-factory.h:
2184 * gst/rtsp-server/rtsp-media.c:
2185 * gst/rtsp-server/rtsp-media.h:
2186 rtsp-media: Add API to directly configure a clock on the media pipelines
2188 2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
2190 * gst/rtsp-server/rtsp-media.c:
2191 rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
2193 2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
2195 * gst/rtsp-server/rtsp-media-factory.c:
2196 rtsp-media-factory: Add FIXME for 2.0
2198 2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
2200 * gst/rtsp-server/rtsp-stream.c:
2201 rtsp-stream: Fix indentation
2203 2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2205 * gst/rtsp-server/rtsp-media.c:
2206 rtsp-media: Do not prepare media after media times out
2207 Deferred calls to start_prepare() can be deferred past the point until
2208 which wait_preroll() and by proxy gst_rtsp_media_get_status() is
2209 prepared to wait. Previously there was no lock and no check for this
2210 situation. This meant that a media could be prepared and unprepared
2211 simultaneously by two different threads. Now a lock is in place and a
2212 suitable check is done.
2213 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
2215 2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
2217 * gst/rtsp-server/rtsp-client.c:
2218 * gst/rtsp-server/rtsp-media-factory.c:
2219 * gst/rtsp-server/rtsp-media-factory.h:
2220 * gst/rtsp-server/rtsp-media.c:
2221 * gst/rtsp-server/rtsp-media.h:
2222 rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
2223 Without TEARDOWN it might be desireable to keep the media running and continue
2224 sending data to the client, even if the RTSP connection itself is
2226 Only do this for session medias that have only UDP transports. If there's at
2227 least on TCP transport, it will stop working and cause problems when the
2228 connection is disconnected.
2229 https://bugzilla.gnome.org/show_bug.cgi?id=758999
2231 2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
2236 === release 1.7.1 ===
2238 2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
2244 * gst-rtsp-server.doap:
2247 2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
2250 configure: Make -Bsymbolic check work with clang.
2251 Update the -Bsymbolic check with the version glib has. This version
2253 https://bugzilla.gnome.org/show_bug.cgi?id=759713
2255 2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
2257 * gst/rtsp-server/rtsp-session-pool.c:
2258 rtsp-session-pool: Avoid dollar sign ($) in session ids
2259 Live555 in VLC strips off dollar signs and then gets very confused,
2260 we don't loose too much entropy by just skipping it.
2262 2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
2264 * gst/rtsp-server/rtsp-address-pool.h:
2265 * gst/rtsp-server/rtsp-auth.h:
2266 * gst/rtsp-server/rtsp-client.h:
2267 * gst/rtsp-server/rtsp-media-factory-uri.h:
2268 * gst/rtsp-server/rtsp-media-factory.h:
2269 * gst/rtsp-server/rtsp-media.h:
2270 * gst/rtsp-server/rtsp-mount-points.h:
2271 * gst/rtsp-server/rtsp-permissions.h:
2272 * gst/rtsp-server/rtsp-server.h:
2273 * gst/rtsp-server/rtsp-session-media.h:
2274 * gst/rtsp-server/rtsp-session-pool.h:
2275 * gst/rtsp-server/rtsp-session.h:
2276 * gst/rtsp-server/rtsp-stream-transport.h:
2277 * gst/rtsp-server/rtsp-stream.h:
2278 * gst/rtsp-server/rtsp-thread-pool.h:
2279 * gst/rtsp-server/rtsp-token.h:
2280 rtsp-server: Add g_autoptr() support to all types
2281 https://bugzilla.gnome.org/show_bug.cgi?id=754464
2283 2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
2285 * gst/rtsp-server/rtsp-stream.c:
2286 rtsp-stream: fixed valgrind error
2287 Fixed the valgrind error in unit test. The UDP source created during
2288 gst_rtsp_stream_join_bin() was not released while destroying the rtp
2290 https://bugzilla.gnome.org/show_bug.cgi?id=759010
2292 2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
2296 Automatic update of common submodule
2297 From b319909 to 86e4663
2299 2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
2301 * gst/rtsp-server/rtsp-client.c:
2302 rtsp-client: suspend media during setup request
2303 SETUP request from clients needs to suspend the media to clear the
2304 prerolled buffers. Otherwise it will not affect the prerolled buffer
2305 and the prerolled buffers will be incorrect (for example block-size
2306 from setup request will not affect the prerolled buffer unless the
2307 media is suspended).
2308 https://bugzilla.gnome.org/show_bug.cgi?id=758268
2310 2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
2312 * gst/rtsp-server/rtsp-stream.c:
2313 rtsp-stream: create stream pipeline based on transport
2314 Based on the protocol, create the rtsp stream pipeline. If only TCP or
2315 only UDP is set as the transport protocol, it will not add the extra tee
2316 or queue element to the pipeline. Both these elements will be added, if
2317 it supports both TCP and UDP protocols. This improves the pipeline
2318 performance when one protocol is present.
2319 https://bugzilla.gnome.org/show_bug.cgi?id=758179
2321 2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
2323 * gst/rtsp-server/rtsp-stream.c:
2324 rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
2325 Adding them when not needed will start some logic inside rtpbin that might be
2326 problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
2327 would start up a rtpjitterbuffer and behave in weird ways.
2328 We still set up the UDP sources for RTP receiving for a sender media to be
2329 able to receive any packets sent by the client for NAT traversal. They will
2330 all go to a fakesink though.
2331 Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
2332 NO_PREROLL, which will cause deadlocks when seeking the media as it will never
2333 receive ASYNC_DONE after a seek.
2334 https://bugzilla.gnome.org/show_bug.cgi?id=758319
2336 2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
2338 * gst/rtsp-server/rtsp-stream.c:
2339 rtsp-stream: Disable multicast loopback for the multicast udp sources too
2340 On POSIX this setting is for sender sockets, on Windows for receiver sockets.
2341 Previously we were only setting this for sender sockets, which caused looped
2342 back packets to be received on Windows if a multicast transport was used.
2344 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
2346 * examples/test-record-auth.c:
2347 * examples/test-record.c:
2348 examples: Actually use the provided port in the record examples
2350 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
2352 * examples/test-record-auth.c:
2353 test-record-auth: Add the option to build in TLS support
2355 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
2357 * examples/test-auth.c:
2358 test-auth: Use an 'anonymous' user for unauthenticated default
2359 There's a comment on one of the resources that 'user' and 'admin'
2360 shouldn't even be able to see it, but they can if the default
2361 token is 'admin2', since that gives them access anyway.
2363 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
2365 * examples/.gitignore:
2366 * examples/Makefile.am:
2367 * examples/test-record-auth.c:
2368 Add test-record-auth example
2370 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
2372 * gst/rtsp-server/rtsp-client.c:
2373 * tests/check/gst/client.c:
2374 rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
2376 2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
2378 * gst/rtsp-server/rtsp-server.c:
2379 rtsp-server: Change the logic so we don't pop a NULL context
2380 When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
2381 will sometimes fail. This call is made before any context is pushed
2382 resulting in an attempt to pop a NULL context.
2383 https://bugzilla.gnome.org/show_bug.cgi?id=757949
2385 2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
2387 * tests/check/gst/rtspserver.c:
2388 rtspserver: Add udp-mcast transport SETUP test
2389 Refactor utility functions in the test file so they can handle
2390 more than UDP and TCP as lower transport.
2391 https://bugzilla.gnome.org/show_bug.cgi?id=756969
2393 2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
2395 * gst/rtsp-server/rtsp-stream.c:
2396 rtsp-stream: Always unref return value of gst_object_get_parent()
2397 Fixes a leak of a GstBin in the udp-mcast case.
2398 https://bugzilla.gnome.org/show_bug.cgi?id=756968
2400 2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
2403 Automatic update of common submodule
2404 From b99800a to b319909
2406 2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
2409 Use new GST_ENABLE_EXTRA_CHECKS #define
2410 https://bugzilla.gnome.org/show_bug.cgi?id=756870
2412 2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
2415 Automatic update of common submodule
2416 From 6babecd to b99800a
2418 2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
2421 Update GLib dependency to 2.40.0
2423 2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
2425 * examples/test-mp4.c:
2426 * gst/rtsp-server/rtsp-stream.c:
2427 stream: listen to sender ssrc signals
2428 https://bugzilla.gnome.org/show_bug.cgi?id=746747
2430 2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
2433 common: update for new suppression
2434 Makes check-valgrind pass with glib 2.46
2436 2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2438 * gst/rtsp-server/rtsp-media.c:
2439 rtsp-media: Take reference to media that will be prepared
2440 default_prepare() takes a transfer-none reference GstRTSPMedia object.
2441 Later on a g_idle_source_new() is created and a pointer to the media
2442 object is passed as user data. If the media is freed before the idle
2443 source is dispatched the media object pointer is invalid, but the idle
2444 source callback expects it to still be valid. To fix this a reference to
2445 the media object is taken when registering the source callback function
2446 and a corresponding release of the reference is done when the souce is
2448 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
2450 2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
2452 * examples/test-launch.c:
2453 * examples/test-mp4.c:
2454 * examples/test-ogg.c:
2455 * examples/test-record.c:
2456 * examples/test-uri.c:
2457 rtsp-server: Fix memory leaks when context parse fails
2458 When g_option_context_parse fails, context and error variables are not getting free'd
2459 which results in memory leaks. Free'ing the same.
2460 And replacing g_error_free with g_clear_error, which checks if the error being passed
2461 is not NULL and sets the variable to NULL on free'ing.
2462 https://bugzilla.gnome.org/show_bug.cgi?id=753863
2464 2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
2469 === release 1.6.0 ===
2471 2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
2477 * gst-rtsp-server.doap:
2480 === release 1.5.91 ===
2482 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
2488 * gst-rtsp-server.doap:
2491 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
2493 * docs/libs/gst-rtsp-server-sections.txt:
2494 * gst/rtsp-server/rtsp-stream.c:
2495 stream: fix docs for recently-added get/set_buffer_size API
2496 https://bugzilla.gnome.org/show_bug.cgi?id=749095
2498 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
2500 * gst/rtsp-server/rtsp-media.c:
2501 rtsp-media: Don't crash on encrypted RTX SDP
2502 In parse_keymgmt(), don't mutate the input string that's been passed
2503 as const, especially since we might need the original value again if
2504 the same key info applies to multiple streams (RTX, for example).
2505 https://bugzilla.gnome.org/show_bug.cgi?id=754753
2507 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
2509 * examples/test-mp4.c:
2510 test-mp4: Support filenames with spaces in them. Error out on too few arguments
2512 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
2514 * examples/test-record.c:
2515 test-record: Check parameter count and print out help
2516 If no launch pipeline was supplied, print out some help
2518 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
2520 * gst/rtsp-server/rtsp-media.c:
2521 * gst/rtsp-server/rtsp-stream.c:
2522 * gst/rtsp-server/rtsp-stream.h:
2523 rtsp-stream: Implement UDP buffer size setting.
2524 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
2526 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
2527 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
2529 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
2531 * gst/rtsp-server/rtsp-media.h:
2532 rtsp-media: Fix small typo causing gtk-doc to complain
2534 === release 1.5.90 ===
2536 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
2542 * gst-rtsp-server.doap:
2545 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
2547 * gst/rtsp-server/rtsp-media-factory.c:
2548 media-factory: get port number through gst_rtsp_url_get_port
2549 https://bugzilla.gnome.org/show_bug.cgi?id=753473
2551 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
2553 * tests/check/gst/media.c:
2554 media-test: Removing unnecessary assertion
2555 https://bugzilla.gnome.org/show_bug.cgi?id=753385
2557 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2559 * gst/rtsp-server/rtsp-server.c:
2560 Document that source keeps a ref on server until it's destroyed
2561 https://bugzilla.gnome.org/show_bug.cgi?id=749227
2563 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
2565 * tests/check/gst/media.c:
2566 media-test: Test for multiple dynamic payload
2567 https://bugzilla.gnome.org/show_bug.cgi?id=753385
2569 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
2571 * gst/rtsp-server/rtsp-media.c:
2572 media: Only add fakesink once per pipeline
2573 The intention is to prevent going PLAYING state before pads are created.
2574 If there was mutilple dynamic payload, it would leak few fakesink and
2575 actually prevent from ever reaching playing state.
2576 https://bugzilla.gnome.org/show_bug.cgi?id=753385
2578 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
2580 * gst/rtsp-server/rtsp-media.c:
2581 Revert "rtsp-media: Only add 1 fakesink per pipeline"
2582 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
2584 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
2586 * gst/rtsp-server/rtsp-media.c:
2587 rtsp-media: Only add 1 fakesink per pipeline
2588 There should be only one fakesink per pipeline, not per dynpay. This
2589 would lead to element naming clash.
2591 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
2593 * gst/rtsp-server/rtsp-media.c:
2594 rtsp-media: assertion error due to wrong condition check
2595 In media to caps function, reserved_keys array is being used for variable i,
2596 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
2597 changed it to variable j
2598 https://bugzilla.gnome.org/show_bug.cgi?id=753009
2600 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
2602 * gst/rtsp-server/rtsp-media.c:
2603 rtsp-media: Strip keys from the fmtp that we use internally in our caps
2604 Skip keys from the fmtp, which we already use ourselves for the
2605 caps. Some software is adding random things like clock-rate into
2606 the fmtp, and we would otherwise here set a string-typed clock-rate
2607 in the caps... and thus fail to create valid RTP caps
2608 https://bugzilla.gnome.org/show_bug.cgi?id=753009
2610 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2612 * gst/rtsp-server/rtsp-thread-pool.c:
2613 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
2614 https://bugzilla.gnome.org/show_bug.cgi?id=752640
2616 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
2619 Automatic update of common submodule
2620 From f74b2df to 9aed1d7
2622 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
2627 === release 1.5.2 ===
2629 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
2635 * gst-rtsp-server.doap:
2638 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
2640 * gst/rtsp-server/rtsp-client.c:
2641 * gst/rtsp-server/rtsp-client.h:
2642 * tests/check/gst/client.c:
2643 rtsp-client: allow application to decide what requirements are supported
2644 Add "check-requirements" signal and vfunc to allow application
2645 (and subclasses) to check the requirements.
2646 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
2647 https://bugzilla.gnome.org/show_bug.cgi?id=749417
2649 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
2652 Automatic update of common submodule
2653 From 6015d26 to f74b2df
2655 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
2657 * gst/rtsp-server/rtsp-media.c:
2658 rtsp-media: Always use real payloader when creating streams
2659 A bin that contains the real payloader might be used as payloader. In this
2660 case we have to get the real payloader for the various properties it provides.
2661 Example use cases for this are bins that payload some media and then have
2662 additional elements that add metadata or RTP extension headers to the stream.
2663 https://bugzilla.gnome.org/show_bug.cgi?id=750800
2665 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
2667 * examples/test-netclock-client.c:
2668 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
2670 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
2672 * examples/test-netclock-client.c:
2673 * examples/test-netclock.c:
2674 test-netclock: Use new ntp-time-source property on rtpbin
2675 Select the clock time to be used as NTP time source. This allows proper
2676 synchronization between receivers, independent of sharing base times, and just
2677 requires them to use the same clock.
2679 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
2681 * examples/test-netclock-client.c:
2682 * examples/test-netclock.c:
2683 test-netclock: Setting the same base time on sender and receiver is not necessary
2684 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
2686 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
2688 * gst/rtsp-server/rtsp-stream.c:
2689 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
2690 https://bugzilla.gnome.org/show_bug.cgi?id=750764
2692 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
2694 * docs/libs/gst-rtsp-server.types:
2695 docs: add missing types
2696 https://bugzilla.gnome.org/show_bug.cgi?id=750764
2698 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
2700 * docs/libs/gst-rtsp-server-sections.txt:
2701 docs: add missing apis
2702 https://bugzilla.gnome.org/show_bug.cgi?id=750764
2704 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
2706 * examples/test-netclock-client.c:
2707 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
2709 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2711 * docs/libs/gst-rtsp-server-sections.txt:
2712 * gst/rtsp-server/rtsp-auth.c:
2713 * gst/rtsp-server/rtsp-auth.h:
2714 GstRTSPAuth: Add client certificate authentication support
2715 https://bugzilla.gnome.org/show_bug.cgi?id=750471
2717 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
2719 * examples/test-netclock-client.c:
2720 test-netclock-client: Use new GstClock API to wait for clock synchronization
2722 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
2724 * examples/test-netclock-client.c:
2725 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
2726 A mainloop is needed to get glimagesink to display something on OSX, and
2727 the source-setup signal just makes things a little bit easier.
2729 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
2732 Automatic update of common submodule
2733 From d9a3353 to 6015d26
2735 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
2738 Automatic update of common submodule
2739 From d37af32 to d9a3353
2741 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
2744 Automatic update of common submodule
2745 From 21ba2e5 to d37af32
2747 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
2750 Automatic update of common submodule
2751 From c408583 to 21ba2e5
2753 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
2755 * docs/libs/Makefile.am:
2756 docs: remove variables that we define in the snippet from common
2757 This is syncing our Makefile.am with upstream gtkdoc.
2759 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
2762 Automatic update of common submodule
2763 From 44a3517 to c408583
2765 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
2770 === release 1.5.1 ===
2772 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
2778 * gst-rtsp-server.doap:
2781 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
2783 * gst/rtsp-server/rtsp-client.c:
2784 rtsp-client: No flush during Teardown.
2785 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
2786 backlog is empty it can happen that just a part of a message will be
2787 sent and rest is in backlog queue. If then flush during teardown
2788 just a part of message will be sent.This can lead to client miss
2789 teardown response since it expect to get the last part of message.
2790 The flushing during teardown was introduced to fix a deadlock that now
2791 is fixed more generally in handle_request by temporary setting backlog
2793 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
2795 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
2797 * tests/check/Makefile.am:
2798 tests: Use AM_TESTS_ENVIRONMENT
2799 Needed by the new automake test runner and the
2800 current version of the common submodule.
2802 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
2804 * gst/rtsp-server/rtsp-media.h:
2805 * gst/rtsp-server/rtsp-stream.h:
2806 rtsp-server: Use single-include rtsp header to make sure we get all definitions
2808 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
2810 * gst/rtsp-server/rtsp-media.c:
2811 rtsp-media: Mark some more functions static
2813 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
2815 * gst/rtsp-server/rtsp-media.c:
2816 rtsp-media: Only unblock the media in suspend() when actually changing the state
2817 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
2819 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
2821 * examples/test-video-rtx.c:
2822 examples: Use AVPF profile for the RTX example
2824 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
2826 * gst/rtsp-server/rtsp-sdp.c:
2827 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
2829 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
2831 * gst/rtsp-server/rtsp-stream.c:
2832 rtsp-stream: get valid clock-rate from last-sample
2833 clock-rate in last-sample's caps is integer, not unsigned.
2834 To get this value properly, variable needs to be type-casted to int.
2835 https://bugzilla.gnome.org/show_bug.cgi?id=747614
2837 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
2841 autogen.sh: only run autopoint if gettext requested in configure.ac
2842 Not just because there happens to be a po directory.
2843 https://bugzilla.gnome.org/show_bug.cgi?id=748058
2845 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
2848 Revert "configure.ac: uncomment gettext version setup"
2849 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
2850 We don't need a gettext setup here and there's no po
2851 directory either, so no reason why autopoint would be
2852 run in the first place.
2853 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
2855 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
2857 * examples/test-multicast.c:
2858 * examples/test-multicast2.c:
2859 * examples/test-sdp.c:
2860 * examples/test-video-rtx.c:
2861 * examples/test-video.c:
2862 * tests/test-cleanup.c:
2863 * tests/test-reuse.c:
2864 Fix timeout function signatures across tests and examples
2866 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
2868 * tests/check/Makefile.am:
2869 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
2870 Make sure the test environment is set up.
2871 https://bugzilla.gnome.org//show_bug.cgi?id=747624
2873 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
2876 configure: bump automake requirement to 1.14 and autoconf to 2.69
2877 This is only required for builds from git, people can still
2878 build tarballs if they only have older autotools.
2879 https://bugzilla.gnome.org//show_bug.cgi?id=747624
2881 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
2884 configure.ac: uncomment gettext version setup
2885 Fixes autogen.sh. It would run autopoint, which would complain
2886 that it could not find the gettext version in configure.ac.
2887 https://bugzilla.gnome.org/show_bug.cgi?id=748058
2889 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
2891 * examples/test-video-rtx.c:
2892 test-video-rtx: set exact payload type to PCMA payloader
2893 Setting wrong payload type causes failure to do retransmission through audio stream
2894 https://bugzilla.gnome.org/show_bug.cgi?id=747839
2896 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
2898 * gst/rtsp-server/rtsp-media.c:
2899 * gst/rtsp-server/rtsp-stream.c:
2900 * gst/rtsp-server/rtsp-stream.h:
2901 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
2902 Because of duplicated g_signal_connect for request-aux-sender signal,
2903 wrong stream pointer is passed to the signal handler.
2904 Instead of passing each stream, pass stream array and get the relevant stream.
2905 https://bugzilla.gnome.org/show_bug.cgi?id=747839
2907 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
2911 Update autogen.sh to latest version from common
2912 Fixes build after aclocal_check etc. helpers have been removed.
2914 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
2917 Automatic update of common submodule
2918 From bc76a8b to c8fb372
2920 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
2922 * gst/rtsp-server/rtsp-stream.c:
2923 rtsp-stream: Limit the queues to 1 buffer
2924 We only need them to be able to pre-roll, queueing up more data here
2925 is only going to harm latency and memory usage.
2927 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
2929 * gst/rtsp-server/rtsp-stream.c:
2930 rtsp-stream: Update comment and ASCII art to the latest code
2931 We have a queue in front of the udpsink too to prevent the pipeline from
2934 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
2936 * gst/rtsp-server/rtsp-stream.c:
2937 rtsp-media: Properly return first rtptime
2938 Instead we where returning first GstBuffer timestamp. This would result
2939 in clock skew and unwanted behaviour in RTSP playback.
2940 https://bugzilla.gnome.org/show_bug.cgi?id=746479
2942 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
2944 * gst/rtsp-server/rtsp-stream.c:
2945 rtsp-stream: Don't leave buffer mapped
2946 If the seq is NULL, the RTP buffer was left mapped. We should always
2949 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
2954 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
2956 * gst/rtsp-server/rtsp-media-factory.c:
2957 * tests/check/gst/client.c:
2958 Fix double semicolons
2960 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
2962 * gst/rtsp-server/rtsp-stream.c:
2963 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
2964 This gives more accurate values than asking the payloader. There might be
2965 queueing happening between the payloader and the sink.
2966 https://bugzilla.gnome.org/show_bug.cgi?id=745704
2968 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
2970 * gst/rtsp-server/rtsp-media.c:
2971 rtsp-media: Don't seek for PLAY if the position will not change
2972 https://bugzilla.gnome.org/show_bug.cgi?id=745704
2974 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
2976 * gst/rtsp-server/rtsp-media.c:
2977 rtsp-media: Don't include payload type in the caps for framesize
2978 When the sdp media attribute framesize are converted to caps
2979 the <payload> should not be included.
2980 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
2981 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
2983 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
2985 * gst/rtsp-server/rtsp-sdp.c:
2986 rtsp-sdp: add payload type to the sdp framesize attribute
2987 The sdp framesize attribute is desribed in RFC6064. It is specified
2988 for payloading of H263 and has the following form
2989 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
2990 should be added to the caps in a payloader and the <payload type> should
2991 be added by the rtsp-server.
2992 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
2994 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
2996 * examples/test-uri.c:
2997 examples: test-uri: fix tainted variable
2998 Insignificant but this keeps Coverity happy.
3001 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
3003 * examples/.gitignore:
3004 * examples/Makefile.am:
3005 * examples/test-netclock-client.c:
3006 * examples/test-netclock.c:
3007 examples: Add a simple example of network synch for live streams.
3008 An example server and client that works for synchronising live streams
3009 only - as it can't support pause/play.
3011 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
3013 * gst/rtsp-server/rtsp-media-factory.c:
3014 * gst/rtsp-server/rtsp-media-factory.h:
3015 rtsp-media-factory: Add functions to set/get the media gtype
3016 Allow specifying the GType of a GstRtspMedia subclass to create
3017 as a simpler way to get the factory to create a custom
3018 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
3020 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
3022 * gst/rtsp-server/rtsp-media.c:
3023 rtsp-media: fix double unlock in _get_buffer_size()
3024 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
3025 because of double g_mutex_unlock () usage.
3026 https://bugzilla.gnome.org/show_bug.cgi?id=745434
3028 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
3030 * gst/rtsp-server/rtsp-session-pool.c:
3031 * gst/rtsp-server/rtsp-session.c:
3032 * gst/rtsp-server/rtsp-session.h:
3033 rtsp-session: Use monotonic time for RTSP session timeout
3034 Changed RTSP session timeout handling to monotonic time
3035 and deprecating the API for current system time.
3036 This fixes timeouts when the system time changes.
3037 https://bugzilla.gnome.org/show_bug.cgi?id=743346
3039 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
3041 * gst/rtsp-server/rtsp-client.c:
3042 * gst/rtsp-server/rtsp-media.c:
3043 rtsp-client: Only error out in PLAY if seeking actually failed
3044 If the media was just not seekable, we continue from whatever position we are
3045 and let the client decide if that is what is wanted or not.
3046 Only if the actual seek failed, we can't really recover and should error out.
3048 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
3050 * gst/rtsp-server/rtsp-stream.c:
3051 rtsp-stream: Add necessary queues between tee and multiudpsink
3052 https://bugzilla.gnome.org/show_bug.cgi?id=744379
3054 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
3056 * gst/rtsp-server/rtsp-client.c:
3057 * gst/rtsp-server/rtsp-media.c:
3058 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
3059 Instead error out properly the same way as if the SEEKING query already
3062 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
3064 * gst/rtsp-server/rtsp-stream.h:
3065 rtsp-stream: minor code formatting fix
3067 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
3069 * gst/rtsp-server/rtsp-media.c:
3070 rtsp-media: fix logic for collect_streams
3071 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
3072 all streams it knows if it got any, and can check if the transport mode is OK.
3075 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
3077 * gst/rtsp-server/rtsp-media.c:
3078 rtsp-media: Don't set the transport mode based on what elements we find
3079 Just print a warning if the one that was set before disagrees with what
3080 elements we found. It must already be set to something before as this
3081 function is called after we received the SDP from ANNOUNCE in RECORD mode,
3082 and we would reject ANNOUNCE if the RECORD flag was not set.
3084 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
3086 * tests/check/gst/rtspserver.c:
3087 tests: rtspserver: rename shadowed variable
3088 We have two different 'sink' variables here,
3089 rename one of them for clarity.
3091 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
3093 * gst/rtsp-server/rtsp-client.c:
3094 rtsp-client: fix awkward if clause
3096 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
3098 * examples/test-uri.c:
3099 examples: test-uri: improve uri argument handling and accept file names
3100 Print an error if the argument passed is not a URI and can't
3101 be converted into one, or no arguments have been provided.
3103 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
3105 * examples/test-uri.c:
3106 examples: test-uri: don't remove mount point after 10 seconds
3107 It's very irritating when trying to test stuff repeatedly
3108 and serves no real purpose other than showing that it can
3111 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
3113 * examples/.gitignore:
3114 examples: add new test-record to .gitignore
3116 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
3118 * examples/test-record.c:
3119 * gst/rtsp-server/rtsp-client.c:
3120 * gst/rtsp-server/rtsp-media-factory.c:
3121 * gst/rtsp-server/rtsp-media-factory.h:
3122 * gst/rtsp-server/rtsp-media.c:
3123 * gst/rtsp-server/rtsp-media.h:
3124 * tests/check/gst/rtspserver.c:
3125 rtsp-media: Use flags to distinguish between PLAY and RECORD media
3127 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
3129 * examples/test-record.c:
3130 test-record: Set latency for playback-style example to 2s instead of 200ms
3132 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
3134 * tests/check/gst/rtspserver.c:
3135 tests: add some unit tests for ANNOUNCE and RECORD
3136 https://bugzilla.gnome.org/show_bug.cgi?id=743175
3138 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
3140 * gst/rtsp-server/rtsp-client.c:
3141 rtsp-client: fix a couple of leaks in handle_announce
3143 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
3145 * gst/rtsp-server/rtsp-media-factory.c:
3146 * gst/rtsp-server/rtsp-media-factory.h:
3147 * gst/rtsp-server/rtsp-media.c:
3148 * gst/rtsp-server/rtsp-media.h:
3149 rtsp-media: Expose latency setting for setting the rtpbin latency
3151 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
3153 * examples/test-record.c:
3154 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
3156 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
3158 * gst/rtsp-server/rtsp-stream.c:
3159 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
3161 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
3163 * examples/Makefile.am:
3164 * examples/test-record.c:
3165 * gst/rtsp-server/rtsp-client.c:
3166 * gst/rtsp-server/rtsp-client.h:
3167 * gst/rtsp-server/rtsp-media-factory.c:
3168 * gst/rtsp-server/rtsp-media-factory.h:
3169 * gst/rtsp-server/rtsp-media.c:
3170 * gst/rtsp-server/rtsp-media.h:
3171 * gst/rtsp-server/rtsp-session-media.c:
3172 * gst/rtsp-server/rtsp-stream.c:
3173 * gst/rtsp-server/rtsp-stream.h:
3174 Add initial support for RECORD
3175 We currently only support media that is RECORD or PLAY only, not both at once.
3176 https://bugzilla.gnome.org/show_bug.cgi?id=743175
3178 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
3180 * gst/rtsp-server/rtsp-stream.c:
3181 rtsp-stream: RTCP and RTP transport cache cookies seperated
3182 RTCP packets were not sent because the same tr_cache_cookie was used for
3183 both RTP and RTCP. So only one of the tr_cache lists were populated
3184 depending on which one was sent first. If the tr_cache list is not
3185 populated then no packets can be sent. Most often this happened to be
3186 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
3187 resulted in both the tr_cache_lists to be populated regardless of which
3189 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
3191 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
3193 * gst/rtsp-server/rtsp-stream.c:
3194 rtsp-stream: fix false compiler warning
3195 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
3197 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
3199 * gst/rtsp-server/rtsp-client.c:
3200 rtsp-client: log interleaved data received
3202 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
3204 * gst/rtsp-server/rtsp-client.c:
3205 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
3207 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
3209 * gst/rtsp-server/rtsp-client.c:
3210 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
3212 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
3214 * gst/rtsp-server/rtsp-client.c:
3215 rtsp-client: Use a random session ID in the SDP
3216 RFC4566 Section 5.2 says that it should make the username, session id,
3217 nettype, addrtype and unicast address tuple globally unique. Always using
3218 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
3219 Instead let's create a 64 bit random number, which at least brings us
3220 closer to the goal of global uniqueness.
3221 https://tools.ietf.org/html/rfc4566#section-5.2
3223 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
3225 * examples/test-launch.c:
3226 * examples/test-mp4.c:
3227 * examples/test-ogg.c:
3228 * examples/test-uri.c:
3229 examples: Don't call gst_init() and gst_get_option_group()
3230 The latter calls the former at the appropriate time.
3232 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
3234 * gst/rtsp-server/rtsp-client.c:
3235 rtsp-client: Drop trailing \0 of RTSP DATA messages
3236 We add a trailing \0 in GstRTSPConnection to make parsing of
3237 string message bodies easier (e.g. the SDP from DESCRIBE) but
3238 for actual data this means we have to drop it or otherwise
3239 create invalid data.
3241 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
3243 * gst/rtsp-server/rtsp-stream.c:
3244 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
3245 Fixes crash when two threads access handle_new_sample() at the same
3246 time, one for RTP, one for RTCP.
3247 Otherwise, when iterating over the transports cache, it might be modified by
3248 another thread at the same time if the transports cookie has changed.
3249 https://bugzilla.gnome.org/show_bug.cgi?id=742954
3251 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
3253 * gst/rtsp-server/rtsp-stream.c:
3254 rtsp-stream: Set format=TIME on our app sources for TCP
3256 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
3258 * gst/rtsp-server/rtsp-session-pool.c:
3259 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
3260 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
3261 RFC 2326 states that session IDs may consist of alphanumeric as well as
3262 the safe characters $-_.+ -- N.B. the percent character is not allowed.
3263 Previously the session ID was URI-escaped, this meant that any character
3264 which was not alphanumeric or any of the characters +-._~ would be
3265 percent encoded. While the RFC (surprisingly) mentions that linear white
3266 space in session IDs should be URI-escaped, it does not say anything
3267 about other characters. Moreover no white space is allowed in the
3268 session ID. Finally the percent character which is the result of
3269 URI-escaping is not allowed in a session ID.
3270 So there is no reason to do any URI-escaping, and now it is removed.
3271 https://bugzilla.gnome.org/show_bug.cgi?id=742869
3273 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
3276 Automatic update of common submodule
3277 From f2c6b95 to bc76a8b
3279 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
3282 Fix 'make check' from top-level directory
3284 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
3286 * examples/test-launch.c:
3287 * examples/test-mp4.c:
3288 * examples/test-ogg.c:
3289 * examples/test-uri.c:
3290 examples: Add command-line parsing and take a 'port' argument
3291 This allows users to run multiple servers on different ports for testing.
3292 Only done for examples that actually take arguments and hence are capable of
3293 outputting different streams for each instance on each port.
3294 https://bugzilla.gnome.org/show_bug.cgi?id=742115
3296 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
3298 * gst/rtsp-server/rtsp-client.c:
3299 * gst/rtsp-server/rtsp-client.h:
3300 rtsp-client: Add a send_message default signal handler
3301 This allows subclasses to easily hook into the response sending
3302 mechanism without doing everything from a signal, which seems
3303 awkward from subclasses.
3305 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
3308 Automatic update of common submodule
3309 From ef1ffdc to f2c6b95
3311 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3315 configure: add --disable-examples switch
3316 https://bugzilla.gnome.org/show_bug.cgi?id=741678
3318 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
3320 * examples/.gitignore:
3321 * examples/Makefile.am:
3322 * examples/test-video-rtx.c:
3323 examples: add a retransmisison example implementing RFC4588
3324 Currently only SSRC-multiplexed rtx streams are supported
3326 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
3328 * gst/rtsp-server/rtsp-stream.c:
3329 rtsp-stream: Fix some minor memory leaks
3331 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
3333 * gst/rtsp-server/rtsp-media.c:
3334 rtsp-media: Some minor cleanup
3336 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
3338 * gst/rtsp-server/rtsp-stream.c:
3339 rtsp-stream: Fix compiler warnings
3340 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
3341 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
3343 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
3344 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
3347 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
3349 * docs/libs/gst-rtsp-server-sections.txt:
3350 * gst/rtsp-server/rtsp-media-factory.c:
3351 * gst/rtsp-server/rtsp-media-factory.h:
3352 * gst/rtsp-server/rtsp-media.c:
3353 * gst/rtsp-server/rtsp-media.h:
3354 * gst/rtsp-server/rtsp-sdp.c:
3355 * gst/rtsp-server/rtsp-stream.c:
3356 * gst/rtsp-server/rtsp-stream.h:
3357 media: implement ssrc-multiplexed retransmission support
3358 based off RFC 4588 and the server-rtpaux example in -good
3360 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
3362 * gst/rtsp-server/rtsp-client.c:
3363 * gst/rtsp-server/rtsp-stream-transport.c:
3364 * gst/rtsp-server/rtsp-stream.c:
3365 rtsp: Ref transports in hash table.
3366 Also ref streams for transports.
3367 This solves a crash when reciving a rtcp after teardown but before
3369 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
3371 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
3374 Automatic update of common submodule
3375 From 7bb2bce to ef1ffdc
3377 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
3379 * gst/rtsp-server/rtsp-client.c:
3380 client: refactor cleanup of cached media
3382 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
3384 * tests/check/gst/client.c:
3386 The session leak is now fixed, lets remove those FIXME comments.
3388 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
3390 * tests/check/gst/rtspserver.c:
3391 tests: Test to setup two sessions on one connection
3392 https://bugzilla.gnome.org/show_bug.cgi?id=739112
3394 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
3396 * tests/check/gst/rtspserver.c:
3397 tests: Test setup with tcp transport
3398 https://bugzilla.gnome.org/show_bug.cgi?id=739112
3400 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
3402 * gst/rtsp-server/rtsp-client.c:
3403 client: Configure transport after creating session media
3404 The default implementation of configure_client_transport() in
3405 rtsp-client uses the session media when it chooses channels for
3406 interleaved traffic.
3407 https://bugzilla.gnome.org/show_bug.cgi?id=739112
3409 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
3411 * gst/rtsp-server/rtsp-client.c:
3412 * gst/rtsp-server/rtsp-session-media.c:
3413 client: Stop caching media in client when doing setup
3414 If the media has been managed by a session media, it should not be
3415 cached in the client any longer. The GstRTSPSessionMedia object is now
3416 responsible for unpreparing the GstRTSPMedia object using
3417 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
3419 https://bugzilla.gnome.org/show_bug.cgi?id=739112
3421 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
3423 * gst/rtsp-server/rtsp-stream.c:
3424 rtsp-stream: unref srtp decoder when leaving bin
3425 https://bugzilla.gnome.org/show_bug.cgi?id=739481
3427 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
3429 * gst/rtsp-server/rtsp-client.c:
3430 rtsp-client: mikey memory leaks
3431 https://bugzilla.gnome.org/show_bug.cgi?id=739383
3433 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
3436 Automatic update of common submodule
3437 From 84d06cd to 7bb2bce
3439 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
3442 Parallelise 'make check-valgrind'
3444 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
3447 Automatic update of common submodule
3448 From a8c8939 to 84d06cd
3450 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
3453 Automatic update of common submodule
3454 From 36388a1 to a8c8939
3456 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
3458 * gst/rtsp-server/rtsp-media.c:
3459 rtsp-media: deactivate media when shutting down from paused
3460 This was only done when going directly from playing.
3461 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
3463 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
3465 * gst/rtsp-server/rtsp-client.c:
3466 * gst/rtsp-server/rtsp-context.h:
3467 rtsp-client: add stream transport to context
3468 We add the stream transport to the context so we can get the configured
3469 client stream transport in the setup request signal.
3470 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
3472 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
3474 * gst/rtsp-server/rtsp-stream.c:
3475 stream: release lock even not all transports have been removed
3476 We don't want to keep the lock even we return FALSE because not all the
3477 transports have been removed. This could lead into a deadlock.
3478 https://bugzilla.gnome.org/show_bug.cgi?id=737797
3480 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
3482 * gst/rtsp-server/rtsp-sdp.c:
3483 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
3484 These were renamed in GstRTPBasePayload in 1.0
3486 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
3488 * gst/rtsp-server/rtsp-client.c:
3489 client: set session media to NULL without the lock
3490 We need to set session medias to NULL without the client lock otherwise
3491 we can end up in a deadlock if another thread is waiting for the lock
3492 and media unprepare is also waiting for that thread to end.
3493 https://bugzilla.gnome.org/show_bug.cgi?id=737690
3495 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
3497 * gst/rtsp-server/rtsp-media.c:
3498 rtsp-media: Set state to UNPREPARING in all cases
3500 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
3502 * gst/rtsp-server/rtsp-media.c:
3503 media: set state to unpreparing when unprepare is initiated
3504 https://bugzilla.gnome.org/show_bug.cgi?id=737675
3506 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
3508 * gst/rtsp-server/rtsp-client.c:
3509 rtsp-client: Remove backlog limit while processings requests
3510 If the backlog limit is kept two cases of deadlocks may be
3511 encountered when streaming over TCP. Without the backlog
3512 limit this deadlocks can not happen, at the expence of
3514 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
3516 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
3518 * gst/rtsp-server/rtsp-client.c:
3519 rtsp-client: do not free main context before rtsp watch
3520 https://bugzilla.gnome.org/show_bug.cgi?id=737110
3522 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
3524 * tests/check/gst/rtspserver.c:
3525 tests: Extend unit test timeout to accomodate for valgrind
3526 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
3528 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
3530 * gst/rtsp-server/rtsp-client.c:
3531 * gst/rtsp-server/rtsp-session.c:
3532 * gst/rtsp-server/rtsp-stream-transport.c:
3533 rtsp-*: Treat sending packets to clients as keepalive
3534 As long as gst-rtsp-server can successfully send RTP/RTCP data to
3535 clients then the client must be reading. This change makes the server
3536 timeout the connection if the client stops reading.
3537 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
3539 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
3541 * gst/rtsp-server/rtsp-client.c:
3542 rtsp-client: Allow backlog to grow while expiring session
3543 Allow the send backlog in the RTSP watch to grow to unlimited size while
3544 attempting to bring the media pipeline to NULL due to a session
3545 expiring. Without this change the appsink element cannot change state
3546 because it is blocked while rendering data in the new_sample callback.
3547 This callback will block until it has successfully put the data into the
3548 send backlog. There is a chance that the send backlog is full at this
3549 point which means that the callback may block for a long time, possibly
3550 forever. Therefore the media pipeline may also be prevented from
3551 changing state for a long time.
3552 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
3554 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
3556 * gst/rtsp-server/rtsp-client.c:
3557 rtsp-client: Make old compilers happy
3558 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
3559 Just in case that guint8 doesn't fit in a pointer. Just in case ...
3561 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
3563 * gst/rtsp-server/rtsp-client.c:
3564 client: raise the backlog limits before pausing
3565 We need to raise the backlog limits before pausing the pipeline or else
3566 the appsink might be blocking in the render method in wait_backlog() and
3567 we would deadlock waiting for paused.
3568 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
3570 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
3572 * gst/rtsp-server/rtsp-client.c:
3573 client: make define for the WATCH_BACKLOG
3574 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
3576 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
3578 * gst/rtsp-server/rtsp-client.c:
3579 client: simplify session transport handling
3580 link/unlink of the transport in a session was done to keep track of all
3581 TCP transports and to send RTP/RTCP data to the streams. We can simplify
3582 that by putting all the TCP transports in a hashtable indexed with the
3584 We also don't need to link/unlink the transports when we pause/resume
3585 the streams. The same effect is already achieved when we pause/play the
3586 media. Indeed, when we pause the media, the transport is removed from
3587 the media and the callbacks will not be called anymore.
3588 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
3590 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
3592 * gst/rtsp-server/rtsp-stream-transport.c:
3593 * gst/rtsp-server/rtsp-stream-transport.h:
3594 stream-transport: make method to handle received data
3595 Make a method to handle the data received on a channel. It sends the
3596 data to the stream of the transport on the RTP or RTCP pads based on
3599 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
3601 * examples/test-mp4.c:
3602 test: add example of dumping RTCP reports
3604 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
3606 * gst/rtsp-server/rtsp-media.c:
3607 * gst/rtsp-server/rtsp-stream.c:
3608 * gst/rtsp-server/rtsp-stream.h:
3609 rtsp-media: Make sure that sequence numbers are monotonic after pause
3610 The sequence number is not monotonic for RTP packets after pause. The
3611 reason is basepayloader generates a randon sequence number when the
3612 pipeline goes from ready to pause. With this fix generation of sequence
3613 number will be monotonic when going from pause to play request.
3614 https://bugzilla.gnome.org/show_bug.cgi?id=736017
3616 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
3618 * gst/rtsp-server/rtsp-client.c:
3619 rtsp-client: Protect saved clients watch with a mutex
3620 Fixes a crash when close() is called while merging clients
3621 in handle_tunnel(). In that case close() would destroy the
3622 watch while it is still being used in handle_tunnel().
3623 https://bugzilla.gnome.org/show_bug.cgi?id=735570
3625 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
3627 * gst/rtsp-server/rtsp-stream.c:
3628 rtsp-stream: Remove the multicast group udp sources when removing from the bin
3630 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
3632 * gst/rtsp-server/rtsp-media.c:
3633 * gst/rtsp-server/rtsp-stream.c:
3634 * gst/rtsp-server/rtsp-stream.h:
3635 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
3636 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
3637 seeking and will always continue counting the time. This leads to
3638 the NPT after a backwards seek to be something completely different
3639 to the actual seek position.
3640 https://bugzilla.gnome.org/show_bug.cgi?id=732644
3642 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
3644 * examples/test-appsrc.c:
3645 examples: fix another reference leak
3646 gst_rtsp_media_get_element() returns a new ref.
3648 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3650 * examples/test-appsrc.c:
3651 examples: unref element after usage
3652 gst_bin_get_by_name_recurse_up() returns an element
3653 reference that must be unreffed after usage.
3654 https://bugzilla.gnome.org/show_bug.cgi?id=734546
3656 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
3658 * gst/rtsp-server/rtsp-media.c:
3659 signals: Fix copy-pasto in target-state signal offset
3661 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
3665 Makefile: Add usage of build-checks step
3666 Allows building checks without running them
3668 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
3670 * gst/rtsp-server/rtsp-stream.c:
3671 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
3672 When a UDP multicast transport is used it is expected that the server listens
3673 for RTP and RTCP packets on the multicast group with the corresponding port.
3674 Without this we will never get RTCP packets from clients in multicast mode.
3675 https://bugzilla.gnome.org/show_bug.cgi?id=732238
3677 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
3682 === release 1.4.0 ===
3684 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
3690 * gst-rtsp-server.doap:
3693 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
3695 * gst/rtsp-server/rtsp-media.h:
3696 media: correct misspelled words in description
3697 https://bugzilla.gnome.org/show_bug.cgi?id=733244
3699 === release 1.3.91 ===
3701 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
3707 * gst-rtsp-server.doap:
3710 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
3712 * docs/libs/gst-rtsp-server-sections.txt:
3715 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
3717 * gst/rtsp-server/rtsp-server.c:
3718 server: implement client REMOVE filter
3720 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
3722 * gst/rtsp-server/rtsp-client.c:
3723 * gst/rtsp-server/rtsp-client.h:
3724 client: expose _close() method
3725 Expose a previously internal close method to close the client
3728 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
3730 * gst/rtsp-server/rtsp-session-pool.c:
3731 session-pool: signal session-removed outside of the lock
3732 Release the lock before emiting the session-removed signal.
3734 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
3736 * gst/rtsp-server/rtsp-client.c:
3737 * gst/rtsp-server/rtsp-server.c:
3738 * gst/rtsp-server/rtsp-session-pool.c:
3739 * gst/rtsp-server/rtsp-session.c:
3740 * gst/rtsp-server/rtsp-stream.c:
3741 filter: Release lock in filter functions
3742 Release the object lock before calling the filter functions. We need to
3743 keep a cookie to detect when the list changed during the filter
3744 callback. We also keep a hashtable to make sure we only call the filter
3745 function once for each object in case of concurrent modification.
3746 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
3748 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
3750 * gst/rtsp-server/rtsp-client.c:
3751 client: check if watch is set in handle_teardown()
3752 The unit tests run without a watch
3754 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
3756 * tests/check/gst/client.c:
3757 client tests: send teardown to cleanup session
3759 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
3761 * tests/check/gst/rtspserver.c:
3762 server tests: send teardown to cleanup session
3764 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
3766 * gst/rtsp-server/rtsp-client.c:
3767 client: keep ref to client for the session removed handler
3768 This extra ref will be dropped when all client sessions have been
3769 removed. A session is removed when a client sends teardown, closes its
3770 endpoint of the TCP connection or the sessions expires.
3771 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
3773 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
3775 * gst/rtsp-server/rtsp-client.c:
3776 * gst/rtsp-server/rtsp-session.c:
3777 * tests/check/gst/client.c:
3778 client: manage media in session as a last step
3779 Once we manage a media in a session, we can't unmanage it anymore
3780 without destroying it. Therefore, first check everything before we
3781 manage the media, otherwise if something is wrong we have no way to
3783 If we created a new session and something went wrong, remove the session
3784 again. Fixes a leak in the unit test.
3786 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
3788 * examples/test-mp4.c:
3789 * examples/test-ogg.c:
3790 examples: print 'stream ready at url' for mp4 and ogg example
3792 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
3794 * gst/rtsp-server/rtsp-client.c:
3795 * gst/rtsp-server/rtsp-sdp.c:
3796 rtsp: fix for MIKEY api change
3798 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
3800 * gst/rtsp-server/rtsp-client.c:
3801 client: free watch context only once
3802 The watch context is freed when the source is destroyed. Avoids
3803 a CRITICAL when we try to unref the context twice.
3805 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
3807 * gst/rtsp-server/rtsp-client.c:
3810 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
3812 * gst/rtsp-server/rtsp-client.c:
3813 client: protect sessions with lock
3814 Protect the list of sessions with the lock.
3815 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
3817 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
3819 * gst/rtsp-server/rtsp-client.c:
3820 Client: keep a ref to the session
3821 Don't just keep a weak ref to the session objects but use a hard ref. We
3822 will be notified when a session is removed from the pool (expired) with
3823 the new session-removed signal.
3824 Don't automatically close the RTSP connection when all the sessions of
3825 a client are removed, a client can continue to operate and it can create
3826 a new session if it wants. If you want to remove the client from the
3827 server, you have to use gst_rtsp_server_client_filter() now.
3828 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
3829 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
3831 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
3833 * gst/rtsp-server/rtsp-session-pool.c:
3834 * gst/rtsp-server/rtsp-session-pool.h:
3835 session-pool: add session-removed signal
3836 Add a signal to be notified when a session is removed from the pool.
3838 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
3840 * gst/rtsp-server/Makefile.am:
3841 * gst/rtsp-server/rtsp-server.h:
3842 Make rtsp-server.h a single-include header, use it for G-I
3843 https://bugzilla.gnome.org/show_bug.cgi?id=732411
3845 === release 1.3.90 ===
3847 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
3853 * gst-rtsp-server.doap:
3856 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
3858 * gst/rtsp-server/rtsp-stream.c:
3859 stream: crypto can be NULL
3861 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
3863 * gst/rtsp-server/rtsp-client.c:
3864 * gst/rtsp-server/rtsp-media.c:
3865 * gst/rtsp-server/rtsp-mount-points.c:
3866 introspection: add missing allow-none annotations
3867 https://bugzilla.gnome.org/show_bug.cgi?id=730952
3869 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
3871 * gst/rtsp-server/rtsp-address-pool.c:
3872 * gst/rtsp-server/rtsp-media.c:
3873 * gst/rtsp-server/rtsp-session-media.c:
3874 * gst/rtsp-server/rtsp-session-pool.c:
3875 * gst/rtsp-server/rtsp-stream-transport.c:
3876 * gst/rtsp-server/rtsp-stream.c:
3877 * gst/rtsp-server/rtsp-token.c:
3878 introspection: add (nullable) annotations to return values
3879 https://bugzilla.gnome.org/show_bug.cgi?id=730952
3881 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
3883 * gst/rtsp-server/rtsp-client.c:
3884 * gst/rtsp-server/rtsp-stream.c:
3885 gi: improve annotations
3886 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
3888 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
3890 * gst/rtsp-server/rtsp-client.c:
3891 * gst/rtsp-server/rtsp-media-factory.c:
3892 * gst/rtsp-server/rtsp-media.c:
3893 * gst/rtsp-server/rtsp-server.c:
3894 signals: use generic marshal function
3895 Use the generic C marshal function.
3896 Use more explicit type instead of G_TYPE_POINTER
3898 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
3900 * gst/rtsp-server/rtsp-context.h:
3901 context: add type macro
3903 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
3905 * gst/rtsp-server/rtsp-client.c:
3906 * gst/rtsp-server/rtsp-sdp.c:
3907 * gst/rtsp-server/rtsp-sdp.h:
3908 sdp: hide key length defines
3909 They don't have a namespace.
3911 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
3916 === release 1.3.3 ===
3918 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
3924 * gst-rtsp-server.doap:
3927 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
3929 * gst/rtsp-server/rtsp-client.c:
3930 * gst/rtsp-server/rtsp-sdp.c:
3931 * gst/rtsp-server/rtsp-sdp.h:
3932 mikey: add different key length parameters
3933 Add encryption and authentication key length parameters to MIKEY. For
3934 the encoders, the key lengths are obtained from the cipher and auth
3935 algorithms set in the caps. For the decoders, they are obtained while
3936 parsing the key management from the client.
3937 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
3939 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
3941 * tests/check/gst/stream.c:
3942 stream tests: Make sure we get right multicast address from stream
3943 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
3945 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
3947 * gst/rtsp-server/rtsp-client.c:
3948 client: ref the context until rtsp watch is alive
3949 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
3951 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
3953 * gst/rtsp-server/rtsp-client.c:
3954 client: Destroy the rtsp watch after connection close
3956 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
3958 * gst/rtsp-server/rtsp-media.c:
3959 media: fix confusing comment
3961 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
3963 * gst/rtsp-server/rtsp-session.c:
3964 rtsp-session: Timeout in header.
3965 Adding the possbilty to always have timout in header.
3966 This is configurabe with setting "timeout-always-visible".
3967 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
3969 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
3974 === release 1.3.2 ===
3976 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
3983 * gst-rtsp-server.doap:
3986 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
3989 Automatic update of common submodule
3990 From 211fa5f to 1f5d3c3
3992 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
3994 * gst/rtsp-server/rtsp-client.c:
3995 client: store TCP ports in transport
3996 Store the TCP ports in the transport when we are doing RTSP over TCP.
3997 This way, we can easily get to the ports from the transport.
3998 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
4000 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4002 * gst/rtsp-server/rtsp-stream.c:
4003 stream: add signals for new RTP/RTCP encoders
4004 New signals to allow the user to configure the dynamically created
4006 https://bugzilla.gnome.org/show_bug.cgi?id=730228
4008 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
4010 * gst/rtsp-server/rtsp-media.c:
4011 * gst/rtsp-server/rtsp-media.h:
4012 media: Make suspend()/unsuspend() virtual
4013 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
4015 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4017 * gst/rtsp-server/rtsp-client.c:
4018 client: fix send-message signal marshaller
4019 Use generic marshalling for the send-message signal. It has
4020 two POINTER arguments, not just one.
4021 https://bugzilla.gnome.org/show_bug.cgi?id=729900
4023 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
4025 * tests/check/gst/media.c:
4026 tests: add and remove pads only once
4027 In this test we simulate a dynamic pad by watching the caps event.
4028 Because of renegotiation in the base payloader now, this caps is sent
4029 multiple times but we can only deal with 1 invocation, use a variable to
4030 only 'add and remove' the pad once.
4032 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
4034 * tests/check/gst/rtspserver.c:
4035 tests: add unit test for correct handling of Require headers
4036 https://bugzilla.gnome.org/show_bug.cgi?id=729426
4038 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
4040 * gst/rtsp-server/rtsp-client.c:
4041 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
4042 Servers must handle Require headers and must report a failure
4043 if they don't handle any of the Required options, see RFC 2326,
4044 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
4045 https://bugzilla.gnome.org/show_bug.cgi?id=729426
4047 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
4052 === release 1.3.1 ===
4054 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
4060 * gst-rtsp-server.doap:
4063 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
4066 Automatic update of common submodule
4067 From bcb1518 to 211fa5f
4069 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
4074 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
4076 * tests/check/gst/sessionmedia.c:
4077 tests: fix memory leak in sessionmedia unit test
4079 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
4081 * gst/rtsp-server/rtsp-client.c:
4082 client: emit a signal before sending a message
4083 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
4085 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
4087 * gst/rtsp-server/rtsp-client.c:
4088 client: pass context to send_message
4089 Pass the current context to send_message, we will need it later.
4091 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
4093 * gst/rtsp-server/rtsp-client.c:
4094 client: fix typo in comment
4096 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
4098 * gst/rtsp-server/rtsp-media.c:
4099 media: Do not stop thread twice if default_prepare() fails
4101 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
4103 * gst/rtsp-server/rtsp-client.c:
4104 client: set the watch to flushing before going to NULL
4105 First set the watch to flushing so that we unblock any current and
4106 future attempt to send data on the watch, Then set the pipeline to
4108 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
4110 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
4112 * gst/rtsp-server/rtsp-session-pool.c:
4113 * tests/check/gst/sessionpool.c:
4114 rtsp-session-pool: Fixes annotation
4115 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
4116 in the sessionpool test.
4117 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
4119 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
4121 * gst/rtsp-server/rtsp-media.c:
4122 * gst/rtsp-server/rtsp-media.h:
4123 media: make media_prepare virtual
4124 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
4126 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
4128 * gst/rtsp-server/rtsp-media.c:
4129 * tests/check/gst/media.c:
4130 media: stop the thread in more error cases
4132 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
4134 * gst/rtsp-server/rtsp-media.c:
4135 * tests/check/gst/media.c:
4136 media: allow NULL as the thread
4137 Use the default context whan passing a NULL thread.
4139 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
4141 * gst/rtsp-server/rtsp-client.c:
4142 rtsp-client: indent cleanup
4143 Coverity was moaning about unreachable code, and I think it was just
4144 confused by { being before the label. We'll see if it pops up again.
4147 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
4149 * gst/rtsp-server/rtsp-client.c:
4150 * gst/rtsp-server/rtsp-media.c:
4151 client: Add drop-backlog property
4152 When we have too many messages queued for a client (currently hardcoded
4153 to 100) we overflow and drop the messages. Add a drop-backlog property
4154 to control this behaviour. Setting this property to FALSE will retry
4155 to send the messages to the client by waiting for more room in the
4157 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
4159 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
4161 * gst/rtsp-server/rtsp-client.c:
4162 client: support for POST before GET when setting up a tunnel
4164 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
4166 * gst/rtsp-server/rtsp-client.c:
4167 client: remove watch of the second client after http tunnel setup
4168 The second client will be freed after the HTTP tunnel has been set up.
4169 Make sure it's RTSP watch is never dispatched again.
4170 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
4172 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
4174 * gst/rtsp-server/rtsp-media.c:
4175 * tests/check/gst/media.c:
4176 media: Make media_prepare() fail if port allocation fails
4177 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
4179 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
4181 * tests/check/gst/media.c:
4182 media test: cleanup the thread pool in tests
4184 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
4186 * gst/rtsp-server/rtsp-media.c:
4187 * tests/check/gst/media.c:
4188 rtsp-media: Unblock blocked streams in unprepare
4189 The streams will be blocked when a live media is prepared.
4190 The streams should be unblocked in gst_rtsp_media_unprepare.
4191 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
4193 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
4195 * gst/rtsp-server/rtsp-media.c:
4196 media: release the state lock when going to NULL
4197 Set our state to UNPREPARING and release the state-lock before
4198 setting the pipeline to the NULL state. This way, any pad-added
4199 callback will be able to take the state-lock and check that we are now
4200 unpreparing instead of deadlocking.
4201 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
4203 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
4205 * gst/rtsp-server/rtsp-media.c:
4206 media: protect status with lock
4207 Make sure we only update the status with the lock.
4209 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
4211 * gst/rtsp-server/rtsp-client.c:
4212 * gst/rtsp-server/rtsp-sdp.c:
4213 rtsp: update for MIKEY API changes
4215 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
4217 * gst/rtsp-server/rtsp-client.c:
4218 client: parse the mikey response from the client
4219 Parse the mikey response from the client and update the policy for
4222 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
4224 * gst/rtsp-server/rtsp-stream.c:
4225 * gst/rtsp-server/rtsp-stream.h:
4226 stream: add method to set crypto info
4227 Make a method to configure the crypto information of a stream.
4228 Set udpsrc in READY instead of PAUSED so that we can configure caps
4231 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
4233 * gst/rtsp-server/rtsp-client.c:
4234 client: cleanup error paths
4236 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
4238 * gst/rtsp-server/rtsp-media.c:
4241 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
4243 * examples/test-video.c:
4244 test: enable SRTP only on RTSPS
4245 We only want to enable SRTP when doing rtsp over TLS so that we can
4246 exchange the keys in a secure way.
4248 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
4250 * examples/test-video.c:
4251 test: print an error on failure
4253 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
4256 * examples/test-video.c:
4257 * gst/rtsp-server/rtsp-sdp.c:
4258 * gst/rtsp-server/rtsp-stream.c:
4259 * tests/check/Makefile.am:
4260 stream: add SRTP support
4261 Install srtp encoder and decoder elements in rtpbin
4264 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4266 * tests/check/Makefile.am:
4267 * tests/check/gst/sessionpool.c:
4268 tests: Add unit tests for sessionpool
4269 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
4271 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4273 * tests/check/gst/threadpool.c:
4274 tests: Improve code coverage of rtsp-threadpool tests
4275 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
4277 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4279 * tests/check/gst/sessionmedia.c:
4280 tests: Improve code coverage for rtsp-session-media
4281 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
4283 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4285 gobject-introspection: Add annotations to support language bindings
4286 In addition a few cosmetic changes:
4287 * Adjust the order of arguments
4288 * Fix typo: occured -> occurred
4289 * Fix indentation after Return:-clauses
4290 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
4292 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4294 * gst/rtsp-server/rtsp-stream.c:
4295 rtsp-stream: Don't mix IPv4 and IPv6 addresses
4296 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
4298 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
4300 * gst/rtsp-server/rtsp-stream.c:
4301 stream: take caps after the session manager
4302 Take the caps for the SDP after they leave the rtpbin so that we can
4303 also get the properties added by rtpbin elements.
4305 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
4307 * gst/rtsp-server/rtsp-stream.c:
4308 stream: release lock while pushing out packets
4309 Keep a cache of the transports and use this to iterate the transport
4310 while pushing packets. This allows us to release the lock early.
4311 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
4313 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
4315 * gst/rtsp-server/rtsp-client.c:
4316 * gst/rtsp-server/rtsp-client.h:
4317 rtsp-client: vmethod for modifying tunnel GET response
4318 Add a vmethod tunnel_http_response where the response to the HTTP GET
4319 for tunneled connections can be modified.
4320 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
4322 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
4324 * gst/rtsp-server/rtsp-sdp.c:
4325 sdp: make 1 media line per profile
4326 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
4327 line in the SDP for each profile. The client is then supposed to pick
4328 one of the profiles in the SETUP request. Because the m= lines have the
4329 same pt, the client also knows that only 1 option is possible.
4331 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
4333 * gst/rtsp-server/rtsp-media-factory.c:
4334 * gst/rtsp-server/rtsp-media-factory.h:
4335 * gst/rtsp-server/rtsp-media.c:
4336 factory: add profile property and pass to media and streams
4338 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
4340 * examples/test-multicast.c:
4341 * gst/rtsp-server/rtsp-sdp.c:
4342 sdp: pass multicast connection for multicast-only stream
4343 Pass the multicast address of the stream in the connection info in the
4344 SDP so that clients try a multicast connection first.
4345 Only allow multicast connections in the test-multicast example. Also
4346 increase the TTL a little.
4348 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4351 .gitignore: Ignore gcov intermediate files
4352 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
4354 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
4356 * gst/rtsp-server/rtsp-stream.c:
4357 stream: release some locks in error cases
4359 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4361 docs: Enable and fix gtk-doc warnings
4362 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
4363 * addresspool/mediafactory: Add missing annotation colon
4364 * stream: Annotate return value
4365 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
4367 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
4370 Automatic update of common submodule
4371 From fe1672e to bcb1518
4373 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
4376 Automatic update of common submodule
4377 From 1a07da9 to fe1672e
4379 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
4381 * examples/Makefile.am:
4382 examples: use LDADD for libs instead of LDFLAGS
4384 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
4387 configure: make sure releases are in .doap file
4389 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
4391 * examples/test-cgroups.c:
4392 examples: test-cgroups: don't put code with side effects into g_assert()
4393 The g_assert() might get compiled out with the right
4394 compiler/preprocessor flags.
4396 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
4398 * examples/.gitignore:
4399 examples: add cgroup test binary to .gitignore
4401 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
4403 * examples/test-cgroups.c:
4404 examples: fix cgroup test build
4405 Fixes build failure caused by compiler warning:
4406 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
4408 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
4411 .gitignore: ignore temp files created in the course of 'make check'
4413 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
4415 * gst/rtsp-server/rtsp-media.c:
4416 rtsp-media: don't loose frames handling new PLAY request
4417 If client supplied a range check if the range specifies the start point.
4418 If not, then do an accurate seek to the current position. If a start
4419 point was specified do do a key unit seek to make sure the streaming
4420 starts with decodeable frames.
4421 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
4423 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
4425 * gst/rtsp-server/rtsp-media.c:
4426 Revert "media: only flush when setting a new start position"
4427 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
4428 We need to do the flush in all cases, demuxer block currently for
4431 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
4433 * gst/rtsp-server/rtsp-media.c:
4434 media: only flush when setting a new start position
4435 Only flush the pipeline when we change the start position with
4437 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
4439 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
4441 * gst/rtsp-server/rtsp-stream.c:
4442 stream: set ttl-mc before adding the socket
4443 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
4444 never be set on socket.
4445 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
4447 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
4449 * gst/rtsp-server/rtsp-media.c:
4450 media: stop thread if media is already prepared
4451 in gst_rtsp_media_prepare() the thread is not used if media is already
4452 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
4454 https://bugzilla.gnome.org/show_bug.cgi?id=724182
4456 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
4459 build: Ship gst-rtsp-server.doap file
4461 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
4463 * tests/check/gst/rtspserver.c:
4464 tests: Fix another compiler warning with gcc
4466 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
4468 * gst/rtsp-server/rtsp-client.c:
4469 * gst/rtsp-server/rtsp-mount-points.c:
4470 * gst/rtsp-server/rtsp-stream.c:
4471 * tests/check/gst/client.c:
4472 rtsp-server: Fix lots of compiler warnings with clang
4474 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
4477 * gst-rtsp-server.doap:
4478 * tests/Makefile.am:
4479 configure: Synchronise with the configure scripts of the other modules
4481 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
4484 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
4486 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
4488 * gst/rtsp-server/rtsp-media.c:
4489 * gst/rtsp-server/rtsp-stream.c:
4490 Revert "rtsp-server: support build against last stable release"
4491 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
4492 Let us require 1.2.3 now, which is going to be released in a few
4495 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
4497 * gst/rtsp-server/rtsp-session-media.c:
4498 * gst/rtsp-server/rtsp-stream-transport.c:
4499 session: improve RTP-Info
4500 Ignore streams that can't generate RTP-Info instead of failing.
4501 Don't return the empty string when all streams are unconfigured but
4502 return NULL so that we don't generate and empty RTP-Info header.
4503 Improve docs a little.
4505 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
4507 * gst/rtsp-server/rtsp-session-media.c:
4508 Don't free rtpinfo GString when it is NULL
4509 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
4511 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
4513 * gst/rtsp-server/rtsp-media.c:
4514 media: only set keyframe flag when modifying start
4515 Only set the keyframe flag when we modify the start position. The
4516 keyframe flag should probably be ignored when no change is requested but
4517 until we can claim this is all documented properly and all demuxer
4518 implement this, avoid setting the flag.
4519 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
4521 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
4523 * gst/rtsp-server/rtsp-thread-pool.c:
4524 thread-pool: Unref source after mainloop has quit to avoid races in GLib
4525 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
4527 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
4529 * gst/rtsp-server/rtsp-stream.c:
4530 stream: handle NULL seqnum and rtptime arguments
4532 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
4534 * gst/rtsp-server/rtsp-thread-pool.c:
4535 * tests/check/gst/threadpool.c:
4536 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
4537 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
4539 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
4541 * gst/rtsp-server/rtsp-stream.c:
4542 stream: add fallback for missing stats property
4543 Use a fallback when the payloader does not have a stats property
4544 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
4546 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
4549 Automatic update of common submodule
4550 From f7bc1c3 to 1a07da9
4552 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
4554 * gst/rtsp-server/rtsp-stream.c:
4555 stream: don't leak stats structure
4556 Don't leak the stats structure and deal with NULL stats.
4558 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
4560 * gst/rtsp-server/rtsp-stream.c:
4561 stream: Get rtpinfo properties atomically from payloader
4562 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
4564 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
4566 * gst/rtsp-server/rtsp-media.c:
4567 media: refactor state change functions and signals
4568 Make functions to set the target state and the pipeline state and emit
4569 the signals from those functions.
4571 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
4573 * gst/rtsp-server/rtsp-media.c:
4574 * gst/rtsp-server/rtsp-media.h:
4575 media: add signal to notify of pending state changes
4577 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
4579 * gst/rtsp-server/rtsp-media.c:
4580 * gst/rtsp-server/rtsp-stream.c:
4581 rtsp-server: support build against last stable release
4582 Until 1.2.3 is out with the new get_type function and we
4585 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
4587 * gst/rtsp-server/rtsp-stream.c:
4588 stream: fix compilation
4590 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
4592 * gst/rtsp-server/rtsp-media.c:
4593 * gst/rtsp-server/rtsp-media.h:
4594 * gst/rtsp-server/rtsp-stream.c:
4595 * gst/rtsp-server/rtsp-stream.h:
4596 stream: add property to configure profiles
4598 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
4600 * gst/rtsp-server/rtsp-client.c:
4601 client: let stream check supported transport
4602 Delegate the check if a transport is allowed to the stream.
4603 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
4605 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
4607 * gst/rtsp-server/rtsp-stream.c:
4608 * gst/rtsp-server/rtsp-stream.h:
4609 stream: add method to check supported transport
4610 Add a method to check if a transport is supported
4612 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
4615 configure.ac: Only check for gstreamer-check, not check
4616 We include check in gstreamer-check since quite some time now.
4618 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
4620 * gst/rtsp-server/rtsp-session-media.c:
4621 * gst/rtsp-server/rtsp-stream-transport.c:
4622 * gst/rtsp-server/rtsp-stream.c:
4623 * gst/rtsp-server/rtsp-stream.h:
4624 stream: return clock-rate from get_rtpinfo
4625 And use it to correct the rtptime to the requested start-time.
4626 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
4628 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
4630 * gst/rtsp-server/rtsp-session-media.c:
4631 * gst/rtsp-server/rtsp-stream-transport.c:
4632 * gst/rtsp-server/rtsp-stream-transport.h:
4633 session-media: calculate start-time
4635 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
4637 * gst/rtsp-server/rtsp-stream-transport.c:
4638 * gst/rtsp-server/rtsp-stream.c:
4639 * gst/rtsp-server/rtsp-stream.h:
4640 stream: also return the running-time
4641 Return the running-time in the rtpinfo as well.
4643 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
4645 * gst/rtsp-server/rtsp-client.c:
4646 * gst/rtsp-server/rtsp-session-media.c:
4647 * gst/rtsp-server/rtsp-session-media.h:
4648 * gst/rtsp-server/rtsp-stream-transport.c:
4649 * gst/rtsp-server/rtsp-stream-transport.h:
4650 session-media: let the session-media make the RTPInfo
4651 Add method to create the RTPInfo for a stream-transport.
4652 Add method to create the RTPInfo for all stream-transports in a
4654 Use the session-media RTPInfo code in client. This allows us to refactor
4655 another method to link the TCP callbacks.
4657 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
4659 mount-points: sort sequence before g_sequence_lookup
4660 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
4661 sort sequence if dirty, otherwise lookup will fail.
4662 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
4664 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
4667 configure: rename package from gst-rtsp to gst-rtsp-server
4668 To match git module name and avoid confusion with the
4669 rtsp lib in gst-plugins-base and rtsp plugin in -good.
4671 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
4674 configure: bump core/base/good requirement to 1.2.0
4675 Bump to released stable version and make implicit
4676 requirements explicit.
4678 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
4683 Fix broken gettext setup which is not used anyway
4685 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
4688 Automatic update of common submodule
4689 From dbedaa0 to d48bed3
4691 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
4693 * gst/rtsp-server/rtsp-client.c:
4694 * gst/rtsp-server/rtsp-media.c:
4695 * gst/rtsp-server/rtsp-media.h:
4696 media: add setup_sdp vmethod
4697 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
4698 gst_rtsp_media_setup_sdp.
4699 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
4701 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
4703 * gst/rtsp-server/rtsp-stream.c:
4704 rtsp-stream: Check return value of sscanf
4705 streamid is only valid if sscanf matched something.
4707 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
4709 * gst/rtsp-server/rtsp-client.c:
4710 rtsp-client: Fix iteration
4711 Wouldn't even enter the code block otherwise (i++ was used as the check
4712 and not the postfix).
4714 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
4716 * gst/rtsp-server/rtsp-client.c:
4717 * gst/rtsp-server/rtsp-client.h:
4718 client: add vmethod to configure media and streams
4719 Implement a vmethod that can be used to configure the media and the
4720 streams based on the current context. Handle the blocksize handling in
4721 the default handler.
4722 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
4724 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
4727 Make git ignore more unit test binaries
4729 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
4731 * gst/rtsp-server/rtsp-address-pool.h:
4732 * gst/rtsp-server/rtsp-auth.h:
4733 * gst/rtsp-server/rtsp-client.h:
4734 * gst/rtsp-server/rtsp-context.h:
4735 * gst/rtsp-server/rtsp-media-factory-uri.h:
4736 * gst/rtsp-server/rtsp-media-factory.h:
4737 * gst/rtsp-server/rtsp-media.h:
4738 * gst/rtsp-server/rtsp-mount-points.h:
4739 * gst/rtsp-server/rtsp-server.h:
4740 * gst/rtsp-server/rtsp-session-media.h:
4741 * gst/rtsp-server/rtsp-session-pool.h:
4742 * gst/rtsp-server/rtsp-session.h:
4743 * gst/rtsp-server/rtsp-stream-transport.h:
4744 * gst/rtsp-server/rtsp-stream.h:
4745 * gst/rtsp-server/rtsp-thread-pool.h:
4746 * gst/rtsp-server/rtsp-token.h:
4747 rtsp-server: add padding to many public structures
4748 Not mini objects though, since they are not subclassable
4749 anyway, nor kept on the stack or inlined in a structure.
4751 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
4753 media: add new create_rtpbin vmethod
4754 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
4755 https://bugzilla.gnome.org/show_bug.cgi?id=719734
4757 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
4759 * tests/check/gst/media.c:
4760 tests: fix memory leak, free test's thread pool
4761 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
4763 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
4765 * gst/rtsp-server/rtsp-stream-transport.c:
4766 stream-transport: free url in finalize
4768 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
4770 * gst/rtsp-server/rtsp-media.c:
4771 media: also do state change in suspended state
4773 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
4775 * gst/rtsp-server/rtsp-client.c:
4776 * gst/rtsp-server/rtsp-media.c:
4777 media: also handle prepare and range in suspended state
4778 When we are suspended, we are already prepared.
4779 We can get the range in the suspended state.
4781 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
4783 * tests/check/Makefile.am:
4784 * tests/check/gst/sessionmedia.c:
4785 check: add test for uri in setup
4786 Added unit tests for the new functionality in GstRTSPStreamTransport.
4787 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
4789 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
4791 * gst/rtsp-server/rtsp-client.c:
4792 client: store setup uri and use in PLAY response
4793 Store the uri used when doing the setup and use that in the PLAY
4795 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
4797 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
4799 * gst/rtsp-server/rtsp-stream-transport.c:
4800 * gst/rtsp-server/rtsp-stream-transport.h:
4801 stream-transport: add method to get/set url
4803 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
4805 * gst/rtsp-server/rtsp-client.c:
4806 client: suspend after SDP and unsuspend before PLAYING
4807 Based on patches by Ognyan Tonchev <ognyan@axis.com>
4808 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
4810 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
4812 * gst/rtsp-server/rtsp-media-factory.c:
4813 * gst/rtsp-server/rtsp-media-factory.h:
4814 * gst/rtsp-server/rtsp-media.c:
4815 * gst/rtsp-server/rtsp-media.h:
4816 * gst/rtsp-server/rtsp-session-media.c:
4817 * gst/rtsp-server/rtsp-session.c:
4818 * tests/check/gst/media.c:
4819 * tests/check/gst/mediafactory.c:
4820 media: add suspend modes
4821 Add support for different suspend modes. The stream is suspended right after
4822 producing the SDP and after PAUSE. Different suspend modes are available that
4823 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
4824 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
4825 state and RESET will bring the pipeline to the NULL state.
4826 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
4827 this means that the pipeline needs to be prerolled again.
4828 Base on patches by Ognyan Tonchev <ognyan@axis.com>
4829 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
4831 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
4833 * gst/rtsp-server/rtsp-media.c:
4834 media: start live streams in blocked state
4835 Start live streams in the blocked state and make them preroll using the
4836 messages. This ensure that no data is played by the sink until we explicitly
4837 unblock the stream right before going to PLAYING.
4838 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
4840 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
4842 * gst/rtsp-server/rtsp-media.c:
4843 media: refactor starting and waiting for preroll
4844 Based on patches from Ognyan Tonchev <ognyan@axis.com>
4845 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
4847 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
4849 * gst/rtsp-server/rtsp-stream.c:
4850 * gst/rtsp-server/rtsp-stream.h:
4851 stream: add API to block streams
4852 Add an API to block on the streams and make it post a message.
4853 Based on patch by Ognyan Tonchev <ognyan@axis.com>
4854 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
4856 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
4858 * docs/libs/Makefile.am:
4859 docs: Specify the override file
4860 Even if it's empty (for now) it avoids make distcheck complaining
4862 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
4864 * gst/rtsp-server/rtsp-media.c:
4865 media: move default implementations to where they are used
4867 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
4869 * gst/rtsp-server/rtsp-media.c:
4870 media: take the right lock in gst_rtsp_media_set_pipeline_state()
4871 We need to take the state_lock when calling this method.
4873 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
4875 * gst/rtsp-server/rtsp-media.c:
4876 media: handle add-added on non-bins too
4877 Handle dynamic payloaders that are not bins, as used in the unit-test.
4879 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4881 * gst/rtsp-server/rtsp-media-factory.c:
4882 * gst/rtsp-server/rtsp-media-factory.h:
4883 * gst/rtsp-server/rtsp-media.c:
4884 rtsp-media/-factory: Fix request pad name comments
4885 These must be escaped for gtk-doc to parse the comments without warnings.
4887 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
4889 rtsp-media: remove transports if media is in error status
4890 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
4891 trying to change to GST_STATE_NULL and media is in error status, we
4892 remove all transports.
4893 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
4895 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
4897 * gst/rtsp-server/rtsp-media.c:
4898 rtsp-media: use element metadata to find payloader
4899 Use the element metadata to find the payloader instead of checking
4901 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
4903 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
4905 rtsp-stream: add getter for payload type
4906 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
4907 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
4908 element and create the stream with this one instead of the dynpay%d
4910 https://bugzilla.gnome.org/show_bug.cgi?id=712396
4912 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4914 * gst/rtsp-server/rtsp-client.c:
4915 * gst/rtsp-server/rtsp-context.h:
4916 * gst/rtsp-server/rtsp-media.c:
4917 * gst/rtsp-server/rtsp-mount-points.c:
4918 * gst/rtsp-server/rtsp-server.c:
4919 * gst/rtsp-server/rtsp-token.c:
4920 rtsp-*: Refer to NULL as a constant in comments
4922 https://bugzilla.gnome.org/show_bug.cgi?id=714988
4924 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4926 rtsp-*: Fix type name typos in comments
4927 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
4928 * rtsp-auth: Refer to part of constant name as text
4929 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
4930 * rtsp-session-media: Fix GstRTSPSessionMedia typo
4931 * rtsp-stream: Fix typo when refering to GstBin
4932 https://bugzilla.gnome.org/show_bug.cgi?id=714988
4934 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4937 * docs/libs/gst-rtsp-server-docs.sgml:
4938 * docs/libs/gst-rtsp-server-sections.txt:
4939 docs: Improve documentation
4940 * Include annotation-glossary to quiet gtk-doc
4941 * Rename remaining ClientState -> Context
4942 * Rename object hierarchy file
4943 * Remove stale chapter references
4944 * Add missing function and object references
4945 * Include missing GstRTSPAddressPoolResult
4946 https://bugzilla.gnome.org/show_bug.cgi?id=714988
4948 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
4950 * gst/rtsp-server/rtsp-client.c:
4951 * gst/rtsp-server/rtsp-server.c:
4952 * gst/rtsp-server/rtsp-session-pool.c:
4953 * gst/rtsp-server/rtsp-session.c:
4954 * gst/rtsp-server/rtsp-stream.c:
4955 rtsp-server: sprinkle some allow-none annotations for g-i
4957 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
4959 * gst/rtsp-server/rtsp-stream.c:
4960 * gst/rtsp-server/rtsp-stream.h:
4961 stream: add method to filter transports
4962 Add a method to safely iterate and collect the stream transports
4963 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
4965 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
4967 * gst/rtsp-server/rtsp-client.c:
4968 * gst/rtsp-server/rtsp-server.c:
4969 * gst/rtsp-server/rtsp-session-pool.c:
4970 * gst/rtsp-server/rtsp-session.c:
4971 rtsp: allow NULL func in filters
4972 Passing a null function make the filters return a list of
4975 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
4977 * gst/rtsp-server/rtsp-address-pool.c:
4978 * tests/check/gst/addresspool.c:
4979 address-pool: fix address increment
4980 Use a guint instead of guint8 to increment the address. It's still not
4981 completely correct because a guint might not be able to hold the complete
4982 address range, but that's an enhacement for later.
4983 Add unit test to test improved behaviour.
4984 https://bugzilla.gnome.org/show_bug.cgi?id=708237
4986 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
4988 * gst/rtsp-server/rtsp-client.c:
4989 * tests/check/gst/client.c:
4990 client: allow absolute path in requests
4991 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
4993 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
4995 * gst/rtsp-server/rtsp-client.c:
4996 * gst/rtsp-server/rtsp-client.h:
4997 client: make make_path_from_uri a vmethod
4999 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
5001 * docs/libs/gst-rtsp-server-sections.txt:
5002 * gst/rtsp-server/rtsp-stream.c:
5003 * gst/rtsp-server/rtsp-stream.h:
5004 * tests/check/Makefile.am:
5005 * tests/check/gst/stream.c:
5006 stream: Add functions to get rtp and rtcp sockets
5007 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
5009 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
5011 * gst/rtsp-server/rtsp-context.c:
5012 * gst/rtsp-server/rtsp-context.h:
5013 context: defing a GType for the context
5014 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
5016 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
5018 * gst/rtsp-server/Makefile.am:
5019 * gst/rtsp-server/rtsp-auth.c:
5020 * gst/rtsp-server/rtsp-context.c:
5021 * gst/rtsp-server/rtsp-media.c:
5022 * gst/rtsp-server/rtsp-mount-points.c:
5023 * gst/rtsp-server/rtsp-server.h:
5024 * gst/rtsp-server/rtsp-session-media.c:
5025 * gst/rtsp-server/rtsp-session.c:
5026 * gst/rtsp-server/rtsp-stream.c:
5027 Fixed several GIR warnings
5029 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
5031 * gst/rtsp-server/rtsp-auth.c:
5034 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5036 * tests/check/Makefile.am:
5037 * tests/check/gst/token.c:
5038 tests: Add unit tests for token
5039 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
5041 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5043 * gst/rtsp-server/rtsp-token.c:
5044 token: Validate args for gst_rtsp_token_is_allowed
5045 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
5047 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5049 * gst/rtsp-server/rtsp-token.c:
5050 token: Fix bug when creating empty token
5051 We always want to have a valid GstStructure in the token.
5052 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
5054 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
5056 * gst/rtsp-server/rtsp-thread-pool.c:
5057 thread-pool: avoid race in shutdown
5058 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
5059 don't actually stop the mainloop ever. Solve this race by adding an idle source
5060 to the mainloop that calls the _quit. This way we immediately exit the mainloop
5061 if quit was called before we started it.
5063 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5065 * tests/check/Makefile.am:
5066 * tests/check/gst/permissions.c:
5067 tests: Add unit tests for permissions
5068 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
5070 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5072 * tests/check/gst/mediafactory.c:
5073 tests: Test mediafactory permissions
5074 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
5076 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5078 * gst/rtsp-server/rtsp-permissions.c:
5079 permissions: Fix refcounting when adding/removing roles
5080 Previously a role that was removed was unreffed twice, and when
5081 replacing an existing role the replaced role was freed while still being
5082 referenced. Both bugs are now fixed.
5083 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
5085 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5087 * tests/check/gst/media.c:
5088 * tests/check/gst/mediafactory.c:
5089 * tests/check/gst/rtspserver.c:
5090 tests: Check gst_rtsp_url_parse return value
5091 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
5093 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
5096 Automatic update of common submodule
5097 From 865aa20 to dbedaa0
5099 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
5101 * gst/rtsp-server/rtsp-server.c:
5102 rtsp-server: Fix socket leak
5103 https://bugzilla.gnome.org/show_bug.cgi?id=710088
5105 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
5107 * gst/rtsp-server/rtsp-session-pool.c:
5108 rtsp-session-pool: Make sure session IDs are properly URI-escaped
5109 https://bugzilla.gnome.org/show_bug.cgi?id=643812
5111 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
5113 * examples/.gitignore:
5114 * examples/test-video.c:
5115 examples: fix compilation when WITH_AUTH is defined
5116 https://bugzilla.gnome.org/show_bug.cgi?id=710228
5118 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
5121 gitignore: Add new test binary
5123 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
5125 * tests/check/Makefile.am:
5126 * tests/check/gst/threadpool.c:
5127 thread-pool: Add unit test for the thread pools
5128 https://bugzilla.gnome.org/show_bug.cgi?id=710228
5130 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
5132 * gst/rtsp-server/rtsp-thread-pool.c:
5133 thread-pool: Fix thread leak when reusing threads
5134 https://bugzilla.gnome.org/show_bug.cgi?id=709730
5136 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
5138 * gst/rtsp-server/rtsp-server.c:
5139 * tests/check/gst/rtspserver.c:
5140 tests: fixed racy behavior in rtspserver tests
5141 https://bugzilla.gnome.org/show_bug.cgi?id=710078
5143 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5145 * tests/check/gst/addresspool.c:
5146 tests: Improve address pool unit tests
5147 Add a range with mixed IPV4 and IPV6 addresses to pool.
5148 Get an IPV4 address from an IPV6-only pool.
5149 Get an IPV6 address from an IPV4-only pool.
5150 Reserve a IPV6 address from an IPV4-only pool.
5151 Check for unicast addresses in multicast-only pool.
5152 Check for unicast addresses in uni-/multicast-mixed pool.
5153 https://bugzilla.gnome.org/show_bug.cgi?id=710128
5155 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5157 * gst/rtsp-server/rtsp-client.c:
5158 client: append query string in PAUSE/PLAY/TEARDOWN as well
5160 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
5162 * gst/rtsp-server/rtsp-client.c:
5163 client: Add query to control path
5164 If the SETUP url contains a query it must be appended to the control
5165 path so that it matches any already created stream in the media. The
5166 query will also be appended to the session media path.
5168 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5170 * gst/rtsp-server/rtsp-media.c:
5171 rtsp-media: remove old line
5173 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
5175 * gst/rtsp-server/rtsp-stream.c:
5176 stream: Correct control comparison
5177 https://bugzilla.gnome.org/show_bug.cgi?id=709176
5179 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5181 * gst/rtsp-server/rtsp-media.c:
5182 media: Check dynamically if the pipeline supports seeking
5183 We should not depend on whether or not the pipeline state change
5184 returned NO_PREROLL or not. A media could dynamically change its
5185 element and switch from seekable to non seekable so it's best to test
5186 the seekable nature of the pipeline dynamically when we try to do a seek.
5188 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5190 * gst/rtsp-server/rtsp-media.c:
5191 media: Return FALSE if seeking is not supported
5193 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5195 * gst/rtsp-server/rtsp-media.c:
5196 rtsp-media: don't seek accurate by default
5197 Accurate seeking is perhaps a little overkill in the most common situation and
5198 causes some formats (mp3) over slow media to seek extremely slowly.
5200 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
5202 * tests/check/gst/rtspserver.c:
5203 tests: fix unit test
5204 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
5206 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
5208 * gst/rtsp-server/rtsp-client.c:
5209 client: Reply 400 if media cannot be constructed
5210 Reply 400 Bad Request instead of 503 Service Unavailable if media
5211 cannot be constructed in SETUP.
5212 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
5214 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
5216 * gst/rtsp-server/rtsp-client.c:
5217 client: Send setup reply once only
5218 If find_media() failed in handle_setup_request() two replies was sent.
5219 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
5221 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
5224 Automatic update of common submodule
5225 From 6b03ba7 to 865aa20
5227 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
5229 * gst/rtsp-server/rtsp-server.c:
5230 server: Emit client-connected signal earlier
5231 Emit client-connected before the client ref is given to a GSource,
5232 otherwise client-connected can be emitted after the client object has
5235 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
5237 * gst/rtsp-server/rtsp-address-pool.c:
5238 * gst/rtsp-server/rtsp-address-pool.h:
5239 * gst/rtsp-server/rtsp-stream.c:
5240 * tests/check/gst/addresspool.c:
5241 addresspool: return reason of failure
5242 Let gst_rtsp_address_pool_reserve_address() return the reason why
5243 the address could not be reserved.
5244 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
5246 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
5249 autogen.sh: Sync behaviour with other GStreamer modules
5250 Allows building from outside of tree amongst other things
5252 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
5255 Automatic update of common submodule
5256 From b613661 to 6b03ba7
5258 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
5261 Automatic update of common submodule
5262 From 74a6857 to b613661
5264 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
5267 Automatic update of common submodule
5268 From 01a7a46 to 74a6857
5270 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
5272 * gst/rtsp-server/rtsp-client.c:
5273 client: Do not read beyond end of path string
5274 If the setup was done without a control url, make sure we don't try to read the
5275 non-existing control string and crash.
5277 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5279 * gst/rtsp-server/rtsp-client.c:
5280 client: Fix RTPInfo header
5281 Refactor the method to make the content_base.
5282 Use the content-base and the control url to construct the RTPInfo
5285 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5287 * gst/rtsp-server/rtsp-client.c:
5288 client: map url to path only in describe
5289 Only map the request url to a path in the DESCRIBE method. The SDP then
5290 contains the base and control urls that should be used to SETUP/PAUSE/
5291 PLAY/TEARDOWN the media.
5293 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5295 * gst/rtsp-server/rtsp-client.c:
5296 Revert "client: map URL to path in requests"
5297 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
5298 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
5299 contains the base and control urls which are used in the SETUP, PLAY,
5300 PAUSE and TEARDOWN requests.
5302 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5304 * gst/rtsp-server/rtsp-client.c:
5305 client: map URL to path in requests
5307 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5309 * gst/rtsp-server/rtsp-client.c:
5310 * gst/rtsp-server/rtsp-mount-points.c:
5311 * gst/rtsp-server/rtsp-mount-points.h:
5312 mount-points: make vmethod to make path from uri
5313 Make a vmethod to transform an url into a path. The path is then used to lookup
5314 the factory. This makes it possible to also use other bits of the url, such as
5315 the query parameters, to locate the factory.
5317 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
5319 * gst/rtsp-server/rtsp-thread-pool.c:
5320 * gst/rtsp-server/rtsp-thread-pool.h:
5321 thread-pool: Add cleanup to wait for the threadpool to finish
5322 Also fix race condition if two threads are asking for the first
5323 thread from the thread pool at once. This would case two internal
5324 GThreadPools to be created.
5325 https://bugzilla.gnome.org/show_bug.cgi?id=707753
5327 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
5329 * gst/rtsp-server/rtsp-client.c:
5330 * tests/check/gst/client.c:
5331 client: free threadpool
5332 https://bugzilla.gnome.org/show_bug.cgi?id=707638
5334 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
5336 * tests/check/gst/mountpoints.c:
5337 mountpoints tests: unref matched factories
5338 https://bugzilla.gnome.org/show_bug.cgi?id=707638
5340 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
5342 * tests/check/gst/media.c:
5343 media tests: unref thread pool and caps
5344 https://bugzilla.gnome.org/show_bug.cgi?id=707638
5346 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
5348 * gst/rtsp-server/rtsp-auth.c:
5349 * gst/rtsp-server/rtsp-media-factory.c:
5350 * gst/rtsp-server/rtsp-media.c:
5351 auth, media, media-factory: unref permissions
5352 https://bugzilla.gnome.org/show_bug.cgi?id=707638
5354 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5356 * examples/Makefile.am:
5357 Makefile: add rule for appsrc example
5359 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5361 * examples/test-appsrc.c:
5362 tests: add appsrc example
5363 Add an example on how to use appsrc to feed the server pipeline with data.
5365 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
5367 * gst/rtsp-server/rtsp-client.c:
5368 rtsp-client: remove query part from content-base string
5369 Make sure that after the control url has been resolved, it's
5370 not a part of the query-string.
5371 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
5373 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5375 * gst/rtsp-server/rtsp-client.c:
5376 client: don't check url in response
5377 There is no url or method in the response to check
5379 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5381 * gst/rtsp-server/rtsp-client.c:
5382 * gst/rtsp-server/rtsp-client.h:
5383 Add handle-response signal for when we receive a GET_PARAMETER response
5385 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5387 * gst/rtsp-server/rtsp-server.c:
5388 Fix gst_rtsp_server_client_filter, using wrong variable type
5390 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
5392 * gst/rtsp-server/rtsp-media-factory-uri.c:
5393 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
5394 For AAC we need to check for framed=true instead of parsed=true.
5395 https://bugzilla.gnome.org/show_bug.cgi?id=701384
5397 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5399 * gst/rtsp-server/rtsp-stream.c:
5400 stream: optimize pipeline for protocols
5401 When TCP is not an allowed protocol for the stream, avoid creating the
5402 appsrc/appsink/queue and tee elements.
5404 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5406 * gst/rtsp-server/rtsp-media.c:
5407 media: set protocols on streams
5409 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5411 * gst/rtsp-server/rtsp-client.c:
5412 client: use protocols supported by stream
5414 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5416 * gst/rtsp-server/rtsp-media-factory.c:
5417 * gst/rtsp-server/rtsp-media.c:
5418 * gst/rtsp-server/rtsp-stream.c:
5419 media-factory: allow all protocols
5421 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5423 * gst/rtsp-server/rtsp-media.c:
5424 media: configure protocols in new streams
5426 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5428 * gst/rtsp-server/rtsp-stream.c:
5429 * gst/rtsp-server/rtsp-stream.h:
5430 stream: add protocols property
5432 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5434 * gst/rtsp-server/rtsp-media.c:
5435 rtsp-media: send state in "new-state" signal
5436 https://bugzilla.gnome.org/show_bug.cgi?id=705110
5438 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
5441 build: add subdir-objects to AM_INIT_AUTOMAKE
5442 Fixes warnings with automake 1.14
5443 https://bugzilla.gnome.org/show_bug.cgi?id=705350
5445 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5447 * docs/libs/gst-rtsp-server-sections.txt:
5448 * gst/rtsp-server/rtsp-client.c:
5449 * gst/rtsp-server/rtsp-server.c:
5450 * gst/rtsp-server/rtsp-server.h:
5451 server: add method to iterate clients of server
5453 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5455 * gst/rtsp-server/rtsp-media.c:
5456 * gst/rtsp-server/rtsp-media.h:
5457 Add vmethod for rtsp-media subclass to access rtpbin
5459 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5461 * gst/rtsp-server/rtsp-client.h:
5462 small documentation fix
5464 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5466 * gst/rtsp-server/rtsp-client.c:
5467 Do not take range header if range is invalid
5469 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5471 * docs/libs/gst-rtsp-server-sections.txt:
5472 * gst/rtsp-server/rtsp-media.c:
5473 media: add docs for new method
5475 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5477 * gst/rtsp-server/rtsp-media.c:
5478 * gst/rtsp-server/rtsp-media.h:
5479 Add API to rtsp-media set the pipeline's state
5481 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5483 * gst/rtsp-server/rtsp-media.c:
5484 Update current position/duration when gst_rtsp_media_get_range_string is called
5486 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5488 * examples/test-cgroups.c:
5489 tests: add some more docs
5491 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5493 * examples/test-cgroups.c:
5494 * gst/rtsp-server/Makefile.am:
5495 * gst/rtsp-server/rtsp-auth.c:
5496 * gst/rtsp-server/rtsp-auth.h:
5497 * gst/rtsp-server/rtsp-client.c:
5498 * gst/rtsp-server/rtsp-client.h:
5499 * gst/rtsp-server/rtsp-context.c:
5500 * gst/rtsp-server/rtsp-context.h:
5501 * gst/rtsp-server/rtsp-params.c:
5502 * gst/rtsp-server/rtsp-params.h:
5503 * gst/rtsp-server/rtsp-server.c:
5504 * gst/rtsp-server/rtsp-thread-pool.c:
5505 * gst/rtsp-server/rtsp-thread-pool.h:
5506 * tests/check/gst/client.c:
5507 ClientState -> Context
5508 Rename the clientstate to context and put the code in a separate file.
5510 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5512 * examples/test-auth.c:
5513 * gst/rtsp-server/rtsp-auth.c:
5514 * gst/rtsp-server/rtsp-auth.h:
5515 auth: add support for default token
5516 The default token is used when the user is not authenticated and can be used to
5517 give minimal permissions.
5519 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5521 * examples/test-auth.c:
5522 * gst/rtsp-server/rtsp-auth.c:
5523 auth: use defines when possible
5525 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5527 * gst/rtsp-server/rtsp-address-pool.c:
5528 address-pool: improve docs
5530 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5532 * gst/rtsp-server/rtsp-permissions.c:
5533 permissions: add the role to the copy
5535 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
5537 * gst/rtsp-server/rtsp-permissions.c:
5538 permissions: Also copy the roles
5540 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
5542 * gst/rtsp-server/rtsp-permissions.c:
5543 permissions: Make it build
5545 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5547 * gst/rtsp-server/rtsp-address-pool.h:
5550 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5552 * docs/libs/gst-rtsp-server-sections.txt:
5553 * gst/rtsp-server/rtsp-auth.c:
5554 * gst/rtsp-server/rtsp-auth.h:
5555 * gst/rtsp-server/rtsp-media.c:
5556 * gst/rtsp-server/rtsp-session-media.c:
5557 * gst/rtsp-server/rtsp-stream-transport.c:
5558 * gst/rtsp-server/rtsp-stream-transport.h:
5559 * gst/rtsp-server/rtsp-stream.c:
5560 * tests/check/gst/client.c:
5563 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5565 * docs/libs/gst-rtsp-server-sections.txt:
5566 * gst/rtsp-server/rtsp-address-pool.c:
5567 * gst/rtsp-server/rtsp-address-pool.h:
5568 * tests/check/gst/addresspool.c:
5569 * tests/check/gst/rtspserver.c:
5570 address-pool: cleanups
5571 Remove redundant method, improve docs.
5573 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5575 * docs/libs/gst-rtsp-server-sections.txt:
5576 * gst/rtsp-server/rtsp-auth.h:
5577 * gst/rtsp-server/rtsp-permissions.c:
5578 * gst/rtsp-server/rtsp-permissions.h:
5579 * gst/rtsp-server/rtsp-token.c:
5582 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5584 * gst/rtsp-server/rtsp-permissions.c:
5585 permissions: implement _remove_role
5587 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5589 * gst/rtsp-server/rtsp-permissions.c:
5590 permissions: update docs
5592 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5594 * tests/check/gst/client.c:
5595 tests: simplify tests
5596 Client settings are now disabled by default so we don't need an auth
5597 module to disable them.
5599 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5601 * gst/rtsp-server/rtsp-auth.c:
5602 auth: add default authorizations
5603 When no auth module is specified, use our table of defaults to look up the
5604 default value of the check instead of always allowing everything. This was
5605 we can disallow client settings by default.
5607 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5610 README: update readme
5612 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5614 * gst/rtsp-server/rtsp-thread-pool.c:
5615 * gst/rtsp-server/rtsp-thread-pool.h:
5616 thread-pool: add more docs
5618 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5620 * gst/rtsp-server/rtsp-thread-pool.c:
5621 * gst/rtsp-server/rtsp-thread-pool.h:
5622 thread-pool: fix race in thread reuse
5623 If we try to reuse a thread right after we made it stop, we end up using a
5624 stopped thread. Catch this case and only reuse threads that are not stopping.
5626 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5628 * gst/rtsp-server/rtsp-server.c:
5629 server: add small debug
5631 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5633 * tests/check/gst/client.c:
5635 Add some permissions to media so we can use the auth and enable
5638 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5640 * gst/rtsp-server/rtsp-client.c:
5641 client: support pushed context in handle_request
5642 If we already have a pushed state, reuse it and add our own things. This makes
5643 it easier to write tests.
5645 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5647 * gst/rtsp-server/rtsp-auth.c:
5648 auth: don't auth on methods
5649 Don't authorize on methods anymore but on the resources that we
5650 try to access, this is more flexible.
5651 Move the authorization checks to where they are needed and let the
5652 check return the response on error.
5654 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5656 * gst/rtsp-server/rtsp-mount-points.c:
5657 mount-points: add some debug
5659 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5661 * tests/check/gst/client.c:
5662 tests: almost fix test
5664 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5666 * gst/rtsp-server/rtsp-auth.c:
5667 * gst/rtsp-server/rtsp-auth.h:
5668 * gst/rtsp-server/rtsp-client.c:
5669 * gst/rtsp-server/rtsp-client.h:
5670 * gst/rtsp-server/rtsp-server.c:
5671 * gst/rtsp-server/rtsp-server.h:
5672 auth: let the auth module check client_settings
5673 Let the auth module decide if client settings are allowed for the
5676 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5678 * gst/rtsp-server/rtsp-token.c:
5679 * gst/rtsp-server/rtsp-token.h:
5680 token: add method to check boolean permission
5682 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5684 * examples/test-auth.c:
5685 * examples/test-cgroups.c:
5686 * gst/rtsp-server/rtsp-token.c:
5687 * gst/rtsp-server/rtsp-token.h:
5688 token: simplify token constructor
5689 Use variable arguments to make easier API.
5691 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5693 * examples/test-auth.c:
5694 * examples/test-cgroups.c:
5695 * gst/rtsp-server/rtsp-media-factory.c:
5696 * gst/rtsp-server/rtsp-media-factory.h:
5697 media-factory: add convenience API for factory
5699 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5701 * examples/test-auth.c:
5702 * examples/test-cgroups.c:
5703 * gst/rtsp-server/rtsp-permissions.c:
5704 * gst/rtsp-server/rtsp-permissions.h:
5705 permissions: simplify API a little
5706 Avoid passing GstStructure in the add_role method, use varargs instead
5707 to construct the structure behind the scenes. We can then also use the
5708 structure name as the role and simplify some more logic.
5710 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5712 * gst/rtsp-server/rtsp-auth.c:
5715 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5717 * gst/rtsp-server/rtsp-auth.c:
5718 * gst/rtsp-server/rtsp-auth.h:
5719 * gst/rtsp-server/rtsp-client.c:
5720 auth: handle unauthorized response
5721 Move handling of the unauthorized response to the auth module, it can add
5722 the appropriate headers to request authorization for the required method
5723 much better than the client.
5725 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5727 * gst/rtsp-server/rtsp-client.c:
5728 * gst/rtsp-server/rtsp-client.h:
5729 client: allow for sending any message, not only requests
5730 Change the _send_request() method to _send_message() so that we
5731 can both send requests and replies.
5733 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5735 * docs/libs/gst-rtsp-server-sections.txt:
5736 * gst/rtsp-server/rtsp-server.h:
5739 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5741 * examples/test-video.c:
5742 * gst/rtsp-server/rtsp-auth.c:
5743 * gst/rtsp-server/rtsp-auth.h:
5744 * gst/rtsp-server/rtsp-server.c:
5745 * gst/rtsp-server/rtsp-server.h:
5746 auth: move TLS handling to auth module
5747 Remove the TLS settings on the server and move it to the auth module because
5748 that is where security related bits go.
5750 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5752 * gst/rtsp-server/rtsp-client.c:
5753 * gst/rtsp-server/rtsp-client.h:
5754 client: add state push/pop
5756 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5758 * gst/rtsp-server/rtsp-client.c:
5759 * gst/rtsp-server/rtsp-client.h:
5760 client: add connection to state
5762 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5764 * gst/rtsp-server/rtsp-mount-points.c:
5765 mount-points: fix debug
5767 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5769 * tests/check/gst/media.c:
5770 tests: fix media test
5772 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5774 * gst/rtsp-server/rtsp-thread-pool.c:
5775 thread-pool: we don't require a state
5777 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5779 * gst/rtsp-server/rtsp-server.c:
5780 server: let context ref the server
5781 So that we don't risk losing the server object early anc crash.
5783 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5785 * tests/check/gst/client.c:
5786 tests: fix client test
5788 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5791 * docs/libs/gst-rtsp-server-docs.sgml:
5792 * docs/libs/gst-rtsp-server-sections.txt:
5793 * gst/rtsp-server/rtsp-address-pool.c:
5794 * gst/rtsp-server/rtsp-auth.c:
5795 * gst/rtsp-server/rtsp-client.c:
5796 * gst/rtsp-server/rtsp-client.h:
5797 * gst/rtsp-server/rtsp-media-factory-uri.c:
5798 * gst/rtsp-server/rtsp-media-factory.c:
5799 * gst/rtsp-server/rtsp-media-factory.h:
5800 * gst/rtsp-server/rtsp-media.c:
5801 * gst/rtsp-server/rtsp-mount-points.c:
5802 * gst/rtsp-server/rtsp-params.c:
5803 * gst/rtsp-server/rtsp-permissions.c:
5804 * gst/rtsp-server/rtsp-sdp.c:
5805 * gst/rtsp-server/rtsp-server.c:
5806 * gst/rtsp-server/rtsp-server.h:
5807 * gst/rtsp-server/rtsp-session-media.c:
5808 * gst/rtsp-server/rtsp-session-pool.c:
5809 * gst/rtsp-server/rtsp-session.c:
5810 * gst/rtsp-server/rtsp-stream-transport.c:
5811 * gst/rtsp-server/rtsp-stream.c:
5812 * gst/rtsp-server/rtsp-thread-pool.c:
5813 * gst/rtsp-server/rtsp-token.c:
5816 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5818 * gst/rtsp-server/rtsp-session-pool.c:
5819 * gst/rtsp-server/rtsp-session-pool.h:
5820 session-pool: make vmethod to create a session
5821 Make a vmethod to create a sessions so that subclasses can create
5822 custom session objects
5824 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5826 * gst/rtsp-server/rtsp-auth.c:
5827 * gst/rtsp-server/rtsp-media-factory.h:
5828 * gst/rtsp-server/rtsp-media.h:
5829 * gst/rtsp-server/rtsp-mount-points.h:
5830 * gst/rtsp-server/rtsp-session-pool.h:
5831 * gst/rtsp-server/rtsp-stream.h:
5834 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5836 * docs/libs/gst-rtsp-server-docs.sgml:
5837 * docs/libs/gst-rtsp-server-sections.txt:
5838 * gst/rtsp-server/rtsp-address-pool.c:
5839 * gst/rtsp-server/rtsp-address-pool.h:
5840 * gst/rtsp-server/rtsp-auth.c:
5841 * gst/rtsp-server/rtsp-client.h:
5842 * gst/rtsp-server/rtsp-media-factory.h:
5843 * gst/rtsp-server/rtsp-media.c:
5844 * gst/rtsp-server/rtsp-media.h:
5845 * gst/rtsp-server/rtsp-permissions.c:
5846 * gst/rtsp-server/rtsp-permissions.h:
5847 * gst/rtsp-server/rtsp-server.h:
5848 * gst/rtsp-server/rtsp-session-media.c:
5849 * gst/rtsp-server/rtsp-session-media.h:
5850 * gst/rtsp-server/rtsp-session-pool.h:
5851 * gst/rtsp-server/rtsp-session.h:
5852 * gst/rtsp-server/rtsp-stream-transport.h:
5853 * gst/rtsp-server/rtsp-stream.c:
5854 * gst/rtsp-server/rtsp-thread-pool.h:
5857 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5860 * examples/Makefile.am:
5861 configure: compile cgroup example conditionally
5862 Only compile the cgroup example when we have libcgroup
5864 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5867 * examples/Makefile.am:
5868 * examples/test-cgroups.c:
5869 examples: add cgroups example
5871 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5873 * tests/check/gst/rtspserver.c:
5874 tests: fix compilation
5876 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5878 * gst/rtsp-server/rtsp-thread-pool.c:
5879 thread-pool: fix vmethod invocation
5881 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5883 * gst/rtsp-server/rtsp-thread-pool.c:
5884 * gst/rtsp-server/rtsp-thread-pool.h:
5885 thread-pool: store thread type in thread
5887 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5889 * gst/rtsp-server/rtsp-client.c:
5890 client: pass thread from pool to media _prepare
5891 Get a thread from the configured threadpool and pass it to the prepare method of
5894 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5896 * gst/rtsp-server/rtsp-media.c:
5897 * gst/rtsp-server/rtsp-media.h:
5898 media: Accept a thread in _prepare
5899 Remove out own threadpool handling and use the provided thread and
5900 maincontext for the bus messages and the state changes.
5902 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5904 * gst/rtsp-server/rtsp-server.c:
5905 server: configure client thread pool
5907 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5909 * gst/rtsp-server/rtsp-client.c:
5910 * gst/rtsp-server/rtsp-client.h:
5911 client: add method to configure thread pool
5913 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5915 * gst/rtsp-server/rtsp-client.h:
5916 * gst/rtsp-server/rtsp-server.c:
5917 * gst/rtsp-server/rtsp-server.h:
5918 server: use thread pool
5919 Use the thread pool instead of doing our own thing.
5921 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5923 * gst/rtsp-server/Makefile.am:
5924 * gst/rtsp-server/rtsp-thread-pool.c:
5925 * gst/rtsp-server/rtsp-thread-pool.h:
5926 thread-pool: add object to manage threads
5927 Add an object to manage the client and media threads.
5929 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5931 * gst/rtsp-server/rtsp-auth.c:
5932 auth: debug authorization check
5934 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5936 * gst/rtsp-server/rtsp-media.c:
5937 media: start media pipeline in context
5938 Start the media pipeline in the provided context (or our default one
5939 when NULL). This makes sure that we run the bus thread in this context and that
5940 all media threads are children of this context.
5942 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5944 * gst/rtsp-server/rtsp-media-factory.c:
5945 factory: pass permissions to media by default
5947 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5949 * examples/test-auth.c:
5950 test: add permissions to auth test
5951 Ass some permissions to the media factory in the test.
5953 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5955 * gst/rtsp-server/rtsp-auth.c:
5956 * gst/rtsp-server/rtsp-auth.h:
5957 * gst/rtsp-server/rtsp-client.c:
5958 auth: simplify auth checks
5959 Remove client from methods, it's now in the state
5960 Perform the check specified by the string, use the information from the
5961 thread local context.
5963 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5965 * gst/rtsp-server/rtsp-client.c:
5966 * gst/rtsp-server/rtsp-client.h:
5967 client: add state to current thread
5968 Add the client to the ClientState object.
5969 Place the ClientState on the current thread.
5971 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5973 * gst/rtsp-server/rtsp-media-factory.c:
5974 * gst/rtsp-server/rtsp-media-factory.h:
5975 * gst/rtsp-server/rtsp-media.c:
5976 * gst/rtsp-server/rtsp-media.h:
5977 media: make it possible to set permissions
5978 Make it possible to set permissions on media and media factory objects
5980 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5982 * gst/rtsp-server/Makefile.am:
5983 * gst/rtsp-server/rtsp-permissions.c:
5984 * gst/rtsp-server/rtsp-permissions.h:
5985 permissions: add permissions object
5986 Add a mini object to store permissions based on a role.
5988 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5990 * examples/test-auth.c:
5991 * gst/rtsp-server/rtsp-auth.c:
5992 * gst/rtsp-server/rtsp-auth.h:
5993 * gst/rtsp-server/rtsp-client.c:
5994 auth: add auth checks
5995 Add an enum with auth checks and implement the checks in the auth object.
5996 Perform the checks from the client.
5998 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6000 * examples/test-auth.c:
6001 * gst/rtsp-server/rtsp-auth.c:
6002 * gst/rtsp-server/rtsp-auth.h:
6003 * gst/rtsp-server/rtsp-client.h:
6004 auth: use the token after authentication
6005 After we authenticated a user, keep the Token around in the state.
6007 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6009 * gst/rtsp-server/rtsp-client.c:
6010 * gst/rtsp-server/rtsp-media.c:
6011 * gst/rtsp-server/rtsp-media.h:
6012 * tests/check/gst/media.c:
6013 media: add optional context for bus messages
6014 Add an optional mainloop to _prepare that will handle the bus messages instead
6015 of always using the shared mainloop.
6017 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6019 * gst/rtsp-server/Makefile.am:
6020 * gst/rtsp-server/rtsp-token.c:
6021 * gst/rtsp-server/rtsp-token.h:
6022 token: add authorization token
6023 Add a simply miniobject that contains the authorizations. The object contains a
6024 GstStructure that hold all authorization fields. When a user is authenticated,
6025 the auth module will create a Token for the user. The token is then used to
6026 check what operations the user is allowed to do and various other configuration
6029 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6031 * examples/test-auth.c:
6032 * gst/rtsp-server/rtsp-auth.c:
6033 * gst/rtsp-server/rtsp-auth.h:
6034 * gst/rtsp-server/rtsp-client.c:
6035 * gst/rtsp-server/rtsp-client.h:
6036 * gst/rtsp-server/rtsp-media-factory.c:
6037 * gst/rtsp-server/rtsp-media-factory.h:
6038 * gst/rtsp-server/rtsp-media.c:
6039 * gst/rtsp-server/rtsp-media.h:
6040 auth: remove auth from media and factory
6041 Remove the auth object from media and factory. We want to have the RTSPClient
6042 authenticate and authorize resources, there is no need to place another auth
6043 manager on the media/factory.
6045 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6047 * examples/test-auth.c:
6048 * gst/rtsp-server/rtsp-auth.c:
6049 * gst/rtsp-server/rtsp-auth.h:
6050 * gst/rtsp-server/rtsp-client.h:
6051 auth: add support for multiple basic auth tokens
6052 Make it possible to add multiple basic authorisation tokens to one authorization
6053 object. Associate with each token an authorization group that will define what
6054 capabilities are allowed.
6056 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6058 * gst/rtsp-server/rtsp-client.c:
6059 client: error out on non-aggregate control
6060 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
6062 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6064 * gst/rtsp-server/rtsp-client.c:
6065 client: rework setup request a little
6066 Cache the media in DESCRIBE based on the longest matching path with the uri
6067 that we can find in the mount points.
6068 Rework the setup request a little to get the media from the session or from
6069 the longest matching path, this way we can derive the control string as
6070 everything after the path instead of hardcoding it.
6071 Find the stream based on the control string and only open a session when all
6074 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6076 * gst/rtsp-server/rtsp-media.c:
6077 * gst/rtsp-server/rtsp-media.h:
6078 media: add method to find a stream by control url
6080 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6082 * gst/rtsp-server/rtsp-stream.c:
6083 * gst/rtsp-server/rtsp-stream.h:
6084 stream: add method to check control url of stream
6086 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6088 * gst/rtsp-server/rtsp-client.c:
6089 * gst/rtsp-server/rtsp-session-media.c:
6090 * gst/rtsp-server/rtsp-session-media.h:
6091 * gst/rtsp-server/rtsp-session.c:
6092 * gst/rtsp-server/rtsp-session.h:
6093 session: use path matching for session media
6094 Use a path string instead of a uri to lookup session media in the sessions. Also
6095 use path matching to find the largest possible path that matches.
6097 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6099 * gst/rtsp-server/rtsp-client.c:
6100 * gst/rtsp-server/rtsp-mount-points.c:
6101 * gst/rtsp-server/rtsp-mount-points.h:
6102 * tests/check/gst/mountpoints.c:
6103 mount-points: remove useless vmethod
6104 Making lookups in the mount points should not be done with a URL, if there is a
6105 mapping to be done from URL to mount points, we'll need to do it somewhere
6108 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6110 * gst/rtsp-server/rtsp-mount-points.c:
6111 * gst/rtsp-server/rtsp-mount-points.h:
6112 * tests/check/gst/mountpoints.c:
6113 mount-points: improve mount point searching
6114 Use a GSequence to keep track of the mount points.
6115 Match a URL to the longest matching registered mount point. This should be the
6116 URL to perform aggreagate control and the remainder is the stream specific
6118 Add some unit tests for this.
6120 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
6122 * gst/rtsp-server/Makefile.am:
6123 rtsp-server: Allow building of static library
6125 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6127 * tests/check/gst/mediafactory.c:
6128 tests: fix compilation
6130 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6132 * gst/rtsp-server/rtsp-sdp.c:
6133 sdp: get control string from stream
6134 Use the control string as configured in the stream.
6136 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6138 * gst/rtsp-server/rtsp-stream.c:
6139 * gst/rtsp-server/rtsp-stream.h:
6140 stream: add methods and property to set control string
6142 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6144 * gst/rtsp-server/rtsp-client.c:
6146 Rename variables for clarity
6147 Keep media in state when we can
6149 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6151 * gst/rtsp-server/rtsp-client.c:
6152 * gst/rtsp-server/rtsp-stream.c:
6153 * gst/rtsp-server/rtsp-stream.h:
6154 stream: add more support for IPv6
6155 Rename _get_address to _get_multicast_address in GstRTSPStream to
6156 make it clear that this function only deals with multicast.
6157 Make it possible to have both an IPv4 and IPv6 multicast address on
6158 a stream. Give the client an IPv4 or IPv6 address depending on the
6159 address it used to connect to the server.
6160 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
6162 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6164 * gst/rtsp-server/rtsp-client.c:
6167 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6169 * gst/rtsp-server/rtsp-stream.c:
6170 stream: handle failed port allocation
6171 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
6172 can't allocate any family at all. Also keep track of what port families we
6174 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
6176 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6178 * gst/rtsp-server/rtsp-stream.c:
6179 stream: improve docs
6181 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6183 * gst/rtsp-server/rtsp-stream-transport.c:
6184 stream-transport: remove old if 0 block
6186 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
6188 * tests/check/gst/client.c:
6190 gst_rtsp_client_get_uri() has been removed
6191 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
6193 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6195 * gst/rtsp-server/rtsp-client.c:
6196 * gst/rtsp-server/rtsp-client.h:
6197 client: add method to filter managed sessions
6198 Add a method to filter the sessions managed by this client connection.
6199 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
6201 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6203 * gst/rtsp-server/rtsp-client.c:
6204 * gst/rtsp-server/rtsp-client.h:
6205 client: remove _get_uri() method
6206 Remove the get_uri() method on the client. A client has no uri, the uri
6207 property is an internal property to manage the last cached media for
6210 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6212 * gst/rtsp-server/rtsp-media-factory.h:
6213 media-factory: fix typo
6215 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
6217 * gst/rtsp-server/rtsp-media.c:
6218 rtsp-media: Do not leak the query in default_query_stop
6219 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
6221 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6223 * gst/rtsp-server/rtsp-media.c:
6224 media: don't unlock when conversion fails
6225 Don't unlock the state lock when conversion fails because it was not locked.
6227 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6229 * gst/rtsp-server/rtsp-media.c:
6230 * gst/rtsp-server/rtsp-media.h:
6231 Add query_position and query_stop vmethods to rtsp-media
6233 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6235 * gst/rtsp-server/rtsp-media.c:
6236 Fix typo in property install for rtsp-media's time-provider
6238 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6240 * gst/rtsp-server/rtsp-client.c:
6241 * gst/rtsp-server/rtsp-client.h:
6242 client: clean some variables
6243 Clean some variables and add some guards to _send_request()
6245 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6247 * gst/rtsp-server/rtsp-client.c:
6248 * gst/rtsp-server/rtsp-client.h:
6249 Add gst_rtsp_client_send_request API
6250 This makes it possible to send arbitrary messages to a client, such as
6251 SET_PARAMETER or GET_PARAMETER
6253 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6255 * gst/rtsp-server/rtsp-media.c:
6256 * gst/rtsp-server/rtsp-media.h:
6257 media: add _get_element() method
6258 Add method to get the element used when creating the media.
6259 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
6261 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6263 * gst/rtsp-server/rtsp-media.c:
6266 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
6268 * gst/rtsp-server/rtsp-stream.c:
6269 * gst/rtsp-server/rtsp-stream.h:
6270 stream: allow access to the rtp session
6271 https://bugzilla.gnome.org/show_bug.cgi?id=703004
6273 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
6275 * gst/rtsp-server/rtsp-stream.c:
6276 * gst/rtsp-server/rtsp-stream.h:
6277 dscp qos support in gst-rtsp-stream
6278 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
6280 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6282 * tests/check/gst/rtspserver.c:
6284 Actually do what the comment says. Also keep the old code around, not sure what
6285 should happen when you get a 454 from a TEARDOWN, does it close the connection?
6286 it currently doesn't.
6288 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6290 * gst/rtsp-server/rtsp-client.c:
6291 client: also watch newly created session
6292 When we newly created a session, start watching it immediately instead of
6293 on the next request.
6295 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
6297 * tests/check/gst/client.c:
6298 tests: add unit test for new-session
6299 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
6301 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6303 * gst/rtsp-server/rtsp-client.c:
6304 client: emit new-session when new session is created
6305 Only emit new-session when we created a new session for a client, not when a
6306 client picked up a previous session.
6307 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
6309 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
6311 * gst/rtsp-server/rtsp-client.c:
6312 client: handle asterisk as path in requests
6313 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
6315 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6317 * gst/rtsp-server/rtsp-media.c:
6318 media: handle segment query format mismatch
6319 It's possible that the segment query returns with a different format than what
6320 we asked for, handle this case also.
6322 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
6324 * gst/rtsp-server/rtsp-media.c:
6325 media: use segment stop in collect_media_stats
6326 Use segment stop instead of duration as range end point.
6327 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
6329 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
6331 * gst/rtsp-server/rtsp-media.c:
6332 * tests/check/gst/media.c:
6333 rtsp-media: Do not leak the element in take_pipeline
6334 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
6336 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
6338 * gst/rtsp-server/rtsp-client.c:
6339 * gst/rtsp-server/rtsp-client.h:
6340 rtsp-client: Make configure_client_transport virtual
6341 This patch makes configure_client_transport virtual. The functionality is
6342 needed to handle some weird clients sending multicast transport settings as url
6344 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
6346 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
6348 * gst/rtsp-server/rtsp-client.c:
6349 * gst/rtsp-server/rtsp-client.h:
6350 rtsp-client: Make param_set and param_get virtual
6351 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
6353 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
6355 * gst/rtsp-server/rtsp-client.c:
6356 * gst/rtsp-server/rtsp-media.c:
6357 * gst/rtsp-server/rtsp-media.h:
6358 media: convert_range replaces get_range_times
6359 get_range_times worked for handling UTC ranges for seeks, but we also
6360 need to convert back from NPT to the requested unit in
6361 get_range_string. convert_range is now used for both.
6362 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
6364 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6366 * gst/rtsp-server/rtsp-client.c:
6367 * gst/rtsp-server/rtsp-sdp.c:
6368 * gst/rtsp-server/rtsp-sdp.h:
6369 sdp: cleanup sdp info
6370 We don't need to pass the proto, we can more easily check a boolean.
6371 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
6373 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
6375 * gst/rtsp-server/rtsp-sdp.c:
6376 use 0.0.0.0 or :: for c= line instead of server address
6378 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
6380 * gst/rtsp-server/rtsp-client.c:
6381 use local address, not remote, in SDP
6382 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
6384 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6387 Automatic update of common submodule
6388 From 098c0d7 to 01a7a46
6390 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
6392 * gst/rtsp-server/rtsp-media.c:
6393 * gst/rtsp-server/rtsp-media.h:
6394 media: possibility to override range time conversion
6395 Make it possible to override the conversion from GstRTSPTimeRange to
6396 GstClockTimes, that is done before seeking on the media
6397 pipeline. Overriding can be useful for UTC ranges, where the default
6398 conversion gives nanoseconds since 1900.
6399 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
6401 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
6403 * gst/rtsp-server/rtsp-server.c:
6404 * gst/rtsp-server/rtsp-server.h:
6405 rtsp-server: Expose the use_client_settings API
6406 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
6408 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
6410 * gst/rtsp-server/rtsp-client.c:
6411 * gst/rtsp-server/rtsp-stream.c:
6412 * gst/rtsp-server/rtsp-stream.h:
6413 rtspstream: handle both ipv4 and ipv6 clients
6414 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
6416 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6418 * gst/rtsp-server/rtsp-sdp.c:
6419 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
6420 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
6421 We already have a way to place extra attributes in the SDP by using a string
6422 property with prefix x- or a- in the caps.
6424 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6426 * gst/rtsp-server/rtsp-sdp.c:
6427 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
6428 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
6429 We already have a way to place extra attributes in the SDP, just make a string
6430 property in the payloader with a- or x- prefix.
6432 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6434 * gst/rtsp-server/rtsp-sdp.c:
6435 rtsp: place a- and x- properties as attributes
6436 application/x-rtp has properties with a- and x- prefixes that should be
6437 placed as attributes in the SDP for the media instead of being added to the
6440 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6442 * examples/Makefile.am:
6443 * examples/test-video.c:
6444 example: add TLS example
6446 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6448 * gst/rtsp-server/rtsp-server.c:
6449 * gst/rtsp-server/rtsp-server.h:
6450 server: add support for TLS
6451 Add methods to set and get a TLS certificate.
6452 Add vmethod to configure a new connection. By default, configure the TLS
6453 certificate in a new connection if needed.
6455 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6457 * gst/rtsp-server/rtsp-server.c:
6458 * gst/rtsp-server/rtsp-server.h:
6459 server: remove accept_client vmethod
6460 This vmethod is not very useful so remove it.
6462 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6464 * gst/rtsp-server/rtsp-server.c:
6465 server: don't crash on NULL GError
6467 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
6469 * gst/rtsp-server/rtsp-session-pool.c:
6470 rtsp-session-pool: corrected session timeout detection
6471 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
6473 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6475 * gst/rtsp-server/rtsp-client.c:
6476 client: improve debug
6478 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6480 * gst/rtsp-server/rtsp-client.c:
6481 * gst/rtsp-server/rtsp-client.h:
6482 * gst/rtsp-server/rtsp-server.c:
6483 server: refactor connection setup
6484 Let the server accept the socket connection and construct a GstRTSPConnection
6485 from it. Remove the code from the client and let the client only deal with
6486 a fully configure GstRTSPConnection object.
6487 We will need this later when the server will configure the connection for
6490 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6492 * gst/rtsp-server/rtsp-stream.c:
6493 stream: keep the transport object alive
6494 Keep the transport object alive while we have it as qdata on the
6497 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
6499 * gst/rtsp-server/rtsp-client.c:
6500 * gst/rtsp-server/rtsp-server.c:
6501 rtsp-server: Do not crash on nmapping of server
6502 * generate error when gst_rtsp_connection_accept fails
6503 * do not stop accepting incoming connections because
6504 accepting a client fails
6505 https://bugzilla.gnome.org/show_bug.cgi?id=701072
6507 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
6509 * gst/rtsp-server/rtsp-client.c:
6510 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
6511 https://bugzilla.gnome.org/show_bug.cgi?id=700953
6513 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
6515 * gst/rtsp-server/rtsp-sdp.c:
6516 rtsp-sdp: Parse framerate caps field and set SDP attribute
6517 The SDP attribute and its format is described in RFC4566.
6518 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
6520 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
6522 * gst/rtsp-server/rtsp-sdp.c:
6523 rtsp-sdp: Parse width/height from caps and set SDP attribute
6524 The SDP attribute and its format is described in RFC6064.
6525 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
6527 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
6529 * gst/rtsp-server/rtsp-sdp.c:
6530 * tests/check/gst/client.c:
6531 rtsp-sdp: add bandwidth line
6532 https://bugzilla.gnome.org/show_bug.cgi?id=699220
6534 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6537 Automatic update of common submodule
6538 From 5edcd85 to 098c0d7
6540 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
6542 * tests/check/gst/media.c:
6543 tests: add dynamic payloader prepare/unprepare check
6545 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6547 * gst/rtsp-server/rtsp-media.c:
6548 media: release lock when removing fakesink
6550 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6552 * gst/rtsp-server/rtsp-stream.c:
6553 stream: set elements to NULL before removing
6554 When removing a stream, set the elements to NULL first. This avoids
6555 element-is-not-in-NULL-state errors when we dispose the elements.
6557 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
6560 Automatic update of common submodule
6561 From 3cb3d3c to 5edcd85
6563 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6565 * gst/rtsp-server/rtsp-media.c:
6566 * gst/rtsp-server/rtsp-media.h:
6567 media: listen to pad-removed signals
6568 Listen to the pad-removed signal and remove the stream associated with the
6570 Add signal to be notified of the removed pad.
6571 Remove the fakesink in unprepare()
6572 Fix signatures of the signal methods
6574 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6576 * examples/test-sdp.c:
6577 tests: add example of reusable pipelines
6579 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
6581 * gst/rtsp-server/rtsp-stream.c:
6582 * gst/rtsp-server/rtsp-stream.h:
6583 stream: add method to get the srcpad
6585 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
6587 * tests/check/gst/media.c:
6588 check: add media prepare/unprepare test
6589 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
6591 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
6593 * gst/rtsp-server/rtsp-media.c:
6594 media: disconnect from signal handlers in unprepare()
6595 We connected to the pad-added and no-more-pads signals in prepare() so
6596 we need to disconnect from them in unprepare().
6597 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
6599 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
6601 * gst/rtsp-server/rtsp-media.c:
6602 media: don't free streams array
6603 Don't free the streams array in the unprepare() method, they were not
6605 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
6607 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
6609 * gst/rtsp-server/rtsp-media.c:
6610 media: don't unref the pipeline in unprepare
6611 Unprepare() should undo what prepare() does. Because the pipeline is
6612 not created in prepare(), we should not unref it in unprepare()
6614 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
6616 * gst/rtsp-server/rtsp-stream.c:
6617 stream: clear session and caps for reuse
6618 Set the session and caps to NULL after unref otherwise we might unref
6620 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
6622 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
6624 * gst/rtsp-server/rtsp-client.c:
6625 client: send out teardown signal before tearing down
6626 The advantage is that in the signal handler you get direct access to
6627 information about what streams are about to get torn down (in the
6628 GstRTSPClientState).
6629 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
6631 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
6633 * gst/rtsp-server/rtsp-client.c:
6634 * gst/rtsp-server/rtsp-client.h:
6635 client: expose connection
6636 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
6638 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
6641 Automatic update of common submodule
6642 From aed87ae to 3cb3d3c
6644 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6646 * gst/rtsp-server/rtsp-media.c:
6647 * gst/rtsp-server/rtsp-media.h:
6648 * gst/rtsp-server/rtsp-session-media.c:
6649 * gst/rtsp-server/rtsp-session-media.h:
6650 media: add method to get the base_time of the pipeline
6651 Together with a shared clock, this base-time could eventually be sent to
6652 the client so that it can reconstruct the exact running-time of the clock
6655 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6657 * gst/rtsp-server/Makefile.am:
6658 * gst/rtsp-server/rtsp-media.c:
6659 * gst/rtsp-server/rtsp-media.h:
6660 * gst/rtsp-server/rtsp-sdp.c:
6661 media: add GstNetTimeProvider support
6662 Add a property to let the media provide a GstNetTimeProvider for its clock.
6663 Make methods to get the clock and nettimeprovider
6664 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
6665 provider and also the current time of the clock. This should make it possible
6666 for (GStreamer) clients to slave their clock to the server clock.
6668 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
6671 Automatic update of common submodule
6672 From 04c7a1e to aed87ae
6674 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6676 * gst/rtsp-server/rtsp-media.c:
6677 media: wait for buffering to complete
6678 Wait for buffering to complete before changing the state to the target state.
6680 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6682 * gst/rtsp-server/rtsp-media.c:
6683 media: small cleanup
6685 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
6687 * tests/check/gst/rtspserver.c:
6688 tests: remove extra unref in test_setup_non_existing_stream
6689 The unref is not needed anymore, teardown runs without it.
6690 https://bugzilla.gnome.org/show_bug.cgi?id=696542
6692 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
6694 * tests/check/gst/rtspserver.c:
6695 tests: GSocketService cleanup in test_bind_already_in_use
6696 Use g_socket_service_stop so the rtspserver test stops listening for
6697 incoming connections in test_bind_already_in_use.
6698 https://bugzilla.gnome.org/show_bug.cgi?id=696541
6700 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
6702 * gst/rtsp-server/rtsp-media-factory.c:
6703 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
6704 Instead use a GWeakRef which is safe to use
6705 This is a known GLib bug, see:
6706 https://bugzilla.gnome.org/show_bug.cgi?id=667145
6708 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
6710 * gst/rtsp-server/rtsp-client.c:
6711 * gst/rtsp-server/rtsp-media.c:
6712 * gst/rtsp-server/rtsp-media.h:
6713 * gst/rtsp-server/rtsp-sdp.c:
6714 * tests/check/gst/media.c:
6715 * tests/check/gst/rtspserver.c:
6716 rtsp-media/client: Reply to PLAY request with same type of Range
6717 Remember the type of Range from the PLAY request and use the same type for
6720 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
6722 * gst/rtsp-server/rtsp-client.c:
6723 * gst/rtsp-server/rtsp-client.h:
6724 * tests/check/gst/client.c:
6725 rtsp-client: expose uri
6727 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
6729 * tests/check/gst/mediafactory.c:
6730 tests: Hold ref while creating second media
6731 To test if the media aren't shared, make sure we keep the first one while creating a second
6732 otherwise the same memory address may be reused.
6734 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
6737 configure: remove out-of-date comment
6739 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
6742 .gitignore: ignore more build files
6744 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
6746 * tests/check/Makefile.am:
6747 tests: use right _LIBS variable for gst-plugins-base libs
6749 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6751 * tests/check/Makefile.am:
6752 check: add librtp to libs
6754 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
6756 * tests/check/gst/rtspserver.c:
6757 tests: Add test to check selecting a port the server will send from
6759 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
6761 * tests/check/gst/rtspserver.c:
6762 tests: Make sure packets are actually received
6764 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
6766 * gst/rtsp-server/rtsp-stream.c:
6767 stream: Select unicast address from pool if appropriate
6769 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
6771 * gst/rtsp-server/rtsp-stream.c:
6772 stream: Properties are always there in Gst 1.0
6774 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
6776 * tests/check/gst/addresspool.c:
6777 tests: Add tests for unicast addresses in pool
6779 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
6781 * gst/rtsp-server/rtsp-address-pool.c:
6782 * tests/check/gst/addresspool.c:
6783 address-pool: Verify that multicast addresses are used for multicast and vice-versa
6785 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
6787 * docs/libs/gst-rtsp-server-sections.txt:
6788 * gst/rtsp-server/rtsp-address-pool.c:
6789 * gst/rtsp-server/rtsp-address-pool.h:
6790 * gst/rtsp-server/rtsp-stream.c:
6791 * tests/check/gst/addresspool.c:
6792 address-pool: Add unicast addresses
6794 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
6797 * gst/rtsp-server/rtsp-server.c:
6798 * tests/check/gst/rtspserver.c:
6799 rtsp-server: Limit the number of threads per server instance
6800 If we exceed the maximum, just round robin the clients over the existing
6803 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
6805 * gst/rtsp-server/rtsp-server.c:
6806 rtsp-server: No need to store the GMainContext in the client context
6808 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
6810 * tests/check/gst/rtspserver.c:
6811 tests: Add test for client disconnection
6813 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
6815 * tests/check/gst/rtspserver.c:
6816 tests: Test client and session timeouts with multiple threads
6818 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
6820 * gst/rtsp-server/rtsp-address-pool.c:
6821 * gst/rtsp-server/rtsp-auth.c:
6822 * gst/rtsp-server/rtsp-client.c:
6823 * gst/rtsp-server/rtsp-media-factory-uri.c:
6824 * gst/rtsp-server/rtsp-media-factory.c:
6825 * gst/rtsp-server/rtsp-media.c:
6826 * gst/rtsp-server/rtsp-mount-points.c:
6827 * gst/rtsp-server/rtsp-server.c:
6828 * gst/rtsp-server/rtsp-session-media.c:
6829 * gst/rtsp-server/rtsp-session-pool.c:
6830 * gst/rtsp-server/rtsp-session.c:
6831 Document locking and its order
6833 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
6835 * tests/check/gst/rtspserver.c:
6836 tests: Test that slow DESCRIBE don't block other clients
6838 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
6840 * tests/check/gst/client.c:
6841 tests: Add tests for client-requested multicast address
6843 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
6845 * docs/libs/gst-rtsp-server-sections.txt:
6846 docs: Put the various functions in the right sections
6848 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
6850 * docs/libs/gst-rtsp-server-docs.sgml:
6851 * docs/libs/gst-rtsp-server-sections.txt:
6852 * gst/rtsp-server/rtsp-address-pool.c:
6853 * gst/rtsp-server/rtsp-address-pool.h:
6854 docs: Generate docs for GstRTSPAddressPool
6856 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
6858 * gst/rtsp-server/rtsp-client.c:
6859 * gst/rtsp-server/rtsp-stream.c:
6860 * gst/rtsp-server/rtsp-stream.h:
6861 client: Check client provided addresses against the address pool
6863 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
6865 * gst/rtsp-server/rtsp-address-pool.c:
6866 * gst/rtsp-server/rtsp-address-pool.h:
6867 * tests/check/gst/addresspool.c:
6868 address-pool: Add API to request a specific address from the pool
6869 Also add relevant unit tests.
6871 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
6873 * tests/check/gst/mediafactory.c:
6874 tests: Check the passing around of a RTSPAddressPool
6875 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
6876 way down to the stream.
6878 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
6880 * tests/check/gst/addresspool.c:
6881 tests: Add more tests for the address pool
6883 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
6885 * gst/rtsp-server/rtsp-address-pool.c:
6886 address-pool: Fix off by one error
6887 When splitting a port range, the port after a skip is not part of range.
6889 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
6892 Automatic update of common submodule
6893 From 2de221c to 04c7a1e
6895 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
6898 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
6899 AM_CONFIG_HEADER was removed in automake 1.13
6900 https://bugzilla.gnome.org/show_bug.cgi?id=693368
6902 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
6905 Automatic update of common submodule
6906 From a942293 to 2de221c
6908 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6910 * gst/rtsp-server/rtsp-client.c:
6911 client: make sure the watch exists while sending data
6912 Protect the send_func with a lock. This allows us to wait for sending
6913 to complete before changing the send_func and user_data. We add an
6914 extra ref to the watch to make sure that it remains valid during
6916 When closing the connection, set the send_func to NULL
6917 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
6919 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6921 * tests/check/Makefile.am:
6922 tests: use GST_*_1_0 environment variables everywhere
6923 The _1_0 suffixed environment variables override the
6924 non-suffixed ones, so if we're in an environment that
6925 sets the _1_0 suffixed ones, such as jhbuild, we need
6926 to set those to make sure ours actually always get
6929 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6932 Automatic update of common submodule
6933 From acb04d9 to a942293
6935 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6937 * gst/rtsp-server/rtsp-client.c:
6938 rtsp-client: set the client backlog
6939 Set the client backlog to a reasonable default
6941 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
6943 * gst/rtsp-server/rtsp-media.c:
6944 rtsp-media: Make the element a constructor parameter
6945 https://bugzilla.gnome.org/show_bug.cgi?id=689594
6947 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
6949 * docs/libs/Makefile.am:
6950 docs: Link with gcov library when gcov is enabled
6951 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
6953 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6955 * gst/rtsp-server/rtsp-media.c:
6956 media: match prepare with unprepare
6957 Really unprepare when there were an equal amount of prepare calls.
6959 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6961 * gst/rtsp-server/rtsp-media.c:
6962 media: media has to be unprepared in finalize
6963 Because unprepare takes away the last ref on the media.
6965 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6967 * gst/rtsp-server/rtsp-client.c:
6968 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
6969 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
6970 We can't use the refcount to trigger unprepare because it is the unprepare call
6971 that removes the last refcount after all messages are consumed. What we should
6972 probably do is make a prepared refcount and only unprepare when the refcount
6975 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6977 * gst/rtsp-server/rtsp-media.c:
6978 media: let the source unref the last media ref
6979 the last ref to the media is held by the source so we don't need to add more ref
6980 and unrefs, we simply destroy the media when the source is gone.
6982 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6984 * gst/rtsp-server/rtsp-media.c:
6985 media: improve debug
6987 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6989 * gst/rtsp-server/rtsp-media.c:
6991 Make sure we are in the right state when collecting the position and duration.
6992 Only make ourselves PREPARED when we were previously PREPARING.
6994 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6996 * gst/rtsp-server/rtsp-media.c:
6997 media: use g_object_ref/unref for GObjects
6999 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
7001 * gst/rtsp-server/rtsp-client.c:
7002 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
7003 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
7004 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
7005 isn't being used anymore.
7007 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
7009 * gst/rtsp-server/rtsp-media.c:
7010 Fix compiler warning
7012 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
7014 * gst/rtsp-server/rtsp-media-factory-uri.c:
7015 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
7017 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7019 * gst/rtsp-server/rtsp-session-media.h:
7022 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7024 * gst/rtsp-server/rtsp-media.c:
7025 * tests/check/gst/media.c:
7026 media: avoid element leak
7028 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7030 * gst/rtsp-server/rtsp-media.c:
7031 media: require an element in media constructor
7033 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7035 * gst/rtsp-server/rtsp-client.c:
7036 Revert "client: TEARDOWN brings that state to Init again"
7037 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
7038 The object is already disposed, there is no point in setting the state.
7040 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7042 * gst/rtsp-server/rtsp-client.c:
7043 client: TEARDOWN brings that state to Init again
7045 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7047 * docs/libs/gst-rtsp-server-sections.txt:
7048 * examples/test-auth.c:
7049 * gst/rtsp-server/rtsp-auth.c:
7050 * gst/rtsp-server/rtsp-auth.h:
7051 * gst/rtsp-server/rtsp-client.c:
7052 * gst/rtsp-server/rtsp-client.h:
7053 * gst/rtsp-server/rtsp-media-factory-uri.c:
7054 * gst/rtsp-server/rtsp-media-factory-uri.h:
7055 * gst/rtsp-server/rtsp-media-factory.c:
7056 * gst/rtsp-server/rtsp-media-factory.h:
7057 * gst/rtsp-server/rtsp-media.c:
7058 * gst/rtsp-server/rtsp-media.h:
7059 * gst/rtsp-server/rtsp-mount-points.c:
7060 * gst/rtsp-server/rtsp-mount-points.h:
7061 * gst/rtsp-server/rtsp-sdp.c:
7062 * gst/rtsp-server/rtsp-server.c:
7063 * gst/rtsp-server/rtsp-server.h:
7064 * gst/rtsp-server/rtsp-session-media.c:
7065 * gst/rtsp-server/rtsp-session-media.h:
7066 * gst/rtsp-server/rtsp-session-pool.c:
7067 * gst/rtsp-server/rtsp-session-pool.h:
7068 * gst/rtsp-server/rtsp-session.c:
7069 * gst/rtsp-server/rtsp-session.h:
7070 * gst/rtsp-server/rtsp-stream-transport.c:
7071 * gst/rtsp-server/rtsp-stream-transport.h:
7072 * gst/rtsp-server/rtsp-stream.c:
7073 * gst/rtsp-server/rtsp-stream.h:
7074 * tests/check/gst/media.c:
7075 rtsp: make object details private
7076 Make all object details private
7077 Add methods to access private bits
7079 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7081 * tests/check/Makefile.am:
7082 * tests/check/gst/media.c:
7083 tests: add media tests
7085 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7087 * gst/rtsp-server/rtsp-media.c:
7088 media: check if prepared for some methods
7089 Check that the media object is prepared before doing seek and getting the
7090 current position etc.
7091 Add some g_return checks.
7093 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7095 * tests/check/Makefile.am:
7096 * tests/check/gst/mediafactory.c:
7097 tests: add mediafactory test
7099 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7101 * gst/rtsp-server/rtsp-stream.c:
7102 stream: improve debug
7104 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7106 * gst/rtsp-server/rtsp-media.c:
7107 * gst/rtsp-server/rtsp-media.h:
7108 media: unref pipeline in finalize to avoid leaking it
7110 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7112 * gst/rtsp-server/rtsp-media-factory-uri.c:
7113 * gst/rtsp-server/rtsp-media.c:
7114 rtsp: use gst_object_unref on GstObjects
7116 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7118 * gst/rtsp-server/rtsp-media-factory.c:
7119 media-factory: require an url
7121 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7123 * examples/test-uri.c:
7124 examples: fix include
7126 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7128 * gst/rtsp-server/rtsp-server.h:
7129 server: remove unused include
7131 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7133 * tests/check/Makefile.am:
7134 * tests/check/gst/mountpoints.c:
7135 tests: add test for mountpoints
7137 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7139 * gst/rtsp-server/rtsp-client.c:
7140 client: fix factory leak
7141 Keep the factory in the state object only for authorization checks and make
7142 sure we unref it on failure. Also don't keep invalid objects in the state
7145 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7147 * gst/rtsp-server/rtsp-mount-points.c:
7148 mounts: add g_return_if guards
7150 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7152 * tests/check/gst/client.c:
7153 tests: add more tests
7155 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7157 * gst/rtsp-server/rtsp-client.c:
7158 client: improve debug
7160 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7162 * gst/rtsp-server/rtsp-client.c:
7163 client: improve debug and fix leaks
7164 Cleanup the uri and session when there is a bad request.
7166 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7171 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7173 * tests/check/gst/client.c:
7174 test: add test for session in options request
7176 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7178 * gst/rtsp-server/rtsp-client.c:
7179 client: use 454 when session can't be found
7180 We should use 454 when a session can't be found because there was no session
7181 pool configured in the server. This is not a server configuration problem
7182 because the server on which the request is done might not be the same one that
7183 will keep the sessions for us and so it does not need to support sessions.
7185 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7187 * gst/rtsp-server/rtsp-client.c:
7188 client: only free connection when there is one
7189 It's possible that the client doesn't have a connection when we try to free it.
7191 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7193 * tests/check/Makefile.am:
7194 * tests/check/gst/client.c:
7195 tests: add unit test for the client object
7197 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7199 * gst/rtsp-server/rtsp-client.c:
7200 client: small cleanup
7202 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7204 * gst/rtsp-server/rtsp-client.h:
7205 client: remove unused include
7207 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7209 * gst/rtsp-server/rtsp-client.c:
7210 client: fix compilation
7212 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7214 * gst/rtsp-server/rtsp-client.c:
7215 client: call destroy without the lock
7217 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7219 * gst/rtsp-server/rtsp-client.c:
7220 * gst/rtsp-server/rtsp-client.h:
7221 client: make the client usable without a socket
7222 Make a method to let the client handle a message and a callback when the client
7223 wants us to send a response message back. This makes it possible to also use the
7224 client object without the sockets, which should make it easier to test.
7226 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7228 * gst/rtsp-server/rtsp-client.c:
7229 * gst/rtsp-server/rtsp-client.h:
7230 client: small cleanup
7232 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7234 * docs/libs/gst-rtsp-server-sections.txt:
7235 * gst/rtsp-server/rtsp-client.c:
7236 * gst/rtsp-server/rtsp-client.h:
7237 * gst/rtsp-server/rtsp-server.c:
7238 client: remove reference to server
7239 We don't need to keep a ref to the server
7241 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7243 * gst/rtsp-server/rtsp-client.c:
7244 * gst/rtsp-server/rtsp-client.h:
7246 Also add some g_return_if()
7248 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7250 * gst/rtsp-server/rtsp-client.c:
7251 client: log more errors
7253 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7255 * gst/rtsp-server/rtsp-client.c:
7256 client: fix compilation
7258 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7260 * gst/rtsp-server/rtsp-client.c:
7261 * gst/rtsp-server/rtsp-client.h:
7262 client: add generic close-after-send support
7263 Add a property to send_response() to close the connection after the response has
7264 been sent to the client.
7266 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7269 * docs/libs/gst-rtsp-server-docs.sgml:
7270 * docs/libs/gst-rtsp-server-sections.txt:
7271 * docs/libs/gst-rtsp-server.types:
7272 * examples/test-auth.c:
7273 * examples/test-launch.c:
7274 * examples/test-mp4.c:
7275 * examples/test-multicast.c:
7276 * examples/test-multicast2.c:
7277 * examples/test-ogg.c:
7278 * examples/test-readme.c:
7279 * examples/test-sdp.c:
7280 * examples/test-uri.c:
7281 * examples/test-video.c:
7282 * gst/rtsp-server/Makefile.am:
7283 * gst/rtsp-server/rtsp-auth.h:
7284 * gst/rtsp-server/rtsp-client.c:
7285 * gst/rtsp-server/rtsp-client.h:
7286 * gst/rtsp-server/rtsp-media-mapping.c:
7287 * gst/rtsp-server/rtsp-media-mapping.h:
7288 * gst/rtsp-server/rtsp-mount-points.c:
7289 * gst/rtsp-server/rtsp-mount-points.h:
7290 * gst/rtsp-server/rtsp-server.c:
7291 * gst/rtsp-server/rtsp-server.h:
7292 * gst/rtsp-server/rtsp-session-media.c:
7293 * gst/rtsp-server/rtsp-session-pool.c:
7294 * gst/rtsp-server/rtsp-session-pool.h:
7295 * tests/check/gst/rtspserver.c:
7296 MediaMapping -> MountPoints
7297 Describes better what the object manages.
7299 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7302 configure: bump required version of -base
7304 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7306 * gst/rtsp-server/rtsp-media.c:
7309 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7311 * gst/rtsp-server/rtsp-media.c:
7312 * gst/rtsp-server/rtsp-media.h:
7313 media: support more Range formats
7314 Use the new -base methods to convert the Range string into a seek start and stop
7317 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7319 * examples/test-launch.c:
7320 examples: fix whitespace
7322 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7324 * examples/test-auth.c:
7325 test-auth: add example of how to remove sessions
7326 Add an example of the session filter api.
7328 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7330 * examples/test-uri.c:
7331 test-uri: remove mapping example
7333 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7335 * examples/test-uri.c:
7336 test-uri: fix callback signature
7338 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7340 * gst/rtsp-server/rtsp-media-factory.c:
7341 factory: keep ref to factory while media active
7342 While the media from a factory is alive, keep a ref to the factory.
7343 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
7345 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7347 * gst/rtsp-server/rtsp-media-factory-uri.c:
7348 factory-uri: add some debug
7350 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7352 * gst/rtsp-server/rtsp-stream.c:
7353 stream: set udp sources to PLAYING
7354 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
7355 so that it doesn't cause our pipeline to produce ASYNC-DONE.
7357 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7359 * gst/rtsp-server/rtsp-media-factory-uri.c:
7360 factory-uri: take ref to factory
7361 Take a ref to the factory that we place in our list.
7363 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7365 * tests/Makefile.am:
7366 * tests/test-reuse.c:
7367 test: add test for server reuse
7368 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
7370 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
7372 * gst/rtsp-server/rtsp-server.c:
7373 server: start and stop multiple times
7374 Stop listening on the RTSP port when the GSource is removed, so clients
7375 can't connect and the server can be started again.
7376 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
7378 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7380 * gst/rtsp-server/rtsp-server.c:
7381 server: fix small leak
7383 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7385 * gst/rtsp-server/rtsp-media.c:
7386 media: unref source in finish_unprepare
7387 The source is created in prepare, unref it in finish_unprepare.
7388 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
7390 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
7392 * gst/rtsp-server/rtsp-client.c:
7393 * gst/rtsp-server/rtsp-media.c:
7394 rtsp-media: remove bus watch before finalizing
7395 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
7396 * An extra media ref is added for the bus watch. This extra ref is unreffed by
7397 the GDestroyNotify function.
7398 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
7399 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
7400 gst_rtsp_media_unprepare before unreffing the media.
7401 This way, the bus watch will be removed before the media is finalized.
7402 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
7404 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
7406 * gst/rtsp-server/rtsp-client.c:
7407 * gst/rtsp-server/rtsp-client.h:
7408 client: wait until the TEARDOWN response is sent to close the connection
7409 Responses can be sent async so we need to wait until the TEARDOWN response has
7410 been written before we close the connection to the client. This avoids the risk
7411 of writing/polling closed sockets.
7412 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
7414 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
7416 * gst/rtsp-server/rtsp-stream.c:
7417 rtsp-stream: plug socket leak
7418 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
7420 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
7423 Automatic update of common submodule
7424 From 6bb6951 to a72faea
7426 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
7428 * gst/rtsp-server/rtsp-media-factory-uri.c:
7429 rtsp-server: don't use deprecated API
7431 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
7433 * gst/rtsp-server/rtsp-client.c:
7434 rtsp-client: fix unused-but-set-variable compiler warning
7435 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
7437 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7440 * docs/libs/gst-rtsp-server-sections.txt:
7441 * gst/rtsp-server/rtsp-client.c:
7444 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7446 * examples/Makefile.am:
7447 * examples/test-multicast2.c:
7448 examples: add another multicast example
7449 Add an example for how to configure separate multicast ranges for each media
7452 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7454 * examples/test-multicast.c:
7457 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7459 * gst/rtsp-server/rtsp-client.c:
7460 * gst/rtsp-server/rtsp-media.c:
7461 * gst/rtsp-server/rtsp-session-media.c:
7462 * gst/rtsp-server/rtsp-session-media.h:
7463 * gst/rtsp-server/rtsp-stream-transport.c:
7464 * gst/rtsp-server/rtsp-stream-transport.h:
7465 stream: use the address managed by the stream
7466 Use the address managed by the stream for multicast. This allows us to have 1
7467 multicast address for each stream.
7468 Because the address is now managed by the stream we don't have to pass it around
7470 Set the address pool on the streams.
7472 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7474 * gst/rtsp-server/rtsp-client.c:
7475 * gst/rtsp-server/rtsp-media.c:
7476 * gst/rtsp-server/rtsp-stream.c:
7479 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7481 * gst/rtsp-server/rtsp-media.c:
7482 * gst/rtsp-server/rtsp-media.h:
7483 media: add signal for new streams
7484 This allows applications to listen for new streams and configure properties on
7485 them, like the address pool.
7487 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7489 * gst/rtsp-server/rtsp-media.c:
7490 media: configure address pool in new streams
7492 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7494 * gst/rtsp-server/rtsp-stream.c:
7495 * gst/rtsp-server/rtsp-stream.h:
7496 stream: add methods to deal with address pool
7497 Add methods to get and set the address pool for the stream
7498 Add method to allocate and get the multicast addresses for this stream.
7500 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7502 * docs/libs/gst-rtsp-server-sections.txt:
7503 * gst/rtsp-server/rtsp-media.c:
7504 * gst/rtsp-server/rtsp-media.h:
7505 media: remove MTU property
7506 It is a stream property
7508 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7510 * gst/rtsp-server/rtsp-client.c:
7511 client: set blocksize only on stream
7512 Set the blocksize only on the current stream.
7514 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7516 * gst/rtsp-server/rtsp-stream.c:
7517 stream: share src and sink sockets
7518 the allocated socket is in the used-socket property, not socket.
7520 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7522 * gst/rtsp-server/rtsp-address-pool.c:
7523 * gst/rtsp-server/rtsp-address-pool.h:
7524 * gst/rtsp-server/rtsp-client.c:
7525 * gst/rtsp-server/rtsp-session-media.c:
7526 * gst/rtsp-server/rtsp-session-media.h:
7527 * gst/rtsp-server/rtsp-stream-transport.c:
7528 * gst/rtsp-server/rtsp-stream-transport.h:
7529 * tests/check/gst/addresspool.c:
7530 rtsp: make address-pool return an address object
7531 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
7532 store more info in the structure and allows us to more easily return the address
7533 to the right pool when no longer needed.
7534 Pass the address to the StreamTransport so that we can return it to the pool
7535 when the stream transport is freed or changed.
7537 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7539 * examples/Makefile.am:
7540 * examples/test-multicast.c:
7541 examples: add multicast example
7542 Show how to set up the multicast address pool so that media can be
7543 server with multicast.
7545 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7547 * gst/rtsp-server/rtsp-client.c:
7548 * gst/rtsp-server/rtsp-media-factory.c:
7549 * gst/rtsp-server/rtsp-media-factory.h:
7550 * gst/rtsp-server/rtsp-media.c:
7551 * gst/rtsp-server/rtsp-media.h:
7552 rtsp: use AddressPool
7553 Remove the multicast_group property.
7554 Use the configured addresspool to allocate multicast addresses.
7556 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7558 * gst/rtsp-server/rtsp-address-pool.c:
7559 * gst/rtsp-server/rtsp-address-pool.h:
7560 address-pool: add clear method
7562 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7564 * gst/rtsp-server/rtsp-address-pool.c:
7565 address-pool: small cleanups
7567 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7569 * tests/check/Makefile.am:
7570 * tests/check/gst/addresspool.c:
7571 tests: add addresspool unit test
7573 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7575 * gst/rtsp-server/Makefile.am:
7576 * gst/rtsp-server/rtsp-address-pool.c:
7577 * gst/rtsp-server/rtsp-address-pool.h:
7578 address-pool: add object to manage multicast addresses
7579 Make an object that can manage a rage of multicast addresses and ports.
7581 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7583 * gst/rtsp-server/rtsp-server.c:
7584 server: set default max-threads property
7586 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7588 * gst/rtsp-server/rtsp-media.c:
7589 media: wait for concurrent _prepare
7590 If a prepare is busy, wait for the result.
7592 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7594 * gst/rtsp-server/rtsp-media.c:
7595 media: add lock around message handler
7596 We don't want to dispatch messages while we are still processing the result of
7599 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7601 * gst/rtsp-server/rtsp-media.c:
7602 * gst/rtsp-server/rtsp-media.h:
7603 media: add lock to protect state changes
7605 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7607 * gst/rtsp-server/rtsp-stream.c:
7608 * gst/rtsp-server/rtsp-stream.h:
7611 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7613 * gst/rtsp-server/rtsp-stream-transport.c:
7614 * gst/rtsp-server/rtsp-stream-transport.h:
7615 * gst/rtsp-server/rtsp-stream.c:
7616 stream-transport: add keep-alive method
7618 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7620 * gst/rtsp-server/rtsp-stream-transport.c:
7621 * gst/rtsp-server/rtsp-stream-transport.h:
7622 * gst/rtsp-server/rtsp-stream.c:
7623 stream-transport: add method to handle RTP/RTCP
7624 Call new methods instead of poking into the structures directly.
7626 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7628 * gst/rtsp-server/rtsp-session-media.c:
7629 * gst/rtsp-server/rtsp-session-media.h:
7630 session-media: add locking
7632 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7634 * gst/rtsp-server/rtsp-session.c:
7635 * gst/rtsp-server/rtsp-session.h:
7636 session: add locking
7638 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7640 * gst/rtsp-server/rtsp-server.c:
7641 server: free old socket
7643 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7645 * gst/rtsp-server/rtsp-media-mapping.c:
7646 * gst/rtsp-server/rtsp-media-mapping.h:
7647 mapping: add locking
7649 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7651 * gst/rtsp-server/rtsp-media-factory.c:
7652 media-factory: add locking
7654 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7656 * gst/rtsp-server/rtsp-auth.c:
7657 * gst/rtsp-server/rtsp-auth.h:
7660 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7662 * gst/rtsp-server/rtsp-server.c:
7663 * gst/rtsp-server/rtsp-server.h:
7664 server: add max-thread property
7666 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7668 * gst/rtsp-server/rtsp-server.c:
7669 * gst/rtsp-server/rtsp-server.h:
7670 server: use a threadpool for the mainloops
7672 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7674 * gst/rtsp-server/rtsp-client.c:
7675 * gst/rtsp-server/rtsp-client.h:
7676 client: rename method
7677 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
7678 don't really create the client from the socket, we use the socket for the
7681 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7683 * gst/rtsp-server/rtsp-client.c:
7684 * gst/rtsp-server/rtsp-client.h:
7685 * gst/rtsp-server/rtsp-server.c:
7686 server: rework maincontext handling in clients
7687 Make a separate method to attach a client to a MainContext.
7688 Let the server decide in what GMainContext the client will operate and give this
7689 context to the client in attach. Then the server can later decide to use a
7690 separate thread for each client or just use the mainthread.
7692 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7694 * gst/rtsp-server/rtsp-client.c:
7695 * gst/rtsp-server/rtsp-session.c:
7696 * gst/rtsp-server/rtsp-session.h:
7697 session: move session header code in session object
7699 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
7703 * examples/test-auth.c:
7704 * examples/test-launch.c:
7705 * examples/test-mp4.c:
7706 * examples/test-ogg.c:
7707 * examples/test-readme.c:
7708 * examples/test-sdp.c:
7709 * examples/test-uri.c:
7710 * examples/test-video.c:
7711 * gst/rtsp-server/rtsp-auth.c:
7712 * gst/rtsp-server/rtsp-auth.h:
7713 * gst/rtsp-server/rtsp-client.c:
7714 * gst/rtsp-server/rtsp-client.h:
7715 * gst/rtsp-server/rtsp-media-factory-uri.c:
7716 * gst/rtsp-server/rtsp-media-factory-uri.h:
7717 * gst/rtsp-server/rtsp-media-factory.c:
7718 * gst/rtsp-server/rtsp-media-factory.h:
7719 * gst/rtsp-server/rtsp-media-mapping.c:
7720 * gst/rtsp-server/rtsp-media-mapping.h:
7721 * gst/rtsp-server/rtsp-media.c:
7722 * gst/rtsp-server/rtsp-media.h:
7723 * gst/rtsp-server/rtsp-params.c:
7724 * gst/rtsp-server/rtsp-params.h:
7725 * gst/rtsp-server/rtsp-sdp.c:
7726 * gst/rtsp-server/rtsp-sdp.h:
7727 * gst/rtsp-server/rtsp-server.c:
7728 * gst/rtsp-server/rtsp-server.h:
7729 * gst/rtsp-server/rtsp-session-media.c:
7730 * gst/rtsp-server/rtsp-session-media.h:
7731 * gst/rtsp-server/rtsp-session-pool.c:
7732 * gst/rtsp-server/rtsp-session-pool.h:
7733 * gst/rtsp-server/rtsp-session.c:
7734 * gst/rtsp-server/rtsp-session.h:
7735 * gst/rtsp-server/rtsp-stream-transport.c:
7736 * gst/rtsp-server/rtsp-stream-transport.h:
7737 * gst/rtsp-server/rtsp-stream.c:
7738 * gst/rtsp-server/rtsp-stream.h:
7739 * tests/check/gst/rtspserver.c:
7740 * tests/test-cleanup.c:
7743 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7745 * gst/rtsp-server/rtsp-media.c:
7746 * gst/rtsp-server/rtsp-session-media.c:
7747 * gst/rtsp-server/rtsp-session.c:
7748 rtsp-server: added annotations to indicate type of ownership transfer of return values
7749 https://bugzilla.gnome.org/show_bug.cgi?id=680777
7751 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
7754 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
7756 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
7759 * bindings/Makefile.am:
7760 * bindings/vala/Makefile.am:
7761 * bindings/vala/gst-rtsp-server-0.10.deps:
7762 * bindings/vala/gst-rtsp-server-0.10.vapi:
7763 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
7764 * bindings/vala/packages/gst-rtsp-server-0.10.files:
7765 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7766 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7767 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
7769 bindings: remove vala bindings
7770 They'll be reunited with the other GStreamer bindings
7771 https://bugzilla.gnome.org/show_bug.cgi?id=680777
7773 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7775 * gst/rtsp-server/rtsp-client.c:
7776 * gst/rtsp-server/rtsp-session-media.c:
7777 * gst/rtsp-server/rtsp-session-media.h:
7778 * gst/rtsp-server/rtsp-stream-transport.c:
7779 * gst/rtsp-server/rtsp-stream-transport.h:
7780 rtsp: only create transport when needed
7781 Only create the StreamTransport when configured.
7783 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7785 * gst/rtsp-server/rtsp-client.c:
7786 client: small cleanup
7788 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7790 * gst/rtsp-server/rtsp-client.c:
7791 * gst/rtsp-server/rtsp-client.h:
7792 * gst/rtsp-server/rtsp-stream-transport.c:
7793 * gst/rtsp-server/rtsp-stream-transport.h:
7794 rtsp: refactor configuration of transport
7795 Move the configuration of the transport to a place where it makes
7798 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7800 * gst/rtsp-server/rtsp-client.c:
7801 client: refactor transport parsing
7803 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7805 * gst/rtsp-server/rtsp-client.c:
7806 client: refuse to change the MTU on shared media
7807 If we change the MTU of chared media, it changes for all clients.
7808 We don't want to set the MTU to something large for clients that
7811 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7813 * examples/test-mp4.c:
7814 * gst/rtsp-server/rtsp-media.c:
7815 small fixes to docs and debug
7817 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7819 * gst/rtsp-server/rtsp-stream.c:
7820 stream: transports must already have been removed
7822 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7824 * gst/rtsp-server/rtsp-media.c:
7825 * gst/rtsp-server/rtsp-stream.c:
7826 * gst/rtsp-server/rtsp-stream.h:
7827 stream: improve join and leave of the pipeline
7829 Do the cleanup properly
7832 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7834 * gst/rtsp-server/rtsp-media.c:
7835 media: move unprepare below default implementation
7836 Makes it easier to find the default implementation
7838 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7840 * gst/rtsp-server/rtsp-media.c:
7841 media: signal unprepared when we actually finish
7843 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7845 * gst/rtsp-server/rtsp-media.c:
7846 media: no need to unlock, unprepare does that when needed
7848 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7850 * docs/libs/gst-rtsp-server-sections.txt:
7851 * gst/rtsp-server/rtsp-media-factory.h:
7852 * gst/rtsp-server/rtsp-media-mapping.c:
7853 * gst/rtsp-server/rtsp-media.h:
7854 * gst/rtsp-server/rtsp-params.c:
7855 * gst/rtsp-server/rtsp-server.c:
7856 * gst/rtsp-server/rtsp-session-pool.h:
7857 * gst/rtsp-server/rtsp-session.c:
7858 * gst/rtsp-server/rtsp-session.h:
7859 * gst/rtsp-server/rtsp-stream-transport.h:
7860 * gst/rtsp-server/rtsp-stream.h:
7863 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7865 * gst/rtsp-server/rtsp-client.c:
7866 * gst/rtsp-server/rtsp-media-mapping.h:
7867 * gst/rtsp-server/rtsp-media.c:
7868 * gst/rtsp-server/rtsp-media.h:
7869 * gst/rtsp-server/rtsp-server.h:
7870 * gst/rtsp-server/rtsp-stream.c:
7871 * gst/rtsp-server/rtsp-stream.h:
7872 rtsp: fix MTU setting
7873 Fix setting of the MTU. There is no need for a vmethod.
7875 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7880 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7883 configure: bump version number after refactoring
7885 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7887 * gst/rtsp-server/Makefile.am:
7888 * gst/rtsp-server/rtsp-client.c:
7889 * gst/rtsp-server/rtsp-client.h:
7890 * gst/rtsp-server/rtsp-media-factory-uri.c:
7891 * gst/rtsp-server/rtsp-media-factory.c:
7892 * gst/rtsp-server/rtsp-media-factory.h:
7893 * gst/rtsp-server/rtsp-media.c:
7894 * gst/rtsp-server/rtsp-media.h:
7895 * gst/rtsp-server/rtsp-sdp.c:
7896 * gst/rtsp-server/rtsp-session-media.c:
7897 * gst/rtsp-server/rtsp-session-media.h:
7898 * gst/rtsp-server/rtsp-session.c:
7899 * gst/rtsp-server/rtsp-session.h:
7900 * gst/rtsp-server/rtsp-stream-transport.c:
7901 * gst/rtsp-server/rtsp-stream-transport.h:
7902 * gst/rtsp-server/rtsp-stream.c:
7903 * gst/rtsp-server/rtsp-stream.h:
7904 rtsp: massive refactoring
7905 Make GObjects from the remaining simple structures.
7906 Remove GstRTSPSessionStream, it's not needed.
7907 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
7908 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
7909 a GstRTSPStream should be transported to a client.
7910 Rename GstRTSPMediaFactory::get_element -> create_element because that
7911 more accurately describes what it does.
7912 Make nice methods instead of poking in the structures.
7913 Move some methods inside the relevant object source code.
7914 Use GPtrArray to store objects instead of plain arrays, it is more
7915 natural and allows us to more easily clean up.
7916 Move the allocation of udp ports to the Stream object. The Stream object
7917 contains the elements needed to stream the media to a client.
7918 Improve the prepare and unprepare methods. Unprepare should now undo
7919 everything prepare did. Improve also async unprepare when doing EOS on
7920 shutdown. Make sure we always unprepare correctly.
7922 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
7924 * gst/rtsp-server/rtsp-client.c:
7925 rtsp-client: Unref server address clients connected to
7926 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
7928 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
7930 * gst/rtsp-server/rtsp-server.c:
7931 rtsp-server: don't ref server socket if it is NULL
7932 Fixes test_bind_already_in_use unit test again after commit 6a497440.
7933 https://bugzilla.gnome.org/show_bug.cgi?id=686644
7935 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
7937 * tests/check/Makefile.am:
7938 tests: Add libgio link dependency
7939 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
7941 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7943 * gst/rtsp-server/rtsp-media-mapping.c:
7944 * gst/rtsp-server/rtsp-media-mapping.h:
7945 rtsp-media-mapping: rename find_media vfunc to find_factory
7946 The virtual method and class method should have the same name
7947 so it is correctly represented in GIR file
7948 https://bugzilla.gnome.org/show_bug.cgi?id=680777
7950 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7952 * gst/rtsp-server/rtsp-auth.c:
7953 * gst/rtsp-server/rtsp-client.c:
7954 * gst/rtsp-server/rtsp-media-factory-uri.c:
7955 * gst/rtsp-server/rtsp-media-factory.c:
7956 * gst/rtsp-server/rtsp-media-mapping.c:
7957 * gst/rtsp-server/rtsp-media.c:
7958 * gst/rtsp-server/rtsp-server.c:
7959 * gst/rtsp-server/rtsp-session-pool.c:
7960 * gst/rtsp-server/rtsp-session.c:
7961 rtsp-server: fixed comments and GIR annotations
7962 https://bugzilla.gnome.org/show_bug.cgi?id=680777
7964 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
7966 * gst/rtsp-server/rtsp-media-mapping.c:
7967 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
7969 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
7971 * gst/rtsp-server/rtsp-server.c:
7972 rtsp-server: allow binding on port 0 (binds on a random port)
7974 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
7976 * gst/rtsp-server/rtsp-server.c:
7977 * gst/rtsp-server/rtsp-server.h:
7978 rtsp-server: add bound-port property
7979 bound-port can be used to retrieve the port number when the server is bound on
7980 port 0, which binds on a random port.
7982 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
7984 * gst/rtsp-server/rtsp-media-factory.c:
7985 * gst/rtsp-server/rtsp-media-factory.h:
7986 rtsp-media-factory: make ::get_element overridable by GI bindings
7987 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
7988 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
7989 as the invoker for ::get_element(), making it overridable by GI generated
7992 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
7994 * gst/rtsp-server/rtsp-media-factory-uri.c:
7995 rtsp-media-factory-uri: don't autoplug parsers in a loop
7996 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
7999 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
8001 * gst/rtsp-server/Makefile.am:
8002 Explicitly link against gio. Fix link error on mac.
8004 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
8006 * gst/rtsp-server/rtsp-session.c:
8007 session: add ttl to the transport header in SETUP
8008 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
8010 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
8012 * gst/rtsp-server/rtsp-client.c:
8013 * gst/rtsp-server/rtsp-client.h:
8014 * gst/rtsp-server/rtsp-media.c:
8015 client: Use client transport settings for multicast if allowed.
8016 This patch makes it possible for the client to send transport settings for
8017 multicast (destination && ttl). Client settings must be explicitly allowed or
8018 the server will use its own settings.
8019 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
8021 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
8024 Automatic update of common submodule
8025 From 6c0b52c to 6bb6951
8027 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
8029 * gst/rtsp-server/rtsp-client.c:
8030 rtsp-client: do not destroy the rtsp watch
8031 Don't destroy the client watch while dispatching. The rtsp watch is
8032 automatically destroyed after the rtsp watch function closed() has
8034 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
8036 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
8039 Automatic update of common submodule
8040 From 4f962f7 to 6c0b52c
8042 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
8044 * gst/rtsp-server/rtsp-media.c:
8045 media: fix check for seekability
8047 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8049 * gst/rtsp-server/rtsp-client.c:
8050 client: use more GIO
8051 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
8053 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8055 * gst/rtsp-server/rtsp-server.c:
8056 server: remove obsolete includes
8058 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
8060 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
8061 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
8062 be available in "on_new_ssrc". The transports are added in
8063 gst_rtsp_media_set_state when going to PLAYING state. However,
8064 "on_new_ssrc" might be called before this happens.
8065 https://bugzilla.gnome.org/show_bug.cgi?id=683304
8067 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
8069 * gst/rtsp-server/rtsp-client.c:
8070 * gst/rtsp-server/rtsp-client.h:
8071 rtsp-client: add signals for rtsp requests (fixes #683287)
8073 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
8075 * gst/rtsp-server/rtsp-client.c:
8076 * gst/rtsp-server/rtsp-client.h:
8077 add new-session signal to rtsp-client (fixes #683058)
8079 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
8082 Automatic update of common submodule
8083 From 668acee to 4f962f7
8085 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
8087 * gst/rtsp-server/rtsp-server.c:
8088 * tests/check/gst/rtspserver.c:
8089 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
8090 Do not assume that *error is set in g_socket_address_enumerator_next.
8091 Added test_bind_already_in_use unit-test.
8092 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
8094 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
8097 Automatic update of common submodule
8098 From 94ccf4c to 668acee
8100 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
8102 * gst/rtsp-server/rtsp-client.c:
8103 * gst/rtsp-server/rtsp-client.h:
8104 rtsp-client: make create_sdp virtual method
8105 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
8107 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8110 Automatic update of common submodule
8111 From 98e386f to 94ccf4c
8113 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8115 * gst/rtsp-server/rtsp-client.c:
8118 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
8120 * gst/rtsp-server/rtsp-client.c:
8121 * gst/rtsp-server/rtsp-client.h:
8122 * gst/rtsp-server/rtsp-server.c:
8123 * gst/rtsp-server/rtsp-server.h:
8124 rtsp-server: use an existing socket to establish HTTP tunnel
8125 Make it possible to transfer a socket from an HTTP server to be used as
8126 an RTSP over HTTP tunnel.
8128 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
8130 * gst/rtsp-server/rtsp-client.c:
8131 * gst/rtsp-server/rtsp-media.c:
8132 * gst/rtsp-server/rtsp-media.h:
8133 rtsp: Handle the blocksize parameter
8134 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
8136 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
8138 * tests/check/Makefile.am:
8139 * tests/check/gst/rtspserver.c:
8140 Have unit test get header from source dir, not installed dir
8141 This makes compilation of unit tests work in a build directory other
8142 than the source directory.
8143 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
8145 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
8147 * gst/rtsp-server/rtsp-media.c:
8148 rtsp-media: update for gst_element_make_from_uri() changes
8150 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
8153 * tests/Makefile.am:
8154 * tests/check/Makefile.am:
8155 * tests/check/gst/rtspserver.c:
8157 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
8159 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
8161 * gst/rtsp-server/rtsp-media.c:
8162 rtsp-media: don't collect media stats when going to NULL
8163 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
8165 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8167 * gst/rtsp-server/rtsp-client.c:
8168 client: don't leak transports
8170 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
8172 * gst/rtsp-server/rtsp-client.c:
8173 rtsp-client: free transport on no_stream in SETUP handler
8175 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
8177 * gst/rtsp-server/rtsp-client.c:
8178 rtsp-client: changed session media iteration
8179 In client_unlink_session: now don't iterate in session->medias
8180 list where items are removed by gst_rtsp_session_release_media.
8181 Instead, repeatedly remove the first item.
8183 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
8185 * gst/rtsp-server/rtsp-client.c:
8186 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
8187 GstRTSPSessionMedia is not a GObject type. When the
8188 GstRTSPSession is freed, it will free the media.
8190 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
8192 * gst/rtsp-server/rtsp-media-factory.c:
8193 factory: plug pad leak in collect_streams
8194 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
8195 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
8196 will take one reference, and the other reference will otherwise
8199 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
8202 configure: suppress some warnings when debug is disabled
8203 Warnings about unused variables should be suppressed if core has the
8204 debug system disabled.
8205 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
8207 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8209 * docs/libs/Makefile.am:
8210 docs: fix build in uninstalled setup
8211 Include gst-plugins-base libs properly.
8213 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
8215 * docs/libs/gst-rtsp-server.types:
8216 docs: include headers defining rtsp-server object types
8217 Fixes compiler warnings during docs build.
8218 https://bugzilla.gnome.org/show_bug.cgi?id=676824
8220 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
8223 configure: Add warning flags for compiler when configuring
8224 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
8226 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
8229 Automatic update of common submodule
8230 From 03a0e57 to 98e386f
8232 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
8235 Automatic update of common submodule
8236 From 1fab359 to 03a0e57
8238 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
8240 * gst/rtsp-server/rtsp-client.c:
8241 client: fix GSocketAddress leak in gst_rtsp_client_accept
8242 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
8244 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
8247 Automatic update of common submodule
8248 From f1b5a96 to 1fab359
8250 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8253 Automatic update of common submodule
8254 From 92b7266 to f1b5a96
8256 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8259 Automatic update of common submodule
8260 From ec1c4a8 to 92b7266
8262 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8265 Automatic update of common submodule
8266 From 3429ba6 to ec1c4a8
8268 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
8270 * gst/rtsp-server/rtsp-auth.c:
8271 * gst/rtsp-server/rtsp-client.c:
8272 * gst/rtsp-server/rtsp-media-factory-uri.c:
8273 * gst/rtsp-server/rtsp-server.c:
8274 rtsp: fix compiler warnings
8275 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
8277 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8280 Automatic update of common submodule
8281 From dc70203 to 3429ba6
8283 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8285 * gst/rtsp-server/rtsp-client.c:
8286 * gst/rtsp-server/rtsp-media-factory.c:
8287 * gst/rtsp-server/rtsp-media-factory.h:
8288 * gst/rtsp-server/rtsp-media.c:
8289 * gst/rtsp-server/rtsp-media.h:
8290 * gst/rtsp-server/rtsp-server.c:
8291 * gst/rtsp-server/rtsp-server.h:
8292 * gst/rtsp-server/rtsp-session-pool.c:
8293 * gst/rtsp-server/rtsp-session-pool.h:
8294 rtsp-server: port to new thread API
8296 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8299 Automatic update of common submodule
8300 From 6db25be to dc70203
8302 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8304 * gst/rtsp-server/rtsp-auth.c:
8305 * gst/rtsp-server/rtsp-auth.h:
8306 * gst/rtsp-server/rtsp-client.c:
8307 rtsp-server: Fix compilation and compiler warnings
8309 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8313 * gst/rtsp-server/Makefile.am:
8314 configure: Modernize autotools setup a bit
8315 Also we now only create tar.bz2 and tar.xz tarballs.
8317 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8320 Automatic update of common submodule
8321 From 464fe15 to 6db25be
8323 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8326 Automatic update of common submodule
8327 From 7fda524 to 464fe15
8329 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8332 * docs/libs/Makefile.am:
8333 * docs/version.entities.in:
8335 * gst/rtsp-server/Makefile.am:
8336 * pkgconfig/Makefile.am:
8337 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
8338 * pkgconfig/gstreamer-rtsp-server.pc.in:
8339 * tests/Makefile.am:
8340 rtsp-server: Update versioning
8342 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8344 Merge remote-tracking branch 'origin/0.10'
8346 gst/rtsp-server/rtsp-session-pool.c
8348 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8350 * gst/rtsp-server/rtsp-session-pool.c:
8351 rtsp-server: Don't use deprecated GLib API
8353 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8355 Replace master with 0.11
8357 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8359 Merge branch 'master' into 0.11
8361 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8363 Merge branch 'master' into 0.11
8365 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
8368 A couple minor typo fixes
8370 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8372 * gst/rtsp-server/rtsp-media.c:
8373 media: fix state of the appqueue
8375 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8377 * gst/rtsp-server/rtsp-media-factory-uri.c:
8378 factory: use videoconvert
8380 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8382 * gst/rtsp-server/rtsp-media-factory-uri.c:
8383 factory: change to new style caps
8385 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8387 * gst/rtsp-server/rtsp-client.c:
8388 * gst/rtsp-server/rtsp-client.h:
8389 * gst/rtsp-server/rtsp-media-factory-uri.c:
8390 * gst/rtsp-server/rtsp-media.c:
8391 * gst/rtsp-server/rtsp-server.c:
8392 * gst/rtsp-server/rtsp-server.h:
8393 * gst/rtsp-server/rtsp-session-pool.c:
8394 rtsp-server: port to GIO
8397 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8400 configure: fix build
8402 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8405 docs: fix for gst_rtsp_server_set_port() -> _set_service()
8406 https://bugzilla.gnome.org/show_bug.cgi?id=666548
8408 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8411 * examples/Makefile.am:
8412 First rule of gst-rtsp-server club: don't talk about gst-phonon
8414 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8417 * pkgconfig/Makefile.am:
8418 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
8419 * pkgconfig/gstreamer-rtsp-server.pc.in:
8420 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
8421 For consistency with all other modules.
8423 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8425 * gst/rtsp-server/rtsp-client.c:
8426 rtsp-client: update for new map API
8428 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8431 * bindings/Makefile.am:
8432 * bindings/python/Makefile.am:
8433 * bindings/python/arg-types.py:
8434 * bindings/python/codegen/Makefile.am:
8435 * bindings/python/codegen/__init__.py:
8436 * bindings/python/codegen/argtypes.py:
8437 * bindings/python/codegen/code-coverage.py:
8438 * bindings/python/codegen/codegen.py:
8439 * bindings/python/codegen/definitions.py:
8440 * bindings/python/codegen/defsparser.py:
8441 * bindings/python/codegen/docextract.py:
8442 * bindings/python/codegen/docgen.py:
8443 * bindings/python/codegen/fileprefix.override:
8444 * bindings/python/codegen/fileprefixmodule.c:
8445 * bindings/python/codegen/h2def.py:
8446 * bindings/python/codegen/mergedefs.py:
8447 * bindings/python/codegen/mkskel.py:
8448 * bindings/python/codegen/override.py:
8449 * bindings/python/codegen/reversewrapper.py:
8450 * bindings/python/codegen/scmexpr.py:
8451 * bindings/python/rtspserver-types.defs:
8452 * bindings/python/rtspserver.defs:
8453 * bindings/python/rtspserver.override:
8454 * bindings/python/rtspservermodule.c:
8455 * bindings/python/test.py:
8457 python: remove pygst-based python bindings
8458 pygi is the future, apparently.
8460 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
8463 Automatic update of common submodule
8464 From c463bc0 to 7fda524
8466 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8469 Automatic update of common submodule
8470 From 2a59016 to c463bc0
8472 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8475 Automatic update of common submodule
8476 From 0807187 to 2a59016
8478 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8481 Automatic update of common submodule
8482 From 11f0cd5 to 0807187
8484 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8486 * examples/test-auth.c:
8487 example: update for new caps
8489 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8491 * examples/test-video.c:
8492 * gst/rtsp-server/rtsp-client.c:
8493 * gst/rtsp-server/rtsp-media-factory-uri.c:
8494 * gst/rtsp-server/rtsp-media.c:
8495 * gst/rtsp-server/rtsp-media.h:
8496 * gst/rtsp-server/rtsp-session.c:
8497 * gst/rtsp-server/rtsp-session.h:
8498 rtsp-server: port some more to 0.11
8500 Remove bufferlist stuff
8502 Add queue before appsink now that preroll-queue-len is gone.
8503 Update for request pad changes.
8505 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8507 Merge branch 'master' into 0.11
8509 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
8511 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8512 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
8513 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
8515 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
8517 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8518 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
8519 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
8521 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8523 Merge branch 'master' into 0.11
8525 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8527 * gst/rtsp-server/rtsp-media.c:
8528 * gst/rtsp-server/rtsp-media.h:
8529 media: add a seekable boolean
8530 Maintain the seekable state with a new variable instead of reusing the
8533 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
8535 * gst/rtsp-server/rtsp-media.c:
8536 Disallow seek in live media
8538 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8540 Merge branch 'master' into 0.11
8542 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
8544 * gst/rtsp-server/rtsp-server.c:
8545 #ifdef statements for windows socket creation were missing
8547 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
8550 Automatic update of common submodule
8551 From a39eb83 to 11f0cd5
8553 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
8556 Automatic update of common submodule
8557 From 605cd9a to a39eb83
8559 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8561 Merge branch 'master' into 0.11
8563 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8565 * gst/rtsp-server/rtsp-client.c:
8566 client: use method to access property
8568 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8570 * gst/rtsp-server/rtsp-media-factory.c:
8571 * gst/rtsp-server/rtsp-media-factory.h:
8572 media-factory: add protocols property
8573 Add a property to configure the allowed protocols in the media created from the
8576 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8578 * gst/rtsp-server/rtsp-media-factory.c:
8579 * gst/rtsp-server/rtsp-media-factory.h:
8580 media-factory: add media-configure signal
8581 Add signal to allow the application to configure the media after it was created
8584 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8586 * gst/rtsp-server/rtsp-client.c:
8587 client: use method to access property
8589 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8591 * gst/rtsp-server/rtsp-media-factory.c:
8592 * gst/rtsp-server/rtsp-media-factory.h:
8593 media-factory: add protocols property
8594 Add a property to configure the allowed protocols in the media created from the
8597 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8599 * gst/rtsp-server/rtsp-media-factory.c:
8600 * gst/rtsp-server/rtsp-media-factory.h:
8601 media-factory: add media-configure signal
8602 Add signal to allow the application to configure the media after it was created
8605 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8607 Merge branch 'master' into 0.11
8609 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8611 * gst/rtsp-server/rtsp-client.c:
8612 client: use media multicast group
8614 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8616 * gst/rtsp-server/rtsp-media-factory.h:
8617 * gst/rtsp-server/rtsp-server.h:
8618 * gst/rtsp-server/rtsp-session-pool.h:
8619 * gst/rtsp-server/rtsp-session.h:
8622 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
8624 * gst/rtsp-server/rtsp-client.c:
8625 * gst/rtsp-server/rtsp-sdp.h:
8626 sdp: copy and free the server ip address
8627 Copy and free the server ip address to make memory management easier later.
8629 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8631 * gst/rtsp-server/rtsp-media-factory.c:
8632 media-factory: configure multicast in media
8634 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8636 * gst/rtsp-server/rtsp-media.c:
8637 * gst/rtsp-server/rtsp-media.h:
8638 media: add property for multicast group
8639 Add a property to configure the multicast group in the media.
8640 Based on patches from Marc Leeman and Robert Krakora.
8642 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8644 * gst/rtsp-server/rtsp-media-factory.c:
8645 * gst/rtsp-server/rtsp-media-factory.h:
8646 media-factory: add property for multicast group
8647 Add a property to configure the multicast group in the media factory.
8648 Based on patches from Marc Leeman and Robert Krakora.
8650 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8652 * gst/rtsp-server/rtsp-client.c:
8653 client: do configuration of transport in one place
8654 Move the configuration of the transport destination address to where we also
8655 configure the other bits.
8657 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8659 * gst/rtsp-server/rtsp-client.c:
8660 client: use media multicast group
8662 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8664 * gst/rtsp-server/rtsp-media-factory.h:
8665 * gst/rtsp-server/rtsp-server.h:
8666 * gst/rtsp-server/rtsp-session-pool.h:
8667 * gst/rtsp-server/rtsp-session.h:
8670 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
8672 * gst/rtsp-server/rtsp-client.c:
8673 * gst/rtsp-server/rtsp-sdp.h:
8674 sdp: copy and free the server ip address
8675 Copy and free the server ip address to make memory management easier later.
8677 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8679 * gst/rtsp-server/rtsp-media-factory.c:
8680 media-factory: configure multicast in media
8682 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8684 * gst/rtsp-server/rtsp-media.c:
8685 * gst/rtsp-server/rtsp-media.h:
8686 media: add property for multicast group
8687 Add a property to configure the multicast group in the media.
8688 Based on patches from Marc Leeman and Robert Krakora.
8690 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8692 * gst/rtsp-server/rtsp-media-factory.c:
8693 * gst/rtsp-server/rtsp-media-factory.h:
8694 media-factory: add property for multicast group
8695 Add a property to configure the multicast group in the media factory.
8696 Based on patches from Marc Leeman and Robert Krakora.
8698 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8700 * gst/rtsp-server/rtsp-client.c:
8701 client: do configuration of transport in one place
8702 Move the configuration of the transport destination address to where we also
8703 configure the other bits.
8705 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8707 Merge branch 'master' into 0.11
8709 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
8711 * gst/rtsp-server/rtsp-client.c:
8712 client: destroy pipeline on client disconnect with no prior TEARDOWN.
8713 The problem occurs when the client abruptly closes the connection without
8714 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
8715 server is where the pipeline gets torn down. Since this handler is not called,
8716 the pipeline remains and is up and running. Subsequent clients get their own
8717 pipelines and if the do not issue TEARDOWNs then those pipelines will also
8718 remain up and running. This is a resource leak.
8720 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8722 Merge branch 'master' into 0.11
8724 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
8726 * gst/rtsp-server/rtsp-media-factory.c:
8727 * gst/rtsp-server/rtsp-media-factory.h:
8728 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
8729 For example, it can be used to retrieve source elements like appsrc, in a more
8730 convenient way than subclassing get_element.
8732 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8734 Merge branch 'master' into 0.11
8736 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
8738 * gst/rtsp-server/rtsp-server.c:
8739 rtsp-server: hold on to reference while using object
8741 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8743 * gst/rtsp-server/rtsp-media.c:
8746 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8749 configure: use unstable api
8751 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
8753 * gst/rtsp-server/rtsp-client.c:
8754 client: fix reference counting
8756 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
8758 * gst/rtsp-server/rtsp-client.c:
8759 * gst/rtsp-server/rtsp-media.c:
8760 fix compiler warnings about unused variables
8762 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
8764 * examples/test-launch.c:
8765 * examples/test-readme.c:
8766 * examples/test-uri.c:
8767 * examples/test-video.c:
8768 examples: tell rtsp uri when ready
8770 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
8773 Automatic update of common submodule
8774 From 69b981f to 605cd9a
8776 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8778 * gst/rtsp-server/rtsp-client.c:
8779 client: update for buffer API change
8781 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
8783 * gst/rtsp-server/Makefile.am:
8784 Makefile.am: 0.10 => @GST_MAJORMINOR@
8786 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
8788 * gst/rtsp-server/rtsp-media-factory-uri.c:
8789 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
8791 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
8793 * gst/rtsp-server/.gitignore:
8794 .gitignore: 0.10 => 0.11
8796 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
8798 * gst/rtsp-server/Makefile.am:
8799 Makefile.am: 0.10 => @GST_MAJORMINOR@
8801 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8803 Merge branch 'master' into 0.11
8805 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
8808 Automatic update of common submodule
8809 From 9e5bbd5 to 69b981f
8811 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
8814 Automatic update of common submodule
8815 From fd35073 to 9e5bbd5
8817 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
8820 Automatic update of common submodule
8821 From 46dfcea to fd35073
8823 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8825 * gst/rtsp-server/rtsp-media-factory-uri.c:
8826 * gst/rtsp-server/rtsp-media.c:
8827 media: port to new caps API
8829 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8831 Merge branch 'master' into 0.11
8833 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
8835 * bindings/vala/gst-rtsp-server-0.10.vapi:
8836 Updated Vala bindings.
8837 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
8839 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
8841 * gst/rtsp-server/rtsp-server.c:
8842 * gst/rtsp-server/rtsp-server.h:
8843 Add a signal for newly connected clients.
8844 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
8846 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
8848 * bindings/python/rtspserver.override:
8849 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
8851 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8853 * gst/rtsp-server/Makefile.am:
8854 * gst/rtsp-server/rtsp-client.c:
8855 * gst/rtsp-server/rtsp-funnel.c:
8856 * gst/rtsp-server/rtsp-funnel.h:
8857 * gst/rtsp-server/rtsp-media.c:
8858 rtsp-server: port to 0.11
8860 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8865 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8867 Merge branch 'master' into 0.11
8872 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8875 Automatic update of common submodule
8876 From c3cafe1 to 46dfcea
8878 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
8880 * bindings/python/Makefile.am:
8881 * bindings/python/rtspserver.defs:
8882 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
8884 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
8886 * bindings/python/arg-types.py:
8887 python bindings: add GstRTSPUrlParam
8888 Needed to implement MediaFactory virtual proxies
8890 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
8892 * bindings/python/arg-types.py:
8893 python bindings: fix returning GstRTSPUrl types
8895 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
8897 * bindings/python/arg-types.py:
8898 python bindings: add arg type for GstRTSPUrl
8900 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
8902 * bindings/python/rtspserver.defs:
8903 python bindings: fix the definition of MediaFactory.collect_stream
8905 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
8908 Automatic update of common submodule
8909 From 1ccbe09 to c3cafe1
8911 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8914 Automatic update of common submodule
8915 From 193b717 to 1ccbe09
8917 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
8920 Automatic update of common submodule
8921 From b77e2bf to 193b717
8923 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8926 build: Include lcov.mak to allow test coverage report generation
8928 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8931 Automatic update of common submodule
8932 From d8814b6 to b77e2bf
8934 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8937 Automatic update of common submodule
8938 From 6aaa286 to d8814b6
8940 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
8943 Automatic update of common submodule
8944 From 6aec6b9 to 6aaa286
8946 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
8949 autogen: wingo signed comment
8951 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
8953 * gst/rtsp-server/rtsp-session-pool.c:
8954 session: use full charset for RTSP session ID
8955 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
8956 session ID more difficult.
8957 https://bugzilla.gnome.org/show_bug.cgi?id=643812
8959 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8961 * gst/rtsp-server/Makefile.am:
8962 rtsp-server: Don't install the funnel header
8964 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
8967 Automatic update of common submodule
8968 From 1de7f6a to 6aec6b9
8970 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8973 configure: require core/base 0.10.31
8974 Needed at least for gst_plugin_feature_rank_compare_func().
8976 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
8979 Automatic update of common submodule
8980 From f94d739 to 1de7f6a
8982 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8984 * gst/rtsp-server/rtsp-media.c:
8985 media: remove more unused code
8987 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8989 * gst/rtsp-server/rtsp-media.c:
8990 * gst/rtsp-server/rtsp-media.h:
8991 media: remove duplicate filtering
8992 Remove the duplicate filtering code now that we have a released -good version.
8993 Give a warning instead.
8995 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8997 * gst/rtsp-server/rtsp-media-factory.c:
8998 * gst/rtsp-server/rtsp-media.c:
8999 media: fix default buffer size
9001 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9003 * gst/rtsp-server/rtsp-media-factory.c:
9004 * gst/rtsp-server/rtsp-media-factory.h:
9005 media-factory: add property to configure the buffer-size
9006 Add a property to configure the kernel UDP buffer size.
9008 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9010 * gst/rtsp-server/rtsp-media.c:
9011 * gst/rtsp-server/rtsp-media.h:
9012 media: add property to configure kernel buffer sizes
9013 Add a property to configure the kernel UDP buffer size.
9015 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9018 configure: set PYGOBJECT_REQ before using it
9019 https://bugzilla.gnome.org/show_bug.cgi?id=640641
9021 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9024 docs: recursive into sub-directories on 'make upload'
9026 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9028 * docs/libs/gst-rtsp-server-docs.sgml:
9029 * docs/version.entities.in:
9030 docs: mention full version these docs are for, not just major-minor
9032 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9037 === release 0.10.8 ===
9039 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9044 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9046 * gst/rtsp-server/rtsp-server.c:
9047 rtsp-server: clarify docs a little
9049 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9051 * gst/rtsp-server/rtsp-media.c:
9052 media: init debug category before starting thread
9054 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9056 * gst/rtsp-server/rtsp-auth.c:
9057 auth: add realm to make it more spec compliant
9059 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9061 * gst/rtsp-server/rtsp-server.c:
9062 * gst/rtsp-server/rtsp-server.h:
9065 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9067 * examples/test-video.c:
9068 example: improve example docs a little
9070 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9072 * gst/rtsp-server/rtsp-server.c:
9073 server: ensure the watch has a ref to the server
9075 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9077 * gst/rtsp-server/rtsp-server.c:
9078 server: simpify channel function
9080 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9082 * gst/rtsp-server/rtsp-server.c:
9083 * gst/rtsp-server/rtsp-server.h:
9084 server: simplify management of channel and source
9085 We don't need to keep around the channel and source objects. Let the mainloop
9086 and the source manage the source and channel respectively.
9088 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9094 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9097 * tests/Makefile.am:
9098 * tests/test-cleanup.c:
9099 tests: add tests directory and cleanup test
9101 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9103 * gst/rtsp-server/rtsp-media-factory-uri.c:
9104 * gst/rtsp-server/rtsp-media-factory.c:
9105 * gst/rtsp-server/rtsp-media-mapping.c:
9106 * gst/rtsp-server/rtsp-media.c:
9107 * gst/rtsp-server/rtsp-session-pool.c:
9108 * gst/rtsp-server/rtsp-session.c:
9109 server: improve debugging in various objects
9111 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9113 * gst/rtsp-server/rtsp-server.c:
9114 server: chain up to the parent finalize
9116 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
9118 * bindings/python/rtspserver-types.defs:
9119 * bindings/python/rtspserver.defs:
9120 * bindings/python/rtspserver.override:
9121 * bindings/python/test.py:
9122 gst-rtsp-server: update python bindings
9124 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9126 * gst/rtsp-server/rtsp-client.c:
9127 client: use the response from the clientstate
9128 Create the response object only once and store in the client state.
9129 Make all methods use the state response,
9131 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9133 * gst/rtsp-server/rtsp-server.c:
9134 server: use signal to keep track of clients
9135 Keep track of all the clients that the server creates and remove them when they
9136 fire the 'closed' signal.
9138 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9140 * gst/rtsp-server/rtsp-client.c:
9141 * gst/rtsp-server/rtsp-client.h:
9142 client: emit signal when closing
9144 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9146 * examples/.gitignore:
9147 * examples/Makefile.am:
9148 * examples/test-auth.c:
9149 * examples/test-video.c:
9150 * gst/rtsp-server/rtsp-auth.c:
9151 * gst/rtsp-server/rtsp-auth.h:
9152 * gst/rtsp-server/rtsp-client.c:
9153 * gst/rtsp-server/rtsp-media-factory.c:
9154 * gst/rtsp-server/rtsp-media.c:
9155 * gst/rtsp-server/rtsp-media.h:
9156 * gst/rtsp-server/rtsp-session-pool.h:
9157 * gst/rtsp-server/rtsp-session.h:
9158 media: enable per factory authorisations
9159 Allow for adding a GstRTSPAuth on the factory and media level and check
9160 permissions when accessing the factory.
9161 Add hints to the auth methods for future more fine grained authorisation.
9162 Add example application for per factory authentication.
9164 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9166 * gst/rtsp-server/rtsp-auth.c:
9167 * gst/rtsp-server/rtsp-auth.h:
9168 * gst/rtsp-server/rtsp-client.c:
9169 * gst/rtsp-server/rtsp-client.h:
9170 * gst/rtsp-server/rtsp-params.c:
9171 * gst/rtsp-server/rtsp-params.h:
9172 rtsp-server: Pass ClientState structure arround
9173 Pass the collected information for the ongoing request in a GstRTSPClientState
9174 structure that we can then pass around to simplify the method arguments. This
9175 will also be handy when we implement logging functionality.
9177 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9179 * gst/rtsp-server/rtsp-media-factory.c:
9180 * gst/rtsp-server/rtsp-media-factory.h:
9181 media-factory: add methods to configure authorisation
9183 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9185 * gst/rtsp-server/rtsp-client.c:
9186 client: unref auth in finalize
9188 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9190 * gst/rtsp-server/rtsp-server.c:
9191 server: unref auth in finalize
9193 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9195 * docs/libs/gst-rtsp-server-docs.sgml:
9196 * docs/libs/gst-rtsp-server-sections.txt:
9197 * docs/libs/gst-rtsp-server.types:
9200 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9202 * gst/rtsp-server/rtsp-server.c:
9203 * gst/rtsp-server/rtsp-server.h:
9204 server: separate create and accept
9205 Create separate create and accept methods so that subclasses can create custom
9207 Configure the server in the client object and prepare for keeping track of
9210 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9212 * gst/rtsp-server/rtsp-client.c:
9213 * gst/rtsp-server/rtsp-client.h:
9214 client: add support for setting the server.
9215 Add support for keeping a ref to the server that started this client
9218 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9220 * gst/rtsp-server/rtsp-auth.c:
9221 auth: fix memleak and add some docs
9222 Fix a memleak of the basic auth token.
9223 Add docs for the helper function
9225 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9227 * gst/rtsp-server/rtsp-auth.c:
9228 * gst/rtsp-server/rtsp-auth.h:
9229 * gst/rtsp-server/rtsp-client.c:
9230 client: delegate setup of auth to the manager
9231 Delegate the configuration of the authentication tokens to the manager object
9234 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9236 * examples/test-video.c:
9237 * gst/rtsp-server/Makefile.am:
9238 * gst/rtsp-server/rtsp-auth.c:
9239 * gst/rtsp-server/rtsp-auth.h:
9240 * gst/rtsp-server/rtsp-client.c:
9241 * gst/rtsp-server/rtsp-client.h:
9242 * gst/rtsp-server/rtsp-server.c:
9243 * gst/rtsp-server/rtsp-server.h:
9244 auth: add authentication object
9245 Add an object that can check the authorization of requests.
9246 Implement basic authentication.
9247 Add example authentication to test-video
9249 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9251 * gst/rtsp-server/rtsp-server.c:
9252 * gst/rtsp-server/rtsp-server.h:
9253 server: move includes back
9254 the includes are needed for sockaddr_in.
9256 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9258 * gst/rtsp-server/rtsp-client.c:
9259 * gst/rtsp-server/rtsp-client.h:
9260 * gst/rtsp-server/rtsp-server.c:
9261 * gst/rtsp-server/rtsp-server.h:
9262 rtsp: move network includes where they are needed
9264 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
9266 * gst/rtsp-server/rtsp-media.h:
9267 rtsp-media.h: Minor corrections in comments.
9270 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
9273 Automatic update of common submodule
9274 From e572c87 to f94d739
9276 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
9280 * docs/libs/.gitignore:
9281 * examples/.gitignore:
9282 * gst/rtsp-server/.gitignore:
9285 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
9287 * docs/libs/Makefile.am:
9288 docs: We don't build ps/pdf for API reference docs
9290 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9293 Automatic update of common submodule
9294 From ccbaa85 to e572c87
9296 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9299 Automatic update of common submodule
9300 From 46445ad to ccbaa85
9302 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9304 * gst/rtsp-server/Makefile.am:
9305 * gst/rtsp-server/rtsp-funnel.c:
9306 * gst/rtsp-server/rtsp-funnel.h:
9307 * gst/rtsp-server/rtsp-media.c:
9308 funnel: rename fsfunnel to rtspfunnel
9309 Rename the funnel to avoid conflicts with the farsight one.
9311 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9313 * gst/rtsp-server/Makefile.am:
9314 * gst/rtsp-server/fs-funnel.c:
9315 * gst/rtsp-server/fs-funnel.h:
9316 * gst/rtsp-server/rtsp-media.c:
9317 rtsp-media: add and use fsfunnel
9318 Add a copy of fsfunnel to the build because input-selector removed the (broken)
9319 select-all property that we need.
9321 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9323 * gst/rtsp-server/Makefile.am:
9324 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
9325 Use PKG_CONFIG_PATH specified at configure time (if any) as well
9326 for the g-ir-compiler, rather than just assuming the env var has
9329 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9336 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
9338 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9341 * gst/rtsp-server/Makefile.am:
9342 gobject-introspection: fix g-i build for uninstalled setup
9343 Requires gst-plugins-base git (> 0.10.31.2).
9345 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9347 * examples/test-uri.c:
9348 examples: add some more options and comments
9350 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9352 * gst/rtsp-server/rtsp-media-factory-uri.c:
9353 factory-uri: use right property type
9355 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9357 * gst/rtsp-server/rtsp-media-factory-uri.c:
9358 factory-uri: attempt to configure buffer-lists
9359 Attempt to configure buffer lists in the payloader for improved performance.
9361 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9363 * gst/rtsp-server/rtsp-media.c:
9364 media: attempt to configure bigger UDP buffers
9365 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
9366 send buffers with high bitrate streams.
9368 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
9370 * gst/rtsp-server/rtsp-client.c:
9371 client: use the socket length from getsockname
9372 Use the length returned by getsockname to perform the getnameinfo call because
9373 the size can depend on the socket type and platform.
9376 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9378 * docs/libs/gst-rtsp-server-docs.sgml:
9379 * docs/libs/gst-rtsp-server-sections.txt:
9380 docs: add uri factory to the docs
9382 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9384 * gst/rtsp-server/rtsp-client.c:
9385 * gst/rtsp-server/rtsp-media.h:
9388 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9390 * gst/rtsp-server/rtsp-client.c:
9391 * gst/rtsp-server/rtsp-media.c:
9392 * gst/rtsp-server/rtsp-media.h:
9393 * gst/rtsp-server/rtsp-session.c:
9394 * gst/rtsp-server/rtsp-session.h:
9395 rtsp-server: add support for buffer lists
9396 Add support for sending bufferlists received from appsink.
9399 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9401 * gst/rtsp-server/rtsp-client.c:
9402 * gst/rtsp-server/rtsp-media.c:
9403 * gst/rtsp-server/rtsp-media.h:
9404 * gst/rtsp-server/rtsp-sdp.c:
9405 media: make method to retrieve the play range
9406 Make a method to retrieve the playback range so that we can conditionally create
9407 a different range for the SDP and the PLAY requests.
9409 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9411 * gst/rtsp-server/rtsp-media.c:
9412 * gst/rtsp-server/rtsp-media.h:
9413 media: add signal to notify of state changes
9415 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9417 * gst/rtsp-server/rtsp-client.h:
9418 client: cleanup headers
9420 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9422 * gst/rtsp-server/rtsp-client.c:
9425 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9427 * gst/rtsp-server/rtsp-media-factory-uri.c:
9428 * gst/rtsp-server/rtsp-media-factory-uri.h:
9429 factory-uri: add support for gstpay
9430 Add an option to prefer gstpay over decoder + raw payloader.
9432 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9434 * gst/rtsp-server/rtsp-media-factory-uri.c:
9435 * gst/rtsp-server/rtsp-media-factory-uri.h:
9436 factory-uri: rework the autoplugger.
9437 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
9440 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9442 * gst/rtsp-server/rtsp-media-factory-uri.c:
9443 factory-uri: use better factory filter
9444 Make better payloader filter based on autoplug rank and RTP use case.
9446 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
9449 Automatic update of common submodule
9450 From 169462a to 46445ad
9452 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9454 * gst/rtsp-server/rtsp-server.c:
9455 server: set SO_REUSEADDR before bind
9456 Set the SO_REUSEADDR _before_ bind() to make it actually work.
9458 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9460 * gst/rtsp-server/rtsp-media.c:
9461 * gst/rtsp-server/rtsp-media.h:
9462 media: emit prepared signal when prepared
9463 Make a 'prepared' signal and emit it when we successfully prepared the element.
9464 This signal can be used to configure the media object after it has been prepared
9467 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
9470 Automatic update of common submodule
9471 From 011bcc8 to 169462a
9473 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
9475 python an optional dependency
9476 * configure.ac: Move up valgrind and g-i checks. Make the python
9477 dependency optional, as it was before.
9479 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9481 Merge branch 'master' into 0.11
9486 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9488 * gst/rtsp-server/rtsp-media.c:
9489 media: update range when active clients changed
9490 When we changed the number of active clients, update the current range
9491 information because we want the second client connecting to a shared resource
9492 continue from where the stream currently.
9494 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9496 * gst/rtsp-server/rtsp-media-factory-uri.c:
9497 * gst/rtsp-server/rtsp-media-factory-uri.h:
9498 factory-uri: add colorspace and fix pt
9499 Rework the way we pass data to the autoplugger.
9500 When we have raw caps, plug a converter element to make pluggin to raw
9501 payloaders more successful.
9502 Make sure all dynamically plugged payloaders have a unique payload types.
9504 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9506 * examples/Makefile.am:
9507 * examples/test-uri.c:
9508 example: add example of the uri factory
9510 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9512 * gst/rtsp-server/Makefile.am:
9513 * gst/rtsp-server/rtsp-media-factory-uri.c:
9514 * gst/rtsp-server/rtsp-media-factory-uri.h:
9515 * gst/rtsp-server/rtsp-server.h:
9516 factory-uri: add a factory to stream any URI
9517 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
9520 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9522 * gst/rtsp-server/rtsp-media.c:
9523 * gst/rtsp-server/rtsp-media.h:
9524 media: ignore spurious ASYNC_DONE messages
9525 When we are dynamically adding pads, the addition of the udpsrc elements will
9526 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
9527 the real ASYNC_DONE when everything is prerolled.
9529 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9531 * gst/rtsp-server/rtsp-media-factory.c:
9532 * gst/rtsp-server/rtsp-media-factory.h:
9533 media-factory: make lock macro
9535 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
9537 * gst/rtsp-server/rtsp-client.c:
9538 rtsp-server: Remove unused variable and dead assignment
9540 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
9542 * examples/test-launch.c:
9543 * examples/test-mp4.c:
9544 * examples/test-ogg.c:
9545 * examples/test-readme.c:
9546 * examples/test-sdp.c:
9547 * examples/test-video.c:
9548 examples: Run gst-indent
9550 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
9552 * gst/rtsp-server/rtsp-client.c:
9553 * gst/rtsp-server/rtsp-media-factory.c:
9554 * gst/rtsp-server/rtsp-media-mapping.c:
9555 * gst/rtsp-server/rtsp-media.c:
9556 * gst/rtsp-server/rtsp-params.c:
9557 * gst/rtsp-server/rtsp-sdp.c:
9558 * gst/rtsp-server/rtsp-server.c:
9559 * gst/rtsp-server/rtsp-session-pool.c:
9560 * gst/rtsp-server/rtsp-session.c:
9561 rtsp-server: Run gst-indent
9562 Since it wasn't using the upstream common previously, there was no
9563 indentation check before commiting.
9565 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
9567 * gst/rtsp-server/rtsp-media-mapping.h:
9568 * gst/rtsp-server/rtsp-media.c:
9569 * gst/rtsp-server/rtsp-media.h:
9570 * gst/rtsp-server/rtsp-sdp.c:
9571 * gst/rtsp-server/rtsp-session-pool.h:
9572 * gst/rtsp-server/rtsp-session.c:
9573 * gst/rtsp-server/rtsp-session.h:
9574 rtsp-server: Some more doc fixups
9576 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
9579 Makefile: Add cruft-cleaning support
9581 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
9586 * docs/libs/Makefile.am:
9587 * docs/libs/gst-rtsp-server-docs.sgml:
9588 * docs/libs/gst-rtsp-server-sections.txt:
9589 * docs/libs/gst-rtsp-server.types:
9590 * docs/version.entities.in:
9591 docs: Add gtk-doc build system
9593 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
9595 * gst/rtsp-server/Makefile.am:
9596 Makefile.am: Use standard GIR make behaviour
9598 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
9602 autogen/configure: Bring more in sync to standard gst module behaviour
9604 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9606 * gst/rtsp-server/rtsp-media.c:
9607 media: warn and fail when gstrtpbin is not found
9609 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9612 configure: open 0.11 branch
9614 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
9618 Add common submodule
9620 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
9623 * common/Makefile.am:
9624 * common/c-to-xml.py:
9626 * common/coverage/coverage-report-entry.pl:
9627 * common/coverage/coverage-report.pl:
9628 * common/coverage/coverage-report.xsl:
9629 * common/coverage/lcov.mak:
9630 * common/gettext.patch:
9631 * common/glib-gen.mak:
9632 * common/gst-autogen.sh:
9633 * common/gst-xmlinspect.py:
9635 * common/gstdoc-scangobj:
9636 * common/gtk-doc-plugins.mak:
9637 * common/gtk-doc.mak:
9638 * common/m4/.gitignore:
9639 * common/m4/Makefile.am:
9641 * common/m4/as-ac-expand.m4:
9642 * common/m4/as-auto-alt.m4:
9643 * common/m4/as-compiler-flag.m4:
9644 * common/m4/as-compiler.m4:
9645 * common/m4/as-docbook.m4:
9646 * common/m4/as-libtool-tags.m4:
9647 * common/m4/as-libtool.m4:
9648 * common/m4/as-python.m4:
9649 * common/m4/as-scrub-include.m4:
9650 * common/m4/as-version.m4:
9651 * common/m4/ax_create_stdint_h.m4:
9652 * common/m4/check.m4:
9653 * common/m4/glib-gettext.m4:
9654 * common/m4/gst-arch.m4:
9655 * common/m4/gst-args.m4:
9656 * common/m4/gst-check.m4:
9657 * common/m4/gst-debuginfo.m4:
9658 * common/m4/gst-default.m4:
9659 * common/m4/gst-doc.m4:
9660 * common/m4/gst-error.m4:
9661 * common/m4/gst-feature.m4:
9662 * common/m4/gst-function.m4:
9663 * common/m4/gst-gettext.m4:
9664 * common/m4/gst-glib2.m4:
9665 * common/m4/gst-libxml2.m4:
9666 * common/m4/gst-plugindir.m4:
9667 * common/m4/gst-valgrind.m4:
9668 * common/m4/gtk-doc.m4:
9669 * common/m4/introspection.m4:
9671 * common/mangle-tmpl.py:
9672 * common/plugins.xsl:
9674 * common/release.mak:
9675 * common/scangobj-merge.py:
9676 * common/upload.mak:
9677 common: Remove static version
9679 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
9681 * common/m4/introspection.m4:
9682 Update introspection.m4 to match usage
9684 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9688 Remove old stuff from the README
9690 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9695 === release 0.10.7 ===
9697 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9702 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9704 * examples/test-ogg.c:
9705 test-ogg: remove parsers
9706 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
9707 buffers with timestamps. Using the parsers also seems to break things.
9709 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9711 * bindings/vala/gst-rtsp-server-0.10.vapi:
9712 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9713 Updated Vala bindings
9715 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9717 * common/m4/introspection.m4:
9719 * gst/rtsp-server/Makefile.am:
9720 Added initial gobject-introspection support
9722 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9724 * gst/rtsp-server/rtsp-media-factory.c:
9725 media-factory: don't use host for shared hash key
9726 When we generate the key to share made between connections, don't include the
9727 host used to connect so that we can share media even if between clients that
9728 connected with localhost and ones with the ip address.
9730 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9732 * bindings/vala/Makefile.am:
9733 build: fix distcheck
9735 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9737 * bindings/vala/gst-rtsp-server-0.10.vapi:
9738 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9739 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9740 Update Vala bindings
9742 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9744 * bindings/vala/Makefile.am:
9746 Fix configure checks and installation location for Vala bindings
9749 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9754 === release 0.10.6 ===
9756 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9759 configure: release 0.10.6
9761 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9763 * gst/rtsp-server/rtsp-media.c:
9764 media: help the compiler a little
9766 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9768 * gst/rtsp-server/rtsp-media.c:
9769 * gst/rtsp-server/rtsp-media.h:
9770 * gst/rtsp-server/rtsp-session.c:
9771 media: cleanup media transport before freeing
9772 Cleanup the media transport data before freeing. In particular, remove the qdata
9773 from the rtpsource object.
9775 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9777 * gst/rtsp-server/rtsp-media-factory.c:
9778 * gst/rtsp-server/rtsp-media-factory.h:
9779 * gst/rtsp-server/rtsp-media.c:
9780 * gst/rtsp-server/rtsp-media.h:
9781 media-factory: add eos-shutdown property
9782 Add an eos-shutdown property that will send an EOS to the pipeline before
9783 shutting it down. This allows for nice cleanup in case of a muxer.
9786 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9788 * gst/rtsp-server/rtsp-media.c:
9789 * gst/rtsp-server/rtsp-media.h:
9790 media: use multiudpsink send-duplicates when we can
9791 If we have a new enough multiudpsink with the send-duplicates property, use this
9792 instead of doing our own filtering. Our custom filtering code should eventually
9793 be removed when we can depend on a released -good.
9795 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9797 * gst/rtsp-server/rtsp-media.c:
9798 media: don't leak destinations
9799 Refactor and cleanup the destinations array when the stream is destroyed.
9801 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9803 * gst/rtsp-server/rtsp-media.c:
9804 * gst/rtsp-server/rtsp-media.h:
9805 media: don't add udp addresses multiple times
9806 Keep track of the udp addresses we added to udpsink and never add the same udp
9807 destination twice. This avoids duplicate packets when using multicast.
9809 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9811 * gst/rtsp-server/rtsp-server.c:
9812 server: disable use of SO_LINGER
9813 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
9814 server close()s the connection.
9816 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9818 * gst/rtsp-server/rtsp-server.c:
9819 server: use 5 second linger period in SO_LINGER
9820 Wait 5 seconds before clearing the send buffers and reseting the connection with
9821 the client when we do a close. This should be enough time to get the message to
9825 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
9827 * gst/rtsp-server/rtsp-server.c:
9828 server: use SO_LINGER
9829 SO_LINGER on the socket will make sure that any pending data on the socket is
9830 flushed ASAP and that the socket connection is reset. This makes sure that the
9831 socket can be reused immediately.
9834 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9837 README: add blurb about shared media factories
9839 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
9841 * gst/rtsp-server/rtsp-media.c:
9842 Add stdlib.h for atoi()
9844 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9846 * bindings/python/Makefile.am:
9847 * bindings/vala/Makefile.am:
9848 build: distcheck fixes
9849 Fix 'make distcheck', somewhat (it still fails because it tries to
9850 install files into /usr/share/vala/vapi/ irrespective of the
9853 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9856 configure: bump core/base requirements to released version
9857 Makes things less confusing for people.
9859 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9862 configure: fail if GStreamer core/base requirements are not met
9864 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9866 * gst/rtsp-server/rtsp-client.c:
9867 client: improve client cleanups
9868 Make sure the session does not timeout when using TCP. We need to do this
9869 because quicktime player does not send RTCP for some reason in tunneled
9871 Refactor some cleanup code.
9874 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9876 * gst/rtsp-server/rtsp-session.c:
9877 * gst/rtsp-server/rtsp-session.h:
9878 session: add support for prevent session timeouts
9879 Add an atomix counter to prevent session timeouts when we are, for example,
9882 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9884 * gst/rtsp-server/rtsp-client.c:
9885 client: fix unlink on session timeouts
9886 When our session times out, make sure we unlink all streams in this
9888 Remove the tunnelid when closing the connection.
9890 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9892 * gst/rtsp-server/rtsp-session.c:
9893 session: small cleanups
9895 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9897 * gst/rtsp-server/rtsp-client.c:
9898 client: handle lost_tunnel callbacks
9899 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
9900 hashtable so that we can reuse it for when the client reopens the POST
9902 Close the connection after a TEARDOWN.
9903 Make sure or watchid is cleared when the watch is removed.
9906 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9908 * gst/rtsp-server/rtsp-client.c:
9909 * gst/rtsp-server/rtsp-media.c:
9910 * gst/rtsp-server/rtsp-sdp.c:
9911 rtsp-server: add more support for multicast
9913 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9916 * gst/rtsp-server/rtsp-media.c:
9917 * gst/rtsp-server/rtsp-media.h:
9918 media: allow configuration of allowed lower transport
9920 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9922 * gst/rtsp-server/rtsp-client.h:
9923 * gst/rtsp-server/rtsp-media.c:
9924 * gst/rtsp-server/rtsp-media.h:
9925 * gst/rtsp-server/rtsp-sdp.c:
9926 * gst/rtsp-server/rtsp-sdp.h:
9927 * gst/rtsp-server/rtsp-server.c:
9928 rtsp: keep track of server ip and ipv6
9929 Keep track of how the client connected to the server and setup the udp ports
9930 with the same protocol.
9931 Copy the server ip address in the SDP so that clients can send RTCP back to
9934 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9936 * gst/rtsp-server/rtsp-session.c:
9939 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9941 * gst/rtsp-server/rtsp-client.c:
9942 client: use right size for malloc
9944 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9946 * gst/rtsp-server/rtsp-server.c:
9947 server: comment ipv6 server listening address
9949 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9951 * gst/rtsp-server/rtsp-media.c:
9952 media: allow for ipv6 sockets
9954 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9956 * gst/rtsp-server/rtsp-server.c:
9957 * gst/rtsp-server/rtsp-server.h:
9958 server: rework server part
9959 Allow setting a bind address, make sure we can deal with ipv6.
9960 Remove the port property and change with the service property.
9962 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9964 * gst/rtsp-server/rtsp-media.h:
9965 media: update comments a little
9967 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9969 * gst/rtsp-server/rtsp-client.c:
9970 client: make content-base better
9971 Use the URI formatting functions to make a content-base. Also make sure that
9972 there is a trailing / at the end.
9974 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9976 * gst/rtsp-server/rtsp-client.c:
9977 client: guard against invalid paths
9979 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9981 * examples/test-video.c:
9982 test: catch server bind errors
9984 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
9986 * gst/rtsp-server/rtsp-media.c:
9987 rtspmedia: emit "unprepared" if _prepare fails.
9988 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
9989 media object is removed from its factory's cache.
9991 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9993 * gst/rtsp-server/rtsp-media.c:
9994 media: collect media position when seek completes
9996 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
9998 * gst/rtsp-server/rtsp-client.c:
9999 client: call unlink_streams in client finalize
10002 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10004 * gst/rtsp-server/rtsp-media.c:
10005 media: limit the time to wait to something huge
10006 Avoid waiting forever but limit the timeout to 20 seconds.
10008 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10010 * gst/rtsp-server/rtsp-sdp.c:
10011 sdp: reindent and check for prepared status
10013 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10015 * gst/rtsp-server/rtsp-media.c:
10016 * gst/rtsp-server/rtsp-media.h:
10017 * gst/rtsp-server/rtsp-session.c:
10018 media: avoid doing _get_state() for state changes
10019 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
10020 until the media is prerolled or in error. This avoids doing a blocking call of
10021 gst_element_get_state() that can cause lockups when there is an error.
10024 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10026 * gst/rtsp-server/rtsp-media.c:
10029 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10031 * gst/rtsp-server/rtsp-media-factory.c:
10032 media-factory: better error handling
10033 Improve the error handling a bit.
10035 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10037 * gst/rtsp-server/rtsp-client.c:
10038 client: rework transport parsing
10039 Rework the transport parsing code so that we can ignore transports we don't
10040 support instead of just picking the first one we can parse.
10041 Configure a (for now hardcoded) destination for multicast transports.
10043 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10045 * gst/rtsp-server/rtsp-media.c:
10046 media: set multicast sink parameters
10047 Disable loop and automatic multicast join on the udpsink elements.
10048 Add some more debug info.
10049 Reset some state variables in the right place.
10050 Use the right port numbers for multicast.
10052 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10054 * gst/rtsp-server/rtsp-session.c:
10055 session: handle transport setup correctly
10056 Handle UDP, MCAST and TCP transport negotiation more correctly.
10057 Store the server session SSRC in the transport.
10059 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10061 * gst/rtsp-server/rtsp-client.c:
10062 rtsp-client: implement error_full
10063 Implement error_full to avoid some segfaults when the rtspconnection calls it.
10066 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10069 * gst/rtsp-server/rtsp-client.c:
10070 * gst/rtsp-server/rtsp-server.c:
10071 docs: update docs and comments
10073 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
10075 * gst/rtsp-server/rtsp-sdp.c:
10076 sdp: make server work better when behind a proxy
10078 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10080 * gst/rtsp-server/rtsp-client.c:
10081 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
10083 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10085 * gst/rtsp-server/rtsp-client.c:
10086 * gst/rtsp-server/rtsp-media-factory.c:
10087 * gst/rtsp-server/rtsp-media-mapping.c:
10088 * gst/rtsp-server/rtsp-media.c:
10089 * gst/rtsp-server/rtsp-server.c:
10090 * gst/rtsp-server/rtsp-session-pool.c:
10091 * gst/rtsp-server/rtsp-session.c:
10092 Use GStreamer's debugging subsystem
10094 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10096 * gst/rtsp-server/rtsp-media-factory.c:
10097 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
10099 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10102 back to development
10104 === release 0.10.5 ===
10106 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10111 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10114 configure: bump required versions
10116 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
10118 * gst/rtsp-server/rtsp-client.c:
10119 client: call weak-unref on client->sessions from finalize
10122 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10124 * gst/rtsp-server/rtsp-media.c:
10125 media: Fixed crasher where caps got unref'ed too often
10127 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10130 * pkgconfig/.gitignore:
10131 * pkgconfig/Makefile.am:
10132 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
10133 Added pkg-config file to use gst-rtsp-server uninstalled
10135 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10137 * gst/rtsp-server/rtsp-media.c:
10138 media: add some docs
10140 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
10142 * gst/rtsp-server/rtsp-client.c:
10143 rtsp: Use gst_rtsp_watch_send_message().
10144 Use gst_rtsp_watch_send_message() since the old API which used
10145 gst_rtsp_watch_queue_message() has been deprecated.
10147 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10150 back to development
10152 === release 0.10.4 ===
10154 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10159 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10161 * gst/rtsp-server/rtsp-client.c:
10162 * gst/rtsp-server/rtsp-session.c:
10163 * gst/rtsp-server/rtsp-session.h:
10164 rtsp: allocate channels in TCP mode
10165 When the client does not provide us with channels in TCP mode, allocate channels
10168 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10170 * gst/rtsp-server/rtsp-client.c:
10171 client: don't crash when tunnelid is missing
10172 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
10173 don't crash but return an error response to the client.
10176 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10178 * bindings/vala/gst-rtsp-server-0.10.vapi:
10179 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
10180 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
10181 bindings: update vala bindings with new method
10183 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10185 * gst/rtsp-server/rtsp-session-pool.c:
10186 * gst/rtsp-server/rtsp-session-pool.h:
10187 sessionpool: add function to filter sessions
10188 Add generic function to retrieve/remove sessions.
10190 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10193 configure: bump core/base requirements to release
10195 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10197 * gst/rtsp-server/rtsp-media.c:
10198 media: fix indentation
10200 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10202 * gst/rtsp-server/rtsp-media.c:
10203 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
10205 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10207 * gst/rtsp-server/rtsp-media.c:
10208 set state and remove elements of media in for loop
10210 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
10212 * bindings/vala/gst-rtsp-server-0.10.vapi:
10213 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
10214 Added gst_rtsp_media_remove_elements function to Vala bindings
10216 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
10218 * gst/rtsp-server/rtsp-media.c:
10219 * gst/rtsp-server/rtsp-media.h:
10220 Added gst_rtsp_media_remove_elements function
10222 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
10224 * gst/rtsp-server/rtsp-media.c:
10225 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
10227 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10229 * bindings/vala/gst-rtsp-server-0.10.vapi:
10230 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
10231 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
10232 Updated Vala bindings
10234 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10236 * gst/rtsp-server/rtsp-media.c:
10237 * gst/rtsp-server/rtsp-media.h:
10238 Added vmethod unprepare to GstRTSPMedia
10239 The default implementation sets the state of the pipeline to GST_STATE_NULL
10241 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10243 * gst/rtsp-server/rtsp-media-factory.c:
10244 * gst/rtsp-server/rtsp-media-factory.h:
10245 Made collect_streams function public
10247 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10249 * gst/rtsp-server/rtsp-media-factory.c:
10250 * gst/rtsp-server/rtsp-media-factory.h:
10251 * gst/rtsp-server/rtsp-media.c:
10252 Added vmethod create_pipeline to GstRTSPMediaFactory
10253 The pipeline is created in this method and the GstRTSPMedia's element is added to it
10255 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10257 * gst/rtsp-server/rtsp-client.c:
10258 client: use g_source_destroy()
10259 We need to use g_source_destroy() because we might have added the source to a
10260 different main context than the default one.
10262 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10264 * gst/rtsp-server/Makefile.am:
10265 * gst/rtsp-server/rtsp-client.c:
10266 * gst/rtsp-server/rtsp-params.c:
10267 * gst/rtsp-server/rtsp-params.h:
10268 rtsp: prepare for handling GET/SET_PARAMETER
10269 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
10271 Fix return codes of handlers.
10273 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10275 * gst/rtsp-server/rtsp-media.c:
10276 media: don't leak session pads
10278 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10280 * gst/rtsp-server/rtsp-media.c:
10281 media: clean up the messages a bit
10283 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10285 * gst/rtsp-server/rtsp-sdp.c:
10286 sdp: warn and skip streams without media
10288 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10290 * bindings/vala/gst-rtsp-server-0.10.vapi:
10291 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
10292 vala: Fixed typo in header file of RTSPMediaStream
10294 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10296 * gst/rtsp-server/rtsp-media.c:
10298 Fix a debug message
10299 Make dumping RTCP stats configurable
10301 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10303 * gst/rtsp-server/rtsp-media.c:
10304 media: be less verbose and leak less
10306 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10308 * gst/rtsp-server/rtsp-media.c:
10309 media: don't leak the destination address
10311 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10313 * gst/rtsp-server/rtsp-client.c:
10314 * gst/rtsp-server/rtsp-media.c:
10315 * gst/rtsp-server/rtsp-media.h:
10316 * gst/rtsp-server/rtsp-session.c:
10317 * gst/rtsp-server/rtsp-session.h:
10318 rtsp: use RTCP to keep the session alive
10319 Use the RTCP rtcp-from stats field to find the associated session and use this
10320 to keep the session alive.
10322 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10324 * gst/rtsp-server/rtsp-session.c:
10325 session: add 5sec to the real session timeout
10326 Allow the session to live 5sec longer before really timing out. This should give
10327 clients some extra time to keep the session active.
10329 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10331 * gst/rtsp-server/rtsp-client.c:
10332 client: replay OK to GET/SET_PARAMETER
10333 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
10334 so that we return OK for those requests.
10336 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10338 * gst/rtsp-server/rtsp-media.c:
10339 * gst/rtsp-server/rtsp-media.h:
10340 media: keep track of active transports
10341 Keep track of which transport is active to avoid closing the connection too
10343 Remove the destination transport also when going to NULL.
10344 Print some stats about the SDES and other RTCP messages we receive from the
10347 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10349 * examples/.gitignore:
10350 * examples/Makefile.am:
10351 * examples/test-sdp.c:
10352 example: add SDP relay example
10354 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10356 * gst/rtsp-server/rtsp-media.c:
10357 media: also count active TCP connections
10359 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10361 * gst/rtsp-server/rtsp-media-factory.c:
10362 * gst/rtsp-server/rtsp-media.c:
10363 * gst/rtsp-server/rtsp-media.h:
10364 rtsp: add support for dynamic elements
10365 Add support for dynamic elements.
10366 Don't set live pipelines back to paused.
10368 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10370 * gst/rtsp-server/rtsp-sdp.c:
10371 sdp: don't add encoding name when absent in caps
10373 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10375 * gst/rtsp-server/rtsp-client.c:
10376 client: warn when we can't do RTP-Info
10378 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10380 * gst/rtsp-server/rtsp-media-factory.c:
10381 factory: factor out the stream construction
10383 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10385 * gst/rtsp-server/rtsp-client.c:
10386 client: only add RTP-Info when we have the info
10387 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
10390 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10393 back to development
10395 === release 0.10.3 ===
10397 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10401 - Fixes a bug where it put the wrong verion in pkgconfig
10402 - Link RTP and RTCP sources
10404 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10406 * gst/rtsp-server/rtsp-media.c:
10407 * gst/rtsp-server/rtsp-media.h:
10408 media: link the RTP udpsrc to the session manager
10409 Link the RTP udpsrc and the appsrc to the session manager so that they don't
10410 shut down when the client sends a packet to open firewalls.
10412 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10414 * pkgconfig/gst-rtsp-server.pc.in:
10415 Don't use hard-coded version number in pkg-config file
10417 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10420 back to development
10422 === release 0.10.2 ===
10424 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10429 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10432 * common/m4/.gitignore:
10433 * examples/.gitignore:
10434 * pkgconfig/.gitignore:
10435 add some .gitignore files
10437 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10439 * gst/rtsp-server/rtsp-media.c:
10440 media: seek to key frames
10442 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10444 * gst/rtsp-server/rtsp-media.c:
10445 media: emit the unprepared signal by id
10446 Emit the unprepared signal by id instead of name and set the media as
10449 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10451 * gst/rtsp-server/rtsp-media.c:
10452 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
10454 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10456 * gst/rtsp-server/rtsp-server.c:
10457 Added finalize function to GstRTPSPServer to unref session pool and media mapping
10459 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10461 * bindings/vala/gst-rtsp-server-0.10.vapi:
10462 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
10463 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
10464 Updated vala bindings
10466 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10468 * gst/rtsp-server/Makefile.am:
10469 * gst/rtsp-server/rtsp-client.c:
10470 * gst/rtsp-server/rtsp-media.c:
10471 server: use appsink and appsrc with the API
10472 Use the appsink/appsrc API instead of the signals for higher
10475 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10477 * examples/test-ogg.c:
10478 tests: set the payload type correctly
10480 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10482 * gst/rtsp-server/rtsp-media-factory.c:
10483 factory: connect to the unprepare signal
10484 Connect to the unprepare signal for non-reusable media so that we can remove
10485 them from the cache.
10487 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10489 * gst/rtsp-server/rtsp-media.c:
10490 * gst/rtsp-server/rtsp-media.h:
10491 media: add signal to notify of unprepare
10493 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10495 * gst/rtsp-server/rtsp-media.c:
10496 * gst/rtsp-server/rtsp-media.h:
10497 media: more work on making the media shared
10498 Add a reusable flag to medias, indicating that they can be reused after a state
10502 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10504 * examples/test-readme.c:
10505 examples: mark the example as shared for testing
10507 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10509 * gst/rtsp-server/rtsp-media.c:
10510 * gst/rtsp-server/rtsp-media.h:
10511 client: support shared media
10512 Always perform the state actions even if the target state of the pipeline is
10513 already correct, we still want to add/remove the transports when we are dealing
10515 Keep a counter of the number of active transports for a media so that we can use
10516 this to perform a state change when needed.
10517 Perform a state change of the pipeline only when the first transport was added
10518 or when there are no active transports.
10520 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10522 * gst/rtsp-server/rtsp-client.c:
10523 client: fix refcounting crasher
10524 Don't need to remove the weak refs in the finalize methods, they are already
10525 removed in the dispose.
10526 Don't register the callback with a DestroyNofity.
10528 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10530 * gst/rtsp-server/rtsp-client.c:
10531 Fix rtsp client refcount management in TCP mode.
10532 Don't unref a client ref we never had. Fixes an unref
10533 of an already-free client object after a client
10534 teardown request for me.
10536 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10538 * gst/rtsp-server/rtsp-session.c:
10539 docs: fix typo in API docs
10541 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10543 * gst/rtsp-server/rtsp-media.c:
10544 More seeking fixes.
10545 Keep the udp sources in playing even if we go to paused. unlock the sources when
10547 Add some more debug info.
10548 Only seek when we need to.
10549 Keep track of the position when we go to paused.
10551 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10553 * gst/rtsp-server/rtsp-client.c:
10554 * gst/rtsp-server/rtsp-media.c:
10555 * gst/rtsp-server/rtsp-media.h:
10556 Add beginnings of seeking.
10557 Parse the Range header and perform a seek on the pipeline for the requested
10558 position. It's disabled currently until I figure out what's going wrong.
10560 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10562 * gst/rtsp-server/rtsp-client.c:
10563 allow pause requests for now.
10566 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10568 * gst/rtsp-server/rtsp-client.c:
10569 Remove weak ref on the session in teardown
10570 We need to remove our weakref from the session when we do a teardown because
10571 else we close the TCP connection prematurely.
10573 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10575 * gst/rtsp-server/rtsp-client.c:
10576 * gst/rtsp-server/rtsp-client.h:
10577 * gst/rtsp-server/rtsp-session-pool.c:
10578 Do some more session cleanup
10579 Make session timeout kill the TCP connection that currently watches the
10581 Remove the client timeout property.
10583 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10585 * gst/rtsp-server/rtsp-client.c:
10586 * gst/rtsp-server/rtsp-client.h:
10587 * gst/rtsp-server/rtsp-media.c:
10588 * gst/rtsp-server/rtsp-media.h:
10589 * gst/rtsp-server/rtsp-server.c:
10590 * gst/rtsp-server/rtsp-session.c:
10591 * gst/rtsp-server/rtsp-session.h:
10593 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
10596 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10598 * examples/Makefile.am:
10599 * examples/test-launch.c:
10600 Add example server that takes launch lines
10601 Add an example server that streams any -launch line.
10603 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10605 * examples/test-readme.c:
10606 * gst/rtsp-server/rtsp-client.c:
10607 * gst/rtsp-server/rtsp-media.c:
10608 * gst/rtsp-server/rtsp-media.h:
10609 Add support for live streams
10610 Add support for live streams and ranges
10611 Start on handling TCP data transfer.
10613 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10615 * gst/rtsp-server/rtsp-media.c:
10616 Free the pipeline before other things
10619 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10621 * gst/rtsp-server/rtsp-client.c:
10622 Only free the pending tunnel if there is one
10625 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10627 * gst/rtsp-server/rtsp-client.c:
10628 * gst/rtsp-server/rtsp-client.h:
10629 * gst/rtsp-server/rtsp-media.c:
10630 rtsp-server: Add support for tunneling
10631 Add support for tunneling over HTTP.
10632 Use new connection methods to retrieve the url.
10633 Dispatch messages based on the message type instead of blindly
10634 assuming it's always a request.
10635 Keep track of the watch id so that we can remove it later.
10636 Set the media pipeline to NULL before unreffing the pipeline.
10638 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10640 * gst/rtsp-server/rtsp-client.c:
10641 * gst/rtsp-server/rtsp-client.h:
10642 Fix for channel -> watch rename in gstreamer
10643 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
10645 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10647 * gst/rtsp-server/rtsp-client.c:
10648 * gst/rtsp-server/rtsp-client.h:
10650 Use the async RTSP channels instead of spawning a new thread for each client.
10651 If a sessionid is specified in a request, fail if we don't have the session.
10653 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10655 * gst/rtsp-server/rtsp-media.c:
10656 Add better debug info
10657 Add some better debug info.
10659 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10661 * examples/test-video.c:
10663 Add support for session timeouts in the example.
10665 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10667 * gst/rtsp-server/rtsp-session-pool.c:
10668 * gst/rtsp-server/rtsp-session-pool.h:
10669 Pass GTimeVal around for performance reasons
10670 Get the current time only once and pass it around so that sessions don't have to
10671 get the current time anymore.
10672 Add experimental support for a GSource that dispatches when the session needs to
10675 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10677 * gst/rtsp-server/rtsp-session.c:
10678 * gst/rtsp-server/rtsp-session.h:
10679 Add better support for session timeouts
10680 Add a method to request the number of milliseconds when a session will timeout.
10682 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10684 * gst/rtsp-server/rtsp-media.c:
10685 * gst/rtsp-server/rtsp-media.h:
10686 Add suport for RTP manager monitoring
10687 Add the first stage in monitoring the rtp manager.
10688 Make sure we don't update the state to something we don't want.
10690 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10692 * gst/rtsp-server/rtsp-client.c:
10693 Add support for session keepalive
10694 Get and update the session timeout for all requests. get the session as early as
10697 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10699 * gst/rtsp-server/rtsp-media-factory.h:
10700 * gst/rtsp-server/rtsp-media.c:
10701 * gst/rtsp-server/rtsp-media.h:
10702 Handle media bus messages
10703 Handle media bus messages in a custom mainloop and dispatch them to the
10704 RTSPMedia objects. Let the default implementation handle some common messages.
10706 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10708 * gst/rtsp-server/rtsp-client.c:
10709 * gst/rtsp-server/rtsp-session-pool.c:
10710 * gst/rtsp-server/rtsp-session.c:
10711 Some more session timeout handling
10712 Move the session header setting code to a central place so that we always add
10713 the timeout parameter too.
10714 Handle timeouts by running the session cleanup code.
10715 Stop media before cleaning up.
10717 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10719 * gst/rtsp-server/rtsp-client.c:
10720 * gst/rtsp-server/rtsp-client.h:
10721 Add timeout property
10722 Add a timeout property ot the client and make the other properties into GObject
10725 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10727 * gst/rtsp-server/rtsp-session-pool.c:
10728 Use getters and setters in property code
10729 Use the getters and setters for the timeout property instead of locking
10732 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10734 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
10736 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10738 * gst/rtsp-server/rtsp-session-pool.c:
10739 * gst/rtsp-server/rtsp-session-pool.h:
10740 * gst/rtsp-server/rtsp-session.c:
10741 * gst/rtsp-server/rtsp-session.h:
10742 Add more timeout stuff
10743 Add method to check if a session is expired.
10744 Add method to perform cleanup on a session pool.
10746 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10748 * gst/rtsp-server/rtsp-client.c:
10749 * gst/rtsp-server/rtsp-session-pool.c:
10750 * gst/rtsp-server/rtsp-session-pool.h:
10751 * gst/rtsp-server/rtsp-session.c:
10752 * gst/rtsp-server/rtsp-session.h:
10753 Add beginnings of session timeouts and limits
10754 Add the timeout value to the Session header for unusual timeout values.
10755 Allow us to configure a limit to the amount of active sessions in a pool. Set a
10756 limit on the amount of retry we do after a sessionid collision.
10757 Add properties to the sessionid and the timeout of a session. Keep track of
10758 creation time and last access time for sessions.
10760 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10762 * gst/rtsp-server/rtsp-client.c:
10763 * gst/rtsp-server/rtsp-media.c:
10764 * gst/rtsp-server/rtsp-media.h:
10765 * gst/rtsp-server/rtsp-sdp.c:
10766 * gst/rtsp-server/rtsp-session-pool.c:
10767 * gst/rtsp-server/rtsp-session.c:
10768 * gst/rtsp-server/rtsp-session.h:
10769 Cleanup of sessions and more
10770 Fix the refcounting of media and sessions in the client. Properly clean up the
10771 session data when the client performs a teardown.
10772 Add Server header to responses.
10773 Allow for multiple uri setups in one session.
10774 Add Range header to the PLAY response and add the range attribute to the SDP
10776 Fix the session pool remove method, it used the wrong key in the hashtable. Also
10777 give the ownership of the sessionid to the session object.
10779 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10781 * gst/rtsp-server/rtsp-server.c:
10782 * gst/rtsp-server/rtsp-server.h:
10784 Rename the 'server_port' variable to simply 'port'.
10786 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10789 * gst/rtsp-server/rtsp-client.c:
10790 * gst/rtsp-server/rtsp-media.c:
10791 * gst/rtsp-server/rtsp-media.h:
10792 * gst/rtsp-server/rtsp-session.c:
10793 * gst/rtsp-server/rtsp-session.h:
10794 Rework the way we handle transports for streams
10795 Make the media accept an array of transports for the streams that we have
10796 configured for the play/pause requests.
10797 Implement server states for a client and its media.
10798 Require 0.10.22.1 (git HEAD) of gstreamer.
10800 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10802 * gst/rtsp-server/rtsp-client.c:
10803 * gst/rtsp-server/rtsp-media-factory.c:
10804 Drop const from functions dealing with urls
10805 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
10806 have the right const in them.
10808 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10810 * gst/rtsp-server/rtsp-client.c:
10811 * gst/rtsp-server/rtsp-media.c:
10812 * gst/rtsp-server/rtsp-sdp.c:
10816 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10818 * gst/rtsp-server/rtsp-client.c:
10819 * gst/rtsp-server/rtsp-media-factory.c:
10820 * gst/rtsp-server/rtsp-media.c:
10821 * gst/rtsp-server/rtsp-media.h:
10823 Don't keep a reference to the GstRTSPMedia in the stream.
10824 Free more things when freeing the GstRTSPMedia.
10826 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10829 * gst/rtsp-server/rtsp-media-factory.c:
10830 * gst/rtsp-server/rtsp-media-factory.h:
10831 * gst/rtsp-server/rtsp-media.c:
10832 * gst/rtsp-server/rtsp-media.h:
10833 * gst/rtsp-server/rtsp-server.c:
10834 * gst/rtsp-server/rtsp-server.h:
10835 More docs and small cleanups
10836 Add some more docs and update the README
10837 Cleanup some method names.
10838 Remove an unneeded idx field in the GstRTSPMediaStream
10840 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10843 * examples/Makefile.am:
10844 * examples/test-readme.c:
10845 Add a README and more example code
10846 Add a README file that contains a small introduction on how to use the server
10847 along with the example code explained in the readme.
10849 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10851 * gst/rtsp-server/rtsp-media.c:
10852 * gst/rtsp-server/rtsp-server.c:
10853 Fix some leaks and change default port
10854 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
10855 we finished the initial preroll. If we keep them locked, setting the pipeline to
10856 NULL will not stop and clean up the sources correctly.
10857 Change the default RTSP port to 8554 aka the official alternative RTSP port.
10859 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10861 * gst/rtsp-server/rtsp-session.c:
10862 * gst/rtsp-server/rtsp-session.h:
10863 Cleanups to the session object
10864 Remove some unneeded variables in the session state of a stream such as the
10865 owner media and the server transport.
10866 Get the configuration of a media stream in a session based on the media_stream
10867 in the original object instead of our cached index.
10868 Free more data in the finalize method.
10870 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10872 * gst/rtsp-server/rtsp-client.c:
10873 * gst/rtsp-server/rtsp-client.h:
10874 Cleanups and reuse media from DESCRIBE
10875 Handle thread create errors.
10876 Rename some internal methods to better match what they actually do.
10877 Handle misconfiguration of session_pool and media_mapping gracefully.
10878 Cache the DESCRIBE media and uri in the client connection and reuse them when
10879 we receive a SETUP request in the same connection for the same uri.
10880 Cleanup the client connection object.
10882 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10884 * gst/rtsp-server/rtsp-media-factory.c:
10885 * gst/rtsp-server/rtsp-media-factory.h:
10886 * gst/rtsp-server/rtsp-media.c:
10887 * gst/rtsp-server/rtsp-media.h:
10888 Add shared properties to media and factory
10889 Add the shared property to media.
10890 Implement some simple caching in the factory depending on if the media is shared
10893 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10895 * gst/rtsp-server/rtsp-client.c:
10896 Add a little comment
10897 Add some comment about the content-base header.
10899 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10901 * examples/Makefile.am:
10902 * examples/test-mp4.c:
10903 * examples/test-ogg.c:
10904 * examples/test-video.c:
10905 * gst/rtsp-server/Makefile.am:
10906 * gst/rtsp-server/rtsp-client.c:
10907 * gst/rtsp-server/rtsp-client.h:
10908 * gst/rtsp-server/rtsp-media-factory.c:
10909 * gst/rtsp-server/rtsp-media-factory.h:
10910 * gst/rtsp-server/rtsp-media.c:
10911 * gst/rtsp-server/rtsp-media.h:
10912 * gst/rtsp-server/rtsp-sdp.c:
10913 * gst/rtsp-server/rtsp-sdp.h:
10914 * gst/rtsp-server/rtsp-server.c:
10915 * gst/rtsp-server/rtsp-server.h:
10916 * gst/rtsp-server/rtsp-session.c:
10917 * gst/rtsp-server/rtsp-session.h:
10918 Reorganize things, prepare for media sharing
10919 Added various other test server examples
10920 Move the SDP message generation to a separate helper.
10921 Refactor common code for finding the session.
10922 Add content-base for realplayer compatibility
10923 Clean up request uris before processing for better vlc compatibility.
10924 Move prerolling and pipeline construction to the RTSPMedia object.
10925 Use multiudpsink for future pipeline reuse.
10927 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10930 Back to development
10933 === release 0.10.1 ===
10935 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10938 Make 0.10.1 release
10941 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10943 * bindings/vala/Makefile.am:
10945 Add more directories and files to the dist.
10947 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10949 * bindings/python/Makefile.am:
10950 * bindings/python/rtspserver.override:
10951 Fixed compile error of python bindings
10953 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10955 * bindings/vala/gst-rtsp-server-0.10.vapi:
10956 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
10957 Marked values as nullable accordingly
10959 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10961 * bindings/vala/gst-rtsp-server-0.10.vapi:
10962 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
10963 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
10964 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
10965 Updated Vala bindings
10967 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10969 * gst/rtsp-server/rtsp-client.c:
10970 * gst/rtsp-server/rtsp-media-mapping.c:
10971 * gst/rtsp-server/rtsp-media-mapping.h:
10972 * gst/rtsp-server/rtsp-media.h:
10973 * gst/rtsp-server/rtsp-session-pool.h:
10974 Cleanups and doc updates
10975 Add some more documentation and do some minor cleanups here and there.
10977 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10979 * gst/rtsp-server/rtsp-client.c:
10980 * gst/rtsp-server/rtsp-media-factory.c:
10981 * gst/rtsp-server/rtsp-media-factory.h:
10982 * gst/rtsp-server/rtsp-media.c:
10983 * gst/rtsp-server/rtsp-media.h:
10984 * gst/rtsp-server/rtsp-session.c:
10985 * gst/rtsp-server/rtsp-session.h:
10987 Rename GstRTSPMediaBin to GstRTSPMedia
10988 Parse the request url into a GstRTSPUri object and pass this object to the
10989 various handlers and methods that require the uri.
10991 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10995 Add some more docs and remove some old code from the example.
10997 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10999 * gst/rtsp-server/rtsp-client.c:
11000 Handle state change failures better
11001 Handle state change failures better when changing the state of the pipeline to
11004 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11006 * gst/rtsp-server/rtsp-media-factory.c:
11007 * gst/rtsp-server/rtsp-media-factory.h:
11008 Make element creation more extendible
11009 Add get_element vmethod to the default MediaFactory so that subclasses can just
11010 override that method and still use the default logic for making a MediaBin from
11013 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11016 * gst/rtsp-server/Makefile.am:
11017 * gst/rtsp-server/rtsp-client.c:
11018 * gst/rtsp-server/rtsp-client.h:
11019 * gst/rtsp-server/rtsp-media-factory.c:
11020 * gst/rtsp-server/rtsp-media-factory.h:
11021 * gst/rtsp-server/rtsp-media-mapping.c:
11022 * gst/rtsp-server/rtsp-media-mapping.h:
11023 * gst/rtsp-server/rtsp-media.c:
11024 * gst/rtsp-server/rtsp-media.h:
11025 * gst/rtsp-server/rtsp-server.c:
11026 * gst/rtsp-server/rtsp-server.h:
11027 * gst/rtsp-server/rtsp-session.c:
11028 * gst/rtsp-server/rtsp-session.h:
11029 Make the server handle arbitrary pipelines
11030 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
11031 The GstMediaBin object has a handle to a bin with elements and to a list of
11032 GstMediaStream objects that this bin produces.
11033 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
11034 with methods to register and remove those mappings.
11035 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
11036 used by the server instance.
11037 Modify the example application so that it shows how to create custom pipelines
11038 attached to a specific mount point.
11039 Various misc cleanps.
11041 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11043 * gst/rtsp-server/rtsp-server.c:
11044 * gst/rtsp-server/rtsp-server.h:
11045 Allow setting a custom media factory for a server
11047 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11049 * gst/rtsp-server/rtsp-client.c:
11050 * gst/rtsp-server/rtsp-client.h:
11051 Allow setting a custom media factory for a client.
11053 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11055 * gst/rtsp-server/Makefile.am:
11056 Add Makefile entry for the media factory
11058 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11060 * gst/rtsp-server/rtsp-media-factory.c:
11061 * gst/rtsp-server/rtsp-media-factory.h:
11062 Add media factory to map urls to media pipeline objects.
11064 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11066 * gst/rtsp-server/rtsp-media.c:
11067 * gst/rtsp-server/rtsp-media.h:
11068 Add comments. Remove unused field
11070 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11072 * gst/rtsp-server/rtsp-session-pool.c:
11073 * gst/rtsp-server/rtsp-session-pool.h:
11074 Allow custom session pools to override the session id allocation algorithms Add some comments.
11076 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11078 * gst/rtsp-server/rtsp-session.h:
11081 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11083 * gst/rtsp-server/rtsp-client.c:
11084 * gst/rtsp-server/rtsp-client.h:
11085 Move the connection code in one place Add some comments
11087 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11089 * gst/rtsp-server/rtsp-server.c:
11090 * gst/rtsp-server/rtsp-server.h:
11091 Make vmethod to create and accept new clients. Add some docs.
11093 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11095 * gst/rtsp-server/rtsp-server.c:
11096 * gst/rtsp-server/rtsp-server.h:
11097 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
11099 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11101 * gst/rtsp-server/rtsp-client.c:
11102 * gst/rtsp-server/rtsp-client.h:
11103 Name the parameters more appropriately.
11105 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11107 * gst/rtsp-server/rtsp-session-pool.c:
11108 Do some more cleanup of the session pool.
11110 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11112 * gst/rtsp-server/Makefile.am:
11113 * gst/rtsp-server/rtsp-client.c:
11114 Check if return value of gst_rtsp_session_get_media is not NULL
11116 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11118 * gst/rtsp-server/Makefile.am:
11119 Install rtsp-session and rtsp-session-pool headers
11121 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11126 * bindings/python/Makefile.am:
11127 * bindings/python/arg-types.py:
11128 * bindings/python/codegen/Makefile.am:
11129 * bindings/python/codegen/__init__.py:
11130 * bindings/python/codegen/argtypes.py:
11131 * bindings/python/codegen/code-coverage.py:
11132 * bindings/python/codegen/codegen.py:
11133 * bindings/python/codegen/definitions.py:
11134 * bindings/python/codegen/defsparser.py:
11135 * bindings/python/codegen/docextract.py:
11136 * bindings/python/codegen/docgen.py:
11137 * bindings/python/codegen/fileprefix.override:
11138 * bindings/python/codegen/fileprefixmodule.c:
11139 * bindings/python/codegen/h2def.py:
11140 * bindings/python/codegen/mergedefs.py:
11141 * bindings/python/codegen/mkskel.py:
11142 * bindings/python/codegen/override.py:
11143 * bindings/python/codegen/reversewrapper.py:
11144 * bindings/python/codegen/scmexpr.py:
11145 * bindings/python/rtspserver-types.defs:
11146 * bindings/python/rtspserver.defs:
11147 * bindings/python/rtspserver.override:
11148 * bindings/python/rtspservermodule.c:
11150 Add python bindings.
11152 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11154 * bindings/Makefile.am:
11156 Don't go into python dir when requirements for python bindings are missing
11158 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11160 * bindings/Makefile.am:
11161 * bindings/vala/Makefile.am:
11163 Install Vala bindings if vala is available
11165 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11167 * bindings/vala/gst-rtsp-server-0.10.deps:
11168 * bindings/vala/gst-rtsp-server-0.10.vapi:
11169 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
11170 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
11171 * bindings/vala/packages/gst-rtsp-server-0.10.files:
11172 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11173 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11174 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
11175 Regenerated Vala bindings
11177 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11179 * bindings/vala/gst-rtsp-server.vapi:
11180 * bindings/vala/packages/gst-rtsp-server.metadata:
11181 Fixed typo in included headers for vala bindings
11183 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11187 * pkgconfig/Makefile.am:
11188 * pkgconfig/gst-rtsp-server.pc.in:
11189 Added pkgconfig file
11191 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
11193 * bindings/vala/gst-rtsp-server.vapi:
11194 * bindings/vala/packages/gst-rtsp-server.excludes:
11195 * bindings/vala/packages/gst-rtsp-server.gi:
11196 * bindings/vala/packages/gst-rtsp-server.metadata:
11197 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
11199 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
11201 * bindings/vala/gst-rtsp-server.vapi:
11202 * bindings/vala/packages/gst-rtsp-server.deps:
11203 * bindings/vala/packages/gst-rtsp-server.files:
11204 * bindings/vala/packages/gst-rtsp-server.gi:
11205 * bindings/vala/packages/gst-rtsp-server.metadata:
11206 * bindings/vala/packages/gst-rtsp-server.namespace:
11207 Added Vala bindings
11209 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
11211 * gst/rtsp-server/rtsp-session.c:
11212 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
11214 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
11216 * examples/Makefile.am:
11217 * gst/rtsp-server/Makefile.am:
11218 Put GStreamer version in library name
11220 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11222 * examples/Makefile.am:
11223 * gst/rtsp-server/Makefile.am:
11224 Fix some issues to pass distcheck
11226 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11228 * gst/rtsp-server/rtsp-server.c:
11229 Added port property to GstRTSPServer class.
11231 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11236 * examples/Makefile.am:
11239 * gst/rtsp-server/Makefile.am:
11240 * gst/rtsp-server/rtsp-client.c:
11241 * gst/rtsp-server/rtsp-client.h:
11242 * gst/rtsp-server/rtsp-media.c:
11243 * gst/rtsp-server/rtsp-media.h:
11244 * gst/rtsp-server/rtsp-server.c:
11245 * gst/rtsp-server/rtsp-server.h:
11246 * gst/rtsp-server/rtsp-session-pool.c:
11247 * gst/rtsp-server/rtsp-session-pool.h:
11248 * gst/rtsp-server/rtsp-session.c:
11249 * gst/rtsp-server/rtsp-session.h:
11251 Split in library and example program
11253 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
11255 * src/rtsp-client.h:
11256 Removed obsolete variable
11258 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
11260 * src/rtsp-client.c:
11261 * src/rtsp-client.h:
11262 Removed pipeline variable GstRTSPClient, because it's only used in one function
11264 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11266 * src/rtsp-media.c:
11267 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
11269 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
11271 * src/rtsp-session.c:
11272 Initialize some more vars.
11274 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
11276 * src/rtsp-session.c:
11277 Initialize variable to avoid compiler warning.
11279 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
11282 Add a reasonable generic .gitignore
projects / platform / upstream / gstreamer.git / blob