3 2018-03-19 20:27:04 +0000 Tim-Philipp Müller <tim@centricular.com>
9 * gst-rtsp-server.doap:
13 === release 1.13.91 ===
15 2018-03-13 19:28:33 +0000 Tim-Philipp Müller <tim@centricular.com>
21 * gst-rtsp-server.doap:
25 2018-03-13 13:30:41 +0000 Tim-Philipp Müller <tim@centricular.com>
27 * gst/rtsp-server/Makefile.am:
28 * gst/rtsp-server/meson.build:
29 * gst/rtsp-server/rtsp-address-pool.h:
30 * gst/rtsp-server/rtsp-auth.h:
31 * gst/rtsp-server/rtsp-client.h:
32 * gst/rtsp-server/rtsp-context.h:
33 * gst/rtsp-server/rtsp-media-factory-uri.h:
34 * gst/rtsp-server/rtsp-media-factory.h:
35 * gst/rtsp-server/rtsp-media.h:
36 * gst/rtsp-server/rtsp-mount-points.h:
37 * gst/rtsp-server/rtsp-onvif-client.h:
38 * gst/rtsp-server/rtsp-onvif-media-factory.h:
39 * gst/rtsp-server/rtsp-onvif-media.h:
40 * gst/rtsp-server/rtsp-onvif-server.h:
41 * gst/rtsp-server/rtsp-params.h:
42 * gst/rtsp-server/rtsp-permissions.h:
43 * gst/rtsp-server/rtsp-sdp.h:
44 * gst/rtsp-server/rtsp-server-prelude.h:
45 * gst/rtsp-server/rtsp-server.h:
46 * gst/rtsp-server/rtsp-session-media.h:
47 * gst/rtsp-server/rtsp-session-pool.h:
48 * gst/rtsp-server/rtsp-session.h:
49 * gst/rtsp-server/rtsp-stream-transport.h:
50 * gst/rtsp-server/rtsp-stream.h:
51 * gst/rtsp-server/rtsp-thread-pool.h:
52 * gst/rtsp-server/rtsp-token.h:
53 rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API
54 We need different export decorators for the different libs.
55 For now no actual change though, just rename before the release,
56 and add prelude headers to define the new decorator to GST_EXPORT.
58 2018-03-07 12:20:05 +0200 Sebastian Dröge <sebastian@centricular.com>
60 * gst/rtsp-server/rtsp-onvif-media-factory.c:
61 rtsp-onvif-media-factory: Document that backchannel pipelines must end with async=false sinks
62 https://bugzilla.gnome.org/show_bug.cgi?id=794143
64 === release 1.13.90 ===
66 2018-03-03 22:49:34 +0000 Tim-Philipp Müller <tim@centricular.com>
72 * gst-rtsp-server.doap:
76 2018-03-02 16:24:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
78 * gst/rtsp-server/rtsp-media-factory.c:
79 * gst/rtsp-server/rtsp-permissions.c:
80 permissions: add Since tags and example for new API
82 2018-03-02 01:36:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
84 * docs/libs/gst-rtsp-server-sections.txt:
85 * gst/rtsp-server/rtsp-media-factory.c:
86 * gst/rtsp-server/rtsp-media-factory.h:
87 * gst/rtsp-server/rtsp-permissions.c:
88 * gst/rtsp-server/rtsp-permissions.h:
89 * tests/check/gst/permissions.c:
90 permissions: more bindings-friendly API
91 https://bugzilla.gnome.org/show_bug.cgi?id=793975
93 2018-03-01 19:28:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
96 meson: enable more warnings
98 2018-02-28 21:12:43 +0200 Sebastian Dröge <sebastian@centricular.com>
100 * gst/rtsp-server/rtsp-client.c:
101 rtsp-client: Place netaddress meta on packets received via TCP
102 This allows us to later map signals from rtpbin/rtpsource back to the
103 corresponding stream transport, and allows to do keep-alive based on
104 RTCP packets in case of TCP media transport.
105 https://bugzilla.gnome.org/show_bug.cgi?id=789646
107 2018-02-27 20:34:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
109 * gst/rtsp-sink/gstrtspclientsink.c:
110 rtspclientsink: if OPEN failed, unqueue next command
111 As READY_TO_PAUSED can no longer return async, the RECORD
112 command will be queued before the OPEN command fails
113 (for example in case the server could not be connected),
114 and record then waits for ever.
115 https://bugzilla.gnome.org/show_bug.cgi?id=793896
117 2018-02-26 22:59:17 +0100 Mathieu Duponchelle <mathieu@centricular.com>
119 * gst/rtsp-sink/gstrtspclientsink.c:
120 rtspclientsink: fix retrieval of custom payloader caps
121 If a bin is passed as the custom payloader, the caps of
122 its factory will be empty, the correct way to obtain the caps
123 is to query its sinkpad.
125 2018-02-26 22:59:00 +0100 Mathieu Duponchelle <mathieu@centricular.com>
127 * gst/rtsp-sink/gstrtspclientsink.c:
128 rtspclientsink: fix extra unref of custom payloader
130 2018-02-26 22:57:39 +0100 Mathieu Duponchelle <mathieu@centricular.com>
132 * gst/rtsp-sink/gstrtspclientsink.c:
133 rspclientsink: fix recent code indentation
135 2018-02-26 20:27:57 +0100 Mathieu Duponchelle <mathieu@centricular.com>
137 * gst/rtsp-sink/gstrtspclientsink.c:
138 rtspclientsink: add missing get_type prototype
140 2018-02-24 03:52:15 +0100 Mathieu Duponchelle <mathieu@centricular.com>
142 * gst/rtsp-sink/gstrtspclientsink.c:
143 rtspclientsink: allow setting payloader as pad property
144 This was a FIXME item, and can be quite useful, also
145 allowing to specify payloader properties from the command
146 line, which is always nice.
147 https://bugzilla.gnome.org/show_bug.cgi?id=793776
149 2018-02-26 14:16:54 +0100 Carlos Rafael Giani <dv@pseudoterminal.org>
151 * gst/rtsp-server/rtsp-media.c:
152 rtsp-media: Replace g_print() log line
153 https://bugzilla.gnome.org/show_bug.cgi?id=793838
155 2018-02-22 20:17:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
157 * gst/rtsp-server/rtsp-media.c:
158 * tests/check/gst/rtspclientsink.c:
159 rtsp-media: fix RECORD getting stuck
160 The test_record case was working because async=false had
161 been added in https://bugzilla.gnome.org/show_bug.cgi?id=757488
162 but that was incorrect, as it should not be needed.
163 Removing async=false made the test fail as expected, this is
164 fixed by not trying to preroll when preparing the media for
165 RECORD, as start_prepare is called upon receiving ANNOUNCE,
166 and our peer will not start sending media until it has received
167 a response to that request, and sent and received a response
168 to RECORD as well, thus obviously preventing preroll.
169 https://bugzilla.gnome.org/show_bug.cgi?id=793738
171 2018-02-23 03:26:21 +0100 Mathieu Duponchelle <mathieu@centricular.com>
173 * gst/rtsp-server/rtsp-auth.c:
174 rtsp-auth: fix set_tls_authentication_mode annotation
176 2018-02-19 11:57:29 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
178 * gst/rtsp-server/rtsp-onvif-media.c:
179 rtp-server: remove redefined variable
180 res is a boolean variable which is defined in the function scope and
181 redefined, with no reason, in the loop scope. This patch removes the
183 https://bugzilla.gnome.org/show_bug.cgi?id=793592
185 2018-02-05 11:49:07 +0100 Ognyan Tonchev <ognyan@axis.com>
187 * gst/rtsp-server/rtsp-media.c:
188 * gst/rtsp-server/rtsp-stream.c:
189 * gst/rtsp-server/rtsp-stream.h:
190 stream: Add functions for checking if stream is receiver or sender
191 ...and replace all checks for RECORD in GstRTSPMedia which are really
192 for "sender-only". This way the code becomes more generic and introducing
193 support for onvif-backchannel later on will require no changes in
196 2017-10-21 14:06:30 +0200 Ognyan Tonchev <ognyan@axis.com>
198 * gst/rtsp-server/rtsp-onvif-media-factory.c:
199 * gst/rtsp-server/rtsp-onvif-media-factory.h:
200 onvif: Make requires_backchannel() public
201 ...in order to let subclasses building the onvif part of the pipeline
202 check whether backchannel shall be included or not.
204 2018-01-22 12:46:34 +0200 Sebastian Dröge <sebastian@centricular.com>
206 * gst/rtsp-server/rtsp-onvif-media.c:
207 rtsp-server: Switch around sendonly/recvonly attributes
208 They are wrong in the ONVIF streaming spec. The backchannel should be
209 recvonly and the normal media should be sendonly: direction is always
210 from the point of view of the SDP offerer (the server) according to
213 2017-09-25 19:41:05 +0300 Sebastian Dröge <sebastian@centricular.com>
215 * docs/libs/gst-rtsp-server-docs.sgml:
216 * docs/libs/gst-rtsp-server-sections.txt:
217 * examples/.gitignore:
218 * examples/Makefile.am:
219 * examples/test-onvif-backchannel.c:
220 * gst/rtsp-server/Makefile.am:
221 * gst/rtsp-server/rtsp-media.h:
222 * gst/rtsp-server/rtsp-onvif-client.c:
223 * gst/rtsp-server/rtsp-onvif-client.h:
224 * gst/rtsp-server/rtsp-onvif-media-factory.c:
225 * gst/rtsp-server/rtsp-onvif-media-factory.h:
226 * gst/rtsp-server/rtsp-onvif-media.c:
227 * gst/rtsp-server/rtsp-onvif-media.h:
228 * gst/rtsp-server/rtsp-onvif-server.c:
229 * gst/rtsp-server/rtsp-onvif-server.h:
230 * gst/rtsp-server/rtsp-sdp.c:
231 * gst/rtsp-server/rtsp-sdp.h:
232 rtsp: Add support for ONVIF backchannel
233 This adds a new RTSP server, client, media-factory and media subclass
234 for handling the specifics of the backchannel. Ideally this later can be
235 extended with other ONVIF specific features.
237 2017-10-12 21:00:16 +0300 Sebastian Dröge <sebastian@centricular.com>
239 * gst/rtsp-server/rtsp-media.c:
240 rtsp-media: Add support for sending+receiving medias
241 We need to add an appsrc/appsink in that case because otherwise the
242 media bin will be a sink and a source for rtpbin, causing a pipeline
244 https://bugzilla.gnome.org/show_bug.cgi?id=788950
246 2018-02-15 19:44:28 +0000 Tim-Philipp Müller <tim@centricular.com>
252 === release 1.13.1 ===
254 2018-02-15 17:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
258 * gst-rtsp-server.doap:
262 2018-02-14 17:11:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
264 * gst/rtsp-server/rtsp-session-pool.c:
265 session-pool: remove nullable return annotation
266 create_watch can only return NULL from the API guards, no
269 2018-02-13 18:59:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
271 * gst/rtsp-server/rtsp-media-factory.c:
272 * gst/rtsp-server/rtsp-media.c:
273 set_clock functions: Add nullable annotations
275 2018-02-10 00:07:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
277 * gst/rtsp-server/rtsp-auth.c:
278 * gst/rtsp-server/rtsp-client.c:
279 * gst/rtsp-server/rtsp-media-factory.c:
280 * gst/rtsp-server/rtsp-media.c:
281 * gst/rtsp-server/rtsp-mount-points.c:
282 * gst/rtsp-server/rtsp-server.c:
283 * gst/rtsp-server/rtsp-session-media.c:
284 * gst/rtsp-server/rtsp-session-pool.c:
285 * gst/rtsp-server/rtsp-session.c:
286 * gst/rtsp-server/rtsp-stream-transport.c:
287 * gst/rtsp-server/rtsp-stream.c:
288 * gst/rtsp-server/rtsp-thread-pool.c:
289 All around: add annotations and API guards
291 2018-02-12 19:12:35 +0100 Mathieu Duponchelle <mathieu@centricular.com>
293 * tests/test-cleanup.c:
294 test-cleanup: bind any port
295 The meson test suite runs tests in parallel, trying to bind
296 a single port made the test fail.
298 2018-02-08 19:15:10 +0000 Tim-Philipp Müller <tim@centricular.com>
301 meson: make version numbers ints and fix int/string comparison
302 WARNING: Trying to compare values of different types (str, int).
303 The result of this is undefined and will become a hard error
304 in a future Meson release.
306 2018-02-06 18:00:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
308 * gst/rtsp-server/rtsp-context.c:
309 gst_rtsp_context_get_current: add (skip) annotation
310 The return value type is defined with G_DEFINE_POINTER_TYPE,
311 and gi emits the following warning:
312 Invalid non-constant return of bare structure or union; register as
315 2018-02-06 17:58:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
317 * gst/rtsp-server/rtsp-client.c:
318 rtsp-client: add type annotations
319 gi doesn't seem to be able to figure out the type of the
320 signal parameters when defined with G_DEFINE_POINTER_TYPE
322 2018-02-04 12:24:09 +0100 Tim-Philipp Müller <tim@centricular.com>
325 autotools: use -fno-strict-aliasing where supported
326 https://bugzilla.gnome.org/show_bug.cgi?id=769183
328 2018-01-30 20:35:21 +0000 Tim-Philipp Müller <tim@centricular.com>
331 meson: use -fno-strict-aliasing where supported
332 https://bugzilla.gnome.org/show_bug.cgi?id=769183
334 2018-01-25 12:09:03 +0000 Tim-Philipp Müller <tim@centricular.com>
336 * gst/rtsp-server/rtsp-mount-points.c:
337 mount-points: bail out of loop again when matching mount points
338 Previous patch led to us iterating the entire sequence. Bail out
339 of the loop again if we have a match but are moving away from it.
340 https://bugzilla.gnome.org/show_bug.cgi?id=771555
342 2018-01-25 12:06:57 +0000 Tim-Philipp Müller <tim@centricular.com>
344 * tests/check/gst/mountpoints.c:
345 tests: mountpoints: add more checks for mount point path matching
346 https://bugzilla.gnome.org/show_bug.cgi?id=771555
348 2016-09-16 20:41:19 +0000 Andrew Bott <andrew.bott@blackmoth.com>
350 * gst/rtsp-server/rtsp-mount-points.c:
351 mount-points: fix matching of paths where there's also an entry with a common prefix
352 e.g. with the following mount points
356 _match() would not match /raw/video and /raw/snapshot correctly.
357 https://bugzilla.gnome.org/show_bug.cgi?id=771555
359 2018-01-18 23:53:20 +0000 Tim-Philipp Müller <tim@centricular.com>
361 * docs/libs/gst-rtsp-server-sections.txt:
362 * gst/rtsp-server/rtsp-permissions.c:
363 * gst/rtsp-server/rtsp-permissions.h:
364 * tests/check/gst/permissions.c:
365 permissions: add some new API to make this usable from bindings
366 https://bugzilla.gnome.org/show_bug.cgi?id=787073
368 2018-01-18 11:32:32 +0000 Tim-Philipp Müller <tim@centricular.com>
370 * gst/rtsp-server/rtsp-token.c:
371 rtsp-token: annotate constructors for bindings
372 This maps _new_empty() to _new(), which also makes RTSPToken()
373 work properly now. Since this API wasn't usable from bindings
374 before, this should hopefully be fine.
375 https://bugzilla.gnome.org/show_bug.cgi?id=787073
377 2018-01-18 11:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
379 * docs/libs/gst-rtsp-server-sections.txt:
380 * gst/rtsp-server/rtsp-token.c:
381 * gst/rtsp-server/rtsp-token.h:
382 * tests/check/gst/token.c:
383 rtsp-token: add some API to set fields from bindings
384 The existing functions are all vararg-based and as such
385 not usable from bindings.
386 https://bugzilla.gnome.org/show_bug.cgi?id=787073
388 2018-01-13 15:02:28 +0000 Tim-Philipp Müller <tim@centricular.com>
390 * tests/check/gst/rtspclientsink.c:
391 * tests/check/gst/rtspserver.c:
392 * tests/check/gst/sessionpool.c:
393 * tests/check/gst/stream.c:
394 tests: fix indentation
397 2018-01-13 14:58:55 +0000 Tim-Philipp Müller <tim@centricular.com>
399 * tests/check/gst/rtspserver.c:
400 tests: rtspserver: fix another ref leak
401 Even if this didn't show up in valgrind.
403 2018-01-13 14:58:00 +0000 Tim-Philipp Müller <tim@centricular.com>
405 * tests/check/gst/rtspclientsink.c:
406 tests: rtspclientsink: fix leak
408 2018-01-02 14:19:31 +0100 Branko Subasic <branko@axis.com>
410 * tests/check/gst/rtspserver.c:
411 test: rtspserver: plug memory leak in test_no_session_timeout
412 In test_no_session_timeout, unref the rtsp session object when the
414 https://bugzilla.gnome.org/show_bug.cgi?id=792127
416 2017-12-20 14:17:02 +0100 Edward Hervey <edward@centricular.com>
418 * gst/rtsp-sink/gstrtspclientsink.c:
419 rtpsclientsink: Initialize and clear newly added mutex and cond
420 While it *did* work, glib would automatically create new mutex and cond
421 ... which never got freed
423 2017-12-19 11:34:37 +0200 Sebastian Dröge <sebastian@centricular.com>
425 * gst/rtsp-server/rtsp-stream.c:
426 rtsp-stream: Set multicast TTL on the multicast sockets
427 And not if we do unicast UDP.
428 https://bugzilla.gnome.org/show_bug.cgi?id=791743
430 2017-12-19 11:14:48 +0200 Sebastian Dröge <sebastian@centricular.com>
432 * gst/rtsp-server/rtsp-stream.c:
433 rtsp-stream: Decide based on the sockets, not the addresses if we already allocated a socket
434 In the multicast case (as in test-multicast, not test-multicast2), the
435 address could be allocated/reserved (and thus set) already without
436 allocating the actual socket. We need to allocate the socket here still
437 instead of just claiming that it was already allocated.
438 See https://bugzilla.gnome.org/show_bug.cgi?id=791743#c2
440 2017-12-16 21:46:53 +0100 Patricia Muscalu <patricia@dovakhiin.com>
442 * gst/rtsp-sink/gstrtspclientsink.c:
443 * gst/rtsp-sink/gstrtspclientsink.h:
444 rtspclientsink: Use the new rtsp-stream API
445 https://bugzilla.gnome.org/show_bug.cgi?id=790412
447 2017-12-16 21:01:43 +0100 Patricia Muscalu <patricia@dovakhiin.com>
449 * gst/rtsp-sink/gstrtspclientsink.c:
450 * gst/rtsp-sink/gstrtspclientsink.h:
451 rtspclientsink: Wait until OPEN has been scheduled
452 Make sure that the sink thread has started opening connection
453 to the server before continuing.
454 https://bugzilla.gnome.org/show_bug.cgi?id=790412
456 2017-12-14 14:53:35 +1100 Matthew Waters <matthew@centricular.com>
459 Automatic update of common submodule
460 From e8c7a71 to 3fa2c9e
462 2017-12-07 16:08:29 +0100 Edward Hervey <edward@centricular.com>
464 * gst/rtsp-server/rtsp-media.c:
465 * gst/rtsp-server/rtsp-session-media.c:
466 * gst/rtsp-server/rtsp-stream.c:
467 rtsp-server: Minor doc fixes
470 2017-12-06 20:47:22 +0000 Tim-Philipp Müller <tim@centricular.com>
474 tests: disable all tests when --disable-tests is used
475 Move conditional subdir include into top level.
476 Based on patch by: Joel Holdsworth
477 https://bugzilla.gnome.org/show_bug.cgi?id=757703
479 2017-12-06 20:42:39 +0000 Tim-Philipp Müller <tim@centricular.com>
484 meson: build more tests and add options to disable tests and examples
486 2017-11-26 13:26:39 -0300 Thibault Saunier <tsaunier@gnome.org>
488 * gst/rtsp-server/rtsp-session.c:
489 Fix build when -Werror=deprecated-declarations is on
490 As gst_rtsp_session_next_timeout is deprecated.
492 ../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]
493 res = (gst_rtsp_session_next_timeout (session, now) == 0);
495 ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-session.c:685:1: note: declared here
496 gst_rtsp_session_next_timeout (GstRTSPSession * session, GTimeVal * now)
497 ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~
500 2017-11-27 20:18:24 +1100 Matthew Waters <matthew@centricular.com>
503 Automatic update of common submodule
504 From 3f4aa96 to e8c7a71
506 2017-11-25 20:34:16 +0100 Patricia Muscalu <patricia@dovakhiin.com>
508 * tests/check/gst/media.c:
509 check/media: Add seekability test case: not all streams are active
510 Media contains two streams but only one is complete and prepared
512 https://bugzilla.gnome.org/show_bug.cgi?id=790674
514 2017-11-25 20:32:02 +0100 Patricia Muscalu <patricia@dovakhiin.com>
516 * gst/rtsp-server/rtsp-stream.c:
517 rtsp-stream: Do not reset 'blocking' if stream is already blocked
518 https://bugzilla.gnome.org/show_bug.cgi?id=790674
520 2017-11-25 20:45:44 +0100 Patricia Muscalu <patricia@dovakhiin.com>
522 * gst/rtsp-server/rtsp-media.c:
523 rtsp-media: Fix missing lock in gst_rtsp_media_seekable()
524 https://bugzilla.gnome.org/show_bug.cgi?id=790674
526 2017-11-26 16:29:49 +0000 Tim-Philipp Müller <tim@centricular.com>
529 meson: remove vs_module_defs_dir variable which is no longer needed
531 2017-11-26 14:46:05 +0000 Tim-Philipp Müller <tim@centricular.com>
533 * gst/rtsp-server/rtsp-session.h:
536 2017-11-26 12:53:42 +0000 Tim-Philipp Müller <tim@centricular.com>
539 * gst/rtsp-server/meson.build:
541 * win32/common/libgstrtspserver.def:
542 win32: remove .def file with exports
543 They're no longer needed, symbol exporting is now explicit
544 via GST_EXPORT in all cases (autotools, meson, incl. MSVC).
546 2017-11-26 12:28:40 +0000 Tim-Philipp Müller <tim@centricular.com>
549 autotools: stop controlling symbol visibility with -export-symbols-regex
550 Instead, use -fvisibility=hidden and explicit exports via GST_EXPORT.
551 This should result in consistent behaviour for the autotools and
554 2017-11-26 12:47:08 +0000 Tim-Philipp Müller <tim@centricular.com>
556 * gst/rtsp-server/rtsp-media.h:
557 * gst/rtsp-server/rtsp-server.h:
558 * gst/rtsp-server/rtsp-session.c:
559 * gst/rtsp-server/rtsp-session.h:
560 rtsp-server: add missing GST_EXPORT and export deprecated funcs
562 2017-11-25 07:53:30 +0100 Edward Hervey <edward@centricular.com>
564 * tests/check/gst/media.c:
565 check: Add seekability testing on medias
566 Make sure that once GstRTSPMedia are prepared they returned
567 the expected seekability results
568 https://bugzilla.gnome.org/show_bug.cgi?id=790674
570 2017-11-24 17:34:31 +0100 Edward Hervey <edward@centricular.com>
572 * docs/libs/gst-rtsp-server-sections.txt:
573 * gst/rtsp-server/rtsp-media.c:
574 * gst/rtsp-server/rtsp-stream.c:
575 * gst/rtsp-server/rtsp-stream.h:
576 * win32/common/libgstrtspserver.def:
577 rtsp-media: Enable seeking query before pipeline is complete
578 SDP are now provided *before* the pipeline is fully complete. In order
579 to know whether a media is seekable or not therefore requires asking
580 the invididual streams.
581 API: gst_rtsp_stream_seekable
582 https://bugzilla.gnome.org/show_bug.cgi?id=790674
584 2017-11-23 20:34:03 +0100 Patricia Muscalu <patricia@axis.com>
586 * gst/rtsp-server/rtsp-media.c:
587 rtsp-media: Fix handling in default_unsuspend()
588 Handle the case when streams are not blocked and media
589 is suspended from PAUSED.
590 Change-Id: I2f3d222ea7b9b20a0732ea5dc81a32d17ab75040
591 https://bugzilla.gnome.org/show_bug.cgi?id=790674
593 2017-11-23 18:51:21 +0100 Patricia Muscalu <patricia@axis.com>
595 * tests/check/gst/media.c:
596 check/media: Fix thread pool leak.
597 Change-Id: I0f92b1caca0ee518ae64a7dacfbd28a214c3eea1
598 https://bugzilla.gnome.org/show_bug.cgi?id=790674
600 2017-11-23 18:39:44 +0100 Patricia Muscalu <patricia@axis.com>
602 * gst/rtsp-server/rtsp-media.c:
603 rtsp-media: Removed fakesink elements
604 There is not need of adding fakesink elements to the media
605 pipeline in the dynamic-payloader case.
606 The media pipeline itself is dynamically updated with
607 the receiver and sender parts that are based on the client
608 transport information known after SETUP has been received.
609 Change-Id: I4e88c9b500c04030669822f0d03b1842913f6cb9
610 https://bugzilla.gnome.org/show_bug.cgi?id=790674
612 2017-11-23 09:10:54 +0100 Patricia Muscalu <patricia@axis.com>
614 * gst/rtsp-server/rtsp-media.c:
615 rtsp-media: Corrected ASYNC_DONE handling
616 Media is complete when all the transport based parts are
617 added to the media pipeline. At this point ASYNC_DONE is
618 posted by the media pipeline and media is ready to enter
620 Change-Id: I50fb8dfed88ebaf057d9a35fca2d7f0a70e9d1fa
621 https://bugzilla.gnome.org/show_bug.cgi?id=790674
623 2017-11-22 12:24:38 +0100 Edward Hervey <bilboed@bilboed.com>
625 * tests/check/gst/media.c:
626 check/media: Check that prepared media can provide a SDP
627 Whenever a RTSPMedia is prepared, it should be able to provide a SDP
629 2017-11-21 09:53:19 +0100 Edward Hervey <edward@centricular.com>
631 * gst/rtsp-server/rtsp-client.c:
632 rtsp-client: Don't leak addr
635 2017-11-21 09:53:08 +0100 Edward Hervey <bilboed@bilboed.com>
637 * gst/rtsp-server/rtsp-client.c:
638 * gst/rtsp-server/rtsp-session-media.c:
639 * gst/rtsp-server/rtsp-stream.c:
642 2017-11-20 18:30:19 +0100 Edward Hervey <bilboed@bilboed.com>
644 * gst/rtsp-server/rtsp-media.c:
645 rtsp-media: Don't unblock with remaining dynamic payloaders
646 If we still have some dynamic paylaoders which haven't posted
647 no-more-pads yet, don't go to PREPARED if one of the streams
649 The risk was that we would end up not exposing/using all specified
651 The downside is that if you have _multiple_ _live_ _dynamic_ payloaders
652 then it will take a bit more time to start. But only if those 3
653 conditions are present.
654 https://bugzilla.gnome.org/show_bug.cgi?id=769521
656 2017-11-20 16:49:29 +0100 Edward Hervey <edward@centricular.com>
658 * gst/rtsp-server/rtsp-media.c:
661 2017-11-20 16:48:55 +0100 Edward Hervey <edward@centricular.com>
663 * gst/rtsp-server/rtsp-media.c:
664 rtsp-media: Don't set float on a gint64 variable
665 Just use 0. Fixes 'undefined' behaviour from clang
667 2017-11-20 18:29:02 +0100 Edward Hervey <edward@centricular.com>
669 * gst/rtsp-server/rtsp-media.c:
670 rtsp-media: Fix previous commit
671 We only want to count dynamic payloaders
673 2017-11-20 09:32:07 +0100 Edward Hervey <edward@centricular.com>
675 * gst/rtsp-server/rtsp-media.c:
676 * tests/check/gst/media.c:
677 rtsp-media: Handle multiple dynamic elements
678 If we have more than one dynamic payloader in the pipeline, we need
679 to wait until the *last* one emits 'no-more-pads' before switching
681 Failure to do so would result in a race where some of the streams
682 wouldn't properly be prepared
683 https://bugzilla.gnome.org/show_bug.cgi?id=769521
685 2017-11-16 12:18:20 +0200 Sebastian Dröge <sebastian@centricular.com>
687 * win32/common/libgstrtspserver.def:
688 win32: Fix exported symbols list
690 2017-11-15 19:52:29 +0200 Sebastian Dröge <sebastian@centricular.com>
692 * gst/rtsp-server/rtsp-stream.c:
693 rtsp-stream: Only update the RTP udpsink if it actually exists
694 For send-only streams it does not exist, but the RTCP udpsink might.
696 2017-11-15 18:15:53 +0200 Sebastian Dröge <sebastian@centricular.com>
698 * win32/common/libgstrtspserver.def:
699 win32: Update exports
701 2017-10-23 09:49:09 +0200 Patricia Muscalu <patricia@axis.com>
703 * gst/rtsp-server/rtsp-media.c:
704 * gst/rtsp-server/rtsp-stream.c:
705 * gst/rtsp-server/rtsp-stream.h:
706 rtsp-media: seek on media pipelines that are complete
707 Make sure that a seek is performed on pipelines that
708 contain at least one sink element.
709 Change-Id: Icf398e10add3191d104b1289de612412da326819
710 https://bugzilla.gnome.org/show_bug.cgi?id=788340
712 2017-10-17 10:44:33 +0200 Patricia Muscalu <patricia@axis.com>
714 * gst/rtsp-server/rtsp-client.c:
715 * gst/rtsp-server/rtsp-media.c:
716 * gst/rtsp-server/rtsp-media.h:
717 * gst/rtsp-server/rtsp-stream.c:
718 * gst/rtsp-server/rtsp-stream.h:
719 * tests/check/gst/client.c:
720 * tests/check/gst/media.c:
721 * tests/check/gst/rtspserver.c:
722 * tests/check/gst/stream.c:
723 Dynamically reconfigure pipeline in PLAY based on transports
724 The initial pipeline does not contain specific transport
725 elements. The receiver and the sender parts are added
727 If the media is shared, the streams are dynamically
728 reconfigured after each PLAY.
729 https://bugzilla.gnome.org/show_bug.cgi?id=788340
731 2017-10-16 12:40:57 +0200 Patricia Muscalu <patricia@axis.com>
733 * gst/rtsp-server/rtsp-stream.c:
734 rtsp-stream: obtain stream position from pad
735 If no sinks have been added yet, obtain the current and
736 the stop position of the stream from the send_src pad.
737 Change-Id: Iacd4ab4bdc69f6b49370d06012880ce48a7d595a
738 https://bugzilla.gnome.org/show_bug.cgi?id=788340
740 2017-10-16 11:35:10 +0200 Patricia Muscalu <patricia@axis.com>
742 * gst/rtsp-server/rtsp-session-media.c:
743 * gst/rtsp-server/rtsp-session-media.h:
744 rtsp-session-media: add function to get a list of transports
745 Change-Id: I817e10624da0f3200f24d1b232cff481099278e3
746 https://bugzilla.gnome.org/show_bug.cgi?id=788340
748 2017-10-16 11:15:55 +0200 Patricia Muscalu <patricia@axis.com>
750 * gst/rtsp-server/rtsp-stream.c:
751 * gst/rtsp-server/rtsp-stream.h:
752 rtsp-stream: add functions to get rtp and rtcp multicast sockets
753 Change-Id: Iddfe6e0bd250cb0159096d5eba9e4202d22b56db
754 https://bugzilla.gnome.org/show_bug.cgi?id=788340
756 2017-10-20 12:21:48 +0200 Patricia Muscalu <patricia@axis.com>
758 * gst/rtsp-server/rtsp-stream.c:
759 stream: set async=sync=false only for RTCP appsink
760 Change-Id: I929a218a9adf4759f61322b6f2063aacc5595f90
761 https://bugzilla.gnome.org/show_bug.cgi?id=788340
763 2017-10-16 10:10:17 +0200 Patricia Muscalu <patricia@axis.com>
765 * gst/rtsp-server/rtsp-media.c:
766 rtsp-media: return minimum value in query position case
767 The minimum position should be returned as we are interested
768 in the whole interval.
769 Change-Id: I30e297fc040c995ae40c25dee8ff56321612fe2b
770 https://bugzilla.gnome.org/show_bug.cgi?id=788340
772 2017-08-09 11:52:38 +0200 Jonathan Karlsson <jonakn@axis.com>
774 * gst/rtsp-server/rtsp-session.c:
775 * tests/check/gst/rtspserver.c:
776 rtsp-session: Handle the case when timeout=0
777 According to the documentation, a timeout of value 0 means
778 that the session never timeouts. This adds handling of that.
779 If timeout=0 we just return with a -1 from
780 gst_rtsp_session_next_timeout_usec ().
781 https://bugzilla.gnome.org/show_bug.cgi?id=785058
783 2017-07-17 17:15:22 +0300 Sebastian Dröge <sebastian@centricular.com>
785 * gst/rtsp-sink/gstrtspclientsink.c:
786 rtspclientsink: Add "accept-certificate" signal for manually checking a TLS certificate for validity
787 https://bugzilla.gnome.org/show_bug.cgi?id=785024
789 2017-10-26 14:43:19 +0200 Mathieu Duponchelle <mathieu@centricular.com>
791 * docs/libs/gst-rtsp-server-sections.txt:
792 * gst/rtsp-server/rtsp-media-factory.c:
793 docs: add media factory transport mode accessors
794 and fix the documentation for the return value of the getter
796 2017-10-09 12:43:01 +0200 Branko Subasic <branko@axis.com>
798 * gst/rtsp-server/rtsp-client.c:
799 rtsp-client: unref 'pipelined_requests' in finalize
800 The hash table priv->pipelined_requests is not unref:ed in the
801 finalize funktion. Make sure it is.
802 https://bugzilla.gnome.org/show_bug.cgi?id=788704
804 2017-10-09 14:44:40 +0200 Thibault Saunier <thibault.saunier@osg.samsung.com>
806 * gst/rtsp-server/rtsp-media.c:
807 rtsp-media: Initialize scalar variable
810 2017-10-06 10:27:34 +0200 Edward Hervey <edward@centricular.com>
812 * win32/common/libgstrtspserver.def:
813 win32: Update export file
815 2017-04-22 09:26:07 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
817 * gst/rtsp-server/rtsp-client.c:
818 * gst/rtsp-server/rtsp-media.c:
819 * gst/rtsp-server/rtsp-media.h:
820 Start support for RTSP 2.0
821 This adds basic support for new 2.0 features, though the protocol is
822 subposdely backward incompatible, most semantics are the sames.
825 * version negotiation
826 * pipelined requests support
827 * Media-Properties support
828 * Accept-Ranges support
830 * gst_rtsp_media_seekable
831 The RTSP methods that have been removed when using 2.0 now return
833 https://bugzilla.gnome.org/show_bug.cgi?id=781446
835 2017-06-02 15:37:54 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
837 * gst/rtsp-server/rtsp-stream.c:
838 stream: Use stream duration as stream-stop if segment was not configured with a stop
839 Allowing client to know stream duration when no seeking happened.
840 https://bugzilla.gnome.org/show_bug.cgi?id=783435
842 2017-09-25 19:40:17 +0300 Sebastian Dröge <sebastian@centricular.com>
844 * gst/rtsp-server/rtsp-media-factory.c:
845 rtsp-media-factory: Don't cache any media if NULL was returned as key
846 The docs already mentioned this, but we actually stored it in the hash
847 table with key==NULL and leaked its reference forever.
849 2017-09-18 19:31:31 +0200 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
851 * gst/rtsp-sink/gstrtspclientsink.c:
852 * gst/rtsp-sink/gstrtspclientsink.h:
853 rtspclientsink: Use a mutex for protecting against concurrent send/receives
854 This is a simple port of:
855 * a722f6e8329032c6eda4865d6a07f4ba5981d7ea
856 * c438545dc9e2f14f657bc0ef261fff726449867b
857 * cd17c71dcea5c9310d21f1347c7520983e5869ac
860 2017-08-31 13:24:15 +0530 Satya Prakash Gupta <sp.gupta@samsung.com>
862 * gst/rtsp-server/rtsp-sdp.c:
863 sdp: fix Memory leak in error case
864 https://bugzilla.gnome.org/show_bug.cgi?id=787059
866 2017-08-18 17:37:01 +0100 Tim-Philipp Müller <tim@centricular.com>
868 * pkgconfig/meson.build:
869 meson: don't install -uninstalled.pc file
870 https://bugzilla.gnome.org/show_bug.cgi?id=786457
872 2017-08-17 12:26:17 +0100 Tim-Philipp Müller <tim@centricular.com>
875 Automatic update of common submodule
876 From 48a5d85 to 3f4aa96
878 2017-08-14 21:04:23 +0300 Sebastian Dröge <sebastian@centricular.com>
880 * gst/rtsp-server/rtsp-client.c:
881 rtsp-client: Fix typo in debug message
883 2017-08-11 14:14:32 +0100 Tim-Philipp Müller <tim@centricular.com>
886 meson: hide symbols by default unless explicitly exported
888 2017-08-10 14:20:12 +0100 Tim-Philipp Müller <tim@centricular.com>
890 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
891 pkgconfig: remove -I@srcdir@/.. which duplicates abs_top_srcdir
892 Fixes meson warning about undefined @srcdir@.
894 2017-07-21 13:36:00 +0100 Tim-Philipp Müller <tim@centricular.com>
897 meson: skip tests on windows for now
898 As we do in the other modules. As libgstcheck is currently not
899 built on windows. Fixes "Fallback variable 'gst_check_dep' in
900 the subproject 'gstreamer' does not exist"" Meson error.
902 2017-06-22 07:25:07 -0700 Julien Isorce <julien.isorce@gmail.com>
904 * gst/rtsp-server/rtsp-stream.c:
905 rtsp-stream: fix connection delay due to wrong assumption on last-sample
906 Commit 852cc09f542af5cadd79ffd7fe79d6475cf57e14 assumed that
907 multiudpsink's last-sample always comes from the payloader. Which
908 is wrong if auxiliary streams are multiplexed in the same stream.
909 So check the buffer's ssrc against the caps'ssrc before to use its
910 seqnum. If not the same ssrc just use the payloader as done prior
911 the commit above or when there is no last-sample yet.
912 https://bugzilla.gnome.org/show_bug.cgi?id=784094
914 2017-06-23 16:19:04 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
917 meson: Allow using glib as a subproject
919 2017-06-26 09:55:49 +0100 Tim-Philipp Müller <tim@centricular.com>
922 meson: fix with-package-name option
923 https://bugzilla.gnome.org/show_bug.cgi?id=784082
925 2017-06-09 20:16:28 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
928 Distribute meson_options.txt
930 2017-06-09 20:11:47 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
933 And config.h.meson is no longer dist either
935 2017-06-09 21:27:09 +0100 Tim-Philipp Müller <tim@centricular.com>
939 meson: config.h.meson is no longer needed
941 2017-06-07 13:04:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
943 * tests/check/meson.build:
945 meson: Fix building tests and activate them again
947 2017-06-07 12:55:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
949 * tests/check/meson.build:
950 meson: Do not use path separator in test names
951 Avoiding warnings like:
952 WARNING: Target "elements/audioamplify" has a path separator in its name.
954 2017-05-20 15:07:31 +0100 Tim-Philipp Müller <tim@centricular.com>
958 meson: add options to set package name and origin
959 https://bugzilla.gnome.org/show_bug.cgi?id=782172
961 2017-05-18 10:35:18 +0100 Tim-Philipp Müller <tim@centricular.com>
963 * gst/rtsp-server/rtsp-address-pool.h:
964 * gst/rtsp-server/rtsp-auth.h:
965 * gst/rtsp-server/rtsp-client.h:
966 * gst/rtsp-server/rtsp-context.h:
967 * gst/rtsp-server/rtsp-media-factory-uri.h:
968 * gst/rtsp-server/rtsp-media-factory.h:
969 * gst/rtsp-server/rtsp-media.h:
970 * gst/rtsp-server/rtsp-mount-points.h:
971 * gst/rtsp-server/rtsp-params.h:
972 * gst/rtsp-server/rtsp-permissions.h:
973 * gst/rtsp-server/rtsp-sdp.h:
974 * gst/rtsp-server/rtsp-server.h:
975 * gst/rtsp-server/rtsp-session-media.h:
976 * gst/rtsp-server/rtsp-session-pool.h:
977 * gst/rtsp-server/rtsp-session.h:
978 * gst/rtsp-server/rtsp-stream-transport.h:
979 * gst/rtsp-server/rtsp-stream.h:
980 * gst/rtsp-server/rtsp-thread-pool.h:
981 * gst/rtsp-server/rtsp-token.h:
982 Mark symbols explicitly for export with GST_EXPORT
984 2017-05-16 14:44:43 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
987 * gst/rtsp-sink/Makefile.am:
988 Remove plugin specific static build option
989 Static and dynamic plugins now have the same interface. The standard
990 --enable-static/--enable-shared toggle are sufficient.
992 2017-05-04 18:59:14 +0300 Sebastian Dröge <sebastian@centricular.com>
998 === release 1.12.0 ===
1000 2017-05-04 15:40:46 +0300 Sebastian Dröge <sebastian@centricular.com>
1006 * gst-rtsp-server.doap:
1010 === release 1.11.91 ===
1012 2017-04-27 17:42:02 +0300 Sebastian Dröge <sebastian@centricular.com>
1018 * gst-rtsp-server.doap:
1022 2017-04-24 20:30:37 +0100 Tim-Philipp Müller <tim@centricular.com>
1025 Automatic update of common submodule
1026 From 60aeef6 to 48a5d85
1028 2017-04-13 14:20:10 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
1030 * gst/rtsp-server/rtsp-media-factory.c:
1031 * gst/rtsp-server/rtsp-media.c:
1032 * gst/rtsp-server/rtsp-session.c:
1033 * gst/rtsp-server/rtsp-stream.c:
1034 gi: Fix some annotations and docstrings
1036 2017-04-13 13:52:26 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
1038 * gst/rtsp-server/meson.build:
1040 * meson_options.txt:
1043 2017-04-10 23:51:12 +0100 Tim-Philipp Müller <tim@centricular.com>
1047 Automatic update of common submodule
1048 From 39ac2f5 to 60aeef6
1050 === release 1.11.90 ===
1052 2017-04-07 16:35:03 +0300 Sebastian Dröge <sebastian@centricular.com>
1058 * gst-rtsp-server.doap:
1062 2017-03-27 18:19:33 +0100 Tim-Philipp Müller <tim@centricular.com>
1064 * examples/test-launch.c:
1065 examples: make test-launch pipeline shared by default as well
1067 2017-02-27 19:10:44 +0200 Sebastian Dröge <sebastian@centricular.com>
1069 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
1070 gstreamer-rtsp-server: Add both srcdir and builddir to the include path
1071 Just the build dir is not going to work for srcdir!=builddir.
1073 2017-02-24 15:59:54 +0200 Sebastian Dröge <sebastian@centricular.com>
1076 meson: Update version
1078 2017-02-24 15:37:49 +0200 Sebastian Dröge <sebastian@centricular.com>
1083 === release 1.11.2 ===
1085 2017-02-24 15:10:07 +0200 Sebastian Dröge <sebastian@centricular.com>
1091 * gst-rtsp-server.doap:
1094 2017-02-14 20:40:26 +0000 Tim-Philipp Müller <tim@centricular.com>
1097 meson: dist meson build files
1098 Ship meson build files in tarballs, so people who use tarballs
1099 in their builds can start playing with meson already.
1101 2017-02-07 23:39:37 +1100 Jan Schmidt <jan@centricular.com>
1103 * examples/test-record.c:
1104 examples/test-record: Add extra line to initial printout
1105 Add an example line of how to deliver a stream to the
1108 2017-01-19 14:57:19 +0200 Sebastian Dröge <sebastian@centricular.com>
1110 * gst/rtsp-server/rtsp-client.c:
1111 rtsp-client: Also handle the (S|G)ET_PARAMETER case of size==0 || !data as keep-alive
1112 If there is no Content-Length header, no body would be allocated and the
1113 '\0' would also not be appended to the body.
1115 2017-01-19 14:24:07 +0200 Sebastian Dröge <sebastian@centricular.com>
1117 * gst/rtsp-server/rtsp-client.c:
1118 rtsp-client: Fix handling of keep-alive GET_PARAMETER/SET_PARAMETER
1119 While they logically have 0 bytes length, GstRTSPConnection is appending
1120 a '\0' to everything making the size be 1 instead.
1122 2017-01-13 12:39:36 +0000 Tim-Philipp Müller <tim@centricular.com>
1127 2017-01-12 19:04:23 +0200 Sebastian Dröge <sebastian@centricular.com>
1129 * gst/rtsp-server/rtsp-session.c:
1130 rtsp-session: Only remove deprecated API if requested to do so, not just when disabling
1131 gst_rtsp_session_is_expired() and gst_rtsp_session_next_timeout() were
1134 2017-01-12 16:32:59 +0200 Sebastian Dröge <sebastian@centricular.com>
1139 === release 1.11.1 ===
1141 2017-01-12 16:14:46 +0200 Sebastian Dröge <sebastian@centricular.com>
1147 * gst-rtsp-server.doap:
1148 * win32/common/libgstrtspserver.def:
1151 2017-01-10 08:34:50 +0100 Patricia Muscalu <patricia@axis.com>
1153 * gst/rtsp-server/rtsp-stream.c:
1154 rtsp-stream: corrected if-statement in _get_server_port()
1155 This bug was accidentally introduced while fixing a segfault
1156 in _get_server_port() function.
1157 https://bugzilla.gnome.org/show_bug.cgi?id=776345
1159 2017-01-09 14:12:05 +0100 Patricia Muscalu <patricia@axis.com>
1161 * gst/rtsp-server/rtsp-stream.c:
1162 * tests/check/gst/stream.c:
1163 rtsp-stream: fixed segmenation fault in _get_server_port()
1164 Calling function gst_rtsp_stream_get_server_port() results in
1165 segmenation fault in the RTP/RTSP/TCP case.
1166 Port that the server will use to receive RTCP makes only
1167 sense in the UDP case, however the function should handle
1168 the TCP case in a nicer way.
1169 https://bugzilla.gnome.org/show_bug.cgi?id=776345
1171 2017-01-09 12:22:40 +0300 Aleksandr Slobodeniuk <alenuke@yandex.ru>
1173 * gst/rtsp-server/rtsp-media-factory.c:
1174 dosc: Fix a little typo
1175 https://bugzilla.gnome.org/show_bug.cgi?id=777037
1177 2017-01-04 16:20:54 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
1179 * pkgconfig/Makefile.am:
1180 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
1181 * pkgconfig/meson.build:
1182 meson: generate pkg-config -uninstalled pc files
1183 Generating those files is useful for users building the GStreamer stack
1184 using meson and having to link it to another project which is still
1185 using the autotools.
1186 https://bugzilla.gnome.org/show_bug.cgi?id=776810
1188 2017-01-04 16:11:08 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
1190 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
1191 pkgconfig: fix -uninstalled pc file
1192 pcfiledir was never defined so the paths were wrong.
1193 https://bugzilla.gnome.org/show_bug.cgi?id=776867
1195 2016-12-21 13:41:50 +0100 Patricia Muscalu <patricia@axis.com>
1197 * gst/rtsp-server/rtsp-stream.c:
1198 * tests/check/gst/rtspserver.c:
1199 rtsp-stream: Fixed TCP transport case
1200 Make sure that the appsink element is actually added to
1201 the bin before trying to link it with the elements in it.
1202 https://bugzilla.gnome.org/show_bug.cgi?id=776343
1204 2016-12-16 17:26:04 +0000 Tim-Philipp Müller <tim@centricular.com>
1210 Remove generated .spec file
1211 Likely extremely bitrotten, and we should not ship this anyway.
1213 2016-12-03 08:21:02 +0100 Edward Hervey <bilboed@bilboed.com>
1216 Automatic update of common submodule
1217 From f980fd9 to 39ac2f5
1219 2016-12-02 15:40:09 +0100 Edward Hervey <edward@centricular.com>
1221 * gst/rtsp-server/rtsp-media.c:
1222 media: Fix pt map caps
1223 Since decryption is handled within rtpbin, all outcoming stream
1224 caps will be application/x-rtp (i.e. regular rtp)
1225 Fixes RECORD with SRTP streams
1227 2016-12-02 15:38:04 +0100 Edward Hervey <edward@centricular.com>
1229 * gst/rtsp-server/rtsp-media-factory.c:
1230 media-factory: Create media objects with the proper transport mode
1231 The function called immediately afterwards (collect_streams()) will
1232 need it to work properly
1234 2016-12-02 14:36:50 +0200 Sebastian Dröge <sebastian@centricular.com>
1236 * gst/rtsp-server/rtsp-auth.c:
1237 rtsp-auth: Don't remove digest-auth nonces that already/still have a client connected
1239 2016-12-01 18:04:34 +0200 Sebastian Dröge <sebastian@centricular.com>
1241 * gst/rtsp-server/rtsp-media-factory.c:
1242 rtsp-media-factory: Don't create a pipeline for the media pipeline string
1243 We're going to put a pipeline into a pipeline otherwise, which is not
1246 2016-10-25 15:41:28 +0300 Kseniia Vasilchuk <vasilchukkseniia@gmail.com>
1248 * gst/rtsp-server/rtsp-media.c:
1249 media: Fix race condition around finish_unprepare() if called multiple time
1250 https://bugzilla.gnome.org/show_bug.cgi?id=755329
1252 2016-11-30 14:06:36 +1100 Jan Schmidt <jan@centricular.com>
1254 * gst/rtsp-sink/gstrtspclientsink.c:
1255 rtspclientsink: Don't leave stale pointer after unref
1256 Fix a warning on shutdown - don't keep a pointer to an
1257 alread-unreffed object.
1259 2016-11-26 11:24:50 +0000 Tim-Philipp Müller <tim@centricular.com>
1262 common: use https protocol for common submodule
1263 https://bugzilla.gnome.org/show_bug.cgi?id=775110
1265 2016-11-21 23:29:56 +1100 Matthew Waters <matthew@centricular.com>
1267 * gst/rtsp-server/rtsp-stream.c:
1268 stream: block the output of rtpbin instead of the source pipeline
1269 85c52e194bcb81928b96614be0ae47d59eccb1ce introduced a more correct
1270 detection of the srtp rollover counter to add to the SDP.
1271 Unfortunately, it was incomplete for live pipelines where the logic
1272 blocks the source bin before creating the SDP and thus would never have
1273 the necessary informaiton to create a correct SDP with srtp encryption.
1274 Move the pad blocks to rtpbin's output pads instead so that the
1275 necessary information can be created before we need the information for
1277 https://bugzilla.gnome.org/show_bug.cgi?id=770239
1279 2016-11-21 16:02:39 +0100 Dag Gullberg <dagg@axis.com>
1281 * gst/rtsp-server/rtsp-client.c:
1282 rtsp-client: add IDLE timeout, before session exists
1283 The RTSP server will not timeout an idle RTSP connection
1284 (note this is different from doing timeout on a RTSP
1286 At least for Apache this is a problem when running RTSP over
1287 HTTPS since it uses one of the threads (there is a rather
1288 limited number) that are available for handling requests.
1289 https://bugzilla.gnome.org/show_bug.cgi?id=771830
1291 2016-11-23 09:45:08 +0000 Tim-Philipp Müller <tim@centricular.com>
1296 2016-11-21 13:05:50 +0100 Göran Jönsson <goranjn@axis.com>
1298 * gst/rtsp-server/rtsp-stream.c:
1299 rtsp-stream: Set close-socket FALSE on UDP src:es
1300 With this RTSP server can use the sockets independent on the udpsrc
1302 When the udp src is finalized it will unref socket and when g_socket
1303 is finalized the socket will be closed.
1304 https://bugzilla.gnome.org/show_bug.cgi?id=765673
1306 2016-11-18 17:47:13 +0200 Sebastian Dröge <sebastian@centricular.com>
1308 * gst/rtsp-sink/gstrtspclientsink.c:
1309 rtspclientsink: Move to new helper function to parse authentication responses
1310 https://bugzilla.gnome.org/show_bug.cgi?id=774416
1312 2016-11-16 08:42:24 +0200 Sebastian Dröge <sebastian@centricular.com>
1314 * examples/Makefile.am:
1315 * examples/test-auth-digest.c:
1316 * gst/rtsp-server/rtsp-auth.c:
1317 * gst/rtsp-server/rtsp-auth.h:
1318 * win32/common/libgstrtspserver.def:
1319 rtsp-auth: Add support for Digest authentication
1320 https://bugzilla.gnome.org/show_bug.cgi?id=774416
1322 2016-11-17 09:41:53 -0800 Scott D Phillips <scott.d.phillips@intel.com>
1325 * gst/rtsp-server/meson.build:
1327 * tests/check/meson.build:
1329 * win32/common/libgstrtspserver.def:
1330 Enable building with MSVC
1331 https://bugzilla.gnome.org/show_bug.cgi?id=774640
1333 2016-11-18 20:23:14 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
1336 meson: gstreamer gst_check_dep does not exist on windows
1338 2016-11-17 09:43:37 -0800 Scott D Phillips <scott.d.phillips@intel.com>
1340 * gst/rtsp-server/rtsp-client.c:
1341 client: update do_send_message to match type GstRTSPClientSendFunc
1342 This type mismatch fails building with MSVC
1343 https://bugzilla.gnome.org/show_bug.cgi?id=774640
1345 2016-11-11 14:42:08 +0200 Sebastian Dröge <sebastian@centricular.com>
1347 * gst/rtsp-server/rtsp-sdp.c:
1348 rtsp-sdp: Fix indentation
1350 2016-11-10 05:16:00 +0000 Neha Arora <arora.neha@samsung.com>
1352 * gst/rtsp-server/rtsp-media.c:
1353 rtsp-media: Only signal "new-state" if the state has actually changed
1354 https://bugzilla.gnome.org/show_bug.cgi?id=774173
1356 2016-08-24 11:39:13 +0200 Branko Subasic <branko@axis.com>
1358 * gst/rtsp-server/rtsp-client.c:
1359 * gst/rtsp-server/rtsp-client.h:
1360 client: emit signal in the beginning of each rtsp request
1361 These signals let the application validate the requests, configure the
1362 media/stream in a certain way and also generate error status code in
1363 case of error or bad request.
1364 https://bugzilla.gnome.org/show_bug.cgi?id=758062
1366 2016-11-01 18:10:35 +0000 Tim-Philipp Müller <tim@centricular.com>
1369 meson: update version
1371 === release 1.11.0 ===
1373 2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com>
1378 === release 1.10.0 ===
1380 2016-11-01 18:06:46 +0200 Sebastian Dröge <sebastian@centricular.com>
1386 * gst-rtsp-server.doap:
1389 2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com>
1391 * tests/check/gst/rtspserver.c:
1392 * tests/check/gst/stream.c:
1393 tests: try to avoid using the same ports in different tests
1394 Causes problems with client multicast tests otherwise if
1395 tests are run in parallel.
1396 https://bugzilla.gnome.org/show_bug.cgi?id=773640
1398 2016-10-28 17:50:59 +0100 Tim-Philipp Müller <tim@centricular.com>
1400 * tests/check/gst/client.c:
1401 tests: client: use fail_unless_equals_foo() for better failure reporting
1403 2016-09-26 11:16:04 +0200 Göran Jönsson <goranjn@axis.com>
1405 * gst/rtsp-server/rtsp-client.c:
1406 rtsp-client: Session filter in unwatch session
1407 Call session filter with filter_session_media as paramer in
1408 client_unwatch_session if using drop_backlog = FALSE.
1409 In client_unwatch_session its allowed to grow the watchs backlog.
1410 If using drop_backlog = FALSE and the backlog is full it will cause
1411 a deadlock when setting session media state to NULL
1412 if the backlog is not allowed to grow.
1413 https://bugzilla.gnome.org/show_bug.cgi?id=771983
1415 2016-10-20 21:40:18 +0100 Tim-Philipp Müller <tim@centricular.com>
1418 meson: add fallbacks for gst modules
1421 2016-09-14 17:48:39 +0300 Nikita Bobkov <NikitaDBobkov@gmail.com>
1423 * gst/rtsp-server/rtsp-client.c:
1424 rtsp-client: Fix factory leaking in find_media() in error cases
1425 https://bugzilla.gnome.org/show_bug.cgi?id=771488
1427 2016-10-06 11:47:50 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1429 * gst/rtsp-server/rtsp-stream.c:
1430 stream: Fix randomly missing streams from SDP with dynamic elements
1431 When using dynamic elements, gst_rtsp_stream_join_bin() is called from
1432 "pad-added" signal. In that case priv->srcpad could already have its caps,
1433 and they'll be sent to priv->send_src[0] pad. That means that when it
1434 connects "notify::caps" signal, that pad could already have received its
1435 caps and the signal won't be emitted anymore.
1436 In that case priv->caps stay to NULL and when building the SDP that stream
1437 gets ignored. Leading to missing video or audio when playing in client side.
1438 https://bugzilla.gnome.org/show_bug.cgi?id=772478
1440 2016-09-30 11:42:08 +0100 Tim-Philipp Müller <tim@centricular.com>
1443 meson: update version
1445 === release 1.9.90 ===
1447 2016-09-30 13:04:12 +0300 Sebastian Dröge <sebastian@centricular.com>
1453 * gst-rtsp-server.doap:
1456 2016-09-17 13:17:19 +0100 Ian Jamison <ian.dev@arkver.com>
1458 * gst/rtsp-server/rtsp-media-factory.c:
1459 * gst/rtsp-server/rtsp-media.c:
1460 * gst/rtsp-server/rtsp-stream.c:
1461 rtsp-server: Hint that set_multicast_iface expects the name of the interface
1462 To prevent any possibly confusion with IPs or anything else.
1463 https://bugzilla.gnome.org/show_bug.cgi?id=771530
1465 2016-09-18 09:58:55 -0400 Sebastian Dröge <sebastian@centricular.com>
1467 * gst/rtsp-server/rtsp-media-factory.c:
1468 * gst/rtsp-server/rtsp-media.c:
1469 rtsp-media: Call g_free() instead of g_object_unref() on multicast-iface strings
1470 https://bugzilla.gnome.org/show_bug.cgi?id=763000#c5
1472 2016-09-14 11:31:15 +0200 Sebastian Dröge <sebastian@centricular.com>
1475 configure: Depend on gstreamer 1.9.2.1
1477 2016-09-10 20:52:31 +1000 Jan Schmidt <jan@centricular.com>
1481 Automatic update of common submodule
1482 From b18d820 to f980fd9
1484 2016-09-10 09:58:31 +1000 Jan Schmidt <jan@centricular.com>
1488 Automatic update of common submodule
1489 From 6f2d209 to b18d820
1491 2016-09-07 18:44:34 +0300 Sebastian Dröge <sebastian@centricular.com>
1493 * gst/rtsp-server/rtsp-stream.c:
1494 rtsp-stream: Remove unused _locked() variant of a function
1495 It was added during refactoring.
1497 2016-09-07 10:21:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1499 * gst/rtsp-server/rtsp-stream.c:
1500 stream: cosmetic cleanup
1501 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1503 2016-09-07 10:16:19 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1505 * gst/rtsp-server/rtsp-stream.c:
1506 stream: Compare IP addresses case insensitive in more places
1507 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1509 2016-09-07 10:12:18 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1512 * gst/rtsp-server/rtsp-stream.c:
1513 stream: Fix leaked joined_bin
1514 There is no need to keep a strong ref on it, and _leave_bin() was
1515 setting it to NULL before calling g_clear_object() so it was leaked.
1516 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1518 2016-09-06 19:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
1520 * gst/rtsp-server/rtsp-stream.c:
1521 rtsp-stream: Compare IP address strings case insensitive
1522 Otherwise IPv6 addresses might fail this comparision.
1524 2016-09-06 19:10:21 +0300 Sebastian Dröge <sebastian@centricular.com>
1526 * gst/rtsp-server/rtsp-stream.c:
1527 rtsp-stream: Bind multicast sockets to ANY as before
1528 https://bugzilla.gnome.org/show_bug.cgi?id=766612#c48
1530 2016-09-05 18:31:36 +0300 Kseniia <vasilchukkseniia@gmail.com>
1532 * gst/rtsp-server/rtsp-session.c:
1533 rtsp-session: Fix segfault when doing keep-alive after removing the session
1534 If keep-alive happens after removing the session but before finalizing the
1535 stream transport, we would segfault.
1536 https://bugzilla.gnome.org/show_bug.cgi?id=750544
1538 2016-09-05 18:04:50 +0300 Sebastian Dröge <sebastian@centricular.com>
1540 * gst/rtsp-server/rtsp-stream.c:
1541 rtsp-stream: Always create multicast UDP elements if the protocol flag is set
1542 Adding them later will cause deadlocks due to
1543 1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
1544 2) adding the multicast sink
1545 3) waiting for it to get data to preroll again
1546 3) never happens because the queues after the tee are full.
1548 2016-09-05 16:32:57 +0300 Sebastian Dröge <sebastian@centricular.com>
1550 * gst/rtsp-server/rtsp-stream.c:
1551 rtsp-stream: Fix up various multicast related issues
1553 2016-09-05 13:40:59 +0300 Sebastian Dröge <sebastian@centricular.com>
1555 * tests/check/gst/stream.c:
1556 tests: Fix compilation
1558 2016-07-28 15:33:05 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1560 * gst/rtsp-server/rtsp-client.c:
1561 * gst/rtsp-server/rtsp-stream.c:
1562 * tests/check/gst/stream.c:
1563 stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast
1564 This is basically reverting changes introduced in commit f62a9a7,
1565 because it was introducing various regressions:
1566 - It introduces a leak of udpsrc elements that got wrongly fixed by adding
1567 an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
1568 ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
1569 - If a mcast client connects, it creates a new socket in SETUP to try to respect
1570 the destination/port given by the client in the transport, and overrides the
1571 socket already set on the udpsink element. That means that if we already had a
1572 client connected, the source address on the udp packets it receives suddenly
1574 - If a 2nd mcast client connects, the destination/port in its transport is
1575 ignored but its transport wasn't updated.
1576 What this patch does:
1577 - Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
1578 - Always have a tee+queue when udp is enabled. This could be optimized
1579 again in a later patch, but is more complicated. If no unicast clients
1580 connects then those elements are useless, this could be also optimized
1582 - When mcast transport is added, it creates a new set of udpsrc/udpsink,
1583 seperated from those for unicast clients. Since we already support only
1584 one mcast address, we also create only one set of elements.
1585 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1587 2016-07-28 15:20:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1589 * gst/rtsp-server/rtsp-stream.c:
1590 stream: factor our plug_src function
1591 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1593 2016-07-21 21:46:16 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1595 * gst/rtsp-server/rtsp-stream.c:
1596 stream: factor out plug_sink function
1597 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1599 2016-07-20 23:05:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1601 * gst/rtsp-server/rtsp-stream.c:
1602 stream: small documentation clarification
1603 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1605 2016-07-20 15:35:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1607 * gst/rtsp-server/rtsp-stream.c:
1608 stream: rename addr_v4/6 to mcast_addr_v4/6 for clarity
1609 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1611 2016-07-14 11:10:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1613 * gst/rtsp-server/rtsp-stream.c:
1614 stream: Keep a ref on joined bin
1615 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1617 2016-07-20 15:11:32 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1619 * gst/rtsp-server/rtsp-stream.c:
1620 stream: code cleanup
1621 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1623 2016-07-20 23:18:23 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1625 * gst/rtsp-server/rtsp-stream.c:
1626 stream: small fix in error code path
1627 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1629 2016-07-20 20:09:57 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1631 * gst/rtsp-server/rtsp-stream.c:
1632 Revert "rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc"
1633 This partly reverts commit cba045e1b19fad6e689e10206f57903e15f1229a,
1634 but keeps unit tests.
1635 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1637 2016-09-01 12:33:00 +0300 Sebastian Dröge <sebastian@centricular.com>
1642 === release 1.9.2 ===
1644 2016-09-01 12:32:51 +0300 Sebastian Dröge <sebastian@centricular.com>
1650 * gst-rtsp-server.doap:
1653 2016-01-27 01:03:52 +0000 Tim-Philipp Müller <tim@centricular.com>
1656 * examples/meson.build:
1658 * gst/rtsp-server/meson.build:
1659 * gst/rtsp-sink/meson.build:
1661 * pkgconfig/meson.build:
1662 * tests/check/meson.build:
1663 * tests/meson.build:
1664 Add support for Meson as alternative/parallel build system
1665 https://github.com/mesonbuild/meson
1667 2016-08-26 21:56:13 +0200 Josep Torra <n770galaxy@gmail.com>
1670 * tests/check/Makefile.am:
1671 build: silence error about pthread for 'make check' in osx
1672 Fixes "clang: error: argument unused during compilation: '-pthread'"
1674 2015-09-25 15:04:00 +0000 Nikita Bobkov <NikitaDBobkov@gmail.com>
1676 * gst/rtsp-server/rtsp-client.c:
1677 rtsp-client: Fix leaking of media in error cases
1678 With additional fixes by Kseniya Vasilchuk <vasilchukkseniia@gmail.com>
1679 and myself to make the media refcounting a bit easier to follow.
1680 https://bugzilla.gnome.org/show_bug.cgi?id=755632
1682 2016-08-02 15:08:22 +0300 Sebastian Dröge <sebastian@centricular.com>
1684 * gst/rtsp-server/rtsp-client.c:
1685 rtsp-client: Fix leaking of session in error cases
1686 https://bugzilla.gnome.org/show_bug.cgi?id=755632
1688 2016-07-11 21:16:04 +0200 Stefan Sauer <ensonic@users.sf.net>
1691 Automatic update of common submodule
1692 From f363b32 to f49c55e
1694 2016-07-06 13:51:15 +0300 Sebastian Dröge <sebastian@centricular.com>
1699 === release 1.9.1 ===
1701 2016-07-06 13:28:12 +0300 Sebastian Dröge <sebastian@centricular.com>
1707 * gst-rtsp-server.doap:
1710 2016-06-24 02:02:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
1713 configure: Need to add -DGST_STATIC_COMPILATION when building only statically
1714 https://bugzilla.gnome.org/show_bug.cgi?id=767463
1716 2016-06-21 11:49:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
1719 Automatic update of common submodule
1720 From ac2f647 to f363b32
1722 2016-04-14 22:56:11 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1724 * gst/rtsp-server/rtsp-sdp.c:
1725 * gst/rtsp-server/rtsp-sdp.h:
1726 * gst/rtsp-server/rtsp-stream.c:
1727 * gst/rtsp-server/rtsp-stream.h:
1728 sdp: add rollover counters for all sender SSRC
1729 We add different crypto sessions in MIKEY, one for each sender
1730 SSRC. Currently, all of them will have the same security policy, 0.
1731 The rollover counters are obtained from the srtpenc element using the
1733 https://bugzilla.gnome.org/show_bug.cgi?id=730539
1735 2016-06-07 20:44:42 +0100 Tim-Philipp Müller <tim@centricular.com>
1737 * gst/rtsp-server/rtsp-media-factory.h:
1738 * gst/rtsp-server/rtsp-server.h:
1739 docs: fix some typos
1741 2016-05-25 10:28:43 +0100 Tim-Philipp Müller <tim@centricular.com>
1743 * gst/rtsp-server/Makefile.am:
1744 g-i: pass compiler env to g-ir-scanner
1745 It's what introspection.mak does as well. Should
1746 fix spurious build failures on gnome-continuous
1747 (caused by g-ir-scanner getting compiler details
1748 via python which is broken in some environments
1749 so passing the compiler details bypasses that).
1751 2016-05-18 16:48:44 +0100 Ian <ian.arkver.dev@gmail.com>
1753 * gst/rtsp-server/rtsp-session.c:
1754 rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header
1755 This works with rtspsrc and live555, but fails with e.g. ffmpeg.
1756 https://bugzilla.gnome.org/show_bug.cgi?id=766619
1758 2016-03-07 14:48:38 +0100 Edward Hervey <bilboed@bilboed.com>
1760 * gst/rtsp-sink/gstrtspclientsink.c:
1761 rtspclientsink: Check return value of sscanf
1762 And just make sure we always have 0/0 if we have an error
1765 2016-04-25 08:55:25 -0400 Jake Foytik <jake.foytik@ipconfigure.com>
1767 * gst/rtsp-server/rtsp-stream.c:
1768 * tests/check/gst/rtspserver.c:
1769 * tests/check/gst/stream.c:
1770 rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
1771 - Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
1772 - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
1773 - Create unit test for shared media.
1774 https://bugzilla.gnome.org/show_bug.cgi?id=764744
1776 2016-04-11 10:55:23 +0300 Sebastian Dröge <sebastian@centricular.com>
1778 * gst/rtsp-server/rtsp-stream.c:
1779 rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
1780 For IPv6 addresses, binding to a multicast group does not work on Linux
1781 either. Always bind to ANY and then later join the multicast group.
1782 https://bugzilla.gnome.org/show_bug.cgi?id=764679
1784 2016-04-14 10:05:02 +0100 Julien Isorce <j.isorce@samsung.com>
1787 Automatic update of common submodule
1788 From 6f2d209 to ac2f647
1790 2016-04-06 10:09:46 +0200 Patricia Muscalu <patricia@axis.com>
1792 * gst/rtsp-server/rtsp-thread-pool.c:
1793 rtsp-thread-pool: explained why GSource is a part of ThreadImpl
1794 Clarified why it is necessary to add source information to
1795 GstRTSPThreadImpl. See the reported bug in GLib:
1796 https://bugzilla.gnome.org/show_bug.cgi?id=720186
1797 for more information.
1798 https://bugzilla.gnome.org/show_bug.cgi?id=761702
1800 2016-04-04 12:58:38 +0300 Sebastian Dröge <sebastian@centricular.com>
1802 * examples/Makefile.am:
1803 examples: Clean up CFLAGS/LDADD even more
1804 The internal .la should come first and is part of LDADD, as is
1807 2016-04-04 12:39:39 +0300 Sebastian Dröge <sebastian@centricular.com>
1809 * examples/Makefile.am:
1810 examples: Clean up CFLAGS/LDADD to link with the correct versions of all libraries
1812 2016-04-03 12:06:29 +0300 Sebastian Dröge <sebastian@centricular.com>
1814 * gst/rtsp-server/Makefile.am:
1815 rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS)
1817 2015-12-30 18:39:05 +0200 Sebastian Dröge <sebastian@centricular.com>
1819 * gst/rtsp-server/rtsp-client.c:
1820 * gst/rtsp-server/rtsp-media-factory.c:
1821 * gst/rtsp-server/rtsp-media-factory.h:
1822 * gst/rtsp-server/rtsp-media.c:
1823 * gst/rtsp-server/rtsp-media.h:
1824 * gst/rtsp-server/rtsp-sdp.c:
1825 * gst/rtsp-server/rtsp-stream.c:
1826 * gst/rtsp-server/rtsp-stream.h:
1827 rtsp-server: Implement clock signalling according to RFC7273
1828 For NTP and PTP clocks we signal the actual clock that is used and signal
1829 the direct media clock offset.
1830 For all other clocks we at least signal that it's the local sender clock.
1831 This allows receivers to know which clock was used to generate the media and
1832 its RTP timestamps. Receivers can then implement network synchronization,
1833 either absolute or at least relative by getting the sender clock rate directly
1834 via NTP/PTP instead of estimating it from RTP timestamps and packet receive
1836 https://bugzilla.gnome.org/show_bug.cgi?id=760005
1838 2016-03-02 19:42:58 +0200 Sebastian Dröge <sebastian@centricular.com>
1840 * gst/rtsp-sink/gstrtspclientsink.c:
1841 rtspclientsink: Add support for setting the multicast interface
1842 https://bugzilla.gnome.org/show_bug.cgi?id=763000
1844 2016-03-02 19:42:13 +0200 Sebastian Dröge <sebastian@centricular.com>
1846 * gst/rtsp-server/rtsp-media-factory.c:
1847 * gst/rtsp-server/rtsp-media-factory.h:
1848 * gst/rtsp-server/rtsp-media.c:
1849 * gst/rtsp-server/rtsp-media.h:
1850 * gst/rtsp-server/rtsp-stream.c:
1851 * gst/rtsp-server/rtsp-stream.h:
1852 rtsp-media: Add support for setting the multicast interface
1853 https://bugzilla.gnome.org/show_bug.cgi?id=763000
1855 2016-03-07 08:50:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
1857 * gst/rtsp-sink/gstrtspclientsink.c:
1858 rtspclientsink: use new gst_element_class_add_static_pad_template()
1859 https://bugzilla.gnome.org/show_bug.cgi?id=763196
1861 2016-03-24 13:33:43 +0200 Sebastian Dröge <sebastian@centricular.com>
1866 === release 1.8.0 ===
1868 2016-03-24 13:00:35 +0200 Sebastian Dröge <sebastian@centricular.com>
1874 * gst-rtsp-server.doap:
1877 2016-03-16 23:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
1879 * gst/rtsp-server/rtsp-stream.c:
1880 rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
1881 This would get us NO_PREROLL in the bin again and break seeking.
1882 Thanks to Carlos Rafael Giani for helping to debug this!
1883 https://bugzilla.gnome.org/show_bug.cgi?id=740509
1885 === release 1.7.91 ===
1887 2016-03-15 12:26:13 +0200 Sebastian Dröge <sebastian@centricular.com>
1893 * gst-rtsp-server.doap:
1896 2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
1898 * gst/rtsp-server/rtsp-stream.c:
1899 rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
1900 Without this, RECORD pipelines are broken because
1901 a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
1902 added later. Previously it was there earlier and due to NO_PREROLL caused the
1903 pipeline to preroll immediately
1904 b) the udpsrc for the pipeline is added later and never set to PLAYING state,
1905 as the corresponding code previously was only for PLAY pipelines.
1906 https://bugzilla.gnome.org/show_bug.cgi?id=763281
1908 2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
1910 * gst/rtsp-server/rtsp-stream.c:
1911 rtsp-stream: Fix typo in the docstring
1912 gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
1914 2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
1916 * gst/rtsp-server/rtsp-stream.c:
1917 rtsp-stream: Disable multicast loopback for all our sockets
1918 On Windows this is a receiver-side setting, on Linux a sender-side setting. As
1919 we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
1920 loopback setting on the socket... while udpsink does which unfortunately has
1921 no effect here on Windows but on Linux.
1922 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1924 2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
1926 * tests/check/gst/stream.c:
1927 stream tests: added new tests
1928 Test a case when the address pool only contains multicast addresses
1929 and the client is requesting unicast udp.
1930 Added tests for multicast ports allocation.
1931 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1933 2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
1935 * gst/rtsp-server/rtsp-stream.c:
1936 rtsp-stream: Only bind multicast sockets to ANY on Windows
1937 On Linux it is still needed to bind to the multicast address
1938 to filter out random other packets, while on Windows binding
1939 to multicast addresses just fails.
1941 2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
1943 * gst/rtsp-server/rtsp-stream.c:
1944 rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
1945 Otherwise we fail to allocate UDP ports if the pool only contains multicast
1946 addresses, which is something that used to work before. For unicast addresses
1947 if the pool contains none, we just allocate them as if there is no pool at
1949 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1951 2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
1953 * gst/rtsp-server/rtsp-client.c:
1954 * gst/rtsp-server/rtsp-stream.c:
1955 rtsp-server: Fix indentation
1957 2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
1959 * gst/rtsp-server/rtsp-stream.c:
1960 rtsp-stream: Don't bind the sockets to multicast addresses
1961 This works on Linux but fails completely on Windows. You're supposed
1962 to bind to ANY and then join the multicast group.
1963 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1965 === release 1.7.90 ===
1967 2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
1973 * gst-rtsp-server.doap:
1976 2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
1979 Automatic update of common submodule
1980 From b64f03f to 6f2d209
1982 2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
1984 * gst/rtsp-sink/gstrtspclientsink.c:
1985 * tests/check/gst/rtspclientsink.c:
1986 rtspsink: Fix some leaks in rtspclientsink and the unit test.
1987 https://bugzilla.gnome.org/show_bug.cgi?id=762525
1989 2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
1991 * tests/check/gst/media.c:
1992 * tests/check/gst/rtspclientsink.c:
1993 * tests/check/gst/rtspserver.c:
1994 * tests/check/gst/stream.c:
1995 tests: unit test fixes
1996 Removed port allocation test from the media suite.
1997 The port allocation failure is now in the stream suite.
1999 Make sure that the media is suspended after the DESCRIBE request
2000 before reconfiguring the UDP sinks.
2002 In the RECORD case we have to set async property to false
2003 for the appsink element in the test in order to make sure
2004 that the media pipeline doesn't hang in start_preroll().
2005 https://bugzilla.gnome.org/show_bug.cgi?id=757488
2007 2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
2009 * gst/rtsp-server/rtsp-client.c:
2010 * gst/rtsp-server/rtsp-stream.c:
2011 * gst/rtsp-server/rtsp-stream.h:
2012 rtsp-stream: postpone UDP socket allocation until SETUP
2013 Postpone the allocation of the UDP sockets until we know
2014 what transport has been chosen by the client.
2015 Both unicast and multicast UDP sources are created in one
2017 https://bugzilla.gnome.org/show_bug.cgi?id=757488
2019 2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
2021 * gst/rtsp-server/rtsp-stream.c:
2022 rtsp-stream: postpone the creation of the UDP sources
2023 Code refactoring: allocate the UDP ports after the sender and
2024 the reciver parts have been created.
2025 We postpone the creation of the UDP sources until the UDP
2026 ports have been allocated.
2027 https://bugzilla.gnome.org/show_bug.cgi?id=757488
2029 2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
2031 * gst/rtsp-server/rtsp-stream.c:
2032 rtsp-stream: added function for setting UDP sources to PLAYING state
2033 Code refactoring: Introduced a function for setting UDP sources
2035 https://bugzilla.gnome.org/show_bug.cgi?id=757488
2037 2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
2039 * gst/rtsp-server/rtsp-stream.c:
2040 rtsp-stream: added function for creating and configuring UDP sources
2041 Code refactoring: create and configure UDP sources in a separate function.
2042 https://bugzilla.gnome.org/show_bug.cgi?id=757488
2044 2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
2046 * gst/rtsp-server/rtsp-stream.c:
2047 rtsp-stream: added function for RTP/RTCP socket configuration
2048 Code refactoring: configure RTP and RTCP sockets for UDP sinks
2049 in a separate function.
2050 https://bugzilla.gnome.org/show_bug.cgi?id=757488
2052 2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
2054 * gst/rtsp-server/rtsp-stream.c:
2055 rtsp-stream: added function for creating and configuring UDP sinks
2056 Code refactoring: create and configure UDP sinks in a separate function.
2057 https://bugzilla.gnome.org/show_bug.cgi?id=757488
2059 2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
2061 * gst/rtsp-server/rtsp-stream.c:
2062 rtsp-stream: added helper function for creating the sender/receiver parts
2063 Code refactoring: introduced helper function for creating
2064 the receiver and the sender parts of the streaming pipeline.
2065 https://bugzilla.gnome.org/show_bug.cgi?id=757488
2067 2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
2072 === release 1.7.2 ===
2074 2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
2080 * gst-rtsp-server.doap:
2083 2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
2085 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
2086 uninstalled.pc: add support for non libtool build systems
2087 Currently the .la path is provided which requires to use libtool as
2088 mentioned in the GStreamer manual section-helloworld-compilerun.html.
2089 It is fine as long as the application is built using libtool.
2090 So currently it is not possible to compile a GStreamer application
2091 within gst-uninstalled with CMake or other build system different
2093 This patch allows to do the following in gst-uninstalled env:
2094 gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
2095 gstreamer-rtsp-server-1.0)
2096 Previously it required to prepend libtool --mode=link
2097 https://bugzilla.gnome.org/show_bug.cgi?id=720778
2099 2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
2101 * gst/rtsp-sink/gstrtspclientsink.c:
2102 rtspclientsink: remove check for impossible condition
2103 Goto error label checks stream to see if it needs to be unreferenced before
2104 returning, but this goto jumps happens before the stream is ever set, so it
2105 will always be NULL in this error label.
2108 2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
2110 * gst/rtsp-sink/gstrtspclientsink.c:
2111 rtspclientsink: clean switch statements
2112 Coverity demands for fallthrough statements to be clearly commented,
2113 to distinguish from accidental fall throughs. And it also needs all
2114 cases to finish with a break, even if the break is never going to be
2115 executed like in the case of a continue jump.
2119 2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
2121 * tests/check/Makefile.am:
2122 tests: extend the AM_TESTS_ENVIRONMENT from check.mak
2123 To get the CK_DEFAULT_TIMEOUT defined for all tests
2124 Also removes a 120 seconds timeout that was set as default
2125 explicitly in this module
2126 https://bugzilla.gnome.org/show_bug.cgi?id=761472
2128 2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
2132 Automatic update of common submodule
2133 From 86e4663 to b64f03f
2135 2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
2137 * gst/rtsp-server/rtsp-media.c:
2138 rtsp-media: fix state_lock not locked again when preroll fails
2139 https://bugzilla.gnome.org/show_bug.cgi?id=761399
2141 2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
2144 configure: Move plugin specific flags below all the others
2145 They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
2146 -no-undefined. And -no-undefined is required on Windows to build DLLs.
2148 2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
2150 * gst/rtsp-sink/gstrtspclientsink.c:
2151 rtspclientsink: Simplify slightly using new -base API
2152 Use the new Mikey and SDP API in the base plugins libs
2153 to simplify some code.
2154 https://bugzilla.gnome.org/show_bug.cgi?id=758180
2156 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
2161 * gst/rtsp-sink/Makefile.am:
2162 * gst/rtsp-sink/gstrtspclientsink.c:
2163 * gst/rtsp-sink/gstrtspclientsink.h:
2164 * gst/rtsp-sink/plugin.c:
2165 * tests/check/Makefile.am:
2166 * tests/check/gst/rtspclientsink.c:
2167 rtspsink: Add rtspclientsink element
2168 Add an rtspclientsink element that accepts streams for which
2169 there is a registered payloader and sends them to
2170 an RTSP server using RECORD.
2171 Sending is synchronised to the pipeline clock. Payload-types
2172 are automatically selected. The 'new-payloader' signal is fired
2173 for custom configuration of payloaders when they are created.
2174 Can now stream a movie like this:
2176 ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
2177 decodebin name=depay1 ! audioconvert ! autoaudiosink )"
2179 gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
2180 queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
2181 https://bugzilla.gnome.org/show_bug.cgi?id=758180
2183 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
2185 * gst/rtsp-server/rtsp-stream.c:
2186 * gst/rtsp-server/rtsp-stream.h:
2187 rtsp-stream: Add functions for using rtsp-stream from the client
2188 Add a boolean to indicate that the rtsp-stream is running on the
2189 'client' side of an RTSP connection, for sending streams via
2190 RECORD. In that case, the roles of the client/server ports
2191 in transport setup are swapped.
2192 https://bugzilla.gnome.org/show_bug.cgi?id=758180
2194 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
2196 * gst/rtsp-server/rtsp-sdp.c:
2197 * gst/rtsp-server/rtsp-sdp.h:
2198 rtsp-sdp: Add gst_rtsp_sdp_from_stream()
2199 A new function that adds info from a GstRTSPStream into an SDP message.
2200 https://bugzilla.gnome.org/show_bug.cgi?id=758180
2202 2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
2204 * gst/rtsp-server/rtsp-media.c:
2205 rtsp-media: Fix mutex beeing unlocked while they should be locked
2206 https://bugzilla.gnome.org/show_bug.cgi?id=761226
2208 2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
2210 * gst/rtsp-server/rtsp-media-factory.c:
2211 rtsp-media-factory: add missing break in "clock" property setter
2214 2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
2216 * gst/rtsp-server/rtsp-stream.c:
2217 rtsp-stream: fixed assert during update transport
2218 When RTSP server trying update transport during multicast, it throws an
2219 assert. The assert is thrown because it is trying to get the parent of
2220 an non-existing funnel element.
2221 https://bugzilla.gnome.org/show_bug.cgi?id=760150
2223 2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
2225 * gst/rtsp-server/rtsp-permissions.h:
2226 * gst/rtsp-server/rtsp-thread-pool.h:
2227 * gst/rtsp-server/rtsp-token.h:
2228 docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
2229 gtk-doc can handle static inline functions just fine these days,
2230 there's no need for this stuff any more.
2232 2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
2234 * gst/rtsp-server/rtsp-media.c:
2235 * gst/rtsp-server/rtsp-sdp.c:
2236 sdp: replace duplicated codes to call new base sdp apis
2237 https://bugzilla.gnome.org/show_bug.cgi?id=745880
2239 2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
2241 * examples/test-netclock.c:
2242 test-netclock: Use the new API to configure a clock directly
2244 2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
2246 * gst/rtsp-server/rtsp-media-factory.c:
2247 * gst/rtsp-server/rtsp-media-factory.h:
2248 * gst/rtsp-server/rtsp-media.c:
2249 * gst/rtsp-server/rtsp-media.h:
2250 rtsp-media: Add API to directly configure a clock on the media pipelines
2252 2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
2254 * gst/rtsp-server/rtsp-media.c:
2255 rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
2257 2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
2259 * gst/rtsp-server/rtsp-media-factory.c:
2260 rtsp-media-factory: Add FIXME for 2.0
2262 2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
2264 * gst/rtsp-server/rtsp-stream.c:
2265 rtsp-stream: Fix indentation
2267 2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2269 * gst/rtsp-server/rtsp-media.c:
2270 rtsp-media: Do not prepare media after media times out
2271 Deferred calls to start_prepare() can be deferred past the point until
2272 which wait_preroll() and by proxy gst_rtsp_media_get_status() is
2273 prepared to wait. Previously there was no lock and no check for this
2274 situation. This meant that a media could be prepared and unprepared
2275 simultaneously by two different threads. Now a lock is in place and a
2276 suitable check is done.
2277 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
2279 2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
2281 * gst/rtsp-server/rtsp-client.c:
2282 * gst/rtsp-server/rtsp-media-factory.c:
2283 * gst/rtsp-server/rtsp-media-factory.h:
2284 * gst/rtsp-server/rtsp-media.c:
2285 * gst/rtsp-server/rtsp-media.h:
2286 rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
2287 Without TEARDOWN it might be desireable to keep the media running and continue
2288 sending data to the client, even if the RTSP connection itself is
2290 Only do this for session medias that have only UDP transports. If there's at
2291 least on TCP transport, it will stop working and cause problems when the
2292 connection is disconnected.
2293 https://bugzilla.gnome.org/show_bug.cgi?id=758999
2295 2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
2300 === release 1.7.1 ===
2302 2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
2308 * gst-rtsp-server.doap:
2311 2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
2314 configure: Make -Bsymbolic check work with clang.
2315 Update the -Bsymbolic check with the version glib has. This version
2317 https://bugzilla.gnome.org/show_bug.cgi?id=759713
2319 2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
2321 * gst/rtsp-server/rtsp-session-pool.c:
2322 rtsp-session-pool: Avoid dollar sign ($) in session ids
2323 Live555 in VLC strips off dollar signs and then gets very confused,
2324 we don't loose too much entropy by just skipping it.
2326 2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
2328 * gst/rtsp-server/rtsp-address-pool.h:
2329 * gst/rtsp-server/rtsp-auth.h:
2330 * gst/rtsp-server/rtsp-client.h:
2331 * gst/rtsp-server/rtsp-media-factory-uri.h:
2332 * gst/rtsp-server/rtsp-media-factory.h:
2333 * gst/rtsp-server/rtsp-media.h:
2334 * gst/rtsp-server/rtsp-mount-points.h:
2335 * gst/rtsp-server/rtsp-permissions.h:
2336 * gst/rtsp-server/rtsp-server.h:
2337 * gst/rtsp-server/rtsp-session-media.h:
2338 * gst/rtsp-server/rtsp-session-pool.h:
2339 * gst/rtsp-server/rtsp-session.h:
2340 * gst/rtsp-server/rtsp-stream-transport.h:
2341 * gst/rtsp-server/rtsp-stream.h:
2342 * gst/rtsp-server/rtsp-thread-pool.h:
2343 * gst/rtsp-server/rtsp-token.h:
2344 rtsp-server: Add g_autoptr() support to all types
2345 https://bugzilla.gnome.org/show_bug.cgi?id=754464
2347 2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
2349 * gst/rtsp-server/rtsp-stream.c:
2350 rtsp-stream: fixed valgrind error
2351 Fixed the valgrind error in unit test. The UDP source created during
2352 gst_rtsp_stream_join_bin() was not released while destroying the rtp
2354 https://bugzilla.gnome.org/show_bug.cgi?id=759010
2356 2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
2360 Automatic update of common submodule
2361 From b319909 to 86e4663
2363 2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
2365 * gst/rtsp-server/rtsp-client.c:
2366 rtsp-client: suspend media during setup request
2367 SETUP request from clients needs to suspend the media to clear the
2368 prerolled buffers. Otherwise it will not affect the prerolled buffer
2369 and the prerolled buffers will be incorrect (for example block-size
2370 from setup request will not affect the prerolled buffer unless the
2371 media is suspended).
2372 https://bugzilla.gnome.org/show_bug.cgi?id=758268
2374 2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
2376 * gst/rtsp-server/rtsp-stream.c:
2377 rtsp-stream: create stream pipeline based on transport
2378 Based on the protocol, create the rtsp stream pipeline. If only TCP or
2379 only UDP is set as the transport protocol, it will not add the extra tee
2380 or queue element to the pipeline. Both these elements will be added, if
2381 it supports both TCP and UDP protocols. This improves the pipeline
2382 performance when one protocol is present.
2383 https://bugzilla.gnome.org/show_bug.cgi?id=758179
2385 2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
2387 * gst/rtsp-server/rtsp-stream.c:
2388 rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
2389 Adding them when not needed will start some logic inside rtpbin that might be
2390 problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
2391 would start up a rtpjitterbuffer and behave in weird ways.
2392 We still set up the UDP sources for RTP receiving for a sender media to be
2393 able to receive any packets sent by the client for NAT traversal. They will
2394 all go to a fakesink though.
2395 Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
2396 NO_PREROLL, which will cause deadlocks when seeking the media as it will never
2397 receive ASYNC_DONE after a seek.
2398 https://bugzilla.gnome.org/show_bug.cgi?id=758319
2400 2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
2402 * gst/rtsp-server/rtsp-stream.c:
2403 rtsp-stream: Disable multicast loopback for the multicast udp sources too
2404 On POSIX this setting is for sender sockets, on Windows for receiver sockets.
2405 Previously we were only setting this for sender sockets, which caused looped
2406 back packets to be received on Windows if a multicast transport was used.
2408 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
2410 * examples/test-record-auth.c:
2411 * examples/test-record.c:
2412 examples: Actually use the provided port in the record examples
2414 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
2416 * examples/test-record-auth.c:
2417 test-record-auth: Add the option to build in TLS support
2419 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
2421 * examples/test-auth.c:
2422 test-auth: Use an 'anonymous' user for unauthenticated default
2423 There's a comment on one of the resources that 'user' and 'admin'
2424 shouldn't even be able to see it, but they can if the default
2425 token is 'admin2', since that gives them access anyway.
2427 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
2429 * examples/.gitignore:
2430 * examples/Makefile.am:
2431 * examples/test-record-auth.c:
2432 Add test-record-auth example
2434 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
2436 * gst/rtsp-server/rtsp-client.c:
2437 * tests/check/gst/client.c:
2438 rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
2440 2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
2442 * gst/rtsp-server/rtsp-server.c:
2443 rtsp-server: Change the logic so we don't pop a NULL context
2444 When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
2445 will sometimes fail. This call is made before any context is pushed
2446 resulting in an attempt to pop a NULL context.
2447 https://bugzilla.gnome.org/show_bug.cgi?id=757949
2449 2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
2451 * tests/check/gst/rtspserver.c:
2452 rtspserver: Add udp-mcast transport SETUP test
2453 Refactor utility functions in the test file so they can handle
2454 more than UDP and TCP as lower transport.
2455 https://bugzilla.gnome.org/show_bug.cgi?id=756969
2457 2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
2459 * gst/rtsp-server/rtsp-stream.c:
2460 rtsp-stream: Always unref return value of gst_object_get_parent()
2461 Fixes a leak of a GstBin in the udp-mcast case.
2462 https://bugzilla.gnome.org/show_bug.cgi?id=756968
2464 2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
2467 Automatic update of common submodule
2468 From b99800a to b319909
2470 2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
2473 Use new GST_ENABLE_EXTRA_CHECKS #define
2474 https://bugzilla.gnome.org/show_bug.cgi?id=756870
2476 2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
2479 Automatic update of common submodule
2480 From 6babecd to b99800a
2482 2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
2485 Update GLib dependency to 2.40.0
2487 2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
2489 * examples/test-mp4.c:
2490 * gst/rtsp-server/rtsp-stream.c:
2491 stream: listen to sender ssrc signals
2492 https://bugzilla.gnome.org/show_bug.cgi?id=746747
2494 2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
2497 common: update for new suppression
2498 Makes check-valgrind pass with glib 2.46
2500 2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2502 * gst/rtsp-server/rtsp-media.c:
2503 rtsp-media: Take reference to media that will be prepared
2504 default_prepare() takes a transfer-none reference GstRTSPMedia object.
2505 Later on a g_idle_source_new() is created and a pointer to the media
2506 object is passed as user data. If the media is freed before the idle
2507 source is dispatched the media object pointer is invalid, but the idle
2508 source callback expects it to still be valid. To fix this a reference to
2509 the media object is taken when registering the source callback function
2510 and a corresponding release of the reference is done when the souce is
2512 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
2514 2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
2516 * examples/test-launch.c:
2517 * examples/test-mp4.c:
2518 * examples/test-ogg.c:
2519 * examples/test-record.c:
2520 * examples/test-uri.c:
2521 rtsp-server: Fix memory leaks when context parse fails
2522 When g_option_context_parse fails, context and error variables are not getting free'd
2523 which results in memory leaks. Free'ing the same.
2524 And replacing g_error_free with g_clear_error, which checks if the error being passed
2525 is not NULL and sets the variable to NULL on free'ing.
2526 https://bugzilla.gnome.org/show_bug.cgi?id=753863
2528 2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
2533 === release 1.6.0 ===
2535 2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
2541 * gst-rtsp-server.doap:
2544 === release 1.5.91 ===
2546 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
2552 * gst-rtsp-server.doap:
2555 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
2557 * docs/libs/gst-rtsp-server-sections.txt:
2558 * gst/rtsp-server/rtsp-stream.c:
2559 stream: fix docs for recently-added get/set_buffer_size API
2560 https://bugzilla.gnome.org/show_bug.cgi?id=749095
2562 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
2564 * gst/rtsp-server/rtsp-media.c:
2565 rtsp-media: Don't crash on encrypted RTX SDP
2566 In parse_keymgmt(), don't mutate the input string that's been passed
2567 as const, especially since we might need the original value again if
2568 the same key info applies to multiple streams (RTX, for example).
2569 https://bugzilla.gnome.org/show_bug.cgi?id=754753
2571 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
2573 * examples/test-mp4.c:
2574 test-mp4: Support filenames with spaces in them. Error out on too few arguments
2576 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
2578 * examples/test-record.c:
2579 test-record: Check parameter count and print out help
2580 If no launch pipeline was supplied, print out some help
2582 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
2584 * gst/rtsp-server/rtsp-media.c:
2585 * gst/rtsp-server/rtsp-stream.c:
2586 * gst/rtsp-server/rtsp-stream.h:
2587 rtsp-stream: Implement UDP buffer size setting.
2588 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
2590 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
2591 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
2593 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
2595 * gst/rtsp-server/rtsp-media.h:
2596 rtsp-media: Fix small typo causing gtk-doc to complain
2598 === release 1.5.90 ===
2600 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
2606 * gst-rtsp-server.doap:
2609 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
2611 * gst/rtsp-server/rtsp-media-factory.c:
2612 media-factory: get port number through gst_rtsp_url_get_port
2613 https://bugzilla.gnome.org/show_bug.cgi?id=753473
2615 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
2617 * tests/check/gst/media.c:
2618 media-test: Removing unnecessary assertion
2619 https://bugzilla.gnome.org/show_bug.cgi?id=753385
2621 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2623 * gst/rtsp-server/rtsp-server.c:
2624 Document that source keeps a ref on server until it's destroyed
2625 https://bugzilla.gnome.org/show_bug.cgi?id=749227
2627 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
2629 * tests/check/gst/media.c:
2630 media-test: Test for multiple dynamic payload
2631 https://bugzilla.gnome.org/show_bug.cgi?id=753385
2633 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
2635 * gst/rtsp-server/rtsp-media.c:
2636 media: Only add fakesink once per pipeline
2637 The intention is to prevent going PLAYING state before pads are created.
2638 If there was mutilple dynamic payload, it would leak few fakesink and
2639 actually prevent from ever reaching playing state.
2640 https://bugzilla.gnome.org/show_bug.cgi?id=753385
2642 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
2644 * gst/rtsp-server/rtsp-media.c:
2645 Revert "rtsp-media: Only add 1 fakesink per pipeline"
2646 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
2648 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
2650 * gst/rtsp-server/rtsp-media.c:
2651 rtsp-media: Only add 1 fakesink per pipeline
2652 There should be only one fakesink per pipeline, not per dynpay. This
2653 would lead to element naming clash.
2655 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
2657 * gst/rtsp-server/rtsp-media.c:
2658 rtsp-media: assertion error due to wrong condition check
2659 In media to caps function, reserved_keys array is being used for variable i,
2660 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
2661 changed it to variable j
2662 https://bugzilla.gnome.org/show_bug.cgi?id=753009
2664 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
2666 * gst/rtsp-server/rtsp-media.c:
2667 rtsp-media: Strip keys from the fmtp that we use internally in our caps
2668 Skip keys from the fmtp, which we already use ourselves for the
2669 caps. Some software is adding random things like clock-rate into
2670 the fmtp, and we would otherwise here set a string-typed clock-rate
2671 in the caps... and thus fail to create valid RTP caps
2672 https://bugzilla.gnome.org/show_bug.cgi?id=753009
2674 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2676 * gst/rtsp-server/rtsp-thread-pool.c:
2677 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
2678 https://bugzilla.gnome.org/show_bug.cgi?id=752640
2680 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
2683 Automatic update of common submodule
2684 From f74b2df to 9aed1d7
2686 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
2691 === release 1.5.2 ===
2693 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
2699 * gst-rtsp-server.doap:
2702 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
2704 * gst/rtsp-server/rtsp-client.c:
2705 * gst/rtsp-server/rtsp-client.h:
2706 * tests/check/gst/client.c:
2707 rtsp-client: allow application to decide what requirements are supported
2708 Add "check-requirements" signal and vfunc to allow application
2709 (and subclasses) to check the requirements.
2710 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
2711 https://bugzilla.gnome.org/show_bug.cgi?id=749417
2713 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
2716 Automatic update of common submodule
2717 From 6015d26 to f74b2df
2719 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
2721 * gst/rtsp-server/rtsp-media.c:
2722 rtsp-media: Always use real payloader when creating streams
2723 A bin that contains the real payloader might be used as payloader. In this
2724 case we have to get the real payloader for the various properties it provides.
2725 Example use cases for this are bins that payload some media and then have
2726 additional elements that add metadata or RTP extension headers to the stream.
2727 https://bugzilla.gnome.org/show_bug.cgi?id=750800
2729 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
2731 * examples/test-netclock-client.c:
2732 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
2734 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
2736 * examples/test-netclock-client.c:
2737 * examples/test-netclock.c:
2738 test-netclock: Use new ntp-time-source property on rtpbin
2739 Select the clock time to be used as NTP time source. This allows proper
2740 synchronization between receivers, independent of sharing base times, and just
2741 requires them to use the same clock.
2743 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
2745 * examples/test-netclock-client.c:
2746 * examples/test-netclock.c:
2747 test-netclock: Setting the same base time on sender and receiver is not necessary
2748 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
2750 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
2752 * gst/rtsp-server/rtsp-stream.c:
2753 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
2754 https://bugzilla.gnome.org/show_bug.cgi?id=750764
2756 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
2758 * docs/libs/gst-rtsp-server.types:
2759 docs: add missing types
2760 https://bugzilla.gnome.org/show_bug.cgi?id=750764
2762 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
2764 * docs/libs/gst-rtsp-server-sections.txt:
2765 docs: add missing apis
2766 https://bugzilla.gnome.org/show_bug.cgi?id=750764
2768 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
2770 * examples/test-netclock-client.c:
2771 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
2773 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2775 * docs/libs/gst-rtsp-server-sections.txt:
2776 * gst/rtsp-server/rtsp-auth.c:
2777 * gst/rtsp-server/rtsp-auth.h:
2778 GstRTSPAuth: Add client certificate authentication support
2779 https://bugzilla.gnome.org/show_bug.cgi?id=750471
2781 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
2783 * examples/test-netclock-client.c:
2784 test-netclock-client: Use new GstClock API to wait for clock synchronization
2786 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
2788 * examples/test-netclock-client.c:
2789 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
2790 A mainloop is needed to get glimagesink to display something on OSX, and
2791 the source-setup signal just makes things a little bit easier.
2793 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
2796 Automatic update of common submodule
2797 From d9a3353 to 6015d26
2799 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
2802 Automatic update of common submodule
2803 From d37af32 to d9a3353
2805 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
2808 Automatic update of common submodule
2809 From 21ba2e5 to d37af32
2811 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
2814 Automatic update of common submodule
2815 From c408583 to 21ba2e5
2817 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
2819 * docs/libs/Makefile.am:
2820 docs: remove variables that we define in the snippet from common
2821 This is syncing our Makefile.am with upstream gtkdoc.
2823 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
2826 Automatic update of common submodule
2827 From 44a3517 to c408583
2829 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
2834 === release 1.5.1 ===
2836 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
2842 * gst-rtsp-server.doap:
2845 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
2847 * gst/rtsp-server/rtsp-client.c:
2848 rtsp-client: No flush during Teardown.
2849 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
2850 backlog is empty it can happen that just a part of a message will be
2851 sent and rest is in backlog queue. If then flush during teardown
2852 just a part of message will be sent.This can lead to client miss
2853 teardown response since it expect to get the last part of message.
2854 The flushing during teardown was introduced to fix a deadlock that now
2855 is fixed more generally in handle_request by temporary setting backlog
2857 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
2859 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
2861 * tests/check/Makefile.am:
2862 tests: Use AM_TESTS_ENVIRONMENT
2863 Needed by the new automake test runner and the
2864 current version of the common submodule.
2866 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
2868 * gst/rtsp-server/rtsp-media.h:
2869 * gst/rtsp-server/rtsp-stream.h:
2870 rtsp-server: Use single-include rtsp header to make sure we get all definitions
2872 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
2874 * gst/rtsp-server/rtsp-media.c:
2875 rtsp-media: Mark some more functions static
2877 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
2879 * gst/rtsp-server/rtsp-media.c:
2880 rtsp-media: Only unblock the media in suspend() when actually changing the state
2881 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
2883 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
2885 * examples/test-video-rtx.c:
2886 examples: Use AVPF profile for the RTX example
2888 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
2890 * gst/rtsp-server/rtsp-sdp.c:
2891 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
2893 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
2895 * gst/rtsp-server/rtsp-stream.c:
2896 rtsp-stream: get valid clock-rate from last-sample
2897 clock-rate in last-sample's caps is integer, not unsigned.
2898 To get this value properly, variable needs to be type-casted to int.
2899 https://bugzilla.gnome.org/show_bug.cgi?id=747614
2901 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
2905 autogen.sh: only run autopoint if gettext requested in configure.ac
2906 Not just because there happens to be a po directory.
2907 https://bugzilla.gnome.org/show_bug.cgi?id=748058
2909 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
2912 Revert "configure.ac: uncomment gettext version setup"
2913 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
2914 We don't need a gettext setup here and there's no po
2915 directory either, so no reason why autopoint would be
2916 run in the first place.
2917 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
2919 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
2921 * examples/test-multicast.c:
2922 * examples/test-multicast2.c:
2923 * examples/test-sdp.c:
2924 * examples/test-video-rtx.c:
2925 * examples/test-video.c:
2926 * tests/test-cleanup.c:
2927 * tests/test-reuse.c:
2928 Fix timeout function signatures across tests and examples
2930 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
2932 * tests/check/Makefile.am:
2933 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
2934 Make sure the test environment is set up.
2935 https://bugzilla.gnome.org//show_bug.cgi?id=747624
2937 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
2940 configure: bump automake requirement to 1.14 and autoconf to 2.69
2941 This is only required for builds from git, people can still
2942 build tarballs if they only have older autotools.
2943 https://bugzilla.gnome.org//show_bug.cgi?id=747624
2945 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
2948 configure.ac: uncomment gettext version setup
2949 Fixes autogen.sh. It would run autopoint, which would complain
2950 that it could not find the gettext version in configure.ac.
2951 https://bugzilla.gnome.org/show_bug.cgi?id=748058
2953 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
2955 * examples/test-video-rtx.c:
2956 test-video-rtx: set exact payload type to PCMA payloader
2957 Setting wrong payload type causes failure to do retransmission through audio stream
2958 https://bugzilla.gnome.org/show_bug.cgi?id=747839
2960 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
2962 * gst/rtsp-server/rtsp-media.c:
2963 * gst/rtsp-server/rtsp-stream.c:
2964 * gst/rtsp-server/rtsp-stream.h:
2965 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
2966 Because of duplicated g_signal_connect for request-aux-sender signal,
2967 wrong stream pointer is passed to the signal handler.
2968 Instead of passing each stream, pass stream array and get the relevant stream.
2969 https://bugzilla.gnome.org/show_bug.cgi?id=747839
2971 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
2975 Update autogen.sh to latest version from common
2976 Fixes build after aclocal_check etc. helpers have been removed.
2978 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
2981 Automatic update of common submodule
2982 From bc76a8b to c8fb372
2984 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
2986 * gst/rtsp-server/rtsp-stream.c:
2987 rtsp-stream: Limit the queues to 1 buffer
2988 We only need them to be able to pre-roll, queueing up more data here
2989 is only going to harm latency and memory usage.
2991 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
2993 * gst/rtsp-server/rtsp-stream.c:
2994 rtsp-stream: Update comment and ASCII art to the latest code
2995 We have a queue in front of the udpsink too to prevent the pipeline from
2998 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
3000 * gst/rtsp-server/rtsp-stream.c:
3001 rtsp-media: Properly return first rtptime
3002 Instead we where returning first GstBuffer timestamp. This would result
3003 in clock skew and unwanted behaviour in RTSP playback.
3004 https://bugzilla.gnome.org/show_bug.cgi?id=746479
3006 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
3008 * gst/rtsp-server/rtsp-stream.c:
3009 rtsp-stream: Don't leave buffer mapped
3010 If the seq is NULL, the RTP buffer was left mapped. We should always
3013 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
3018 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
3020 * gst/rtsp-server/rtsp-media-factory.c:
3021 * tests/check/gst/client.c:
3022 Fix double semicolons
3024 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
3026 * gst/rtsp-server/rtsp-stream.c:
3027 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
3028 This gives more accurate values than asking the payloader. There might be
3029 queueing happening between the payloader and the sink.
3030 https://bugzilla.gnome.org/show_bug.cgi?id=745704
3032 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
3034 * gst/rtsp-server/rtsp-media.c:
3035 rtsp-media: Don't seek for PLAY if the position will not change
3036 https://bugzilla.gnome.org/show_bug.cgi?id=745704
3038 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
3040 * gst/rtsp-server/rtsp-media.c:
3041 rtsp-media: Don't include payload type in the caps for framesize
3042 When the sdp media attribute framesize are converted to caps
3043 the <payload> should not be included.
3044 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
3045 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
3047 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
3049 * gst/rtsp-server/rtsp-sdp.c:
3050 rtsp-sdp: add payload type to the sdp framesize attribute
3051 The sdp framesize attribute is desribed in RFC6064. It is specified
3052 for payloading of H263 and has the following form
3053 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
3054 should be added to the caps in a payloader and the <payload type> should
3055 be added by the rtsp-server.
3056 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
3058 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
3060 * examples/test-uri.c:
3061 examples: test-uri: fix tainted variable
3062 Insignificant but this keeps Coverity happy.
3065 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
3067 * examples/.gitignore:
3068 * examples/Makefile.am:
3069 * examples/test-netclock-client.c:
3070 * examples/test-netclock.c:
3071 examples: Add a simple example of network synch for live streams.
3072 An example server and client that works for synchronising live streams
3073 only - as it can't support pause/play.
3075 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
3077 * gst/rtsp-server/rtsp-media-factory.c:
3078 * gst/rtsp-server/rtsp-media-factory.h:
3079 rtsp-media-factory: Add functions to set/get the media gtype
3080 Allow specifying the GType of a GstRtspMedia subclass to create
3081 as a simpler way to get the factory to create a custom
3082 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
3084 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
3086 * gst/rtsp-server/rtsp-media.c:
3087 rtsp-media: fix double unlock in _get_buffer_size()
3088 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
3089 because of double g_mutex_unlock () usage.
3090 https://bugzilla.gnome.org/show_bug.cgi?id=745434
3092 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
3094 * gst/rtsp-server/rtsp-session-pool.c:
3095 * gst/rtsp-server/rtsp-session.c:
3096 * gst/rtsp-server/rtsp-session.h:
3097 rtsp-session: Use monotonic time for RTSP session timeout
3098 Changed RTSP session timeout handling to monotonic time
3099 and deprecating the API for current system time.
3100 This fixes timeouts when the system time changes.
3101 https://bugzilla.gnome.org/show_bug.cgi?id=743346
3103 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
3105 * gst/rtsp-server/rtsp-client.c:
3106 * gst/rtsp-server/rtsp-media.c:
3107 rtsp-client: Only error out in PLAY if seeking actually failed
3108 If the media was just not seekable, we continue from whatever position we are
3109 and let the client decide if that is what is wanted or not.
3110 Only if the actual seek failed, we can't really recover and should error out.
3112 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
3114 * gst/rtsp-server/rtsp-stream.c:
3115 rtsp-stream: Add necessary queues between tee and multiudpsink
3116 https://bugzilla.gnome.org/show_bug.cgi?id=744379
3118 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
3120 * gst/rtsp-server/rtsp-client.c:
3121 * gst/rtsp-server/rtsp-media.c:
3122 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
3123 Instead error out properly the same way as if the SEEKING query already
3126 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
3128 * gst/rtsp-server/rtsp-stream.h:
3129 rtsp-stream: minor code formatting fix
3131 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
3133 * gst/rtsp-server/rtsp-media.c:
3134 rtsp-media: fix logic for collect_streams
3135 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
3136 all streams it knows if it got any, and can check if the transport mode is OK.
3139 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
3141 * gst/rtsp-server/rtsp-media.c:
3142 rtsp-media: Don't set the transport mode based on what elements we find
3143 Just print a warning if the one that was set before disagrees with what
3144 elements we found. It must already be set to something before as this
3145 function is called after we received the SDP from ANNOUNCE in RECORD mode,
3146 and we would reject ANNOUNCE if the RECORD flag was not set.
3148 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
3150 * tests/check/gst/rtspserver.c:
3151 tests: rtspserver: rename shadowed variable
3152 We have two different 'sink' variables here,
3153 rename one of them for clarity.
3155 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
3157 * gst/rtsp-server/rtsp-client.c:
3158 rtsp-client: fix awkward if clause
3160 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
3162 * examples/test-uri.c:
3163 examples: test-uri: improve uri argument handling and accept file names
3164 Print an error if the argument passed is not a URI and can't
3165 be converted into one, or no arguments have been provided.
3167 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
3169 * examples/test-uri.c:
3170 examples: test-uri: don't remove mount point after 10 seconds
3171 It's very irritating when trying to test stuff repeatedly
3172 and serves no real purpose other than showing that it can
3175 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
3177 * examples/.gitignore:
3178 examples: add new test-record to .gitignore
3180 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
3182 * examples/test-record.c:
3183 * gst/rtsp-server/rtsp-client.c:
3184 * gst/rtsp-server/rtsp-media-factory.c:
3185 * gst/rtsp-server/rtsp-media-factory.h:
3186 * gst/rtsp-server/rtsp-media.c:
3187 * gst/rtsp-server/rtsp-media.h:
3188 * tests/check/gst/rtspserver.c:
3189 rtsp-media: Use flags to distinguish between PLAY and RECORD media
3191 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
3193 * examples/test-record.c:
3194 test-record: Set latency for playback-style example to 2s instead of 200ms
3196 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
3198 * tests/check/gst/rtspserver.c:
3199 tests: add some unit tests for ANNOUNCE and RECORD
3200 https://bugzilla.gnome.org/show_bug.cgi?id=743175
3202 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
3204 * gst/rtsp-server/rtsp-client.c:
3205 rtsp-client: fix a couple of leaks in handle_announce
3207 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
3209 * gst/rtsp-server/rtsp-media-factory.c:
3210 * gst/rtsp-server/rtsp-media-factory.h:
3211 * gst/rtsp-server/rtsp-media.c:
3212 * gst/rtsp-server/rtsp-media.h:
3213 rtsp-media: Expose latency setting for setting the rtpbin latency
3215 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
3217 * examples/test-record.c:
3218 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
3220 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
3222 * gst/rtsp-server/rtsp-stream.c:
3223 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
3225 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
3227 * examples/Makefile.am:
3228 * examples/test-record.c:
3229 * gst/rtsp-server/rtsp-client.c:
3230 * gst/rtsp-server/rtsp-client.h:
3231 * gst/rtsp-server/rtsp-media-factory.c:
3232 * gst/rtsp-server/rtsp-media-factory.h:
3233 * gst/rtsp-server/rtsp-media.c:
3234 * gst/rtsp-server/rtsp-media.h:
3235 * gst/rtsp-server/rtsp-session-media.c:
3236 * gst/rtsp-server/rtsp-stream.c:
3237 * gst/rtsp-server/rtsp-stream.h:
3238 Add initial support for RECORD
3239 We currently only support media that is RECORD or PLAY only, not both at once.
3240 https://bugzilla.gnome.org/show_bug.cgi?id=743175
3242 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
3244 * gst/rtsp-server/rtsp-stream.c:
3245 rtsp-stream: RTCP and RTP transport cache cookies seperated
3246 RTCP packets were not sent because the same tr_cache_cookie was used for
3247 both RTP and RTCP. So only one of the tr_cache lists were populated
3248 depending on which one was sent first. If the tr_cache list is not
3249 populated then no packets can be sent. Most often this happened to be
3250 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
3251 resulted in both the tr_cache_lists to be populated regardless of which
3253 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
3255 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
3257 * gst/rtsp-server/rtsp-stream.c:
3258 rtsp-stream: fix false compiler warning
3259 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
3261 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
3263 * gst/rtsp-server/rtsp-client.c:
3264 rtsp-client: log interleaved data received
3266 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
3268 * gst/rtsp-server/rtsp-client.c:
3269 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
3271 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
3273 * gst/rtsp-server/rtsp-client.c:
3274 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
3276 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
3278 * gst/rtsp-server/rtsp-client.c:
3279 rtsp-client: Use a random session ID in the SDP
3280 RFC4566 Section 5.2 says that it should make the username, session id,
3281 nettype, addrtype and unicast address tuple globally unique. Always using
3282 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
3283 Instead let's create a 64 bit random number, which at least brings us
3284 closer to the goal of global uniqueness.
3285 https://tools.ietf.org/html/rfc4566#section-5.2
3287 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
3289 * examples/test-launch.c:
3290 * examples/test-mp4.c:
3291 * examples/test-ogg.c:
3292 * examples/test-uri.c:
3293 examples: Don't call gst_init() and gst_get_option_group()
3294 The latter calls the former at the appropriate time.
3296 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
3298 * gst/rtsp-server/rtsp-client.c:
3299 rtsp-client: Drop trailing \0 of RTSP DATA messages
3300 We add a trailing \0 in GstRTSPConnection to make parsing of
3301 string message bodies easier (e.g. the SDP from DESCRIBE) but
3302 for actual data this means we have to drop it or otherwise
3303 create invalid data.
3305 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
3307 * gst/rtsp-server/rtsp-stream.c:
3308 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
3309 Fixes crash when two threads access handle_new_sample() at the same
3310 time, one for RTP, one for RTCP.
3311 Otherwise, when iterating over the transports cache, it might be modified by
3312 another thread at the same time if the transports cookie has changed.
3313 https://bugzilla.gnome.org/show_bug.cgi?id=742954
3315 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
3317 * gst/rtsp-server/rtsp-stream.c:
3318 rtsp-stream: Set format=TIME on our app sources for TCP
3320 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
3322 * gst/rtsp-server/rtsp-session-pool.c:
3323 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
3324 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
3325 RFC 2326 states that session IDs may consist of alphanumeric as well as
3326 the safe characters $-_.+ -- N.B. the percent character is not allowed.
3327 Previously the session ID was URI-escaped, this meant that any character
3328 which was not alphanumeric or any of the characters +-._~ would be
3329 percent encoded. While the RFC (surprisingly) mentions that linear white
3330 space in session IDs should be URI-escaped, it does not say anything
3331 about other characters. Moreover no white space is allowed in the
3332 session ID. Finally the percent character which is the result of
3333 URI-escaping is not allowed in a session ID.
3334 So there is no reason to do any URI-escaping, and now it is removed.
3335 https://bugzilla.gnome.org/show_bug.cgi?id=742869
3337 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
3340 Automatic update of common submodule
3341 From f2c6b95 to bc76a8b
3343 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
3346 Fix 'make check' from top-level directory
3348 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
3350 * examples/test-launch.c:
3351 * examples/test-mp4.c:
3352 * examples/test-ogg.c:
3353 * examples/test-uri.c:
3354 examples: Add command-line parsing and take a 'port' argument
3355 This allows users to run multiple servers on different ports for testing.
3356 Only done for examples that actually take arguments and hence are capable of
3357 outputting different streams for each instance on each port.
3358 https://bugzilla.gnome.org/show_bug.cgi?id=742115
3360 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
3362 * gst/rtsp-server/rtsp-client.c:
3363 * gst/rtsp-server/rtsp-client.h:
3364 rtsp-client: Add a send_message default signal handler
3365 This allows subclasses to easily hook into the response sending
3366 mechanism without doing everything from a signal, which seems
3367 awkward from subclasses.
3369 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
3372 Automatic update of common submodule
3373 From ef1ffdc to f2c6b95
3375 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3379 configure: add --disable-examples switch
3380 https://bugzilla.gnome.org/show_bug.cgi?id=741678
3382 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
3384 * examples/.gitignore:
3385 * examples/Makefile.am:
3386 * examples/test-video-rtx.c:
3387 examples: add a retransmisison example implementing RFC4588
3388 Currently only SSRC-multiplexed rtx streams are supported
3390 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
3392 * gst/rtsp-server/rtsp-stream.c:
3393 rtsp-stream: Fix some minor memory leaks
3395 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
3397 * gst/rtsp-server/rtsp-media.c:
3398 rtsp-media: Some minor cleanup
3400 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
3402 * gst/rtsp-server/rtsp-stream.c:
3403 rtsp-stream: Fix compiler warnings
3404 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
3405 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
3407 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
3408 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
3411 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
3413 * docs/libs/gst-rtsp-server-sections.txt:
3414 * gst/rtsp-server/rtsp-media-factory.c:
3415 * gst/rtsp-server/rtsp-media-factory.h:
3416 * gst/rtsp-server/rtsp-media.c:
3417 * gst/rtsp-server/rtsp-media.h:
3418 * gst/rtsp-server/rtsp-sdp.c:
3419 * gst/rtsp-server/rtsp-stream.c:
3420 * gst/rtsp-server/rtsp-stream.h:
3421 media: implement ssrc-multiplexed retransmission support
3422 based off RFC 4588 and the server-rtpaux example in -good
3424 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
3426 * gst/rtsp-server/rtsp-client.c:
3427 * gst/rtsp-server/rtsp-stream-transport.c:
3428 * gst/rtsp-server/rtsp-stream.c:
3429 rtsp: Ref transports in hash table.
3430 Also ref streams for transports.
3431 This solves a crash when reciving a rtcp after teardown but before
3433 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
3435 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
3438 Automatic update of common submodule
3439 From 7bb2bce to ef1ffdc
3441 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
3443 * gst/rtsp-server/rtsp-client.c:
3444 client: refactor cleanup of cached media
3446 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
3448 * tests/check/gst/client.c:
3450 The session leak is now fixed, lets remove those FIXME comments.
3452 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
3454 * tests/check/gst/rtspserver.c:
3455 tests: Test to setup two sessions on one connection
3456 https://bugzilla.gnome.org/show_bug.cgi?id=739112
3458 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
3460 * tests/check/gst/rtspserver.c:
3461 tests: Test setup with tcp transport
3462 https://bugzilla.gnome.org/show_bug.cgi?id=739112
3464 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
3466 * gst/rtsp-server/rtsp-client.c:
3467 client: Configure transport after creating session media
3468 The default implementation of configure_client_transport() in
3469 rtsp-client uses the session media when it chooses channels for
3470 interleaved traffic.
3471 https://bugzilla.gnome.org/show_bug.cgi?id=739112
3473 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
3475 * gst/rtsp-server/rtsp-client.c:
3476 * gst/rtsp-server/rtsp-session-media.c:
3477 client: Stop caching media in client when doing setup
3478 If the media has been managed by a session media, it should not be
3479 cached in the client any longer. The GstRTSPSessionMedia object is now
3480 responsible for unpreparing the GstRTSPMedia object using
3481 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
3483 https://bugzilla.gnome.org/show_bug.cgi?id=739112
3485 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
3487 * gst/rtsp-server/rtsp-stream.c:
3488 rtsp-stream: unref srtp decoder when leaving bin
3489 https://bugzilla.gnome.org/show_bug.cgi?id=739481
3491 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
3493 * gst/rtsp-server/rtsp-client.c:
3494 rtsp-client: mikey memory leaks
3495 https://bugzilla.gnome.org/show_bug.cgi?id=739383
3497 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
3500 Automatic update of common submodule
3501 From 84d06cd to 7bb2bce
3503 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
3506 Parallelise 'make check-valgrind'
3508 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
3511 Automatic update of common submodule
3512 From a8c8939 to 84d06cd
3514 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
3517 Automatic update of common submodule
3518 From 36388a1 to a8c8939
3520 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
3522 * gst/rtsp-server/rtsp-media.c:
3523 rtsp-media: deactivate media when shutting down from paused
3524 This was only done when going directly from playing.
3525 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
3527 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
3529 * gst/rtsp-server/rtsp-client.c:
3530 * gst/rtsp-server/rtsp-context.h:
3531 rtsp-client: add stream transport to context
3532 We add the stream transport to the context so we can get the configured
3533 client stream transport in the setup request signal.
3534 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
3536 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
3538 * gst/rtsp-server/rtsp-stream.c:
3539 stream: release lock even not all transports have been removed
3540 We don't want to keep the lock even we return FALSE because not all the
3541 transports have been removed. This could lead into a deadlock.
3542 https://bugzilla.gnome.org/show_bug.cgi?id=737797
3544 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
3546 * gst/rtsp-server/rtsp-sdp.c:
3547 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
3548 These were renamed in GstRTPBasePayload in 1.0
3550 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
3552 * gst/rtsp-server/rtsp-client.c:
3553 client: set session media to NULL without the lock
3554 We need to set session medias to NULL without the client lock otherwise
3555 we can end up in a deadlock if another thread is waiting for the lock
3556 and media unprepare is also waiting for that thread to end.
3557 https://bugzilla.gnome.org/show_bug.cgi?id=737690
3559 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
3561 * gst/rtsp-server/rtsp-media.c:
3562 rtsp-media: Set state to UNPREPARING in all cases
3564 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
3566 * gst/rtsp-server/rtsp-media.c:
3567 media: set state to unpreparing when unprepare is initiated
3568 https://bugzilla.gnome.org/show_bug.cgi?id=737675
3570 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
3572 * gst/rtsp-server/rtsp-client.c:
3573 rtsp-client: Remove backlog limit while processings requests
3574 If the backlog limit is kept two cases of deadlocks may be
3575 encountered when streaming over TCP. Without the backlog
3576 limit this deadlocks can not happen, at the expence of
3578 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
3580 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
3582 * gst/rtsp-server/rtsp-client.c:
3583 rtsp-client: do not free main context before rtsp watch
3584 https://bugzilla.gnome.org/show_bug.cgi?id=737110
3586 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
3588 * tests/check/gst/rtspserver.c:
3589 tests: Extend unit test timeout to accomodate for valgrind
3590 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
3592 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
3594 * gst/rtsp-server/rtsp-client.c:
3595 * gst/rtsp-server/rtsp-session.c:
3596 * gst/rtsp-server/rtsp-stream-transport.c:
3597 rtsp-*: Treat sending packets to clients as keepalive
3598 As long as gst-rtsp-server can successfully send RTP/RTCP data to
3599 clients then the client must be reading. This change makes the server
3600 timeout the connection if the client stops reading.
3601 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
3603 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
3605 * gst/rtsp-server/rtsp-client.c:
3606 rtsp-client: Allow backlog to grow while expiring session
3607 Allow the send backlog in the RTSP watch to grow to unlimited size while
3608 attempting to bring the media pipeline to NULL due to a session
3609 expiring. Without this change the appsink element cannot change state
3610 because it is blocked while rendering data in the new_sample callback.
3611 This callback will block until it has successfully put the data into the
3612 send backlog. There is a chance that the send backlog is full at this
3613 point which means that the callback may block for a long time, possibly
3614 forever. Therefore the media pipeline may also be prevented from
3615 changing state for a long time.
3616 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
3618 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
3620 * gst/rtsp-server/rtsp-client.c:
3621 rtsp-client: Make old compilers happy
3622 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
3623 Just in case that guint8 doesn't fit in a pointer. Just in case ...
3625 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
3627 * gst/rtsp-server/rtsp-client.c:
3628 client: raise the backlog limits before pausing
3629 We need to raise the backlog limits before pausing the pipeline or else
3630 the appsink might be blocking in the render method in wait_backlog() and
3631 we would deadlock waiting for paused.
3632 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
3634 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
3636 * gst/rtsp-server/rtsp-client.c:
3637 client: make define for the WATCH_BACKLOG
3638 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
3640 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
3642 * gst/rtsp-server/rtsp-client.c:
3643 client: simplify session transport handling
3644 link/unlink of the transport in a session was done to keep track of all
3645 TCP transports and to send RTP/RTCP data to the streams. We can simplify
3646 that by putting all the TCP transports in a hashtable indexed with the
3648 We also don't need to link/unlink the transports when we pause/resume
3649 the streams. The same effect is already achieved when we pause/play the
3650 media. Indeed, when we pause the media, the transport is removed from
3651 the media and the callbacks will not be called anymore.
3652 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
3654 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
3656 * gst/rtsp-server/rtsp-stream-transport.c:
3657 * gst/rtsp-server/rtsp-stream-transport.h:
3658 stream-transport: make method to handle received data
3659 Make a method to handle the data received on a channel. It sends the
3660 data to the stream of the transport on the RTP or RTCP pads based on
3663 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
3665 * examples/test-mp4.c:
3666 test: add example of dumping RTCP reports
3668 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
3670 * gst/rtsp-server/rtsp-media.c:
3671 * gst/rtsp-server/rtsp-stream.c:
3672 * gst/rtsp-server/rtsp-stream.h:
3673 rtsp-media: Make sure that sequence numbers are monotonic after pause
3674 The sequence number is not monotonic for RTP packets after pause. The
3675 reason is basepayloader generates a randon sequence number when the
3676 pipeline goes from ready to pause. With this fix generation of sequence
3677 number will be monotonic when going from pause to play request.
3678 https://bugzilla.gnome.org/show_bug.cgi?id=736017
3680 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
3682 * gst/rtsp-server/rtsp-client.c:
3683 rtsp-client: Protect saved clients watch with a mutex
3684 Fixes a crash when close() is called while merging clients
3685 in handle_tunnel(). In that case close() would destroy the
3686 watch while it is still being used in handle_tunnel().
3687 https://bugzilla.gnome.org/show_bug.cgi?id=735570
3689 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
3691 * gst/rtsp-server/rtsp-stream.c:
3692 rtsp-stream: Remove the multicast group udp sources when removing from the bin
3694 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
3696 * gst/rtsp-server/rtsp-media.c:
3697 * gst/rtsp-server/rtsp-stream.c:
3698 * gst/rtsp-server/rtsp-stream.h:
3699 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
3700 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
3701 seeking and will always continue counting the time. This leads to
3702 the NPT after a backwards seek to be something completely different
3703 to the actual seek position.
3704 https://bugzilla.gnome.org/show_bug.cgi?id=732644
3706 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
3708 * examples/test-appsrc.c:
3709 examples: fix another reference leak
3710 gst_rtsp_media_get_element() returns a new ref.
3712 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3714 * examples/test-appsrc.c:
3715 examples: unref element after usage
3716 gst_bin_get_by_name_recurse_up() returns an element
3717 reference that must be unreffed after usage.
3718 https://bugzilla.gnome.org/show_bug.cgi?id=734546
3720 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
3722 * gst/rtsp-server/rtsp-media.c:
3723 signals: Fix copy-pasto in target-state signal offset
3725 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
3729 Makefile: Add usage of build-checks step
3730 Allows building checks without running them
3732 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
3734 * gst/rtsp-server/rtsp-stream.c:
3735 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
3736 When a UDP multicast transport is used it is expected that the server listens
3737 for RTP and RTCP packets on the multicast group with the corresponding port.
3738 Without this we will never get RTCP packets from clients in multicast mode.
3739 https://bugzilla.gnome.org/show_bug.cgi?id=732238
3741 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
3746 === release 1.4.0 ===
3748 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
3754 * gst-rtsp-server.doap:
3757 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
3759 * gst/rtsp-server/rtsp-media.h:
3760 media: correct misspelled words in description
3761 https://bugzilla.gnome.org/show_bug.cgi?id=733244
3763 === release 1.3.91 ===
3765 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
3771 * gst-rtsp-server.doap:
3774 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
3776 * docs/libs/gst-rtsp-server-sections.txt:
3779 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
3781 * gst/rtsp-server/rtsp-server.c:
3782 server: implement client REMOVE filter
3784 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
3786 * gst/rtsp-server/rtsp-client.c:
3787 * gst/rtsp-server/rtsp-client.h:
3788 client: expose _close() method
3789 Expose a previously internal close method to close the client
3792 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
3794 * gst/rtsp-server/rtsp-session-pool.c:
3795 session-pool: signal session-removed outside of the lock
3796 Release the lock before emiting the session-removed signal.
3798 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
3800 * gst/rtsp-server/rtsp-client.c:
3801 * gst/rtsp-server/rtsp-server.c:
3802 * gst/rtsp-server/rtsp-session-pool.c:
3803 * gst/rtsp-server/rtsp-session.c:
3804 * gst/rtsp-server/rtsp-stream.c:
3805 filter: Release lock in filter functions
3806 Release the object lock before calling the filter functions. We need to
3807 keep a cookie to detect when the list changed during the filter
3808 callback. We also keep a hashtable to make sure we only call the filter
3809 function once for each object in case of concurrent modification.
3810 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
3812 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
3814 * gst/rtsp-server/rtsp-client.c:
3815 client: check if watch is set in handle_teardown()
3816 The unit tests run without a watch
3818 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
3820 * tests/check/gst/client.c:
3821 client tests: send teardown to cleanup session
3823 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
3825 * tests/check/gst/rtspserver.c:
3826 server tests: send teardown to cleanup session
3828 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
3830 * gst/rtsp-server/rtsp-client.c:
3831 client: keep ref to client for the session removed handler
3832 This extra ref will be dropped when all client sessions have been
3833 removed. A session is removed when a client sends teardown, closes its
3834 endpoint of the TCP connection or the sessions expires.
3835 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
3837 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
3839 * gst/rtsp-server/rtsp-client.c:
3840 * gst/rtsp-server/rtsp-session.c:
3841 * tests/check/gst/client.c:
3842 client: manage media in session as a last step
3843 Once we manage a media in a session, we can't unmanage it anymore
3844 without destroying it. Therefore, first check everything before we
3845 manage the media, otherwise if something is wrong we have no way to
3847 If we created a new session and something went wrong, remove the session
3848 again. Fixes a leak in the unit test.
3850 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
3852 * examples/test-mp4.c:
3853 * examples/test-ogg.c:
3854 examples: print 'stream ready at url' for mp4 and ogg example
3856 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
3858 * gst/rtsp-server/rtsp-client.c:
3859 * gst/rtsp-server/rtsp-sdp.c:
3860 rtsp: fix for MIKEY api change
3862 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
3864 * gst/rtsp-server/rtsp-client.c:
3865 client: free watch context only once
3866 The watch context is freed when the source is destroyed. Avoids
3867 a CRITICAL when we try to unref the context twice.
3869 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
3871 * gst/rtsp-server/rtsp-client.c:
3874 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
3876 * gst/rtsp-server/rtsp-client.c:
3877 client: protect sessions with lock
3878 Protect the list of sessions with the lock.
3879 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
3881 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
3883 * gst/rtsp-server/rtsp-client.c:
3884 Client: keep a ref to the session
3885 Don't just keep a weak ref to the session objects but use a hard ref. We
3886 will be notified when a session is removed from the pool (expired) with
3887 the new session-removed signal.
3888 Don't automatically close the RTSP connection when all the sessions of
3889 a client are removed, a client can continue to operate and it can create
3890 a new session if it wants. If you want to remove the client from the
3891 server, you have to use gst_rtsp_server_client_filter() now.
3892 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
3893 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
3895 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
3897 * gst/rtsp-server/rtsp-session-pool.c:
3898 * gst/rtsp-server/rtsp-session-pool.h:
3899 session-pool: add session-removed signal
3900 Add a signal to be notified when a session is removed from the pool.
3902 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
3904 * gst/rtsp-server/Makefile.am:
3905 * gst/rtsp-server/rtsp-server.h:
3906 Make rtsp-server.h a single-include header, use it for G-I
3907 https://bugzilla.gnome.org/show_bug.cgi?id=732411
3909 === release 1.3.90 ===
3911 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
3917 * gst-rtsp-server.doap:
3920 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
3922 * gst/rtsp-server/rtsp-stream.c:
3923 stream: crypto can be NULL
3925 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
3927 * gst/rtsp-server/rtsp-client.c:
3928 * gst/rtsp-server/rtsp-media.c:
3929 * gst/rtsp-server/rtsp-mount-points.c:
3930 introspection: add missing allow-none annotations
3931 https://bugzilla.gnome.org/show_bug.cgi?id=730952
3933 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
3935 * gst/rtsp-server/rtsp-address-pool.c:
3936 * gst/rtsp-server/rtsp-media.c:
3937 * gst/rtsp-server/rtsp-session-media.c:
3938 * gst/rtsp-server/rtsp-session-pool.c:
3939 * gst/rtsp-server/rtsp-stream-transport.c:
3940 * gst/rtsp-server/rtsp-stream.c:
3941 * gst/rtsp-server/rtsp-token.c:
3942 introspection: add (nullable) annotations to return values
3943 https://bugzilla.gnome.org/show_bug.cgi?id=730952
3945 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
3947 * gst/rtsp-server/rtsp-client.c:
3948 * gst/rtsp-server/rtsp-stream.c:
3949 gi: improve annotations
3950 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
3952 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
3954 * gst/rtsp-server/rtsp-client.c:
3955 * gst/rtsp-server/rtsp-media-factory.c:
3956 * gst/rtsp-server/rtsp-media.c:
3957 * gst/rtsp-server/rtsp-server.c:
3958 signals: use generic marshal function
3959 Use the generic C marshal function.
3960 Use more explicit type instead of G_TYPE_POINTER
3962 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
3964 * gst/rtsp-server/rtsp-context.h:
3965 context: add type macro
3967 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
3969 * gst/rtsp-server/rtsp-client.c:
3970 * gst/rtsp-server/rtsp-sdp.c:
3971 * gst/rtsp-server/rtsp-sdp.h:
3972 sdp: hide key length defines
3973 They don't have a namespace.
3975 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
3980 === release 1.3.3 ===
3982 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
3988 * gst-rtsp-server.doap:
3991 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
3993 * gst/rtsp-server/rtsp-client.c:
3994 * gst/rtsp-server/rtsp-sdp.c:
3995 * gst/rtsp-server/rtsp-sdp.h:
3996 mikey: add different key length parameters
3997 Add encryption and authentication key length parameters to MIKEY. For
3998 the encoders, the key lengths are obtained from the cipher and auth
3999 algorithms set in the caps. For the decoders, they are obtained while
4000 parsing the key management from the client.
4001 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
4003 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
4005 * tests/check/gst/stream.c:
4006 stream tests: Make sure we get right multicast address from stream
4007 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
4009 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
4011 * gst/rtsp-server/rtsp-client.c:
4012 client: ref the context until rtsp watch is alive
4013 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
4015 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
4017 * gst/rtsp-server/rtsp-client.c:
4018 client: Destroy the rtsp watch after connection close
4020 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
4022 * gst/rtsp-server/rtsp-media.c:
4023 media: fix confusing comment
4025 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
4027 * gst/rtsp-server/rtsp-session.c:
4028 rtsp-session: Timeout in header.
4029 Adding the possbilty to always have timout in header.
4030 This is configurabe with setting "timeout-always-visible".
4031 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
4033 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
4038 === release 1.3.2 ===
4040 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
4047 * gst-rtsp-server.doap:
4050 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
4053 Automatic update of common submodule
4054 From 211fa5f to 1f5d3c3
4056 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
4058 * gst/rtsp-server/rtsp-client.c:
4059 client: store TCP ports in transport
4060 Store the TCP ports in the transport when we are doing RTSP over TCP.
4061 This way, we can easily get to the ports from the transport.
4062 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
4064 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4066 * gst/rtsp-server/rtsp-stream.c:
4067 stream: add signals for new RTP/RTCP encoders
4068 New signals to allow the user to configure the dynamically created
4070 https://bugzilla.gnome.org/show_bug.cgi?id=730228
4072 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
4074 * gst/rtsp-server/rtsp-media.c:
4075 * gst/rtsp-server/rtsp-media.h:
4076 media: Make suspend()/unsuspend() virtual
4077 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
4079 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4081 * gst/rtsp-server/rtsp-client.c:
4082 client: fix send-message signal marshaller
4083 Use generic marshalling for the send-message signal. It has
4084 two POINTER arguments, not just one.
4085 https://bugzilla.gnome.org/show_bug.cgi?id=729900
4087 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
4089 * tests/check/gst/media.c:
4090 tests: add and remove pads only once
4091 In this test we simulate a dynamic pad by watching the caps event.
4092 Because of renegotiation in the base payloader now, this caps is sent
4093 multiple times but we can only deal with 1 invocation, use a variable to
4094 only 'add and remove' the pad once.
4096 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
4098 * tests/check/gst/rtspserver.c:
4099 tests: add unit test for correct handling of Require headers
4100 https://bugzilla.gnome.org/show_bug.cgi?id=729426
4102 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
4104 * gst/rtsp-server/rtsp-client.c:
4105 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
4106 Servers must handle Require headers and must report a failure
4107 if they don't handle any of the Required options, see RFC 2326,
4108 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
4109 https://bugzilla.gnome.org/show_bug.cgi?id=729426
4111 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
4116 === release 1.3.1 ===
4118 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
4124 * gst-rtsp-server.doap:
4127 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
4130 Automatic update of common submodule
4131 From bcb1518 to 211fa5f
4133 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
4138 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
4140 * tests/check/gst/sessionmedia.c:
4141 tests: fix memory leak in sessionmedia unit test
4143 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
4145 * gst/rtsp-server/rtsp-client.c:
4146 client: emit a signal before sending a message
4147 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
4149 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
4151 * gst/rtsp-server/rtsp-client.c:
4152 client: pass context to send_message
4153 Pass the current context to send_message, we will need it later.
4155 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
4157 * gst/rtsp-server/rtsp-client.c:
4158 client: fix typo in comment
4160 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
4162 * gst/rtsp-server/rtsp-media.c:
4163 media: Do not stop thread twice if default_prepare() fails
4165 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
4167 * gst/rtsp-server/rtsp-client.c:
4168 client: set the watch to flushing before going to NULL
4169 First set the watch to flushing so that we unblock any current and
4170 future attempt to send data on the watch, Then set the pipeline to
4172 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
4174 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
4176 * gst/rtsp-server/rtsp-session-pool.c:
4177 * tests/check/gst/sessionpool.c:
4178 rtsp-session-pool: Fixes annotation
4179 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
4180 in the sessionpool test.
4181 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
4183 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
4185 * gst/rtsp-server/rtsp-media.c:
4186 * gst/rtsp-server/rtsp-media.h:
4187 media: make media_prepare virtual
4188 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
4190 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
4192 * gst/rtsp-server/rtsp-media.c:
4193 * tests/check/gst/media.c:
4194 media: stop the thread in more error cases
4196 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
4198 * gst/rtsp-server/rtsp-media.c:
4199 * tests/check/gst/media.c:
4200 media: allow NULL as the thread
4201 Use the default context whan passing a NULL thread.
4203 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
4205 * gst/rtsp-server/rtsp-client.c:
4206 rtsp-client: indent cleanup
4207 Coverity was moaning about unreachable code, and I think it was just
4208 confused by { being before the label. We'll see if it pops up again.
4211 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
4213 * gst/rtsp-server/rtsp-client.c:
4214 * gst/rtsp-server/rtsp-media.c:
4215 client: Add drop-backlog property
4216 When we have too many messages queued for a client (currently hardcoded
4217 to 100) we overflow and drop the messages. Add a drop-backlog property
4218 to control this behaviour. Setting this property to FALSE will retry
4219 to send the messages to the client by waiting for more room in the
4221 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
4223 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
4225 * gst/rtsp-server/rtsp-client.c:
4226 client: support for POST before GET when setting up a tunnel
4228 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
4230 * gst/rtsp-server/rtsp-client.c:
4231 client: remove watch of the second client after http tunnel setup
4232 The second client will be freed after the HTTP tunnel has been set up.
4233 Make sure it's RTSP watch is never dispatched again.
4234 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
4236 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
4238 * gst/rtsp-server/rtsp-media.c:
4239 * tests/check/gst/media.c:
4240 media: Make media_prepare() fail if port allocation fails
4241 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
4243 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
4245 * tests/check/gst/media.c:
4246 media test: cleanup the thread pool in tests
4248 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
4250 * gst/rtsp-server/rtsp-media.c:
4251 * tests/check/gst/media.c:
4252 rtsp-media: Unblock blocked streams in unprepare
4253 The streams will be blocked when a live media is prepared.
4254 The streams should be unblocked in gst_rtsp_media_unprepare.
4255 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
4257 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
4259 * gst/rtsp-server/rtsp-media.c:
4260 media: release the state lock when going to NULL
4261 Set our state to UNPREPARING and release the state-lock before
4262 setting the pipeline to the NULL state. This way, any pad-added
4263 callback will be able to take the state-lock and check that we are now
4264 unpreparing instead of deadlocking.
4265 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
4267 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
4269 * gst/rtsp-server/rtsp-media.c:
4270 media: protect status with lock
4271 Make sure we only update the status with the lock.
4273 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
4275 * gst/rtsp-server/rtsp-client.c:
4276 * gst/rtsp-server/rtsp-sdp.c:
4277 rtsp: update for MIKEY API changes
4279 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
4281 * gst/rtsp-server/rtsp-client.c:
4282 client: parse the mikey response from the client
4283 Parse the mikey response from the client and update the policy for
4286 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
4288 * gst/rtsp-server/rtsp-stream.c:
4289 * gst/rtsp-server/rtsp-stream.h:
4290 stream: add method to set crypto info
4291 Make a method to configure the crypto information of a stream.
4292 Set udpsrc in READY instead of PAUSED so that we can configure caps
4295 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
4297 * gst/rtsp-server/rtsp-client.c:
4298 client: cleanup error paths
4300 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
4302 * gst/rtsp-server/rtsp-media.c:
4305 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
4307 * examples/test-video.c:
4308 test: enable SRTP only on RTSPS
4309 We only want to enable SRTP when doing rtsp over TLS so that we can
4310 exchange the keys in a secure way.
4312 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
4314 * examples/test-video.c:
4315 test: print an error on failure
4317 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
4320 * examples/test-video.c:
4321 * gst/rtsp-server/rtsp-sdp.c:
4322 * gst/rtsp-server/rtsp-stream.c:
4323 * tests/check/Makefile.am:
4324 stream: add SRTP support
4325 Install srtp encoder and decoder elements in rtpbin
4328 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4330 * tests/check/Makefile.am:
4331 * tests/check/gst/sessionpool.c:
4332 tests: Add unit tests for sessionpool
4333 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
4335 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4337 * tests/check/gst/threadpool.c:
4338 tests: Improve code coverage of rtsp-threadpool tests
4339 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
4341 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4343 * tests/check/gst/sessionmedia.c:
4344 tests: Improve code coverage for rtsp-session-media
4345 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
4347 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4349 gobject-introspection: Add annotations to support language bindings
4350 In addition a few cosmetic changes:
4351 * Adjust the order of arguments
4352 * Fix typo: occured -> occurred
4353 * Fix indentation after Return:-clauses
4354 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
4356 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4358 * gst/rtsp-server/rtsp-stream.c:
4359 rtsp-stream: Don't mix IPv4 and IPv6 addresses
4360 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
4362 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
4364 * gst/rtsp-server/rtsp-stream.c:
4365 stream: take caps after the session manager
4366 Take the caps for the SDP after they leave the rtpbin so that we can
4367 also get the properties added by rtpbin elements.
4369 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
4371 * gst/rtsp-server/rtsp-stream.c:
4372 stream: release lock while pushing out packets
4373 Keep a cache of the transports and use this to iterate the transport
4374 while pushing packets. This allows us to release the lock early.
4375 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
4377 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
4379 * gst/rtsp-server/rtsp-client.c:
4380 * gst/rtsp-server/rtsp-client.h:
4381 rtsp-client: vmethod for modifying tunnel GET response
4382 Add a vmethod tunnel_http_response where the response to the HTTP GET
4383 for tunneled connections can be modified.
4384 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
4386 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
4388 * gst/rtsp-server/rtsp-sdp.c:
4389 sdp: make 1 media line per profile
4390 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
4391 line in the SDP for each profile. The client is then supposed to pick
4392 one of the profiles in the SETUP request. Because the m= lines have the
4393 same pt, the client also knows that only 1 option is possible.
4395 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
4397 * gst/rtsp-server/rtsp-media-factory.c:
4398 * gst/rtsp-server/rtsp-media-factory.h:
4399 * gst/rtsp-server/rtsp-media.c:
4400 factory: add profile property and pass to media and streams
4402 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
4404 * examples/test-multicast.c:
4405 * gst/rtsp-server/rtsp-sdp.c:
4406 sdp: pass multicast connection for multicast-only stream
4407 Pass the multicast address of the stream in the connection info in the
4408 SDP so that clients try a multicast connection first.
4409 Only allow multicast connections in the test-multicast example. Also
4410 increase the TTL a little.
4412 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4415 .gitignore: Ignore gcov intermediate files
4416 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
4418 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
4420 * gst/rtsp-server/rtsp-stream.c:
4421 stream: release some locks in error cases
4423 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4425 docs: Enable and fix gtk-doc warnings
4426 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
4427 * addresspool/mediafactory: Add missing annotation colon
4428 * stream: Annotate return value
4429 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
4431 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
4434 Automatic update of common submodule
4435 From fe1672e to bcb1518
4437 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
4440 Automatic update of common submodule
4441 From 1a07da9 to fe1672e
4443 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
4445 * examples/Makefile.am:
4446 examples: use LDADD for libs instead of LDFLAGS
4448 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
4451 configure: make sure releases are in .doap file
4453 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
4455 * examples/test-cgroups.c:
4456 examples: test-cgroups: don't put code with side effects into g_assert()
4457 The g_assert() might get compiled out with the right
4458 compiler/preprocessor flags.
4460 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
4462 * examples/.gitignore:
4463 examples: add cgroup test binary to .gitignore
4465 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
4467 * examples/test-cgroups.c:
4468 examples: fix cgroup test build
4469 Fixes build failure caused by compiler warning:
4470 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
4472 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
4475 .gitignore: ignore temp files created in the course of 'make check'
4477 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
4479 * gst/rtsp-server/rtsp-media.c:
4480 rtsp-media: don't loose frames handling new PLAY request
4481 If client supplied a range check if the range specifies the start point.
4482 If not, then do an accurate seek to the current position. If a start
4483 point was specified do do a key unit seek to make sure the streaming
4484 starts with decodeable frames.
4485 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
4487 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
4489 * gst/rtsp-server/rtsp-media.c:
4490 Revert "media: only flush when setting a new start position"
4491 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
4492 We need to do the flush in all cases, demuxer block currently for
4495 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
4497 * gst/rtsp-server/rtsp-media.c:
4498 media: only flush when setting a new start position
4499 Only flush the pipeline when we change the start position with
4501 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
4503 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
4505 * gst/rtsp-server/rtsp-stream.c:
4506 stream: set ttl-mc before adding the socket
4507 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
4508 never be set on socket.
4509 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
4511 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
4513 * gst/rtsp-server/rtsp-media.c:
4514 media: stop thread if media is already prepared
4515 in gst_rtsp_media_prepare() the thread is not used if media is already
4516 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
4518 https://bugzilla.gnome.org/show_bug.cgi?id=724182
4520 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
4523 build: Ship gst-rtsp-server.doap file
4525 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
4527 * tests/check/gst/rtspserver.c:
4528 tests: Fix another compiler warning with gcc
4530 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
4532 * gst/rtsp-server/rtsp-client.c:
4533 * gst/rtsp-server/rtsp-mount-points.c:
4534 * gst/rtsp-server/rtsp-stream.c:
4535 * tests/check/gst/client.c:
4536 rtsp-server: Fix lots of compiler warnings with clang
4538 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
4541 * gst-rtsp-server.doap:
4542 * tests/Makefile.am:
4543 configure: Synchronise with the configure scripts of the other modules
4545 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
4548 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
4550 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
4552 * gst/rtsp-server/rtsp-media.c:
4553 * gst/rtsp-server/rtsp-stream.c:
4554 Revert "rtsp-server: support build against last stable release"
4555 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
4556 Let us require 1.2.3 now, which is going to be released in a few
4559 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
4561 * gst/rtsp-server/rtsp-session-media.c:
4562 * gst/rtsp-server/rtsp-stream-transport.c:
4563 session: improve RTP-Info
4564 Ignore streams that can't generate RTP-Info instead of failing.
4565 Don't return the empty string when all streams are unconfigured but
4566 return NULL so that we don't generate and empty RTP-Info header.
4567 Improve docs a little.
4569 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
4571 * gst/rtsp-server/rtsp-session-media.c:
4572 Don't free rtpinfo GString when it is NULL
4573 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
4575 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
4577 * gst/rtsp-server/rtsp-media.c:
4578 media: only set keyframe flag when modifying start
4579 Only set the keyframe flag when we modify the start position. The
4580 keyframe flag should probably be ignored when no change is requested but
4581 until we can claim this is all documented properly and all demuxer
4582 implement this, avoid setting the flag.
4583 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
4585 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
4587 * gst/rtsp-server/rtsp-thread-pool.c:
4588 thread-pool: Unref source after mainloop has quit to avoid races in GLib
4589 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
4591 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
4593 * gst/rtsp-server/rtsp-stream.c:
4594 stream: handle NULL seqnum and rtptime arguments
4596 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
4598 * gst/rtsp-server/rtsp-thread-pool.c:
4599 * tests/check/gst/threadpool.c:
4600 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
4601 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
4603 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
4605 * gst/rtsp-server/rtsp-stream.c:
4606 stream: add fallback for missing stats property
4607 Use a fallback when the payloader does not have a stats property
4608 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
4610 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
4613 Automatic update of common submodule
4614 From f7bc1c3 to 1a07da9
4616 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
4618 * gst/rtsp-server/rtsp-stream.c:
4619 stream: don't leak stats structure
4620 Don't leak the stats structure and deal with NULL stats.
4622 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
4624 * gst/rtsp-server/rtsp-stream.c:
4625 stream: Get rtpinfo properties atomically from payloader
4626 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
4628 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
4630 * gst/rtsp-server/rtsp-media.c:
4631 media: refactor state change functions and signals
4632 Make functions to set the target state and the pipeline state and emit
4633 the signals from those functions.
4635 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
4637 * gst/rtsp-server/rtsp-media.c:
4638 * gst/rtsp-server/rtsp-media.h:
4639 media: add signal to notify of pending state changes
4641 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
4643 * gst/rtsp-server/rtsp-media.c:
4644 * gst/rtsp-server/rtsp-stream.c:
4645 rtsp-server: support build against last stable release
4646 Until 1.2.3 is out with the new get_type function and we
4649 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
4651 * gst/rtsp-server/rtsp-stream.c:
4652 stream: fix compilation
4654 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
4656 * gst/rtsp-server/rtsp-media.c:
4657 * gst/rtsp-server/rtsp-media.h:
4658 * gst/rtsp-server/rtsp-stream.c:
4659 * gst/rtsp-server/rtsp-stream.h:
4660 stream: add property to configure profiles
4662 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
4664 * gst/rtsp-server/rtsp-client.c:
4665 client: let stream check supported transport
4666 Delegate the check if a transport is allowed to the stream.
4667 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
4669 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
4671 * gst/rtsp-server/rtsp-stream.c:
4672 * gst/rtsp-server/rtsp-stream.h:
4673 stream: add method to check supported transport
4674 Add a method to check if a transport is supported
4676 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
4679 configure.ac: Only check for gstreamer-check, not check
4680 We include check in gstreamer-check since quite some time now.
4682 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
4684 * gst/rtsp-server/rtsp-session-media.c:
4685 * gst/rtsp-server/rtsp-stream-transport.c:
4686 * gst/rtsp-server/rtsp-stream.c:
4687 * gst/rtsp-server/rtsp-stream.h:
4688 stream: return clock-rate from get_rtpinfo
4689 And use it to correct the rtptime to the requested start-time.
4690 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
4692 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
4694 * gst/rtsp-server/rtsp-session-media.c:
4695 * gst/rtsp-server/rtsp-stream-transport.c:
4696 * gst/rtsp-server/rtsp-stream-transport.h:
4697 session-media: calculate start-time
4699 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
4701 * gst/rtsp-server/rtsp-stream-transport.c:
4702 * gst/rtsp-server/rtsp-stream.c:
4703 * gst/rtsp-server/rtsp-stream.h:
4704 stream: also return the running-time
4705 Return the running-time in the rtpinfo as well.
4707 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
4709 * gst/rtsp-server/rtsp-client.c:
4710 * gst/rtsp-server/rtsp-session-media.c:
4711 * gst/rtsp-server/rtsp-session-media.h:
4712 * gst/rtsp-server/rtsp-stream-transport.c:
4713 * gst/rtsp-server/rtsp-stream-transport.h:
4714 session-media: let the session-media make the RTPInfo
4715 Add method to create the RTPInfo for a stream-transport.
4716 Add method to create the RTPInfo for all stream-transports in a
4718 Use the session-media RTPInfo code in client. This allows us to refactor
4719 another method to link the TCP callbacks.
4721 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
4723 mount-points: sort sequence before g_sequence_lookup
4724 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
4725 sort sequence if dirty, otherwise lookup will fail.
4726 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
4728 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
4731 configure: rename package from gst-rtsp to gst-rtsp-server
4732 To match git module name and avoid confusion with the
4733 rtsp lib in gst-plugins-base and rtsp plugin in -good.
4735 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
4738 configure: bump core/base/good requirement to 1.2.0
4739 Bump to released stable version and make implicit
4740 requirements explicit.
4742 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
4747 Fix broken gettext setup which is not used anyway
4749 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
4752 Automatic update of common submodule
4753 From dbedaa0 to d48bed3
4755 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
4757 * gst/rtsp-server/rtsp-client.c:
4758 * gst/rtsp-server/rtsp-media.c:
4759 * gst/rtsp-server/rtsp-media.h:
4760 media: add setup_sdp vmethod
4761 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
4762 gst_rtsp_media_setup_sdp.
4763 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
4765 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
4767 * gst/rtsp-server/rtsp-stream.c:
4768 rtsp-stream: Check return value of sscanf
4769 streamid is only valid if sscanf matched something.
4771 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
4773 * gst/rtsp-server/rtsp-client.c:
4774 rtsp-client: Fix iteration
4775 Wouldn't even enter the code block otherwise (i++ was used as the check
4776 and not the postfix).
4778 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
4780 * gst/rtsp-server/rtsp-client.c:
4781 * gst/rtsp-server/rtsp-client.h:
4782 client: add vmethod to configure media and streams
4783 Implement a vmethod that can be used to configure the media and the
4784 streams based on the current context. Handle the blocksize handling in
4785 the default handler.
4786 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
4788 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
4791 Make git ignore more unit test binaries
4793 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
4795 * gst/rtsp-server/rtsp-address-pool.h:
4796 * gst/rtsp-server/rtsp-auth.h:
4797 * gst/rtsp-server/rtsp-client.h:
4798 * gst/rtsp-server/rtsp-context.h:
4799 * gst/rtsp-server/rtsp-media-factory-uri.h:
4800 * gst/rtsp-server/rtsp-media-factory.h:
4801 * gst/rtsp-server/rtsp-media.h:
4802 * gst/rtsp-server/rtsp-mount-points.h:
4803 * gst/rtsp-server/rtsp-server.h:
4804 * gst/rtsp-server/rtsp-session-media.h:
4805 * gst/rtsp-server/rtsp-session-pool.h:
4806 * gst/rtsp-server/rtsp-session.h:
4807 * gst/rtsp-server/rtsp-stream-transport.h:
4808 * gst/rtsp-server/rtsp-stream.h:
4809 * gst/rtsp-server/rtsp-thread-pool.h:
4810 * gst/rtsp-server/rtsp-token.h:
4811 rtsp-server: add padding to many public structures
4812 Not mini objects though, since they are not subclassable
4813 anyway, nor kept on the stack or inlined in a structure.
4815 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
4817 media: add new create_rtpbin vmethod
4818 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
4819 https://bugzilla.gnome.org/show_bug.cgi?id=719734
4821 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
4823 * tests/check/gst/media.c:
4824 tests: fix memory leak, free test's thread pool
4825 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
4827 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
4829 * gst/rtsp-server/rtsp-stream-transport.c:
4830 stream-transport: free url in finalize
4832 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
4834 * gst/rtsp-server/rtsp-media.c:
4835 media: also do state change in suspended state
4837 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
4839 * gst/rtsp-server/rtsp-client.c:
4840 * gst/rtsp-server/rtsp-media.c:
4841 media: also handle prepare and range in suspended state
4842 When we are suspended, we are already prepared.
4843 We can get the range in the suspended state.
4845 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
4847 * tests/check/Makefile.am:
4848 * tests/check/gst/sessionmedia.c:
4849 check: add test for uri in setup
4850 Added unit tests for the new functionality in GstRTSPStreamTransport.
4851 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
4853 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
4855 * gst/rtsp-server/rtsp-client.c:
4856 client: store setup uri and use in PLAY response
4857 Store the uri used when doing the setup and use that in the PLAY
4859 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
4861 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
4863 * gst/rtsp-server/rtsp-stream-transport.c:
4864 * gst/rtsp-server/rtsp-stream-transport.h:
4865 stream-transport: add method to get/set url
4867 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
4869 * gst/rtsp-server/rtsp-client.c:
4870 client: suspend after SDP and unsuspend before PLAYING
4871 Based on patches by Ognyan Tonchev <ognyan@axis.com>
4872 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
4874 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
4876 * gst/rtsp-server/rtsp-media-factory.c:
4877 * gst/rtsp-server/rtsp-media-factory.h:
4878 * gst/rtsp-server/rtsp-media.c:
4879 * gst/rtsp-server/rtsp-media.h:
4880 * gst/rtsp-server/rtsp-session-media.c:
4881 * gst/rtsp-server/rtsp-session.c:
4882 * tests/check/gst/media.c:
4883 * tests/check/gst/mediafactory.c:
4884 media: add suspend modes
4885 Add support for different suspend modes. The stream is suspended right after
4886 producing the SDP and after PAUSE. Different suspend modes are available that
4887 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
4888 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
4889 state and RESET will bring the pipeline to the NULL state.
4890 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
4891 this means that the pipeline needs to be prerolled again.
4892 Base on patches by Ognyan Tonchev <ognyan@axis.com>
4893 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
4895 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
4897 * gst/rtsp-server/rtsp-media.c:
4898 media: start live streams in blocked state
4899 Start live streams in the blocked state and make them preroll using the
4900 messages. This ensure that no data is played by the sink until we explicitly
4901 unblock the stream right before going to PLAYING.
4902 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
4904 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
4906 * gst/rtsp-server/rtsp-media.c:
4907 media: refactor starting and waiting for preroll
4908 Based on patches from Ognyan Tonchev <ognyan@axis.com>
4909 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
4911 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
4913 * gst/rtsp-server/rtsp-stream.c:
4914 * gst/rtsp-server/rtsp-stream.h:
4915 stream: add API to block streams
4916 Add an API to block on the streams and make it post a message.
4917 Based on patch by Ognyan Tonchev <ognyan@axis.com>
4918 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
4920 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
4922 * docs/libs/Makefile.am:
4923 docs: Specify the override file
4924 Even if it's empty (for now) it avoids make distcheck complaining
4926 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
4928 * gst/rtsp-server/rtsp-media.c:
4929 media: move default implementations to where they are used
4931 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
4933 * gst/rtsp-server/rtsp-media.c:
4934 media: take the right lock in gst_rtsp_media_set_pipeline_state()
4935 We need to take the state_lock when calling this method.
4937 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
4939 * gst/rtsp-server/rtsp-media.c:
4940 media: handle add-added on non-bins too
4941 Handle dynamic payloaders that are not bins, as used in the unit-test.
4943 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4945 * gst/rtsp-server/rtsp-media-factory.c:
4946 * gst/rtsp-server/rtsp-media-factory.h:
4947 * gst/rtsp-server/rtsp-media.c:
4948 rtsp-media/-factory: Fix request pad name comments
4949 These must be escaped for gtk-doc to parse the comments without warnings.
4951 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
4953 rtsp-media: remove transports if media is in error status
4954 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
4955 trying to change to GST_STATE_NULL and media is in error status, we
4956 remove all transports.
4957 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
4959 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
4961 * gst/rtsp-server/rtsp-media.c:
4962 rtsp-media: use element metadata to find payloader
4963 Use the element metadata to find the payloader instead of checking
4965 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
4967 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
4969 rtsp-stream: add getter for payload type
4970 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
4971 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
4972 element and create the stream with this one instead of the dynpay%d
4974 https://bugzilla.gnome.org/show_bug.cgi?id=712396
4976 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4978 * gst/rtsp-server/rtsp-client.c:
4979 * gst/rtsp-server/rtsp-context.h:
4980 * gst/rtsp-server/rtsp-media.c:
4981 * gst/rtsp-server/rtsp-mount-points.c:
4982 * gst/rtsp-server/rtsp-server.c:
4983 * gst/rtsp-server/rtsp-token.c:
4984 rtsp-*: Refer to NULL as a constant in comments
4986 https://bugzilla.gnome.org/show_bug.cgi?id=714988
4988 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4990 rtsp-*: Fix type name typos in comments
4991 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
4992 * rtsp-auth: Refer to part of constant name as text
4993 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
4994 * rtsp-session-media: Fix GstRTSPSessionMedia typo
4995 * rtsp-stream: Fix typo when refering to GstBin
4996 https://bugzilla.gnome.org/show_bug.cgi?id=714988
4998 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5001 * docs/libs/gst-rtsp-server-docs.sgml:
5002 * docs/libs/gst-rtsp-server-sections.txt:
5003 docs: Improve documentation
5004 * Include annotation-glossary to quiet gtk-doc
5005 * Rename remaining ClientState -> Context
5006 * Rename object hierarchy file
5007 * Remove stale chapter references
5008 * Add missing function and object references
5009 * Include missing GstRTSPAddressPoolResult
5010 https://bugzilla.gnome.org/show_bug.cgi?id=714988
5012 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
5014 * gst/rtsp-server/rtsp-client.c:
5015 * gst/rtsp-server/rtsp-server.c:
5016 * gst/rtsp-server/rtsp-session-pool.c:
5017 * gst/rtsp-server/rtsp-session.c:
5018 * gst/rtsp-server/rtsp-stream.c:
5019 rtsp-server: sprinkle some allow-none annotations for g-i
5021 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
5023 * gst/rtsp-server/rtsp-stream.c:
5024 * gst/rtsp-server/rtsp-stream.h:
5025 stream: add method to filter transports
5026 Add a method to safely iterate and collect the stream transports
5027 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
5029 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
5031 * gst/rtsp-server/rtsp-client.c:
5032 * gst/rtsp-server/rtsp-server.c:
5033 * gst/rtsp-server/rtsp-session-pool.c:
5034 * gst/rtsp-server/rtsp-session.c:
5035 rtsp: allow NULL func in filters
5036 Passing a null function make the filters return a list of
5039 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
5041 * gst/rtsp-server/rtsp-address-pool.c:
5042 * tests/check/gst/addresspool.c:
5043 address-pool: fix address increment
5044 Use a guint instead of guint8 to increment the address. It's still not
5045 completely correct because a guint might not be able to hold the complete
5046 address range, but that's an enhacement for later.
5047 Add unit test to test improved behaviour.
5048 https://bugzilla.gnome.org/show_bug.cgi?id=708237
5050 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
5052 * gst/rtsp-server/rtsp-client.c:
5053 * tests/check/gst/client.c:
5054 client: allow absolute path in requests
5055 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
5057 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
5059 * gst/rtsp-server/rtsp-client.c:
5060 * gst/rtsp-server/rtsp-client.h:
5061 client: make make_path_from_uri a vmethod
5063 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
5065 * docs/libs/gst-rtsp-server-sections.txt:
5066 * gst/rtsp-server/rtsp-stream.c:
5067 * gst/rtsp-server/rtsp-stream.h:
5068 * tests/check/Makefile.am:
5069 * tests/check/gst/stream.c:
5070 stream: Add functions to get rtp and rtcp sockets
5071 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
5073 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
5075 * gst/rtsp-server/rtsp-context.c:
5076 * gst/rtsp-server/rtsp-context.h:
5077 context: defing a GType for the context
5078 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
5080 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
5082 * gst/rtsp-server/Makefile.am:
5083 * gst/rtsp-server/rtsp-auth.c:
5084 * gst/rtsp-server/rtsp-context.c:
5085 * gst/rtsp-server/rtsp-media.c:
5086 * gst/rtsp-server/rtsp-mount-points.c:
5087 * gst/rtsp-server/rtsp-server.h:
5088 * gst/rtsp-server/rtsp-session-media.c:
5089 * gst/rtsp-server/rtsp-session.c:
5090 * gst/rtsp-server/rtsp-stream.c:
5091 Fixed several GIR warnings
5093 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
5095 * gst/rtsp-server/rtsp-auth.c:
5098 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5100 * tests/check/Makefile.am:
5101 * tests/check/gst/token.c:
5102 tests: Add unit tests for token
5103 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
5105 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5107 * gst/rtsp-server/rtsp-token.c:
5108 token: Validate args for gst_rtsp_token_is_allowed
5109 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
5111 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5113 * gst/rtsp-server/rtsp-token.c:
5114 token: Fix bug when creating empty token
5115 We always want to have a valid GstStructure in the token.
5116 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
5118 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
5120 * gst/rtsp-server/rtsp-thread-pool.c:
5121 thread-pool: avoid race in shutdown
5122 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
5123 don't actually stop the mainloop ever. Solve this race by adding an idle source
5124 to the mainloop that calls the _quit. This way we immediately exit the mainloop
5125 if quit was called before we started it.
5127 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5129 * tests/check/Makefile.am:
5130 * tests/check/gst/permissions.c:
5131 tests: Add unit tests for permissions
5132 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
5134 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5136 * tests/check/gst/mediafactory.c:
5137 tests: Test mediafactory permissions
5138 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
5140 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5142 * gst/rtsp-server/rtsp-permissions.c:
5143 permissions: Fix refcounting when adding/removing roles
5144 Previously a role that was removed was unreffed twice, and when
5145 replacing an existing role the replaced role was freed while still being
5146 referenced. Both bugs are now fixed.
5147 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
5149 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5151 * tests/check/gst/media.c:
5152 * tests/check/gst/mediafactory.c:
5153 * tests/check/gst/rtspserver.c:
5154 tests: Check gst_rtsp_url_parse return value
5155 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
5157 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
5160 Automatic update of common submodule
5161 From 865aa20 to dbedaa0
5163 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
5165 * gst/rtsp-server/rtsp-server.c:
5166 rtsp-server: Fix socket leak
5167 https://bugzilla.gnome.org/show_bug.cgi?id=710088
5169 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
5171 * gst/rtsp-server/rtsp-session-pool.c:
5172 rtsp-session-pool: Make sure session IDs are properly URI-escaped
5173 https://bugzilla.gnome.org/show_bug.cgi?id=643812
5175 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
5177 * examples/.gitignore:
5178 * examples/test-video.c:
5179 examples: fix compilation when WITH_AUTH is defined
5180 https://bugzilla.gnome.org/show_bug.cgi?id=710228
5182 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
5185 gitignore: Add new test binary
5187 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
5189 * tests/check/Makefile.am:
5190 * tests/check/gst/threadpool.c:
5191 thread-pool: Add unit test for the thread pools
5192 https://bugzilla.gnome.org/show_bug.cgi?id=710228
5194 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
5196 * gst/rtsp-server/rtsp-thread-pool.c:
5197 thread-pool: Fix thread leak when reusing threads
5198 https://bugzilla.gnome.org/show_bug.cgi?id=709730
5200 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
5202 * gst/rtsp-server/rtsp-server.c:
5203 * tests/check/gst/rtspserver.c:
5204 tests: fixed racy behavior in rtspserver tests
5205 https://bugzilla.gnome.org/show_bug.cgi?id=710078
5207 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5209 * tests/check/gst/addresspool.c:
5210 tests: Improve address pool unit tests
5211 Add a range with mixed IPV4 and IPV6 addresses to pool.
5212 Get an IPV4 address from an IPV6-only pool.
5213 Get an IPV6 address from an IPV4-only pool.
5214 Reserve a IPV6 address from an IPV4-only pool.
5215 Check for unicast addresses in multicast-only pool.
5216 Check for unicast addresses in uni-/multicast-mixed pool.
5217 https://bugzilla.gnome.org/show_bug.cgi?id=710128
5219 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5221 * gst/rtsp-server/rtsp-client.c:
5222 client: append query string in PAUSE/PLAY/TEARDOWN as well
5224 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
5226 * gst/rtsp-server/rtsp-client.c:
5227 client: Add query to control path
5228 If the SETUP url contains a query it must be appended to the control
5229 path so that it matches any already created stream in the media. The
5230 query will also be appended to the session media path.
5232 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5234 * gst/rtsp-server/rtsp-media.c:
5235 rtsp-media: remove old line
5237 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
5239 * gst/rtsp-server/rtsp-stream.c:
5240 stream: Correct control comparison
5241 https://bugzilla.gnome.org/show_bug.cgi?id=709176
5243 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5245 * gst/rtsp-server/rtsp-media.c:
5246 media: Check dynamically if the pipeline supports seeking
5247 We should not depend on whether or not the pipeline state change
5248 returned NO_PREROLL or not. A media could dynamically change its
5249 element and switch from seekable to non seekable so it's best to test
5250 the seekable nature of the pipeline dynamically when we try to do a seek.
5252 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5254 * gst/rtsp-server/rtsp-media.c:
5255 media: Return FALSE if seeking is not supported
5257 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5259 * gst/rtsp-server/rtsp-media.c:
5260 rtsp-media: don't seek accurate by default
5261 Accurate seeking is perhaps a little overkill in the most common situation and
5262 causes some formats (mp3) over slow media to seek extremely slowly.
5264 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
5266 * tests/check/gst/rtspserver.c:
5267 tests: fix unit test
5268 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
5270 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
5272 * gst/rtsp-server/rtsp-client.c:
5273 client: Reply 400 if media cannot be constructed
5274 Reply 400 Bad Request instead of 503 Service Unavailable if media
5275 cannot be constructed in SETUP.
5276 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
5278 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
5280 * gst/rtsp-server/rtsp-client.c:
5281 client: Send setup reply once only
5282 If find_media() failed in handle_setup_request() two replies was sent.
5283 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
5285 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
5288 Automatic update of common submodule
5289 From 6b03ba7 to 865aa20
5291 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
5293 * gst/rtsp-server/rtsp-server.c:
5294 server: Emit client-connected signal earlier
5295 Emit client-connected before the client ref is given to a GSource,
5296 otherwise client-connected can be emitted after the client object has
5299 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
5301 * gst/rtsp-server/rtsp-address-pool.c:
5302 * gst/rtsp-server/rtsp-address-pool.h:
5303 * gst/rtsp-server/rtsp-stream.c:
5304 * tests/check/gst/addresspool.c:
5305 addresspool: return reason of failure
5306 Let gst_rtsp_address_pool_reserve_address() return the reason why
5307 the address could not be reserved.
5308 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
5310 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
5313 autogen.sh: Sync behaviour with other GStreamer modules
5314 Allows building from outside of tree amongst other things
5316 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
5319 Automatic update of common submodule
5320 From b613661 to 6b03ba7
5322 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
5325 Automatic update of common submodule
5326 From 74a6857 to b613661
5328 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
5331 Automatic update of common submodule
5332 From 01a7a46 to 74a6857
5334 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
5336 * gst/rtsp-server/rtsp-client.c:
5337 client: Do not read beyond end of path string
5338 If the setup was done without a control url, make sure we don't try to read the
5339 non-existing control string and crash.
5341 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5343 * gst/rtsp-server/rtsp-client.c:
5344 client: Fix RTPInfo header
5345 Refactor the method to make the content_base.
5346 Use the content-base and the control url to construct the RTPInfo
5349 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5351 * gst/rtsp-server/rtsp-client.c:
5352 client: map url to path only in describe
5353 Only map the request url to a path in the DESCRIBE method. The SDP then
5354 contains the base and control urls that should be used to SETUP/PAUSE/
5355 PLAY/TEARDOWN the media.
5357 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5359 * gst/rtsp-server/rtsp-client.c:
5360 Revert "client: map URL to path in requests"
5361 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
5362 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
5363 contains the base and control urls which are used in the SETUP, PLAY,
5364 PAUSE and TEARDOWN requests.
5366 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5368 * gst/rtsp-server/rtsp-client.c:
5369 client: map URL to path in requests
5371 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5373 * gst/rtsp-server/rtsp-client.c:
5374 * gst/rtsp-server/rtsp-mount-points.c:
5375 * gst/rtsp-server/rtsp-mount-points.h:
5376 mount-points: make vmethod to make path from uri
5377 Make a vmethod to transform an url into a path. The path is then used to lookup
5378 the factory. This makes it possible to also use other bits of the url, such as
5379 the query parameters, to locate the factory.
5381 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
5383 * gst/rtsp-server/rtsp-thread-pool.c:
5384 * gst/rtsp-server/rtsp-thread-pool.h:
5385 thread-pool: Add cleanup to wait for the threadpool to finish
5386 Also fix race condition if two threads are asking for the first
5387 thread from the thread pool at once. This would case two internal
5388 GThreadPools to be created.
5389 https://bugzilla.gnome.org/show_bug.cgi?id=707753
5391 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
5393 * gst/rtsp-server/rtsp-client.c:
5394 * tests/check/gst/client.c:
5395 client: free threadpool
5396 https://bugzilla.gnome.org/show_bug.cgi?id=707638
5398 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
5400 * tests/check/gst/mountpoints.c:
5401 mountpoints tests: unref matched factories
5402 https://bugzilla.gnome.org/show_bug.cgi?id=707638
5404 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
5406 * tests/check/gst/media.c:
5407 media tests: unref thread pool and caps
5408 https://bugzilla.gnome.org/show_bug.cgi?id=707638
5410 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
5412 * gst/rtsp-server/rtsp-auth.c:
5413 * gst/rtsp-server/rtsp-media-factory.c:
5414 * gst/rtsp-server/rtsp-media.c:
5415 auth, media, media-factory: unref permissions
5416 https://bugzilla.gnome.org/show_bug.cgi?id=707638
5418 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5420 * examples/Makefile.am:
5421 Makefile: add rule for appsrc example
5423 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5425 * examples/test-appsrc.c:
5426 tests: add appsrc example
5427 Add an example on how to use appsrc to feed the server pipeline with data.
5429 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
5431 * gst/rtsp-server/rtsp-client.c:
5432 rtsp-client: remove query part from content-base string
5433 Make sure that after the control url has been resolved, it's
5434 not a part of the query-string.
5435 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
5437 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5439 * gst/rtsp-server/rtsp-client.c:
5440 client: don't check url in response
5441 There is no url or method in the response to check
5443 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5445 * gst/rtsp-server/rtsp-client.c:
5446 * gst/rtsp-server/rtsp-client.h:
5447 Add handle-response signal for when we receive a GET_PARAMETER response
5449 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5451 * gst/rtsp-server/rtsp-server.c:
5452 Fix gst_rtsp_server_client_filter, using wrong variable type
5454 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
5456 * gst/rtsp-server/rtsp-media-factory-uri.c:
5457 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
5458 For AAC we need to check for framed=true instead of parsed=true.
5459 https://bugzilla.gnome.org/show_bug.cgi?id=701384
5461 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5463 * gst/rtsp-server/rtsp-stream.c:
5464 stream: optimize pipeline for protocols
5465 When TCP is not an allowed protocol for the stream, avoid creating the
5466 appsrc/appsink/queue and tee elements.
5468 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5470 * gst/rtsp-server/rtsp-media.c:
5471 media: set protocols on streams
5473 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5475 * gst/rtsp-server/rtsp-client.c:
5476 client: use protocols supported by stream
5478 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5480 * gst/rtsp-server/rtsp-media-factory.c:
5481 * gst/rtsp-server/rtsp-media.c:
5482 * gst/rtsp-server/rtsp-stream.c:
5483 media-factory: allow all protocols
5485 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5487 * gst/rtsp-server/rtsp-media.c:
5488 media: configure protocols in new streams
5490 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5492 * gst/rtsp-server/rtsp-stream.c:
5493 * gst/rtsp-server/rtsp-stream.h:
5494 stream: add protocols property
5496 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5498 * gst/rtsp-server/rtsp-media.c:
5499 rtsp-media: send state in "new-state" signal
5500 https://bugzilla.gnome.org/show_bug.cgi?id=705110
5502 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
5505 build: add subdir-objects to AM_INIT_AUTOMAKE
5506 Fixes warnings with automake 1.14
5507 https://bugzilla.gnome.org/show_bug.cgi?id=705350
5509 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5511 * docs/libs/gst-rtsp-server-sections.txt:
5512 * gst/rtsp-server/rtsp-client.c:
5513 * gst/rtsp-server/rtsp-server.c:
5514 * gst/rtsp-server/rtsp-server.h:
5515 server: add method to iterate clients of server
5517 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5519 * gst/rtsp-server/rtsp-media.c:
5520 * gst/rtsp-server/rtsp-media.h:
5521 Add vmethod for rtsp-media subclass to access rtpbin
5523 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5525 * gst/rtsp-server/rtsp-client.h:
5526 small documentation fix
5528 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5530 * gst/rtsp-server/rtsp-client.c:
5531 Do not take range header if range is invalid
5533 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5535 * docs/libs/gst-rtsp-server-sections.txt:
5536 * gst/rtsp-server/rtsp-media.c:
5537 media: add docs for new method
5539 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5541 * gst/rtsp-server/rtsp-media.c:
5542 * gst/rtsp-server/rtsp-media.h:
5543 Add API to rtsp-media set the pipeline's state
5545 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5547 * gst/rtsp-server/rtsp-media.c:
5548 Update current position/duration when gst_rtsp_media_get_range_string is called
5550 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5552 * examples/test-cgroups.c:
5553 tests: add some more docs
5555 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5557 * examples/test-cgroups.c:
5558 * gst/rtsp-server/Makefile.am:
5559 * gst/rtsp-server/rtsp-auth.c:
5560 * gst/rtsp-server/rtsp-auth.h:
5561 * gst/rtsp-server/rtsp-client.c:
5562 * gst/rtsp-server/rtsp-client.h:
5563 * gst/rtsp-server/rtsp-context.c:
5564 * gst/rtsp-server/rtsp-context.h:
5565 * gst/rtsp-server/rtsp-params.c:
5566 * gst/rtsp-server/rtsp-params.h:
5567 * gst/rtsp-server/rtsp-server.c:
5568 * gst/rtsp-server/rtsp-thread-pool.c:
5569 * gst/rtsp-server/rtsp-thread-pool.h:
5570 * tests/check/gst/client.c:
5571 ClientState -> Context
5572 Rename the clientstate to context and put the code in a separate file.
5574 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5576 * examples/test-auth.c:
5577 * gst/rtsp-server/rtsp-auth.c:
5578 * gst/rtsp-server/rtsp-auth.h:
5579 auth: add support for default token
5580 The default token is used when the user is not authenticated and can be used to
5581 give minimal permissions.
5583 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5585 * examples/test-auth.c:
5586 * gst/rtsp-server/rtsp-auth.c:
5587 auth: use defines when possible
5589 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5591 * gst/rtsp-server/rtsp-address-pool.c:
5592 address-pool: improve docs
5594 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5596 * gst/rtsp-server/rtsp-permissions.c:
5597 permissions: add the role to the copy
5599 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
5601 * gst/rtsp-server/rtsp-permissions.c:
5602 permissions: Also copy the roles
5604 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
5606 * gst/rtsp-server/rtsp-permissions.c:
5607 permissions: Make it build
5609 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5611 * gst/rtsp-server/rtsp-address-pool.h:
5614 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5616 * docs/libs/gst-rtsp-server-sections.txt:
5617 * gst/rtsp-server/rtsp-auth.c:
5618 * gst/rtsp-server/rtsp-auth.h:
5619 * gst/rtsp-server/rtsp-media.c:
5620 * gst/rtsp-server/rtsp-session-media.c:
5621 * gst/rtsp-server/rtsp-stream-transport.c:
5622 * gst/rtsp-server/rtsp-stream-transport.h:
5623 * gst/rtsp-server/rtsp-stream.c:
5624 * tests/check/gst/client.c:
5627 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5629 * docs/libs/gst-rtsp-server-sections.txt:
5630 * gst/rtsp-server/rtsp-address-pool.c:
5631 * gst/rtsp-server/rtsp-address-pool.h:
5632 * tests/check/gst/addresspool.c:
5633 * tests/check/gst/rtspserver.c:
5634 address-pool: cleanups
5635 Remove redundant method, improve docs.
5637 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5639 * docs/libs/gst-rtsp-server-sections.txt:
5640 * gst/rtsp-server/rtsp-auth.h:
5641 * gst/rtsp-server/rtsp-permissions.c:
5642 * gst/rtsp-server/rtsp-permissions.h:
5643 * gst/rtsp-server/rtsp-token.c:
5646 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5648 * gst/rtsp-server/rtsp-permissions.c:
5649 permissions: implement _remove_role
5651 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5653 * gst/rtsp-server/rtsp-permissions.c:
5654 permissions: update docs
5656 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5658 * tests/check/gst/client.c:
5659 tests: simplify tests
5660 Client settings are now disabled by default so we don't need an auth
5661 module to disable them.
5663 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5665 * gst/rtsp-server/rtsp-auth.c:
5666 auth: add default authorizations
5667 When no auth module is specified, use our table of defaults to look up the
5668 default value of the check instead of always allowing everything. This was
5669 we can disallow client settings by default.
5671 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5674 README: update readme
5676 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5678 * gst/rtsp-server/rtsp-thread-pool.c:
5679 * gst/rtsp-server/rtsp-thread-pool.h:
5680 thread-pool: add more docs
5682 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5684 * gst/rtsp-server/rtsp-thread-pool.c:
5685 * gst/rtsp-server/rtsp-thread-pool.h:
5686 thread-pool: fix race in thread reuse
5687 If we try to reuse a thread right after we made it stop, we end up using a
5688 stopped thread. Catch this case and only reuse threads that are not stopping.
5690 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5692 * gst/rtsp-server/rtsp-server.c:
5693 server: add small debug
5695 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5697 * tests/check/gst/client.c:
5699 Add some permissions to media so we can use the auth and enable
5702 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5704 * gst/rtsp-server/rtsp-client.c:
5705 client: support pushed context in handle_request
5706 If we already have a pushed state, reuse it and add our own things. This makes
5707 it easier to write tests.
5709 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5711 * gst/rtsp-server/rtsp-auth.c:
5712 auth: don't auth on methods
5713 Don't authorize on methods anymore but on the resources that we
5714 try to access, this is more flexible.
5715 Move the authorization checks to where they are needed and let the
5716 check return the response on error.
5718 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5720 * gst/rtsp-server/rtsp-mount-points.c:
5721 mount-points: add some debug
5723 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5725 * tests/check/gst/client.c:
5726 tests: almost fix test
5728 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5730 * gst/rtsp-server/rtsp-auth.c:
5731 * gst/rtsp-server/rtsp-auth.h:
5732 * gst/rtsp-server/rtsp-client.c:
5733 * gst/rtsp-server/rtsp-client.h:
5734 * gst/rtsp-server/rtsp-server.c:
5735 * gst/rtsp-server/rtsp-server.h:
5736 auth: let the auth module check client_settings
5737 Let the auth module decide if client settings are allowed for the
5740 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5742 * gst/rtsp-server/rtsp-token.c:
5743 * gst/rtsp-server/rtsp-token.h:
5744 token: add method to check boolean permission
5746 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5748 * examples/test-auth.c:
5749 * examples/test-cgroups.c:
5750 * gst/rtsp-server/rtsp-token.c:
5751 * gst/rtsp-server/rtsp-token.h:
5752 token: simplify token constructor
5753 Use variable arguments to make easier API.
5755 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5757 * examples/test-auth.c:
5758 * examples/test-cgroups.c:
5759 * gst/rtsp-server/rtsp-media-factory.c:
5760 * gst/rtsp-server/rtsp-media-factory.h:
5761 media-factory: add convenience API for factory
5763 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5765 * examples/test-auth.c:
5766 * examples/test-cgroups.c:
5767 * gst/rtsp-server/rtsp-permissions.c:
5768 * gst/rtsp-server/rtsp-permissions.h:
5769 permissions: simplify API a little
5770 Avoid passing GstStructure in the add_role method, use varargs instead
5771 to construct the structure behind the scenes. We can then also use the
5772 structure name as the role and simplify some more logic.
5774 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5776 * gst/rtsp-server/rtsp-auth.c:
5779 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5781 * gst/rtsp-server/rtsp-auth.c:
5782 * gst/rtsp-server/rtsp-auth.h:
5783 * gst/rtsp-server/rtsp-client.c:
5784 auth: handle unauthorized response
5785 Move handling of the unauthorized response to the auth module, it can add
5786 the appropriate headers to request authorization for the required method
5787 much better than the client.
5789 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5791 * gst/rtsp-server/rtsp-client.c:
5792 * gst/rtsp-server/rtsp-client.h:
5793 client: allow for sending any message, not only requests
5794 Change the _send_request() method to _send_message() so that we
5795 can both send requests and replies.
5797 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5799 * docs/libs/gst-rtsp-server-sections.txt:
5800 * gst/rtsp-server/rtsp-server.h:
5803 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5805 * examples/test-video.c:
5806 * gst/rtsp-server/rtsp-auth.c:
5807 * gst/rtsp-server/rtsp-auth.h:
5808 * gst/rtsp-server/rtsp-server.c:
5809 * gst/rtsp-server/rtsp-server.h:
5810 auth: move TLS handling to auth module
5811 Remove the TLS settings on the server and move it to the auth module because
5812 that is where security related bits go.
5814 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5816 * gst/rtsp-server/rtsp-client.c:
5817 * gst/rtsp-server/rtsp-client.h:
5818 client: add state push/pop
5820 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5822 * gst/rtsp-server/rtsp-client.c:
5823 * gst/rtsp-server/rtsp-client.h:
5824 client: add connection to state
5826 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5828 * gst/rtsp-server/rtsp-mount-points.c:
5829 mount-points: fix debug
5831 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5833 * tests/check/gst/media.c:
5834 tests: fix media test
5836 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5838 * gst/rtsp-server/rtsp-thread-pool.c:
5839 thread-pool: we don't require a state
5841 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5843 * gst/rtsp-server/rtsp-server.c:
5844 server: let context ref the server
5845 So that we don't risk losing the server object early anc crash.
5847 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5849 * tests/check/gst/client.c:
5850 tests: fix client test
5852 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5855 * docs/libs/gst-rtsp-server-docs.sgml:
5856 * docs/libs/gst-rtsp-server-sections.txt:
5857 * gst/rtsp-server/rtsp-address-pool.c:
5858 * gst/rtsp-server/rtsp-auth.c:
5859 * gst/rtsp-server/rtsp-client.c:
5860 * gst/rtsp-server/rtsp-client.h:
5861 * gst/rtsp-server/rtsp-media-factory-uri.c:
5862 * gst/rtsp-server/rtsp-media-factory.c:
5863 * gst/rtsp-server/rtsp-media-factory.h:
5864 * gst/rtsp-server/rtsp-media.c:
5865 * gst/rtsp-server/rtsp-mount-points.c:
5866 * gst/rtsp-server/rtsp-params.c:
5867 * gst/rtsp-server/rtsp-permissions.c:
5868 * gst/rtsp-server/rtsp-sdp.c:
5869 * gst/rtsp-server/rtsp-server.c:
5870 * gst/rtsp-server/rtsp-server.h:
5871 * gst/rtsp-server/rtsp-session-media.c:
5872 * gst/rtsp-server/rtsp-session-pool.c:
5873 * gst/rtsp-server/rtsp-session.c:
5874 * gst/rtsp-server/rtsp-stream-transport.c:
5875 * gst/rtsp-server/rtsp-stream.c:
5876 * gst/rtsp-server/rtsp-thread-pool.c:
5877 * gst/rtsp-server/rtsp-token.c:
5880 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5882 * gst/rtsp-server/rtsp-session-pool.c:
5883 * gst/rtsp-server/rtsp-session-pool.h:
5884 session-pool: make vmethod to create a session
5885 Make a vmethod to create a sessions so that subclasses can create
5886 custom session objects
5888 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5890 * gst/rtsp-server/rtsp-auth.c:
5891 * gst/rtsp-server/rtsp-media-factory.h:
5892 * gst/rtsp-server/rtsp-media.h:
5893 * gst/rtsp-server/rtsp-mount-points.h:
5894 * gst/rtsp-server/rtsp-session-pool.h:
5895 * gst/rtsp-server/rtsp-stream.h:
5898 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5900 * docs/libs/gst-rtsp-server-docs.sgml:
5901 * docs/libs/gst-rtsp-server-sections.txt:
5902 * gst/rtsp-server/rtsp-address-pool.c:
5903 * gst/rtsp-server/rtsp-address-pool.h:
5904 * gst/rtsp-server/rtsp-auth.c:
5905 * gst/rtsp-server/rtsp-client.h:
5906 * gst/rtsp-server/rtsp-media-factory.h:
5907 * gst/rtsp-server/rtsp-media.c:
5908 * gst/rtsp-server/rtsp-media.h:
5909 * gst/rtsp-server/rtsp-permissions.c:
5910 * gst/rtsp-server/rtsp-permissions.h:
5911 * gst/rtsp-server/rtsp-server.h:
5912 * gst/rtsp-server/rtsp-session-media.c:
5913 * gst/rtsp-server/rtsp-session-media.h:
5914 * gst/rtsp-server/rtsp-session-pool.h:
5915 * gst/rtsp-server/rtsp-session.h:
5916 * gst/rtsp-server/rtsp-stream-transport.h:
5917 * gst/rtsp-server/rtsp-stream.c:
5918 * gst/rtsp-server/rtsp-thread-pool.h:
5921 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5924 * examples/Makefile.am:
5925 configure: compile cgroup example conditionally
5926 Only compile the cgroup example when we have libcgroup
5928 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5931 * examples/Makefile.am:
5932 * examples/test-cgroups.c:
5933 examples: add cgroups example
5935 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5937 * tests/check/gst/rtspserver.c:
5938 tests: fix compilation
5940 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5942 * gst/rtsp-server/rtsp-thread-pool.c:
5943 thread-pool: fix vmethod invocation
5945 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5947 * gst/rtsp-server/rtsp-thread-pool.c:
5948 * gst/rtsp-server/rtsp-thread-pool.h:
5949 thread-pool: store thread type in thread
5951 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5953 * gst/rtsp-server/rtsp-client.c:
5954 client: pass thread from pool to media _prepare
5955 Get a thread from the configured threadpool and pass it to the prepare method of
5958 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5960 * gst/rtsp-server/rtsp-media.c:
5961 * gst/rtsp-server/rtsp-media.h:
5962 media: Accept a thread in _prepare
5963 Remove out own threadpool handling and use the provided thread and
5964 maincontext for the bus messages and the state changes.
5966 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5968 * gst/rtsp-server/rtsp-server.c:
5969 server: configure client thread pool
5971 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5973 * gst/rtsp-server/rtsp-client.c:
5974 * gst/rtsp-server/rtsp-client.h:
5975 client: add method to configure thread pool
5977 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5979 * gst/rtsp-server/rtsp-client.h:
5980 * gst/rtsp-server/rtsp-server.c:
5981 * gst/rtsp-server/rtsp-server.h:
5982 server: use thread pool
5983 Use the thread pool instead of doing our own thing.
5985 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5987 * gst/rtsp-server/Makefile.am:
5988 * gst/rtsp-server/rtsp-thread-pool.c:
5989 * gst/rtsp-server/rtsp-thread-pool.h:
5990 thread-pool: add object to manage threads
5991 Add an object to manage the client and media threads.
5993 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5995 * gst/rtsp-server/rtsp-auth.c:
5996 auth: debug authorization check
5998 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6000 * gst/rtsp-server/rtsp-media.c:
6001 media: start media pipeline in context
6002 Start the media pipeline in the provided context (or our default one
6003 when NULL). This makes sure that we run the bus thread in this context and that
6004 all media threads are children of this context.
6006 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6008 * gst/rtsp-server/rtsp-media-factory.c:
6009 factory: pass permissions to media by default
6011 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6013 * examples/test-auth.c:
6014 test: add permissions to auth test
6015 Ass some permissions to the media factory in the test.
6017 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6019 * gst/rtsp-server/rtsp-auth.c:
6020 * gst/rtsp-server/rtsp-auth.h:
6021 * gst/rtsp-server/rtsp-client.c:
6022 auth: simplify auth checks
6023 Remove client from methods, it's now in the state
6024 Perform the check specified by the string, use the information from the
6025 thread local context.
6027 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6029 * gst/rtsp-server/rtsp-client.c:
6030 * gst/rtsp-server/rtsp-client.h:
6031 client: add state to current thread
6032 Add the client to the ClientState object.
6033 Place the ClientState on the current thread.
6035 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6037 * gst/rtsp-server/rtsp-media-factory.c:
6038 * gst/rtsp-server/rtsp-media-factory.h:
6039 * gst/rtsp-server/rtsp-media.c:
6040 * gst/rtsp-server/rtsp-media.h:
6041 media: make it possible to set permissions
6042 Make it possible to set permissions on media and media factory objects
6044 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6046 * gst/rtsp-server/Makefile.am:
6047 * gst/rtsp-server/rtsp-permissions.c:
6048 * gst/rtsp-server/rtsp-permissions.h:
6049 permissions: add permissions object
6050 Add a mini object to store permissions based on a role.
6052 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6054 * examples/test-auth.c:
6055 * gst/rtsp-server/rtsp-auth.c:
6056 * gst/rtsp-server/rtsp-auth.h:
6057 * gst/rtsp-server/rtsp-client.c:
6058 auth: add auth checks
6059 Add an enum with auth checks and implement the checks in the auth object.
6060 Perform the checks from the client.
6062 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6064 * examples/test-auth.c:
6065 * gst/rtsp-server/rtsp-auth.c:
6066 * gst/rtsp-server/rtsp-auth.h:
6067 * gst/rtsp-server/rtsp-client.h:
6068 auth: use the token after authentication
6069 After we authenticated a user, keep the Token around in the state.
6071 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6073 * gst/rtsp-server/rtsp-client.c:
6074 * gst/rtsp-server/rtsp-media.c:
6075 * gst/rtsp-server/rtsp-media.h:
6076 * tests/check/gst/media.c:
6077 media: add optional context for bus messages
6078 Add an optional mainloop to _prepare that will handle the bus messages instead
6079 of always using the shared mainloop.
6081 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6083 * gst/rtsp-server/Makefile.am:
6084 * gst/rtsp-server/rtsp-token.c:
6085 * gst/rtsp-server/rtsp-token.h:
6086 token: add authorization token
6087 Add a simply miniobject that contains the authorizations. The object contains a
6088 GstStructure that hold all authorization fields. When a user is authenticated,
6089 the auth module will create a Token for the user. The token is then used to
6090 check what operations the user is allowed to do and various other configuration
6093 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6095 * examples/test-auth.c:
6096 * gst/rtsp-server/rtsp-auth.c:
6097 * gst/rtsp-server/rtsp-auth.h:
6098 * gst/rtsp-server/rtsp-client.c:
6099 * gst/rtsp-server/rtsp-client.h:
6100 * gst/rtsp-server/rtsp-media-factory.c:
6101 * gst/rtsp-server/rtsp-media-factory.h:
6102 * gst/rtsp-server/rtsp-media.c:
6103 * gst/rtsp-server/rtsp-media.h:
6104 auth: remove auth from media and factory
6105 Remove the auth object from media and factory. We want to have the RTSPClient
6106 authenticate and authorize resources, there is no need to place another auth
6107 manager on the media/factory.
6109 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6111 * examples/test-auth.c:
6112 * gst/rtsp-server/rtsp-auth.c:
6113 * gst/rtsp-server/rtsp-auth.h:
6114 * gst/rtsp-server/rtsp-client.h:
6115 auth: add support for multiple basic auth tokens
6116 Make it possible to add multiple basic authorisation tokens to one authorization
6117 object. Associate with each token an authorization group that will define what
6118 capabilities are allowed.
6120 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6122 * gst/rtsp-server/rtsp-client.c:
6123 client: error out on non-aggregate control
6124 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
6126 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6128 * gst/rtsp-server/rtsp-client.c:
6129 client: rework setup request a little
6130 Cache the media in DESCRIBE based on the longest matching path with the uri
6131 that we can find in the mount points.
6132 Rework the setup request a little to get the media from the session or from
6133 the longest matching path, this way we can derive the control string as
6134 everything after the path instead of hardcoding it.
6135 Find the stream based on the control string and only open a session when all
6138 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6140 * gst/rtsp-server/rtsp-media.c:
6141 * gst/rtsp-server/rtsp-media.h:
6142 media: add method to find a stream by control url
6144 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6146 * gst/rtsp-server/rtsp-stream.c:
6147 * gst/rtsp-server/rtsp-stream.h:
6148 stream: add method to check control url of stream
6150 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6152 * gst/rtsp-server/rtsp-client.c:
6153 * gst/rtsp-server/rtsp-session-media.c:
6154 * gst/rtsp-server/rtsp-session-media.h:
6155 * gst/rtsp-server/rtsp-session.c:
6156 * gst/rtsp-server/rtsp-session.h:
6157 session: use path matching for session media
6158 Use a path string instead of a uri to lookup session media in the sessions. Also
6159 use path matching to find the largest possible path that matches.
6161 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6163 * gst/rtsp-server/rtsp-client.c:
6164 * gst/rtsp-server/rtsp-mount-points.c:
6165 * gst/rtsp-server/rtsp-mount-points.h:
6166 * tests/check/gst/mountpoints.c:
6167 mount-points: remove useless vmethod
6168 Making lookups in the mount points should not be done with a URL, if there is a
6169 mapping to be done from URL to mount points, we'll need to do it somewhere
6172 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6174 * gst/rtsp-server/rtsp-mount-points.c:
6175 * gst/rtsp-server/rtsp-mount-points.h:
6176 * tests/check/gst/mountpoints.c:
6177 mount-points: improve mount point searching
6178 Use a GSequence to keep track of the mount points.
6179 Match a URL to the longest matching registered mount point. This should be the
6180 URL to perform aggreagate control and the remainder is the stream specific
6182 Add some unit tests for this.
6184 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
6186 * gst/rtsp-server/Makefile.am:
6187 rtsp-server: Allow building of static library
6189 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6191 * tests/check/gst/mediafactory.c:
6192 tests: fix compilation
6194 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6196 * gst/rtsp-server/rtsp-sdp.c:
6197 sdp: get control string from stream
6198 Use the control string as configured in the stream.
6200 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6202 * gst/rtsp-server/rtsp-stream.c:
6203 * gst/rtsp-server/rtsp-stream.h:
6204 stream: add methods and property to set control string
6206 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6208 * gst/rtsp-server/rtsp-client.c:
6210 Rename variables for clarity
6211 Keep media in state when we can
6213 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6215 * gst/rtsp-server/rtsp-client.c:
6216 * gst/rtsp-server/rtsp-stream.c:
6217 * gst/rtsp-server/rtsp-stream.h:
6218 stream: add more support for IPv6
6219 Rename _get_address to _get_multicast_address in GstRTSPStream to
6220 make it clear that this function only deals with multicast.
6221 Make it possible to have both an IPv4 and IPv6 multicast address on
6222 a stream. Give the client an IPv4 or IPv6 address depending on the
6223 address it used to connect to the server.
6224 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
6226 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6228 * gst/rtsp-server/rtsp-client.c:
6231 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6233 * gst/rtsp-server/rtsp-stream.c:
6234 stream: handle failed port allocation
6235 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
6236 can't allocate any family at all. Also keep track of what port families we
6238 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
6240 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6242 * gst/rtsp-server/rtsp-stream.c:
6243 stream: improve docs
6245 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6247 * gst/rtsp-server/rtsp-stream-transport.c:
6248 stream-transport: remove old if 0 block
6250 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
6252 * tests/check/gst/client.c:
6254 gst_rtsp_client_get_uri() has been removed
6255 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
6257 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6259 * gst/rtsp-server/rtsp-client.c:
6260 * gst/rtsp-server/rtsp-client.h:
6261 client: add method to filter managed sessions
6262 Add a method to filter the sessions managed by this client connection.
6263 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
6265 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6267 * gst/rtsp-server/rtsp-client.c:
6268 * gst/rtsp-server/rtsp-client.h:
6269 client: remove _get_uri() method
6270 Remove the get_uri() method on the client. A client has no uri, the uri
6271 property is an internal property to manage the last cached media for
6274 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6276 * gst/rtsp-server/rtsp-media-factory.h:
6277 media-factory: fix typo
6279 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
6281 * gst/rtsp-server/rtsp-media.c:
6282 rtsp-media: Do not leak the query in default_query_stop
6283 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
6285 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6287 * gst/rtsp-server/rtsp-media.c:
6288 media: don't unlock when conversion fails
6289 Don't unlock the state lock when conversion fails because it was not locked.
6291 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6293 * gst/rtsp-server/rtsp-media.c:
6294 * gst/rtsp-server/rtsp-media.h:
6295 Add query_position and query_stop vmethods to rtsp-media
6297 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6299 * gst/rtsp-server/rtsp-media.c:
6300 Fix typo in property install for rtsp-media's time-provider
6302 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6304 * gst/rtsp-server/rtsp-client.c:
6305 * gst/rtsp-server/rtsp-client.h:
6306 client: clean some variables
6307 Clean some variables and add some guards to _send_request()
6309 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6311 * gst/rtsp-server/rtsp-client.c:
6312 * gst/rtsp-server/rtsp-client.h:
6313 Add gst_rtsp_client_send_request API
6314 This makes it possible to send arbitrary messages to a client, such as
6315 SET_PARAMETER or GET_PARAMETER
6317 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6319 * gst/rtsp-server/rtsp-media.c:
6320 * gst/rtsp-server/rtsp-media.h:
6321 media: add _get_element() method
6322 Add method to get the element used when creating the media.
6323 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
6325 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6327 * gst/rtsp-server/rtsp-media.c:
6330 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
6332 * gst/rtsp-server/rtsp-stream.c:
6333 * gst/rtsp-server/rtsp-stream.h:
6334 stream: allow access to the rtp session
6335 https://bugzilla.gnome.org/show_bug.cgi?id=703004
6337 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
6339 * gst/rtsp-server/rtsp-stream.c:
6340 * gst/rtsp-server/rtsp-stream.h:
6341 dscp qos support in gst-rtsp-stream
6342 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
6344 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6346 * tests/check/gst/rtspserver.c:
6348 Actually do what the comment says. Also keep the old code around, not sure what
6349 should happen when you get a 454 from a TEARDOWN, does it close the connection?
6350 it currently doesn't.
6352 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6354 * gst/rtsp-server/rtsp-client.c:
6355 client: also watch newly created session
6356 When we newly created a session, start watching it immediately instead of
6357 on the next request.
6359 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
6361 * tests/check/gst/client.c:
6362 tests: add unit test for new-session
6363 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
6365 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6367 * gst/rtsp-server/rtsp-client.c:
6368 client: emit new-session when new session is created
6369 Only emit new-session when we created a new session for a client, not when a
6370 client picked up a previous session.
6371 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
6373 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
6375 * gst/rtsp-server/rtsp-client.c:
6376 client: handle asterisk as path in requests
6377 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
6379 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6381 * gst/rtsp-server/rtsp-media.c:
6382 media: handle segment query format mismatch
6383 It's possible that the segment query returns with a different format than what
6384 we asked for, handle this case also.
6386 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
6388 * gst/rtsp-server/rtsp-media.c:
6389 media: use segment stop in collect_media_stats
6390 Use segment stop instead of duration as range end point.
6391 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
6393 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
6395 * gst/rtsp-server/rtsp-media.c:
6396 * tests/check/gst/media.c:
6397 rtsp-media: Do not leak the element in take_pipeline
6398 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
6400 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
6402 * gst/rtsp-server/rtsp-client.c:
6403 * gst/rtsp-server/rtsp-client.h:
6404 rtsp-client: Make configure_client_transport virtual
6405 This patch makes configure_client_transport virtual. The functionality is
6406 needed to handle some weird clients sending multicast transport settings as url
6408 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
6410 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
6412 * gst/rtsp-server/rtsp-client.c:
6413 * gst/rtsp-server/rtsp-client.h:
6414 rtsp-client: Make param_set and param_get virtual
6415 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
6417 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
6419 * gst/rtsp-server/rtsp-client.c:
6420 * gst/rtsp-server/rtsp-media.c:
6421 * gst/rtsp-server/rtsp-media.h:
6422 media: convert_range replaces get_range_times
6423 get_range_times worked for handling UTC ranges for seeks, but we also
6424 need to convert back from NPT to the requested unit in
6425 get_range_string. convert_range is now used for both.
6426 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
6428 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6430 * gst/rtsp-server/rtsp-client.c:
6431 * gst/rtsp-server/rtsp-sdp.c:
6432 * gst/rtsp-server/rtsp-sdp.h:
6433 sdp: cleanup sdp info
6434 We don't need to pass the proto, we can more easily check a boolean.
6435 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
6437 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
6439 * gst/rtsp-server/rtsp-sdp.c:
6440 use 0.0.0.0 or :: for c= line instead of server address
6442 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
6444 * gst/rtsp-server/rtsp-client.c:
6445 use local address, not remote, in SDP
6446 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
6448 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6451 Automatic update of common submodule
6452 From 098c0d7 to 01a7a46
6454 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
6456 * gst/rtsp-server/rtsp-media.c:
6457 * gst/rtsp-server/rtsp-media.h:
6458 media: possibility to override range time conversion
6459 Make it possible to override the conversion from GstRTSPTimeRange to
6460 GstClockTimes, that is done before seeking on the media
6461 pipeline. Overriding can be useful for UTC ranges, where the default
6462 conversion gives nanoseconds since 1900.
6463 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
6465 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
6467 * gst/rtsp-server/rtsp-server.c:
6468 * gst/rtsp-server/rtsp-server.h:
6469 rtsp-server: Expose the use_client_settings API
6470 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
6472 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
6474 * gst/rtsp-server/rtsp-client.c:
6475 * gst/rtsp-server/rtsp-stream.c:
6476 * gst/rtsp-server/rtsp-stream.h:
6477 rtspstream: handle both ipv4 and ipv6 clients
6478 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
6480 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6482 * gst/rtsp-server/rtsp-sdp.c:
6483 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
6484 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
6485 We already have a way to place extra attributes in the SDP by using a string
6486 property with prefix x- or a- in the caps.
6488 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6490 * gst/rtsp-server/rtsp-sdp.c:
6491 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
6492 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
6493 We already have a way to place extra attributes in the SDP, just make a string
6494 property in the payloader with a- or x- prefix.
6496 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6498 * gst/rtsp-server/rtsp-sdp.c:
6499 rtsp: place a- and x- properties as attributes
6500 application/x-rtp has properties with a- and x- prefixes that should be
6501 placed as attributes in the SDP for the media instead of being added to the
6504 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6506 * examples/Makefile.am:
6507 * examples/test-video.c:
6508 example: add TLS example
6510 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6512 * gst/rtsp-server/rtsp-server.c:
6513 * gst/rtsp-server/rtsp-server.h:
6514 server: add support for TLS
6515 Add methods to set and get a TLS certificate.
6516 Add vmethod to configure a new connection. By default, configure the TLS
6517 certificate in a new connection if needed.
6519 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6521 * gst/rtsp-server/rtsp-server.c:
6522 * gst/rtsp-server/rtsp-server.h:
6523 server: remove accept_client vmethod
6524 This vmethod is not very useful so remove it.
6526 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6528 * gst/rtsp-server/rtsp-server.c:
6529 server: don't crash on NULL GError
6531 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
6533 * gst/rtsp-server/rtsp-session-pool.c:
6534 rtsp-session-pool: corrected session timeout detection
6535 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
6537 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6539 * gst/rtsp-server/rtsp-client.c:
6540 client: improve debug
6542 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6544 * gst/rtsp-server/rtsp-client.c:
6545 * gst/rtsp-server/rtsp-client.h:
6546 * gst/rtsp-server/rtsp-server.c:
6547 server: refactor connection setup
6548 Let the server accept the socket connection and construct a GstRTSPConnection
6549 from it. Remove the code from the client and let the client only deal with
6550 a fully configure GstRTSPConnection object.
6551 We will need this later when the server will configure the connection for
6554 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6556 * gst/rtsp-server/rtsp-stream.c:
6557 stream: keep the transport object alive
6558 Keep the transport object alive while we have it as qdata on the
6561 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
6563 * gst/rtsp-server/rtsp-client.c:
6564 * gst/rtsp-server/rtsp-server.c:
6565 rtsp-server: Do not crash on nmapping of server
6566 * generate error when gst_rtsp_connection_accept fails
6567 * do not stop accepting incoming connections because
6568 accepting a client fails
6569 https://bugzilla.gnome.org/show_bug.cgi?id=701072
6571 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
6573 * gst/rtsp-server/rtsp-client.c:
6574 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
6575 https://bugzilla.gnome.org/show_bug.cgi?id=700953
6577 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
6579 * gst/rtsp-server/rtsp-sdp.c:
6580 rtsp-sdp: Parse framerate caps field and set SDP attribute
6581 The SDP attribute and its format is described in RFC4566.
6582 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
6584 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
6586 * gst/rtsp-server/rtsp-sdp.c:
6587 rtsp-sdp: Parse width/height from caps and set SDP attribute
6588 The SDP attribute and its format is described in RFC6064.
6589 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
6591 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
6593 * gst/rtsp-server/rtsp-sdp.c:
6594 * tests/check/gst/client.c:
6595 rtsp-sdp: add bandwidth line
6596 https://bugzilla.gnome.org/show_bug.cgi?id=699220
6598 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6601 Automatic update of common submodule
6602 From 5edcd85 to 098c0d7
6604 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
6606 * tests/check/gst/media.c:
6607 tests: add dynamic payloader prepare/unprepare check
6609 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6611 * gst/rtsp-server/rtsp-media.c:
6612 media: release lock when removing fakesink
6614 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6616 * gst/rtsp-server/rtsp-stream.c:
6617 stream: set elements to NULL before removing
6618 When removing a stream, set the elements to NULL first. This avoids
6619 element-is-not-in-NULL-state errors when we dispose the elements.
6621 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
6624 Automatic update of common submodule
6625 From 3cb3d3c to 5edcd85
6627 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6629 * gst/rtsp-server/rtsp-media.c:
6630 * gst/rtsp-server/rtsp-media.h:
6631 media: listen to pad-removed signals
6632 Listen to the pad-removed signal and remove the stream associated with the
6634 Add signal to be notified of the removed pad.
6635 Remove the fakesink in unprepare()
6636 Fix signatures of the signal methods
6638 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6640 * examples/test-sdp.c:
6641 tests: add example of reusable pipelines
6643 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
6645 * gst/rtsp-server/rtsp-stream.c:
6646 * gst/rtsp-server/rtsp-stream.h:
6647 stream: add method to get the srcpad
6649 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
6651 * tests/check/gst/media.c:
6652 check: add media prepare/unprepare test
6653 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
6655 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
6657 * gst/rtsp-server/rtsp-media.c:
6658 media: disconnect from signal handlers in unprepare()
6659 We connected to the pad-added and no-more-pads signals in prepare() so
6660 we need to disconnect from them in unprepare().
6661 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
6663 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
6665 * gst/rtsp-server/rtsp-media.c:
6666 media: don't free streams array
6667 Don't free the streams array in the unprepare() method, they were not
6669 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
6671 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
6673 * gst/rtsp-server/rtsp-media.c:
6674 media: don't unref the pipeline in unprepare
6675 Unprepare() should undo what prepare() does. Because the pipeline is
6676 not created in prepare(), we should not unref it in unprepare()
6678 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
6680 * gst/rtsp-server/rtsp-stream.c:
6681 stream: clear session and caps for reuse
6682 Set the session and caps to NULL after unref otherwise we might unref
6684 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
6686 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
6688 * gst/rtsp-server/rtsp-client.c:
6689 client: send out teardown signal before tearing down
6690 The advantage is that in the signal handler you get direct access to
6691 information about what streams are about to get torn down (in the
6692 GstRTSPClientState).
6693 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
6695 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
6697 * gst/rtsp-server/rtsp-client.c:
6698 * gst/rtsp-server/rtsp-client.h:
6699 client: expose connection
6700 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
6702 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
6705 Automatic update of common submodule
6706 From aed87ae to 3cb3d3c
6708 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6710 * gst/rtsp-server/rtsp-media.c:
6711 * gst/rtsp-server/rtsp-media.h:
6712 * gst/rtsp-server/rtsp-session-media.c:
6713 * gst/rtsp-server/rtsp-session-media.h:
6714 media: add method to get the base_time of the pipeline
6715 Together with a shared clock, this base-time could eventually be sent to
6716 the client so that it can reconstruct the exact running-time of the clock
6719 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6721 * gst/rtsp-server/Makefile.am:
6722 * gst/rtsp-server/rtsp-media.c:
6723 * gst/rtsp-server/rtsp-media.h:
6724 * gst/rtsp-server/rtsp-sdp.c:
6725 media: add GstNetTimeProvider support
6726 Add a property to let the media provide a GstNetTimeProvider for its clock.
6727 Make methods to get the clock and nettimeprovider
6728 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
6729 provider and also the current time of the clock. This should make it possible
6730 for (GStreamer) clients to slave their clock to the server clock.
6732 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
6735 Automatic update of common submodule
6736 From 04c7a1e to aed87ae
6738 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6740 * gst/rtsp-server/rtsp-media.c:
6741 media: wait for buffering to complete
6742 Wait for buffering to complete before changing the state to the target state.
6744 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6746 * gst/rtsp-server/rtsp-media.c:
6747 media: small cleanup
6749 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
6751 * tests/check/gst/rtspserver.c:
6752 tests: remove extra unref in test_setup_non_existing_stream
6753 The unref is not needed anymore, teardown runs without it.
6754 https://bugzilla.gnome.org/show_bug.cgi?id=696542
6756 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
6758 * tests/check/gst/rtspserver.c:
6759 tests: GSocketService cleanup in test_bind_already_in_use
6760 Use g_socket_service_stop so the rtspserver test stops listening for
6761 incoming connections in test_bind_already_in_use.
6762 https://bugzilla.gnome.org/show_bug.cgi?id=696541
6764 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
6766 * gst/rtsp-server/rtsp-media-factory.c:
6767 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
6768 Instead use a GWeakRef which is safe to use
6769 This is a known GLib bug, see:
6770 https://bugzilla.gnome.org/show_bug.cgi?id=667145
6772 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
6774 * gst/rtsp-server/rtsp-client.c:
6775 * gst/rtsp-server/rtsp-media.c:
6776 * gst/rtsp-server/rtsp-media.h:
6777 * gst/rtsp-server/rtsp-sdp.c:
6778 * tests/check/gst/media.c:
6779 * tests/check/gst/rtspserver.c:
6780 rtsp-media/client: Reply to PLAY request with same type of Range
6781 Remember the type of Range from the PLAY request and use the same type for
6784 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
6786 * gst/rtsp-server/rtsp-client.c:
6787 * gst/rtsp-server/rtsp-client.h:
6788 * tests/check/gst/client.c:
6789 rtsp-client: expose uri
6791 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
6793 * tests/check/gst/mediafactory.c:
6794 tests: Hold ref while creating second media
6795 To test if the media aren't shared, make sure we keep the first one while creating a second
6796 otherwise the same memory address may be reused.
6798 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
6801 configure: remove out-of-date comment
6803 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
6806 .gitignore: ignore more build files
6808 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
6810 * tests/check/Makefile.am:
6811 tests: use right _LIBS variable for gst-plugins-base libs
6813 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6815 * tests/check/Makefile.am:
6816 check: add librtp to libs
6818 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
6820 * tests/check/gst/rtspserver.c:
6821 tests: Add test to check selecting a port the server will send from
6823 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
6825 * tests/check/gst/rtspserver.c:
6826 tests: Make sure packets are actually received
6828 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
6830 * gst/rtsp-server/rtsp-stream.c:
6831 stream: Select unicast address from pool if appropriate
6833 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
6835 * gst/rtsp-server/rtsp-stream.c:
6836 stream: Properties are always there in Gst 1.0
6838 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
6840 * tests/check/gst/addresspool.c:
6841 tests: Add tests for unicast addresses in pool
6843 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
6845 * gst/rtsp-server/rtsp-address-pool.c:
6846 * tests/check/gst/addresspool.c:
6847 address-pool: Verify that multicast addresses are used for multicast and vice-versa
6849 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
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 * gst/rtsp-server/rtsp-stream.c:
6855 * tests/check/gst/addresspool.c:
6856 address-pool: Add unicast addresses
6858 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
6861 * gst/rtsp-server/rtsp-server.c:
6862 * tests/check/gst/rtspserver.c:
6863 rtsp-server: Limit the number of threads per server instance
6864 If we exceed the maximum, just round robin the clients over the existing
6867 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
6869 * gst/rtsp-server/rtsp-server.c:
6870 rtsp-server: No need to store the GMainContext in the client context
6872 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
6874 * tests/check/gst/rtspserver.c:
6875 tests: Add test for client disconnection
6877 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
6879 * tests/check/gst/rtspserver.c:
6880 tests: Test client and session timeouts with multiple threads
6882 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
6884 * gst/rtsp-server/rtsp-address-pool.c:
6885 * gst/rtsp-server/rtsp-auth.c:
6886 * gst/rtsp-server/rtsp-client.c:
6887 * gst/rtsp-server/rtsp-media-factory-uri.c:
6888 * gst/rtsp-server/rtsp-media-factory.c:
6889 * gst/rtsp-server/rtsp-media.c:
6890 * gst/rtsp-server/rtsp-mount-points.c:
6891 * gst/rtsp-server/rtsp-server.c:
6892 * gst/rtsp-server/rtsp-session-media.c:
6893 * gst/rtsp-server/rtsp-session-pool.c:
6894 * gst/rtsp-server/rtsp-session.c:
6895 Document locking and its order
6897 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
6899 * tests/check/gst/rtspserver.c:
6900 tests: Test that slow DESCRIBE don't block other clients
6902 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
6904 * tests/check/gst/client.c:
6905 tests: Add tests for client-requested multicast address
6907 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
6909 * docs/libs/gst-rtsp-server-sections.txt:
6910 docs: Put the various functions in the right sections
6912 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
6914 * docs/libs/gst-rtsp-server-docs.sgml:
6915 * docs/libs/gst-rtsp-server-sections.txt:
6916 * gst/rtsp-server/rtsp-address-pool.c:
6917 * gst/rtsp-server/rtsp-address-pool.h:
6918 docs: Generate docs for GstRTSPAddressPool
6920 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
6922 * gst/rtsp-server/rtsp-client.c:
6923 * gst/rtsp-server/rtsp-stream.c:
6924 * gst/rtsp-server/rtsp-stream.h:
6925 client: Check client provided addresses against the address pool
6927 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
6929 * gst/rtsp-server/rtsp-address-pool.c:
6930 * gst/rtsp-server/rtsp-address-pool.h:
6931 * tests/check/gst/addresspool.c:
6932 address-pool: Add API to request a specific address from the pool
6933 Also add relevant unit tests.
6935 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
6937 * tests/check/gst/mediafactory.c:
6938 tests: Check the passing around of a RTSPAddressPool
6939 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
6940 way down to the stream.
6942 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
6944 * tests/check/gst/addresspool.c:
6945 tests: Add more tests for the address pool
6947 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
6949 * gst/rtsp-server/rtsp-address-pool.c:
6950 address-pool: Fix off by one error
6951 When splitting a port range, the port after a skip is not part of range.
6953 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
6956 Automatic update of common submodule
6957 From 2de221c to 04c7a1e
6959 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
6962 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
6963 AM_CONFIG_HEADER was removed in automake 1.13
6964 https://bugzilla.gnome.org/show_bug.cgi?id=693368
6966 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
6969 Automatic update of common submodule
6970 From a942293 to 2de221c
6972 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6974 * gst/rtsp-server/rtsp-client.c:
6975 client: make sure the watch exists while sending data
6976 Protect the send_func with a lock. This allows us to wait for sending
6977 to complete before changing the send_func and user_data. We add an
6978 extra ref to the watch to make sure that it remains valid during
6980 When closing the connection, set the send_func to NULL
6981 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
6983 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6985 * tests/check/Makefile.am:
6986 tests: use GST_*_1_0 environment variables everywhere
6987 The _1_0 suffixed environment variables override the
6988 non-suffixed ones, so if we're in an environment that
6989 sets the _1_0 suffixed ones, such as jhbuild, we need
6990 to set those to make sure ours actually always get
6993 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6996 Automatic update of common submodule
6997 From acb04d9 to a942293
6999 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7001 * gst/rtsp-server/rtsp-client.c:
7002 rtsp-client: set the client backlog
7003 Set the client backlog to a reasonable default
7005 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
7007 * gst/rtsp-server/rtsp-media.c:
7008 rtsp-media: Make the element a constructor parameter
7009 https://bugzilla.gnome.org/show_bug.cgi?id=689594
7011 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7013 * docs/libs/Makefile.am:
7014 docs: Link with gcov library when gcov is enabled
7015 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
7017 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7019 * gst/rtsp-server/rtsp-media.c:
7020 media: match prepare with unprepare
7021 Really unprepare when there were an equal amount of prepare calls.
7023 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7025 * gst/rtsp-server/rtsp-media.c:
7026 media: media has to be unprepared in finalize
7027 Because unprepare takes away the last ref on the media.
7029 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7031 * gst/rtsp-server/rtsp-client.c:
7032 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
7033 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
7034 We can't use the refcount to trigger unprepare because it is the unprepare call
7035 that removes the last refcount after all messages are consumed. What we should
7036 probably do is make a prepared refcount and only unprepare when the refcount
7039 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7041 * gst/rtsp-server/rtsp-media.c:
7042 media: let the source unref the last media ref
7043 the last ref to the media is held by the source so we don't need to add more ref
7044 and unrefs, we simply destroy the media when the source is gone.
7046 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7048 * gst/rtsp-server/rtsp-media.c:
7049 media: improve debug
7051 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7053 * gst/rtsp-server/rtsp-media.c:
7055 Make sure we are in the right state when collecting the position and duration.
7056 Only make ourselves PREPARED when we were previously PREPARING.
7058 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7060 * gst/rtsp-server/rtsp-media.c:
7061 media: use g_object_ref/unref for GObjects
7063 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
7065 * gst/rtsp-server/rtsp-client.c:
7066 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
7067 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
7068 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
7069 isn't being used anymore.
7071 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
7073 * gst/rtsp-server/rtsp-media.c:
7074 Fix compiler warning
7076 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
7078 * gst/rtsp-server/rtsp-media-factory-uri.c:
7079 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
7081 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7083 * gst/rtsp-server/rtsp-session-media.h:
7086 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7088 * gst/rtsp-server/rtsp-media.c:
7089 * tests/check/gst/media.c:
7090 media: avoid element leak
7092 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7094 * gst/rtsp-server/rtsp-media.c:
7095 media: require an element in media constructor
7097 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7099 * gst/rtsp-server/rtsp-client.c:
7100 Revert "client: TEARDOWN brings that state to Init again"
7101 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
7102 The object is already disposed, there is no point in setting the state.
7104 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7106 * gst/rtsp-server/rtsp-client.c:
7107 client: TEARDOWN brings that state to Init again
7109 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7111 * docs/libs/gst-rtsp-server-sections.txt:
7112 * examples/test-auth.c:
7113 * gst/rtsp-server/rtsp-auth.c:
7114 * gst/rtsp-server/rtsp-auth.h:
7115 * gst/rtsp-server/rtsp-client.c:
7116 * gst/rtsp-server/rtsp-client.h:
7117 * gst/rtsp-server/rtsp-media-factory-uri.c:
7118 * gst/rtsp-server/rtsp-media-factory-uri.h:
7119 * gst/rtsp-server/rtsp-media-factory.c:
7120 * gst/rtsp-server/rtsp-media-factory.h:
7121 * gst/rtsp-server/rtsp-media.c:
7122 * gst/rtsp-server/rtsp-media.h:
7123 * gst/rtsp-server/rtsp-mount-points.c:
7124 * gst/rtsp-server/rtsp-mount-points.h:
7125 * gst/rtsp-server/rtsp-sdp.c:
7126 * gst/rtsp-server/rtsp-server.c:
7127 * gst/rtsp-server/rtsp-server.h:
7128 * gst/rtsp-server/rtsp-session-media.c:
7129 * gst/rtsp-server/rtsp-session-media.h:
7130 * gst/rtsp-server/rtsp-session-pool.c:
7131 * gst/rtsp-server/rtsp-session-pool.h:
7132 * gst/rtsp-server/rtsp-session.c:
7133 * gst/rtsp-server/rtsp-session.h:
7134 * gst/rtsp-server/rtsp-stream-transport.c:
7135 * gst/rtsp-server/rtsp-stream-transport.h:
7136 * gst/rtsp-server/rtsp-stream.c:
7137 * gst/rtsp-server/rtsp-stream.h:
7138 * tests/check/gst/media.c:
7139 rtsp: make object details private
7140 Make all object details private
7141 Add methods to access private bits
7143 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7145 * tests/check/Makefile.am:
7146 * tests/check/gst/media.c:
7147 tests: add media tests
7149 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7151 * gst/rtsp-server/rtsp-media.c:
7152 media: check if prepared for some methods
7153 Check that the media object is prepared before doing seek and getting the
7154 current position etc.
7155 Add some g_return checks.
7157 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7159 * tests/check/Makefile.am:
7160 * tests/check/gst/mediafactory.c:
7161 tests: add mediafactory test
7163 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7165 * gst/rtsp-server/rtsp-stream.c:
7166 stream: improve debug
7168 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7170 * gst/rtsp-server/rtsp-media.c:
7171 * gst/rtsp-server/rtsp-media.h:
7172 media: unref pipeline in finalize to avoid leaking it
7174 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7176 * gst/rtsp-server/rtsp-media-factory-uri.c:
7177 * gst/rtsp-server/rtsp-media.c:
7178 rtsp: use gst_object_unref on GstObjects
7180 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7182 * gst/rtsp-server/rtsp-media-factory.c:
7183 media-factory: require an url
7185 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7187 * examples/test-uri.c:
7188 examples: fix include
7190 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7192 * gst/rtsp-server/rtsp-server.h:
7193 server: remove unused include
7195 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7197 * tests/check/Makefile.am:
7198 * tests/check/gst/mountpoints.c:
7199 tests: add test for mountpoints
7201 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7203 * gst/rtsp-server/rtsp-client.c:
7204 client: fix factory leak
7205 Keep the factory in the state object only for authorization checks and make
7206 sure we unref it on failure. Also don't keep invalid objects in the state
7209 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7211 * gst/rtsp-server/rtsp-mount-points.c:
7212 mounts: add g_return_if guards
7214 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7216 * tests/check/gst/client.c:
7217 tests: add more tests
7219 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7221 * gst/rtsp-server/rtsp-client.c:
7222 client: improve debug
7224 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7226 * gst/rtsp-server/rtsp-client.c:
7227 client: improve debug and fix leaks
7228 Cleanup the uri and session when there is a bad request.
7230 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7235 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7237 * tests/check/gst/client.c:
7238 test: add test for session in options request
7240 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7242 * gst/rtsp-server/rtsp-client.c:
7243 client: use 454 when session can't be found
7244 We should use 454 when a session can't be found because there was no session
7245 pool configured in the server. This is not a server configuration problem
7246 because the server on which the request is done might not be the same one that
7247 will keep the sessions for us and so it does not need to support sessions.
7249 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7251 * gst/rtsp-server/rtsp-client.c:
7252 client: only free connection when there is one
7253 It's possible that the client doesn't have a connection when we try to free it.
7255 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7257 * tests/check/Makefile.am:
7258 * tests/check/gst/client.c:
7259 tests: add unit test for the client object
7261 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7263 * gst/rtsp-server/rtsp-client.c:
7264 client: small cleanup
7266 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7268 * gst/rtsp-server/rtsp-client.h:
7269 client: remove unused include
7271 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7273 * gst/rtsp-server/rtsp-client.c:
7274 client: fix compilation
7276 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7278 * gst/rtsp-server/rtsp-client.c:
7279 client: call destroy without the lock
7281 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7283 * gst/rtsp-server/rtsp-client.c:
7284 * gst/rtsp-server/rtsp-client.h:
7285 client: make the client usable without a socket
7286 Make a method to let the client handle a message and a callback when the client
7287 wants us to send a response message back. This makes it possible to also use the
7288 client object without the sockets, which should make it easier to test.
7290 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7292 * gst/rtsp-server/rtsp-client.c:
7293 * gst/rtsp-server/rtsp-client.h:
7294 client: small cleanup
7296 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7298 * docs/libs/gst-rtsp-server-sections.txt:
7299 * gst/rtsp-server/rtsp-client.c:
7300 * gst/rtsp-server/rtsp-client.h:
7301 * gst/rtsp-server/rtsp-server.c:
7302 client: remove reference to server
7303 We don't need to keep a ref to the server
7305 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7307 * gst/rtsp-server/rtsp-client.c:
7308 * gst/rtsp-server/rtsp-client.h:
7310 Also add some g_return_if()
7312 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7314 * gst/rtsp-server/rtsp-client.c:
7315 client: log more errors
7317 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7319 * gst/rtsp-server/rtsp-client.c:
7320 client: fix compilation
7322 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7324 * gst/rtsp-server/rtsp-client.c:
7325 * gst/rtsp-server/rtsp-client.h:
7326 client: add generic close-after-send support
7327 Add a property to send_response() to close the connection after the response has
7328 been sent to the client.
7330 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7333 * docs/libs/gst-rtsp-server-docs.sgml:
7334 * docs/libs/gst-rtsp-server-sections.txt:
7335 * docs/libs/gst-rtsp-server.types:
7336 * examples/test-auth.c:
7337 * examples/test-launch.c:
7338 * examples/test-mp4.c:
7339 * examples/test-multicast.c:
7340 * examples/test-multicast2.c:
7341 * examples/test-ogg.c:
7342 * examples/test-readme.c:
7343 * examples/test-sdp.c:
7344 * examples/test-uri.c:
7345 * examples/test-video.c:
7346 * gst/rtsp-server/Makefile.am:
7347 * gst/rtsp-server/rtsp-auth.h:
7348 * gst/rtsp-server/rtsp-client.c:
7349 * gst/rtsp-server/rtsp-client.h:
7350 * gst/rtsp-server/rtsp-media-mapping.c:
7351 * gst/rtsp-server/rtsp-media-mapping.h:
7352 * gst/rtsp-server/rtsp-mount-points.c:
7353 * gst/rtsp-server/rtsp-mount-points.h:
7354 * gst/rtsp-server/rtsp-server.c:
7355 * gst/rtsp-server/rtsp-server.h:
7356 * gst/rtsp-server/rtsp-session-media.c:
7357 * gst/rtsp-server/rtsp-session-pool.c:
7358 * gst/rtsp-server/rtsp-session-pool.h:
7359 * tests/check/gst/rtspserver.c:
7360 MediaMapping -> MountPoints
7361 Describes better what the object manages.
7363 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7366 configure: bump required version of -base
7368 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7370 * gst/rtsp-server/rtsp-media.c:
7373 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7375 * gst/rtsp-server/rtsp-media.c:
7376 * gst/rtsp-server/rtsp-media.h:
7377 media: support more Range formats
7378 Use the new -base methods to convert the Range string into a seek start and stop
7381 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7383 * examples/test-launch.c:
7384 examples: fix whitespace
7386 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7388 * examples/test-auth.c:
7389 test-auth: add example of how to remove sessions
7390 Add an example of the session filter api.
7392 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7394 * examples/test-uri.c:
7395 test-uri: remove mapping example
7397 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7399 * examples/test-uri.c:
7400 test-uri: fix callback signature
7402 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7404 * gst/rtsp-server/rtsp-media-factory.c:
7405 factory: keep ref to factory while media active
7406 While the media from a factory is alive, keep a ref to the factory.
7407 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
7409 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7411 * gst/rtsp-server/rtsp-media-factory-uri.c:
7412 factory-uri: add some debug
7414 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7416 * gst/rtsp-server/rtsp-stream.c:
7417 stream: set udp sources to PLAYING
7418 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
7419 so that it doesn't cause our pipeline to produce ASYNC-DONE.
7421 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7423 * gst/rtsp-server/rtsp-media-factory-uri.c:
7424 factory-uri: take ref to factory
7425 Take a ref to the factory that we place in our list.
7427 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7429 * tests/Makefile.am:
7430 * tests/test-reuse.c:
7431 test: add test for server reuse
7432 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
7434 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
7436 * gst/rtsp-server/rtsp-server.c:
7437 server: start and stop multiple times
7438 Stop listening on the RTSP port when the GSource is removed, so clients
7439 can't connect and the server can be started again.
7440 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
7442 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7444 * gst/rtsp-server/rtsp-server.c:
7445 server: fix small leak
7447 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7449 * gst/rtsp-server/rtsp-media.c:
7450 media: unref source in finish_unprepare
7451 The source is created in prepare, unref it in finish_unprepare.
7452 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
7454 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
7456 * gst/rtsp-server/rtsp-client.c:
7457 * gst/rtsp-server/rtsp-media.c:
7458 rtsp-media: remove bus watch before finalizing
7459 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
7460 * An extra media ref is added for the bus watch. This extra ref is unreffed by
7461 the GDestroyNotify function.
7462 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
7463 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
7464 gst_rtsp_media_unprepare before unreffing the media.
7465 This way, the bus watch will be removed before the media is finalized.
7466 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
7468 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
7470 * gst/rtsp-server/rtsp-client.c:
7471 * gst/rtsp-server/rtsp-client.h:
7472 client: wait until the TEARDOWN response is sent to close the connection
7473 Responses can be sent async so we need to wait until the TEARDOWN response has
7474 been written before we close the connection to the client. This avoids the risk
7475 of writing/polling closed sockets.
7476 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
7478 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
7480 * gst/rtsp-server/rtsp-stream.c:
7481 rtsp-stream: plug socket leak
7482 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
7484 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
7487 Automatic update of common submodule
7488 From 6bb6951 to a72faea
7490 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
7492 * gst/rtsp-server/rtsp-media-factory-uri.c:
7493 rtsp-server: don't use deprecated API
7495 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
7497 * gst/rtsp-server/rtsp-client.c:
7498 rtsp-client: fix unused-but-set-variable compiler warning
7499 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
7501 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7504 * docs/libs/gst-rtsp-server-sections.txt:
7505 * gst/rtsp-server/rtsp-client.c:
7508 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7510 * examples/Makefile.am:
7511 * examples/test-multicast2.c:
7512 examples: add another multicast example
7513 Add an example for how to configure separate multicast ranges for each media
7516 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7518 * examples/test-multicast.c:
7521 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7523 * gst/rtsp-server/rtsp-client.c:
7524 * gst/rtsp-server/rtsp-media.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 stream: use the address managed by the stream
7530 Use the address managed by the stream for multicast. This allows us to have 1
7531 multicast address for each stream.
7532 Because the address is now managed by the stream we don't have to pass it around
7534 Set the address pool on the streams.
7536 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7538 * gst/rtsp-server/rtsp-client.c:
7539 * gst/rtsp-server/rtsp-media.c:
7540 * gst/rtsp-server/rtsp-stream.c:
7543 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7545 * gst/rtsp-server/rtsp-media.c:
7546 * gst/rtsp-server/rtsp-media.h:
7547 media: add signal for new streams
7548 This allows applications to listen for new streams and configure properties on
7549 them, like the address pool.
7551 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7553 * gst/rtsp-server/rtsp-media.c:
7554 media: configure address pool in new streams
7556 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7558 * gst/rtsp-server/rtsp-stream.c:
7559 * gst/rtsp-server/rtsp-stream.h:
7560 stream: add methods to deal with address pool
7561 Add methods to get and set the address pool for the stream
7562 Add method to allocate and get the multicast addresses for this stream.
7564 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7566 * docs/libs/gst-rtsp-server-sections.txt:
7567 * gst/rtsp-server/rtsp-media.c:
7568 * gst/rtsp-server/rtsp-media.h:
7569 media: remove MTU property
7570 It is a stream property
7572 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7574 * gst/rtsp-server/rtsp-client.c:
7575 client: set blocksize only on stream
7576 Set the blocksize only on the current stream.
7578 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7580 * gst/rtsp-server/rtsp-stream.c:
7581 stream: share src and sink sockets
7582 the allocated socket is in the used-socket property, not socket.
7584 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7586 * gst/rtsp-server/rtsp-address-pool.c:
7587 * gst/rtsp-server/rtsp-address-pool.h:
7588 * gst/rtsp-server/rtsp-client.c:
7589 * gst/rtsp-server/rtsp-session-media.c:
7590 * gst/rtsp-server/rtsp-session-media.h:
7591 * gst/rtsp-server/rtsp-stream-transport.c:
7592 * gst/rtsp-server/rtsp-stream-transport.h:
7593 * tests/check/gst/addresspool.c:
7594 rtsp: make address-pool return an address object
7595 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
7596 store more info in the structure and allows us to more easily return the address
7597 to the right pool when no longer needed.
7598 Pass the address to the StreamTransport so that we can return it to the pool
7599 when the stream transport is freed or changed.
7601 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7603 * examples/Makefile.am:
7604 * examples/test-multicast.c:
7605 examples: add multicast example
7606 Show how to set up the multicast address pool so that media can be
7607 server with multicast.
7609 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7611 * gst/rtsp-server/rtsp-client.c:
7612 * gst/rtsp-server/rtsp-media-factory.c:
7613 * gst/rtsp-server/rtsp-media-factory.h:
7614 * gst/rtsp-server/rtsp-media.c:
7615 * gst/rtsp-server/rtsp-media.h:
7616 rtsp: use AddressPool
7617 Remove the multicast_group property.
7618 Use the configured addresspool to allocate multicast addresses.
7620 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7622 * gst/rtsp-server/rtsp-address-pool.c:
7623 * gst/rtsp-server/rtsp-address-pool.h:
7624 address-pool: add clear method
7626 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7628 * gst/rtsp-server/rtsp-address-pool.c:
7629 address-pool: small cleanups
7631 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7633 * tests/check/Makefile.am:
7634 * tests/check/gst/addresspool.c:
7635 tests: add addresspool unit test
7637 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7639 * gst/rtsp-server/Makefile.am:
7640 * gst/rtsp-server/rtsp-address-pool.c:
7641 * gst/rtsp-server/rtsp-address-pool.h:
7642 address-pool: add object to manage multicast addresses
7643 Make an object that can manage a rage of multicast addresses and ports.
7645 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7647 * gst/rtsp-server/rtsp-server.c:
7648 server: set default max-threads property
7650 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7652 * gst/rtsp-server/rtsp-media.c:
7653 media: wait for concurrent _prepare
7654 If a prepare is busy, wait for the result.
7656 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7658 * gst/rtsp-server/rtsp-media.c:
7659 media: add lock around message handler
7660 We don't want to dispatch messages while we are still processing the result of
7663 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7665 * gst/rtsp-server/rtsp-media.c:
7666 * gst/rtsp-server/rtsp-media.h:
7667 media: add lock to protect state changes
7669 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7671 * gst/rtsp-server/rtsp-stream.c:
7672 * gst/rtsp-server/rtsp-stream.h:
7675 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7677 * gst/rtsp-server/rtsp-stream-transport.c:
7678 * gst/rtsp-server/rtsp-stream-transport.h:
7679 * gst/rtsp-server/rtsp-stream.c:
7680 stream-transport: add keep-alive method
7682 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7684 * gst/rtsp-server/rtsp-stream-transport.c:
7685 * gst/rtsp-server/rtsp-stream-transport.h:
7686 * gst/rtsp-server/rtsp-stream.c:
7687 stream-transport: add method to handle RTP/RTCP
7688 Call new methods instead of poking into the structures directly.
7690 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7692 * gst/rtsp-server/rtsp-session-media.c:
7693 * gst/rtsp-server/rtsp-session-media.h:
7694 session-media: add locking
7696 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7698 * gst/rtsp-server/rtsp-session.c:
7699 * gst/rtsp-server/rtsp-session.h:
7700 session: add locking
7702 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7704 * gst/rtsp-server/rtsp-server.c:
7705 server: free old socket
7707 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7709 * gst/rtsp-server/rtsp-media-mapping.c:
7710 * gst/rtsp-server/rtsp-media-mapping.h:
7711 mapping: add locking
7713 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7715 * gst/rtsp-server/rtsp-media-factory.c:
7716 media-factory: add locking
7718 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7720 * gst/rtsp-server/rtsp-auth.c:
7721 * gst/rtsp-server/rtsp-auth.h:
7724 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7726 * gst/rtsp-server/rtsp-server.c:
7727 * gst/rtsp-server/rtsp-server.h:
7728 server: add max-thread property
7730 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7732 * gst/rtsp-server/rtsp-server.c:
7733 * gst/rtsp-server/rtsp-server.h:
7734 server: use a threadpool for the mainloops
7736 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7738 * gst/rtsp-server/rtsp-client.c:
7739 * gst/rtsp-server/rtsp-client.h:
7740 client: rename method
7741 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
7742 don't really create the client from the socket, we use the socket for the
7745 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7747 * gst/rtsp-server/rtsp-client.c:
7748 * gst/rtsp-server/rtsp-client.h:
7749 * gst/rtsp-server/rtsp-server.c:
7750 server: rework maincontext handling in clients
7751 Make a separate method to attach a client to a MainContext.
7752 Let the server decide in what GMainContext the client will operate and give this
7753 context to the client in attach. Then the server can later decide to use a
7754 separate thread for each client or just use the mainthread.
7756 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7758 * gst/rtsp-server/rtsp-client.c:
7759 * gst/rtsp-server/rtsp-session.c:
7760 * gst/rtsp-server/rtsp-session.h:
7761 session: move session header code in session object
7763 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
7767 * examples/test-auth.c:
7768 * examples/test-launch.c:
7769 * examples/test-mp4.c:
7770 * examples/test-ogg.c:
7771 * examples/test-readme.c:
7772 * examples/test-sdp.c:
7773 * examples/test-uri.c:
7774 * examples/test-video.c:
7775 * gst/rtsp-server/rtsp-auth.c:
7776 * gst/rtsp-server/rtsp-auth.h:
7777 * gst/rtsp-server/rtsp-client.c:
7778 * gst/rtsp-server/rtsp-client.h:
7779 * gst/rtsp-server/rtsp-media-factory-uri.c:
7780 * gst/rtsp-server/rtsp-media-factory-uri.h:
7781 * gst/rtsp-server/rtsp-media-factory.c:
7782 * gst/rtsp-server/rtsp-media-factory.h:
7783 * gst/rtsp-server/rtsp-media-mapping.c:
7784 * gst/rtsp-server/rtsp-media-mapping.h:
7785 * gst/rtsp-server/rtsp-media.c:
7786 * gst/rtsp-server/rtsp-media.h:
7787 * gst/rtsp-server/rtsp-params.c:
7788 * gst/rtsp-server/rtsp-params.h:
7789 * gst/rtsp-server/rtsp-sdp.c:
7790 * gst/rtsp-server/rtsp-sdp.h:
7791 * gst/rtsp-server/rtsp-server.c:
7792 * gst/rtsp-server/rtsp-server.h:
7793 * gst/rtsp-server/rtsp-session-media.c:
7794 * gst/rtsp-server/rtsp-session-media.h:
7795 * gst/rtsp-server/rtsp-session-pool.c:
7796 * gst/rtsp-server/rtsp-session-pool.h:
7797 * gst/rtsp-server/rtsp-session.c:
7798 * gst/rtsp-server/rtsp-session.h:
7799 * gst/rtsp-server/rtsp-stream-transport.c:
7800 * gst/rtsp-server/rtsp-stream-transport.h:
7801 * gst/rtsp-server/rtsp-stream.c:
7802 * gst/rtsp-server/rtsp-stream.h:
7803 * tests/check/gst/rtspserver.c:
7804 * tests/test-cleanup.c:
7807 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7809 * gst/rtsp-server/rtsp-media.c:
7810 * gst/rtsp-server/rtsp-session-media.c:
7811 * gst/rtsp-server/rtsp-session.c:
7812 rtsp-server: added annotations to indicate type of ownership transfer of return values
7813 https://bugzilla.gnome.org/show_bug.cgi?id=680777
7815 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
7818 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
7820 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
7823 * bindings/Makefile.am:
7824 * bindings/vala/Makefile.am:
7825 * bindings/vala/gst-rtsp-server-0.10.deps:
7826 * bindings/vala/gst-rtsp-server-0.10.vapi:
7827 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
7828 * bindings/vala/packages/gst-rtsp-server-0.10.files:
7829 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7830 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7831 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
7833 bindings: remove vala bindings
7834 They'll be reunited with the other GStreamer bindings
7835 https://bugzilla.gnome.org/show_bug.cgi?id=680777
7837 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7839 * gst/rtsp-server/rtsp-client.c:
7840 * gst/rtsp-server/rtsp-session-media.c:
7841 * gst/rtsp-server/rtsp-session-media.h:
7842 * gst/rtsp-server/rtsp-stream-transport.c:
7843 * gst/rtsp-server/rtsp-stream-transport.h:
7844 rtsp: only create transport when needed
7845 Only create the StreamTransport when configured.
7847 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7849 * gst/rtsp-server/rtsp-client.c:
7850 client: small cleanup
7852 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7854 * gst/rtsp-server/rtsp-client.c:
7855 * gst/rtsp-server/rtsp-client.h:
7856 * gst/rtsp-server/rtsp-stream-transport.c:
7857 * gst/rtsp-server/rtsp-stream-transport.h:
7858 rtsp: refactor configuration of transport
7859 Move the configuration of the transport to a place where it makes
7862 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7864 * gst/rtsp-server/rtsp-client.c:
7865 client: refactor transport parsing
7867 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7869 * gst/rtsp-server/rtsp-client.c:
7870 client: refuse to change the MTU on shared media
7871 If we change the MTU of chared media, it changes for all clients.
7872 We don't want to set the MTU to something large for clients that
7875 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7877 * examples/test-mp4.c:
7878 * gst/rtsp-server/rtsp-media.c:
7879 small fixes to docs and debug
7881 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7883 * gst/rtsp-server/rtsp-stream.c:
7884 stream: transports must already have been removed
7886 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7888 * gst/rtsp-server/rtsp-media.c:
7889 * gst/rtsp-server/rtsp-stream.c:
7890 * gst/rtsp-server/rtsp-stream.h:
7891 stream: improve join and leave of the pipeline
7893 Do the cleanup properly
7896 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7898 * gst/rtsp-server/rtsp-media.c:
7899 media: move unprepare below default implementation
7900 Makes it easier to find the default implementation
7902 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7904 * gst/rtsp-server/rtsp-media.c:
7905 media: signal unprepared when we actually finish
7907 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7909 * gst/rtsp-server/rtsp-media.c:
7910 media: no need to unlock, unprepare does that when needed
7912 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7914 * docs/libs/gst-rtsp-server-sections.txt:
7915 * gst/rtsp-server/rtsp-media-factory.h:
7916 * gst/rtsp-server/rtsp-media-mapping.c:
7917 * gst/rtsp-server/rtsp-media.h:
7918 * gst/rtsp-server/rtsp-params.c:
7919 * gst/rtsp-server/rtsp-server.c:
7920 * gst/rtsp-server/rtsp-session-pool.h:
7921 * gst/rtsp-server/rtsp-session.c:
7922 * gst/rtsp-server/rtsp-session.h:
7923 * gst/rtsp-server/rtsp-stream-transport.h:
7924 * gst/rtsp-server/rtsp-stream.h:
7927 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7929 * gst/rtsp-server/rtsp-client.c:
7930 * gst/rtsp-server/rtsp-media-mapping.h:
7931 * gst/rtsp-server/rtsp-media.c:
7932 * gst/rtsp-server/rtsp-media.h:
7933 * gst/rtsp-server/rtsp-server.h:
7934 * gst/rtsp-server/rtsp-stream.c:
7935 * gst/rtsp-server/rtsp-stream.h:
7936 rtsp: fix MTU setting
7937 Fix setting of the MTU. There is no need for a vmethod.
7939 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7944 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7947 configure: bump version number after refactoring
7949 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7951 * gst/rtsp-server/Makefile.am:
7952 * gst/rtsp-server/rtsp-client.c:
7953 * gst/rtsp-server/rtsp-client.h:
7954 * gst/rtsp-server/rtsp-media-factory-uri.c:
7955 * gst/rtsp-server/rtsp-media-factory.c:
7956 * gst/rtsp-server/rtsp-media-factory.h:
7957 * gst/rtsp-server/rtsp-media.c:
7958 * gst/rtsp-server/rtsp-media.h:
7959 * gst/rtsp-server/rtsp-sdp.c:
7960 * gst/rtsp-server/rtsp-session-media.c:
7961 * gst/rtsp-server/rtsp-session-media.h:
7962 * gst/rtsp-server/rtsp-session.c:
7963 * gst/rtsp-server/rtsp-session.h:
7964 * gst/rtsp-server/rtsp-stream-transport.c:
7965 * gst/rtsp-server/rtsp-stream-transport.h:
7966 * gst/rtsp-server/rtsp-stream.c:
7967 * gst/rtsp-server/rtsp-stream.h:
7968 rtsp: massive refactoring
7969 Make GObjects from the remaining simple structures.
7970 Remove GstRTSPSessionStream, it's not needed.
7971 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
7972 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
7973 a GstRTSPStream should be transported to a client.
7974 Rename GstRTSPMediaFactory::get_element -> create_element because that
7975 more accurately describes what it does.
7976 Make nice methods instead of poking in the structures.
7977 Move some methods inside the relevant object source code.
7978 Use GPtrArray to store objects instead of plain arrays, it is more
7979 natural and allows us to more easily clean up.
7980 Move the allocation of udp ports to the Stream object. The Stream object
7981 contains the elements needed to stream the media to a client.
7982 Improve the prepare and unprepare methods. Unprepare should now undo
7983 everything prepare did. Improve also async unprepare when doing EOS on
7984 shutdown. Make sure we always unprepare correctly.
7986 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
7988 * gst/rtsp-server/rtsp-client.c:
7989 rtsp-client: Unref server address clients connected to
7990 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
7992 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
7994 * gst/rtsp-server/rtsp-server.c:
7995 rtsp-server: don't ref server socket if it is NULL
7996 Fixes test_bind_already_in_use unit test again after commit 6a497440.
7997 https://bugzilla.gnome.org/show_bug.cgi?id=686644
7999 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
8001 * tests/check/Makefile.am:
8002 tests: Add libgio link dependency
8003 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
8005 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8007 * gst/rtsp-server/rtsp-media-mapping.c:
8008 * gst/rtsp-server/rtsp-media-mapping.h:
8009 rtsp-media-mapping: rename find_media vfunc to find_factory
8010 The virtual method and class method should have the same name
8011 so it is correctly represented in GIR file
8012 https://bugzilla.gnome.org/show_bug.cgi?id=680777
8014 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8016 * gst/rtsp-server/rtsp-auth.c:
8017 * gst/rtsp-server/rtsp-client.c:
8018 * gst/rtsp-server/rtsp-media-factory-uri.c:
8019 * gst/rtsp-server/rtsp-media-factory.c:
8020 * gst/rtsp-server/rtsp-media-mapping.c:
8021 * gst/rtsp-server/rtsp-media.c:
8022 * gst/rtsp-server/rtsp-server.c:
8023 * gst/rtsp-server/rtsp-session-pool.c:
8024 * gst/rtsp-server/rtsp-session.c:
8025 rtsp-server: fixed comments and GIR annotations
8026 https://bugzilla.gnome.org/show_bug.cgi?id=680777
8028 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
8030 * gst/rtsp-server/rtsp-media-mapping.c:
8031 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
8033 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
8035 * gst/rtsp-server/rtsp-server.c:
8036 rtsp-server: allow binding on port 0 (binds on a random port)
8038 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
8040 * gst/rtsp-server/rtsp-server.c:
8041 * gst/rtsp-server/rtsp-server.h:
8042 rtsp-server: add bound-port property
8043 bound-port can be used to retrieve the port number when the server is bound on
8044 port 0, which binds on a random port.
8046 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
8048 * gst/rtsp-server/rtsp-media-factory.c:
8049 * gst/rtsp-server/rtsp-media-factory.h:
8050 rtsp-media-factory: make ::get_element overridable by GI bindings
8051 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
8052 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
8053 as the invoker for ::get_element(), making it overridable by GI generated
8056 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
8058 * gst/rtsp-server/rtsp-media-factory-uri.c:
8059 rtsp-media-factory-uri: don't autoplug parsers in a loop
8060 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
8063 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
8065 * gst/rtsp-server/Makefile.am:
8066 Explicitly link against gio. Fix link error on mac.
8068 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
8070 * gst/rtsp-server/rtsp-session.c:
8071 session: add ttl to the transport header in SETUP
8072 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
8074 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
8076 * gst/rtsp-server/rtsp-client.c:
8077 * gst/rtsp-server/rtsp-client.h:
8078 * gst/rtsp-server/rtsp-media.c:
8079 client: Use client transport settings for multicast if allowed.
8080 This patch makes it possible for the client to send transport settings for
8081 multicast (destination && ttl). Client settings must be explicitly allowed or
8082 the server will use its own settings.
8083 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
8085 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
8088 Automatic update of common submodule
8089 From 6c0b52c to 6bb6951
8091 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
8093 * gst/rtsp-server/rtsp-client.c:
8094 rtsp-client: do not destroy the rtsp watch
8095 Don't destroy the client watch while dispatching. The rtsp watch is
8096 automatically destroyed after the rtsp watch function closed() has
8098 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
8100 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
8103 Automatic update of common submodule
8104 From 4f962f7 to 6c0b52c
8106 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
8108 * gst/rtsp-server/rtsp-media.c:
8109 media: fix check for seekability
8111 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8113 * gst/rtsp-server/rtsp-client.c:
8114 client: use more GIO
8115 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
8117 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8119 * gst/rtsp-server/rtsp-server.c:
8120 server: remove obsolete includes
8122 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
8124 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
8125 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
8126 be available in "on_new_ssrc". The transports are added in
8127 gst_rtsp_media_set_state when going to PLAYING state. However,
8128 "on_new_ssrc" might be called before this happens.
8129 https://bugzilla.gnome.org/show_bug.cgi?id=683304
8131 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
8133 * gst/rtsp-server/rtsp-client.c:
8134 * gst/rtsp-server/rtsp-client.h:
8135 rtsp-client: add signals for rtsp requests (fixes #683287)
8137 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
8139 * gst/rtsp-server/rtsp-client.c:
8140 * gst/rtsp-server/rtsp-client.h:
8141 add new-session signal to rtsp-client (fixes #683058)
8143 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
8146 Automatic update of common submodule
8147 From 668acee to 4f962f7
8149 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
8151 * gst/rtsp-server/rtsp-server.c:
8152 * tests/check/gst/rtspserver.c:
8153 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
8154 Do not assume that *error is set in g_socket_address_enumerator_next.
8155 Added test_bind_already_in_use unit-test.
8156 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
8158 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
8161 Automatic update of common submodule
8162 From 94ccf4c to 668acee
8164 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
8166 * gst/rtsp-server/rtsp-client.c:
8167 * gst/rtsp-server/rtsp-client.h:
8168 rtsp-client: make create_sdp virtual method
8169 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
8171 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8174 Automatic update of common submodule
8175 From 98e386f to 94ccf4c
8177 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8179 * gst/rtsp-server/rtsp-client.c:
8182 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
8184 * gst/rtsp-server/rtsp-client.c:
8185 * gst/rtsp-server/rtsp-client.h:
8186 * gst/rtsp-server/rtsp-server.c:
8187 * gst/rtsp-server/rtsp-server.h:
8188 rtsp-server: use an existing socket to establish HTTP tunnel
8189 Make it possible to transfer a socket from an HTTP server to be used as
8190 an RTSP over HTTP tunnel.
8192 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
8194 * gst/rtsp-server/rtsp-client.c:
8195 * gst/rtsp-server/rtsp-media.c:
8196 * gst/rtsp-server/rtsp-media.h:
8197 rtsp: Handle the blocksize parameter
8198 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
8200 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
8202 * tests/check/Makefile.am:
8203 * tests/check/gst/rtspserver.c:
8204 Have unit test get header from source dir, not installed dir
8205 This makes compilation of unit tests work in a build directory other
8206 than the source directory.
8207 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
8209 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
8211 * gst/rtsp-server/rtsp-media.c:
8212 rtsp-media: update for gst_element_make_from_uri() changes
8214 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
8217 * tests/Makefile.am:
8218 * tests/check/Makefile.am:
8219 * tests/check/gst/rtspserver.c:
8221 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
8223 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
8225 * gst/rtsp-server/rtsp-media.c:
8226 rtsp-media: don't collect media stats when going to NULL
8227 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
8229 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8231 * gst/rtsp-server/rtsp-client.c:
8232 client: don't leak transports
8234 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
8236 * gst/rtsp-server/rtsp-client.c:
8237 rtsp-client: free transport on no_stream in SETUP handler
8239 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
8241 * gst/rtsp-server/rtsp-client.c:
8242 rtsp-client: changed session media iteration
8243 In client_unlink_session: now don't iterate in session->medias
8244 list where items are removed by gst_rtsp_session_release_media.
8245 Instead, repeatedly remove the first item.
8247 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
8249 * gst/rtsp-server/rtsp-client.c:
8250 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
8251 GstRTSPSessionMedia is not a GObject type. When the
8252 GstRTSPSession is freed, it will free the media.
8254 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
8256 * gst/rtsp-server/rtsp-media-factory.c:
8257 factory: plug pad leak in collect_streams
8258 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
8259 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
8260 will take one reference, and the other reference will otherwise
8263 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
8266 configure: suppress some warnings when debug is disabled
8267 Warnings about unused variables should be suppressed if core has the
8268 debug system disabled.
8269 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
8271 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8273 * docs/libs/Makefile.am:
8274 docs: fix build in uninstalled setup
8275 Include gst-plugins-base libs properly.
8277 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
8279 * docs/libs/gst-rtsp-server.types:
8280 docs: include headers defining rtsp-server object types
8281 Fixes compiler warnings during docs build.
8282 https://bugzilla.gnome.org/show_bug.cgi?id=676824
8284 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
8287 configure: Add warning flags for compiler when configuring
8288 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
8290 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
8293 Automatic update of common submodule
8294 From 03a0e57 to 98e386f
8296 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
8299 Automatic update of common submodule
8300 From 1fab359 to 03a0e57
8302 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
8304 * gst/rtsp-server/rtsp-client.c:
8305 client: fix GSocketAddress leak in gst_rtsp_client_accept
8306 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
8308 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
8311 Automatic update of common submodule
8312 From f1b5a96 to 1fab359
8314 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8317 Automatic update of common submodule
8318 From 92b7266 to f1b5a96
8320 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8323 Automatic update of common submodule
8324 From ec1c4a8 to 92b7266
8326 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8329 Automatic update of common submodule
8330 From 3429ba6 to ec1c4a8
8332 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
8334 * gst/rtsp-server/rtsp-auth.c:
8335 * gst/rtsp-server/rtsp-client.c:
8336 * gst/rtsp-server/rtsp-media-factory-uri.c:
8337 * gst/rtsp-server/rtsp-server.c:
8338 rtsp: fix compiler warnings
8339 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
8341 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8344 Automatic update of common submodule
8345 From dc70203 to 3429ba6
8347 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8349 * gst/rtsp-server/rtsp-client.c:
8350 * gst/rtsp-server/rtsp-media-factory.c:
8351 * gst/rtsp-server/rtsp-media-factory.h:
8352 * gst/rtsp-server/rtsp-media.c:
8353 * gst/rtsp-server/rtsp-media.h:
8354 * gst/rtsp-server/rtsp-server.c:
8355 * gst/rtsp-server/rtsp-server.h:
8356 * gst/rtsp-server/rtsp-session-pool.c:
8357 * gst/rtsp-server/rtsp-session-pool.h:
8358 rtsp-server: port to new thread API
8360 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8363 Automatic update of common submodule
8364 From 6db25be to dc70203
8366 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8368 * gst/rtsp-server/rtsp-auth.c:
8369 * gst/rtsp-server/rtsp-auth.h:
8370 * gst/rtsp-server/rtsp-client.c:
8371 rtsp-server: Fix compilation and compiler warnings
8373 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8377 * gst/rtsp-server/Makefile.am:
8378 configure: Modernize autotools setup a bit
8379 Also we now only create tar.bz2 and tar.xz tarballs.
8381 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8384 Automatic update of common submodule
8385 From 464fe15 to 6db25be
8387 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8390 Automatic update of common submodule
8391 From 7fda524 to 464fe15
8393 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8396 * docs/libs/Makefile.am:
8397 * docs/version.entities.in:
8399 * gst/rtsp-server/Makefile.am:
8400 * pkgconfig/Makefile.am:
8401 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
8402 * pkgconfig/gstreamer-rtsp-server.pc.in:
8403 * tests/Makefile.am:
8404 rtsp-server: Update versioning
8406 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8408 Merge remote-tracking branch 'origin/0.10'
8410 gst/rtsp-server/rtsp-session-pool.c
8412 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8414 * gst/rtsp-server/rtsp-session-pool.c:
8415 rtsp-server: Don't use deprecated GLib API
8417 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8419 Replace master with 0.11
8421 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8423 Merge branch 'master' into 0.11
8425 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8427 Merge branch 'master' into 0.11
8429 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
8432 A couple minor typo fixes
8434 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8436 * gst/rtsp-server/rtsp-media.c:
8437 media: fix state of the appqueue
8439 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8441 * gst/rtsp-server/rtsp-media-factory-uri.c:
8442 factory: use videoconvert
8444 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8446 * gst/rtsp-server/rtsp-media-factory-uri.c:
8447 factory: change to new style caps
8449 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8451 * gst/rtsp-server/rtsp-client.c:
8452 * gst/rtsp-server/rtsp-client.h:
8453 * gst/rtsp-server/rtsp-media-factory-uri.c:
8454 * gst/rtsp-server/rtsp-media.c:
8455 * gst/rtsp-server/rtsp-server.c:
8456 * gst/rtsp-server/rtsp-server.h:
8457 * gst/rtsp-server/rtsp-session-pool.c:
8458 rtsp-server: port to GIO
8461 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8464 configure: fix build
8466 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8469 docs: fix for gst_rtsp_server_set_port() -> _set_service()
8470 https://bugzilla.gnome.org/show_bug.cgi?id=666548
8472 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8475 * examples/Makefile.am:
8476 First rule of gst-rtsp-server club: don't talk about gst-phonon
8478 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8481 * pkgconfig/Makefile.am:
8482 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
8483 * pkgconfig/gstreamer-rtsp-server.pc.in:
8484 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
8485 For consistency with all other modules.
8487 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8489 * gst/rtsp-server/rtsp-client.c:
8490 rtsp-client: update for new map API
8492 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8495 * bindings/Makefile.am:
8496 * bindings/python/Makefile.am:
8497 * bindings/python/arg-types.py:
8498 * bindings/python/codegen/Makefile.am:
8499 * bindings/python/codegen/__init__.py:
8500 * bindings/python/codegen/argtypes.py:
8501 * bindings/python/codegen/code-coverage.py:
8502 * bindings/python/codegen/codegen.py:
8503 * bindings/python/codegen/definitions.py:
8504 * bindings/python/codegen/defsparser.py:
8505 * bindings/python/codegen/docextract.py:
8506 * bindings/python/codegen/docgen.py:
8507 * bindings/python/codegen/fileprefix.override:
8508 * bindings/python/codegen/fileprefixmodule.c:
8509 * bindings/python/codegen/h2def.py:
8510 * bindings/python/codegen/mergedefs.py:
8511 * bindings/python/codegen/mkskel.py:
8512 * bindings/python/codegen/override.py:
8513 * bindings/python/codegen/reversewrapper.py:
8514 * bindings/python/codegen/scmexpr.py:
8515 * bindings/python/rtspserver-types.defs:
8516 * bindings/python/rtspserver.defs:
8517 * bindings/python/rtspserver.override:
8518 * bindings/python/rtspservermodule.c:
8519 * bindings/python/test.py:
8521 python: remove pygst-based python bindings
8522 pygi is the future, apparently.
8524 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
8527 Automatic update of common submodule
8528 From c463bc0 to 7fda524
8530 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8533 Automatic update of common submodule
8534 From 2a59016 to c463bc0
8536 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8539 Automatic update of common submodule
8540 From 0807187 to 2a59016
8542 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8545 Automatic update of common submodule
8546 From 11f0cd5 to 0807187
8548 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8550 * examples/test-auth.c:
8551 example: update for new caps
8553 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8555 * examples/test-video.c:
8556 * gst/rtsp-server/rtsp-client.c:
8557 * gst/rtsp-server/rtsp-media-factory-uri.c:
8558 * gst/rtsp-server/rtsp-media.c:
8559 * gst/rtsp-server/rtsp-media.h:
8560 * gst/rtsp-server/rtsp-session.c:
8561 * gst/rtsp-server/rtsp-session.h:
8562 rtsp-server: port some more to 0.11
8564 Remove bufferlist stuff
8566 Add queue before appsink now that preroll-queue-len is gone.
8567 Update for request pad changes.
8569 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8571 Merge branch 'master' into 0.11
8573 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
8575 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8576 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
8577 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
8579 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
8581 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8582 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
8583 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
8585 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8587 Merge branch 'master' into 0.11
8589 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8591 * gst/rtsp-server/rtsp-media.c:
8592 * gst/rtsp-server/rtsp-media.h:
8593 media: add a seekable boolean
8594 Maintain the seekable state with a new variable instead of reusing the
8597 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
8599 * gst/rtsp-server/rtsp-media.c:
8600 Disallow seek in live media
8602 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8604 Merge branch 'master' into 0.11
8606 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
8608 * gst/rtsp-server/rtsp-server.c:
8609 #ifdef statements for windows socket creation were missing
8611 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
8614 Automatic update of common submodule
8615 From a39eb83 to 11f0cd5
8617 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
8620 Automatic update of common submodule
8621 From 605cd9a to a39eb83
8623 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8625 Merge branch 'master' into 0.11
8627 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8629 * gst/rtsp-server/rtsp-client.c:
8630 client: use method to access property
8632 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8634 * gst/rtsp-server/rtsp-media-factory.c:
8635 * gst/rtsp-server/rtsp-media-factory.h:
8636 media-factory: add protocols property
8637 Add a property to configure the allowed protocols in the media created from the
8640 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8642 * gst/rtsp-server/rtsp-media-factory.c:
8643 * gst/rtsp-server/rtsp-media-factory.h:
8644 media-factory: add media-configure signal
8645 Add signal to allow the application to configure the media after it was created
8648 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8650 * gst/rtsp-server/rtsp-client.c:
8651 client: use method to access property
8653 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8655 * gst/rtsp-server/rtsp-media-factory.c:
8656 * gst/rtsp-server/rtsp-media-factory.h:
8657 media-factory: add protocols property
8658 Add a property to configure the allowed protocols in the media created from the
8661 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8663 * gst/rtsp-server/rtsp-media-factory.c:
8664 * gst/rtsp-server/rtsp-media-factory.h:
8665 media-factory: add media-configure signal
8666 Add signal to allow the application to configure the media after it was created
8669 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8671 Merge branch 'master' into 0.11
8673 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8675 * gst/rtsp-server/rtsp-client.c:
8676 client: use media multicast group
8678 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8680 * gst/rtsp-server/rtsp-media-factory.h:
8681 * gst/rtsp-server/rtsp-server.h:
8682 * gst/rtsp-server/rtsp-session-pool.h:
8683 * gst/rtsp-server/rtsp-session.h:
8686 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
8688 * gst/rtsp-server/rtsp-client.c:
8689 * gst/rtsp-server/rtsp-sdp.h:
8690 sdp: copy and free the server ip address
8691 Copy and free the server ip address to make memory management easier later.
8693 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8695 * gst/rtsp-server/rtsp-media-factory.c:
8696 media-factory: configure multicast in media
8698 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8700 * gst/rtsp-server/rtsp-media.c:
8701 * gst/rtsp-server/rtsp-media.h:
8702 media: add property for multicast group
8703 Add a property to configure the multicast group in the media.
8704 Based on patches from Marc Leeman and Robert Krakora.
8706 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8708 * gst/rtsp-server/rtsp-media-factory.c:
8709 * gst/rtsp-server/rtsp-media-factory.h:
8710 media-factory: add property for multicast group
8711 Add a property to configure the multicast group in the media factory.
8712 Based on patches from Marc Leeman and Robert Krakora.
8714 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8716 * gst/rtsp-server/rtsp-client.c:
8717 client: do configuration of transport in one place
8718 Move the configuration of the transport destination address to where we also
8719 configure the other bits.
8721 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8723 * gst/rtsp-server/rtsp-client.c:
8724 client: use media multicast group
8726 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8728 * gst/rtsp-server/rtsp-media-factory.h:
8729 * gst/rtsp-server/rtsp-server.h:
8730 * gst/rtsp-server/rtsp-session-pool.h:
8731 * gst/rtsp-server/rtsp-session.h:
8734 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
8736 * gst/rtsp-server/rtsp-client.c:
8737 * gst/rtsp-server/rtsp-sdp.h:
8738 sdp: copy and free the server ip address
8739 Copy and free the server ip address to make memory management easier later.
8741 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8743 * gst/rtsp-server/rtsp-media-factory.c:
8744 media-factory: configure multicast in media
8746 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8748 * gst/rtsp-server/rtsp-media.c:
8749 * gst/rtsp-server/rtsp-media.h:
8750 media: add property for multicast group
8751 Add a property to configure the multicast group in the media.
8752 Based on patches from Marc Leeman and Robert Krakora.
8754 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8756 * gst/rtsp-server/rtsp-media-factory.c:
8757 * gst/rtsp-server/rtsp-media-factory.h:
8758 media-factory: add property for multicast group
8759 Add a property to configure the multicast group in the media factory.
8760 Based on patches from Marc Leeman and Robert Krakora.
8762 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8764 * gst/rtsp-server/rtsp-client.c:
8765 client: do configuration of transport in one place
8766 Move the configuration of the transport destination address to where we also
8767 configure the other bits.
8769 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8771 Merge branch 'master' into 0.11
8773 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
8775 * gst/rtsp-server/rtsp-client.c:
8776 client: destroy pipeline on client disconnect with no prior TEARDOWN.
8777 The problem occurs when the client abruptly closes the connection without
8778 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
8779 server is where the pipeline gets torn down. Since this handler is not called,
8780 the pipeline remains and is up and running. Subsequent clients get their own
8781 pipelines and if the do not issue TEARDOWNs then those pipelines will also
8782 remain up and running. This is a resource leak.
8784 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8786 Merge branch 'master' into 0.11
8788 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
8790 * gst/rtsp-server/rtsp-media-factory.c:
8791 * gst/rtsp-server/rtsp-media-factory.h:
8792 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
8793 For example, it can be used to retrieve source elements like appsrc, in a more
8794 convenient way than subclassing get_element.
8796 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8798 Merge branch 'master' into 0.11
8800 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
8802 * gst/rtsp-server/rtsp-server.c:
8803 rtsp-server: hold on to reference while using object
8805 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8807 * gst/rtsp-server/rtsp-media.c:
8810 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8813 configure: use unstable api
8815 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
8817 * gst/rtsp-server/rtsp-client.c:
8818 client: fix reference counting
8820 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
8822 * gst/rtsp-server/rtsp-client.c:
8823 * gst/rtsp-server/rtsp-media.c:
8824 fix compiler warnings about unused variables
8826 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
8828 * examples/test-launch.c:
8829 * examples/test-readme.c:
8830 * examples/test-uri.c:
8831 * examples/test-video.c:
8832 examples: tell rtsp uri when ready
8834 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
8837 Automatic update of common submodule
8838 From 69b981f to 605cd9a
8840 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8842 * gst/rtsp-server/rtsp-client.c:
8843 client: update for buffer API change
8845 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
8847 * gst/rtsp-server/Makefile.am:
8848 Makefile.am: 0.10 => @GST_MAJORMINOR@
8850 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
8852 * gst/rtsp-server/rtsp-media-factory-uri.c:
8853 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
8855 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
8857 * gst/rtsp-server/.gitignore:
8858 .gitignore: 0.10 => 0.11
8860 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
8862 * gst/rtsp-server/Makefile.am:
8863 Makefile.am: 0.10 => @GST_MAJORMINOR@
8865 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8867 Merge branch 'master' into 0.11
8869 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
8872 Automatic update of common submodule
8873 From 9e5bbd5 to 69b981f
8875 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
8878 Automatic update of common submodule
8879 From fd35073 to 9e5bbd5
8881 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
8884 Automatic update of common submodule
8885 From 46dfcea to fd35073
8887 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8889 * gst/rtsp-server/rtsp-media-factory-uri.c:
8890 * gst/rtsp-server/rtsp-media.c:
8891 media: port to new caps API
8893 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8895 Merge branch 'master' into 0.11
8897 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
8899 * bindings/vala/gst-rtsp-server-0.10.vapi:
8900 Updated Vala bindings.
8901 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
8903 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
8905 * gst/rtsp-server/rtsp-server.c:
8906 * gst/rtsp-server/rtsp-server.h:
8907 Add a signal for newly connected clients.
8908 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
8910 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
8912 * bindings/python/rtspserver.override:
8913 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
8915 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8917 * gst/rtsp-server/Makefile.am:
8918 * gst/rtsp-server/rtsp-client.c:
8919 * gst/rtsp-server/rtsp-funnel.c:
8920 * gst/rtsp-server/rtsp-funnel.h:
8921 * gst/rtsp-server/rtsp-media.c:
8922 rtsp-server: port to 0.11
8924 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8929 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8931 Merge branch 'master' into 0.11
8936 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8939 Automatic update of common submodule
8940 From c3cafe1 to 46dfcea
8942 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
8944 * bindings/python/Makefile.am:
8945 * bindings/python/rtspserver.defs:
8946 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
8948 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
8950 * bindings/python/arg-types.py:
8951 python bindings: add GstRTSPUrlParam
8952 Needed to implement MediaFactory virtual proxies
8954 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
8956 * bindings/python/arg-types.py:
8957 python bindings: fix returning GstRTSPUrl types
8959 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
8961 * bindings/python/arg-types.py:
8962 python bindings: add arg type for GstRTSPUrl
8964 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
8966 * bindings/python/rtspserver.defs:
8967 python bindings: fix the definition of MediaFactory.collect_stream
8969 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
8972 Automatic update of common submodule
8973 From 1ccbe09 to c3cafe1
8975 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8978 Automatic update of common submodule
8979 From 193b717 to 1ccbe09
8981 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
8984 Automatic update of common submodule
8985 From b77e2bf to 193b717
8987 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8990 build: Include lcov.mak to allow test coverage report generation
8992 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8995 Automatic update of common submodule
8996 From d8814b6 to b77e2bf
8998 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9001 Automatic update of common submodule
9002 From 6aaa286 to d8814b6
9004 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
9007 Automatic update of common submodule
9008 From 6aec6b9 to 6aaa286
9010 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
9013 autogen: wingo signed comment
9015 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
9017 * gst/rtsp-server/rtsp-session-pool.c:
9018 session: use full charset for RTSP session ID
9019 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
9020 session ID more difficult.
9021 https://bugzilla.gnome.org/show_bug.cgi?id=643812
9023 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9025 * gst/rtsp-server/Makefile.am:
9026 rtsp-server: Don't install the funnel header
9028 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
9031 Automatic update of common submodule
9032 From 1de7f6a to 6aec6b9
9034 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9037 configure: require core/base 0.10.31
9038 Needed at least for gst_plugin_feature_rank_compare_func().
9040 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
9043 Automatic update of common submodule
9044 From f94d739 to 1de7f6a
9046 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9048 * gst/rtsp-server/rtsp-media.c:
9049 media: remove more unused code
9051 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9053 * gst/rtsp-server/rtsp-media.c:
9054 * gst/rtsp-server/rtsp-media.h:
9055 media: remove duplicate filtering
9056 Remove the duplicate filtering code now that we have a released -good version.
9057 Give a warning instead.
9059 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9061 * gst/rtsp-server/rtsp-media-factory.c:
9062 * gst/rtsp-server/rtsp-media.c:
9063 media: fix default buffer size
9065 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9067 * gst/rtsp-server/rtsp-media-factory.c:
9068 * gst/rtsp-server/rtsp-media-factory.h:
9069 media-factory: add property to configure the buffer-size
9070 Add a property to configure the kernel UDP buffer size.
9072 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9074 * gst/rtsp-server/rtsp-media.c:
9075 * gst/rtsp-server/rtsp-media.h:
9076 media: add property to configure kernel buffer sizes
9077 Add a property to configure the kernel UDP buffer size.
9079 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9082 configure: set PYGOBJECT_REQ before using it
9083 https://bugzilla.gnome.org/show_bug.cgi?id=640641
9085 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9088 docs: recursive into sub-directories on 'make upload'
9090 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9092 * docs/libs/gst-rtsp-server-docs.sgml:
9093 * docs/version.entities.in:
9094 docs: mention full version these docs are for, not just major-minor
9096 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9101 === release 0.10.8 ===
9103 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9108 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9110 * gst/rtsp-server/rtsp-server.c:
9111 rtsp-server: clarify docs a little
9113 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9115 * gst/rtsp-server/rtsp-media.c:
9116 media: init debug category before starting thread
9118 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9120 * gst/rtsp-server/rtsp-auth.c:
9121 auth: add realm to make it more spec compliant
9123 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9125 * gst/rtsp-server/rtsp-server.c:
9126 * gst/rtsp-server/rtsp-server.h:
9129 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9131 * examples/test-video.c:
9132 example: improve example docs a little
9134 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9136 * gst/rtsp-server/rtsp-server.c:
9137 server: ensure the watch has a ref to the server
9139 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9141 * gst/rtsp-server/rtsp-server.c:
9142 server: simpify channel function
9144 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9146 * gst/rtsp-server/rtsp-server.c:
9147 * gst/rtsp-server/rtsp-server.h:
9148 server: simplify management of channel and source
9149 We don't need to keep around the channel and source objects. Let the mainloop
9150 and the source manage the source and channel respectively.
9152 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9158 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9161 * tests/Makefile.am:
9162 * tests/test-cleanup.c:
9163 tests: add tests directory and cleanup test
9165 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9167 * gst/rtsp-server/rtsp-media-factory-uri.c:
9168 * gst/rtsp-server/rtsp-media-factory.c:
9169 * gst/rtsp-server/rtsp-media-mapping.c:
9170 * gst/rtsp-server/rtsp-media.c:
9171 * gst/rtsp-server/rtsp-session-pool.c:
9172 * gst/rtsp-server/rtsp-session.c:
9173 server: improve debugging in various objects
9175 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9177 * gst/rtsp-server/rtsp-server.c:
9178 server: chain up to the parent finalize
9180 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
9182 * bindings/python/rtspserver-types.defs:
9183 * bindings/python/rtspserver.defs:
9184 * bindings/python/rtspserver.override:
9185 * bindings/python/test.py:
9186 gst-rtsp-server: update python bindings
9188 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9190 * gst/rtsp-server/rtsp-client.c:
9191 client: use the response from the clientstate
9192 Create the response object only once and store in the client state.
9193 Make all methods use the state response,
9195 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9197 * gst/rtsp-server/rtsp-server.c:
9198 server: use signal to keep track of clients
9199 Keep track of all the clients that the server creates and remove them when they
9200 fire the 'closed' signal.
9202 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9204 * gst/rtsp-server/rtsp-client.c:
9205 * gst/rtsp-server/rtsp-client.h:
9206 client: emit signal when closing
9208 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9210 * examples/.gitignore:
9211 * examples/Makefile.am:
9212 * examples/test-auth.c:
9213 * examples/test-video.c:
9214 * gst/rtsp-server/rtsp-auth.c:
9215 * gst/rtsp-server/rtsp-auth.h:
9216 * gst/rtsp-server/rtsp-client.c:
9217 * gst/rtsp-server/rtsp-media-factory.c:
9218 * gst/rtsp-server/rtsp-media.c:
9219 * gst/rtsp-server/rtsp-media.h:
9220 * gst/rtsp-server/rtsp-session-pool.h:
9221 * gst/rtsp-server/rtsp-session.h:
9222 media: enable per factory authorisations
9223 Allow for adding a GstRTSPAuth on the factory and media level and check
9224 permissions when accessing the factory.
9225 Add hints to the auth methods for future more fine grained authorisation.
9226 Add example application for per factory authentication.
9228 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9230 * gst/rtsp-server/rtsp-auth.c:
9231 * gst/rtsp-server/rtsp-auth.h:
9232 * gst/rtsp-server/rtsp-client.c:
9233 * gst/rtsp-server/rtsp-client.h:
9234 * gst/rtsp-server/rtsp-params.c:
9235 * gst/rtsp-server/rtsp-params.h:
9236 rtsp-server: Pass ClientState structure arround
9237 Pass the collected information for the ongoing request in a GstRTSPClientState
9238 structure that we can then pass around to simplify the method arguments. This
9239 will also be handy when we implement logging functionality.
9241 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9243 * gst/rtsp-server/rtsp-media-factory.c:
9244 * gst/rtsp-server/rtsp-media-factory.h:
9245 media-factory: add methods to configure authorisation
9247 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9249 * gst/rtsp-server/rtsp-client.c:
9250 client: unref auth in finalize
9252 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9254 * gst/rtsp-server/rtsp-server.c:
9255 server: unref auth in finalize
9257 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9259 * docs/libs/gst-rtsp-server-docs.sgml:
9260 * docs/libs/gst-rtsp-server-sections.txt:
9261 * docs/libs/gst-rtsp-server.types:
9264 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9266 * gst/rtsp-server/rtsp-server.c:
9267 * gst/rtsp-server/rtsp-server.h:
9268 server: separate create and accept
9269 Create separate create and accept methods so that subclasses can create custom
9271 Configure the server in the client object and prepare for keeping track of
9274 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9276 * gst/rtsp-server/rtsp-client.c:
9277 * gst/rtsp-server/rtsp-client.h:
9278 client: add support for setting the server.
9279 Add support for keeping a ref to the server that started this client
9282 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9284 * gst/rtsp-server/rtsp-auth.c:
9285 auth: fix memleak and add some docs
9286 Fix a memleak of the basic auth token.
9287 Add docs for the helper function
9289 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9291 * gst/rtsp-server/rtsp-auth.c:
9292 * gst/rtsp-server/rtsp-auth.h:
9293 * gst/rtsp-server/rtsp-client.c:
9294 client: delegate setup of auth to the manager
9295 Delegate the configuration of the authentication tokens to the manager object
9298 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9300 * examples/test-video.c:
9301 * gst/rtsp-server/Makefile.am:
9302 * gst/rtsp-server/rtsp-auth.c:
9303 * gst/rtsp-server/rtsp-auth.h:
9304 * gst/rtsp-server/rtsp-client.c:
9305 * gst/rtsp-server/rtsp-client.h:
9306 * gst/rtsp-server/rtsp-server.c:
9307 * gst/rtsp-server/rtsp-server.h:
9308 auth: add authentication object
9309 Add an object that can check the authorization of requests.
9310 Implement basic authentication.
9311 Add example authentication to test-video
9313 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9315 * gst/rtsp-server/rtsp-server.c:
9316 * gst/rtsp-server/rtsp-server.h:
9317 server: move includes back
9318 the includes are needed for sockaddr_in.
9320 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9322 * gst/rtsp-server/rtsp-client.c:
9323 * gst/rtsp-server/rtsp-client.h:
9324 * gst/rtsp-server/rtsp-server.c:
9325 * gst/rtsp-server/rtsp-server.h:
9326 rtsp: move network includes where they are needed
9328 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
9330 * gst/rtsp-server/rtsp-media.h:
9331 rtsp-media.h: Minor corrections in comments.
9334 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
9337 Automatic update of common submodule
9338 From e572c87 to f94d739
9340 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
9344 * docs/libs/.gitignore:
9345 * examples/.gitignore:
9346 * gst/rtsp-server/.gitignore:
9349 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
9351 * docs/libs/Makefile.am:
9352 docs: We don't build ps/pdf for API reference docs
9354 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9357 Automatic update of common submodule
9358 From ccbaa85 to e572c87
9360 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9363 Automatic update of common submodule
9364 From 46445ad to ccbaa85
9366 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9368 * gst/rtsp-server/Makefile.am:
9369 * gst/rtsp-server/rtsp-funnel.c:
9370 * gst/rtsp-server/rtsp-funnel.h:
9371 * gst/rtsp-server/rtsp-media.c:
9372 funnel: rename fsfunnel to rtspfunnel
9373 Rename the funnel to avoid conflicts with the farsight one.
9375 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9377 * gst/rtsp-server/Makefile.am:
9378 * gst/rtsp-server/fs-funnel.c:
9379 * gst/rtsp-server/fs-funnel.h:
9380 * gst/rtsp-server/rtsp-media.c:
9381 rtsp-media: add and use fsfunnel
9382 Add a copy of fsfunnel to the build because input-selector removed the (broken)
9383 select-all property that we need.
9385 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9387 * gst/rtsp-server/Makefile.am:
9388 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
9389 Use PKG_CONFIG_PATH specified at configure time (if any) as well
9390 for the g-ir-compiler, rather than just assuming the env var has
9393 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9400 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
9402 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9405 * gst/rtsp-server/Makefile.am:
9406 gobject-introspection: fix g-i build for uninstalled setup
9407 Requires gst-plugins-base git (> 0.10.31.2).
9409 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9411 * examples/test-uri.c:
9412 examples: add some more options and comments
9414 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9416 * gst/rtsp-server/rtsp-media-factory-uri.c:
9417 factory-uri: use right property type
9419 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9421 * gst/rtsp-server/rtsp-media-factory-uri.c:
9422 factory-uri: attempt to configure buffer-lists
9423 Attempt to configure buffer lists in the payloader for improved performance.
9425 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9427 * gst/rtsp-server/rtsp-media.c:
9428 media: attempt to configure bigger UDP buffers
9429 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
9430 send buffers with high bitrate streams.
9432 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
9434 * gst/rtsp-server/rtsp-client.c:
9435 client: use the socket length from getsockname
9436 Use the length returned by getsockname to perform the getnameinfo call because
9437 the size can depend on the socket type and platform.
9440 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9442 * docs/libs/gst-rtsp-server-docs.sgml:
9443 * docs/libs/gst-rtsp-server-sections.txt:
9444 docs: add uri factory to the docs
9446 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9448 * gst/rtsp-server/rtsp-client.c:
9449 * gst/rtsp-server/rtsp-media.h:
9452 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9454 * gst/rtsp-server/rtsp-client.c:
9455 * gst/rtsp-server/rtsp-media.c:
9456 * gst/rtsp-server/rtsp-media.h:
9457 * gst/rtsp-server/rtsp-session.c:
9458 * gst/rtsp-server/rtsp-session.h:
9459 rtsp-server: add support for buffer lists
9460 Add support for sending bufferlists received from appsink.
9463 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9465 * gst/rtsp-server/rtsp-client.c:
9466 * gst/rtsp-server/rtsp-media.c:
9467 * gst/rtsp-server/rtsp-media.h:
9468 * gst/rtsp-server/rtsp-sdp.c:
9469 media: make method to retrieve the play range
9470 Make a method to retrieve the playback range so that we can conditionally create
9471 a different range for the SDP and the PLAY requests.
9473 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9475 * gst/rtsp-server/rtsp-media.c:
9476 * gst/rtsp-server/rtsp-media.h:
9477 media: add signal to notify of state changes
9479 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9481 * gst/rtsp-server/rtsp-client.h:
9482 client: cleanup headers
9484 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9486 * gst/rtsp-server/rtsp-client.c:
9489 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9491 * gst/rtsp-server/rtsp-media-factory-uri.c:
9492 * gst/rtsp-server/rtsp-media-factory-uri.h:
9493 factory-uri: add support for gstpay
9494 Add an option to prefer gstpay over decoder + raw payloader.
9496 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9498 * gst/rtsp-server/rtsp-media-factory-uri.c:
9499 * gst/rtsp-server/rtsp-media-factory-uri.h:
9500 factory-uri: rework the autoplugger.
9501 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
9504 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9506 * gst/rtsp-server/rtsp-media-factory-uri.c:
9507 factory-uri: use better factory filter
9508 Make better payloader filter based on autoplug rank and RTP use case.
9510 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
9513 Automatic update of common submodule
9514 From 169462a to 46445ad
9516 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9518 * gst/rtsp-server/rtsp-server.c:
9519 server: set SO_REUSEADDR before bind
9520 Set the SO_REUSEADDR _before_ bind() to make it actually work.
9522 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9524 * gst/rtsp-server/rtsp-media.c:
9525 * gst/rtsp-server/rtsp-media.h:
9526 media: emit prepared signal when prepared
9527 Make a 'prepared' signal and emit it when we successfully prepared the element.
9528 This signal can be used to configure the media object after it has been prepared
9531 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
9534 Automatic update of common submodule
9535 From 011bcc8 to 169462a
9537 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
9539 python an optional dependency
9540 * configure.ac: Move up valgrind and g-i checks. Make the python
9541 dependency optional, as it was before.
9543 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9545 Merge branch 'master' into 0.11
9550 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9552 * gst/rtsp-server/rtsp-media.c:
9553 media: update range when active clients changed
9554 When we changed the number of active clients, update the current range
9555 information because we want the second client connecting to a shared resource
9556 continue from where the stream currently.
9558 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9560 * gst/rtsp-server/rtsp-media-factory-uri.c:
9561 * gst/rtsp-server/rtsp-media-factory-uri.h:
9562 factory-uri: add colorspace and fix pt
9563 Rework the way we pass data to the autoplugger.
9564 When we have raw caps, plug a converter element to make pluggin to raw
9565 payloaders more successful.
9566 Make sure all dynamically plugged payloaders have a unique payload types.
9568 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9570 * examples/Makefile.am:
9571 * examples/test-uri.c:
9572 example: add example of the uri factory
9574 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9576 * gst/rtsp-server/Makefile.am:
9577 * gst/rtsp-server/rtsp-media-factory-uri.c:
9578 * gst/rtsp-server/rtsp-media-factory-uri.h:
9579 * gst/rtsp-server/rtsp-server.h:
9580 factory-uri: add a factory to stream any URI
9581 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
9584 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9586 * gst/rtsp-server/rtsp-media.c:
9587 * gst/rtsp-server/rtsp-media.h:
9588 media: ignore spurious ASYNC_DONE messages
9589 When we are dynamically adding pads, the addition of the udpsrc elements will
9590 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
9591 the real ASYNC_DONE when everything is prerolled.
9593 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9595 * gst/rtsp-server/rtsp-media-factory.c:
9596 * gst/rtsp-server/rtsp-media-factory.h:
9597 media-factory: make lock macro
9599 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
9601 * gst/rtsp-server/rtsp-client.c:
9602 rtsp-server: Remove unused variable and dead assignment
9604 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
9606 * examples/test-launch.c:
9607 * examples/test-mp4.c:
9608 * examples/test-ogg.c:
9609 * examples/test-readme.c:
9610 * examples/test-sdp.c:
9611 * examples/test-video.c:
9612 examples: Run gst-indent
9614 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
9616 * gst/rtsp-server/rtsp-client.c:
9617 * gst/rtsp-server/rtsp-media-factory.c:
9618 * gst/rtsp-server/rtsp-media-mapping.c:
9619 * gst/rtsp-server/rtsp-media.c:
9620 * gst/rtsp-server/rtsp-params.c:
9621 * gst/rtsp-server/rtsp-sdp.c:
9622 * gst/rtsp-server/rtsp-server.c:
9623 * gst/rtsp-server/rtsp-session-pool.c:
9624 * gst/rtsp-server/rtsp-session.c:
9625 rtsp-server: Run gst-indent
9626 Since it wasn't using the upstream common previously, there was no
9627 indentation check before commiting.
9629 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
9631 * gst/rtsp-server/rtsp-media-mapping.h:
9632 * gst/rtsp-server/rtsp-media.c:
9633 * gst/rtsp-server/rtsp-media.h:
9634 * gst/rtsp-server/rtsp-sdp.c:
9635 * gst/rtsp-server/rtsp-session-pool.h:
9636 * gst/rtsp-server/rtsp-session.c:
9637 * gst/rtsp-server/rtsp-session.h:
9638 rtsp-server: Some more doc fixups
9640 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
9643 Makefile: Add cruft-cleaning support
9645 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
9650 * docs/libs/Makefile.am:
9651 * docs/libs/gst-rtsp-server-docs.sgml:
9652 * docs/libs/gst-rtsp-server-sections.txt:
9653 * docs/libs/gst-rtsp-server.types:
9654 * docs/version.entities.in:
9655 docs: Add gtk-doc build system
9657 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
9659 * gst/rtsp-server/Makefile.am:
9660 Makefile.am: Use standard GIR make behaviour
9662 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
9666 autogen/configure: Bring more in sync to standard gst module behaviour
9668 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9670 * gst/rtsp-server/rtsp-media.c:
9671 media: warn and fail when gstrtpbin is not found
9673 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9676 configure: open 0.11 branch
9678 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
9682 Add common submodule
9684 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
9687 * common/Makefile.am:
9688 * common/c-to-xml.py:
9690 * common/coverage/coverage-report-entry.pl:
9691 * common/coverage/coverage-report.pl:
9692 * common/coverage/coverage-report.xsl:
9693 * common/coverage/lcov.mak:
9694 * common/gettext.patch:
9695 * common/glib-gen.mak:
9696 * common/gst-autogen.sh:
9697 * common/gst-xmlinspect.py:
9699 * common/gstdoc-scangobj:
9700 * common/gtk-doc-plugins.mak:
9701 * common/gtk-doc.mak:
9702 * common/m4/.gitignore:
9703 * common/m4/Makefile.am:
9705 * common/m4/as-ac-expand.m4:
9706 * common/m4/as-auto-alt.m4:
9707 * common/m4/as-compiler-flag.m4:
9708 * common/m4/as-compiler.m4:
9709 * common/m4/as-docbook.m4:
9710 * common/m4/as-libtool-tags.m4:
9711 * common/m4/as-libtool.m4:
9712 * common/m4/as-python.m4:
9713 * common/m4/as-scrub-include.m4:
9714 * common/m4/as-version.m4:
9715 * common/m4/ax_create_stdint_h.m4:
9716 * common/m4/check.m4:
9717 * common/m4/glib-gettext.m4:
9718 * common/m4/gst-arch.m4:
9719 * common/m4/gst-args.m4:
9720 * common/m4/gst-check.m4:
9721 * common/m4/gst-debuginfo.m4:
9722 * common/m4/gst-default.m4:
9723 * common/m4/gst-doc.m4:
9724 * common/m4/gst-error.m4:
9725 * common/m4/gst-feature.m4:
9726 * common/m4/gst-function.m4:
9727 * common/m4/gst-gettext.m4:
9728 * common/m4/gst-glib2.m4:
9729 * common/m4/gst-libxml2.m4:
9730 * common/m4/gst-plugindir.m4:
9731 * common/m4/gst-valgrind.m4:
9732 * common/m4/gtk-doc.m4:
9733 * common/m4/introspection.m4:
9735 * common/mangle-tmpl.py:
9736 * common/plugins.xsl:
9738 * common/release.mak:
9739 * common/scangobj-merge.py:
9740 * common/upload.mak:
9741 common: Remove static version
9743 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
9745 * common/m4/introspection.m4:
9746 Update introspection.m4 to match usage
9748 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9752 Remove old stuff from the README
9754 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9759 === release 0.10.7 ===
9761 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9766 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9768 * examples/test-ogg.c:
9769 test-ogg: remove parsers
9770 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
9771 buffers with timestamps. Using the parsers also seems to break things.
9773 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9775 * bindings/vala/gst-rtsp-server-0.10.vapi:
9776 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9777 Updated Vala bindings
9779 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9781 * common/m4/introspection.m4:
9783 * gst/rtsp-server/Makefile.am:
9784 Added initial gobject-introspection support
9786 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9788 * gst/rtsp-server/rtsp-media-factory.c:
9789 media-factory: don't use host for shared hash key
9790 When we generate the key to share made between connections, don't include the
9791 host used to connect so that we can share media even if between clients that
9792 connected with localhost and ones with the ip address.
9794 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9796 * bindings/vala/Makefile.am:
9797 build: fix distcheck
9799 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9801 * bindings/vala/gst-rtsp-server-0.10.vapi:
9802 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9803 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9804 Update Vala bindings
9806 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9808 * bindings/vala/Makefile.am:
9810 Fix configure checks and installation location for Vala bindings
9813 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9818 === release 0.10.6 ===
9820 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9823 configure: release 0.10.6
9825 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9827 * gst/rtsp-server/rtsp-media.c:
9828 media: help the compiler a little
9830 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9832 * gst/rtsp-server/rtsp-media.c:
9833 * gst/rtsp-server/rtsp-media.h:
9834 * gst/rtsp-server/rtsp-session.c:
9835 media: cleanup media transport before freeing
9836 Cleanup the media transport data before freeing. In particular, remove the qdata
9837 from the rtpsource object.
9839 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9841 * gst/rtsp-server/rtsp-media-factory.c:
9842 * gst/rtsp-server/rtsp-media-factory.h:
9843 * gst/rtsp-server/rtsp-media.c:
9844 * gst/rtsp-server/rtsp-media.h:
9845 media-factory: add eos-shutdown property
9846 Add an eos-shutdown property that will send an EOS to the pipeline before
9847 shutting it down. This allows for nice cleanup in case of a muxer.
9850 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9852 * gst/rtsp-server/rtsp-media.c:
9853 * gst/rtsp-server/rtsp-media.h:
9854 media: use multiudpsink send-duplicates when we can
9855 If we have a new enough multiudpsink with the send-duplicates property, use this
9856 instead of doing our own filtering. Our custom filtering code should eventually
9857 be removed when we can depend on a released -good.
9859 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9861 * gst/rtsp-server/rtsp-media.c:
9862 media: don't leak destinations
9863 Refactor and cleanup the destinations array when the stream is destroyed.
9865 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9867 * gst/rtsp-server/rtsp-media.c:
9868 * gst/rtsp-server/rtsp-media.h:
9869 media: don't add udp addresses multiple times
9870 Keep track of the udp addresses we added to udpsink and never add the same udp
9871 destination twice. This avoids duplicate packets when using multicast.
9873 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9875 * gst/rtsp-server/rtsp-server.c:
9876 server: disable use of SO_LINGER
9877 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
9878 server close()s the connection.
9880 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9882 * gst/rtsp-server/rtsp-server.c:
9883 server: use 5 second linger period in SO_LINGER
9884 Wait 5 seconds before clearing the send buffers and reseting the connection with
9885 the client when we do a close. This should be enough time to get the message to
9889 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
9891 * gst/rtsp-server/rtsp-server.c:
9892 server: use SO_LINGER
9893 SO_LINGER on the socket will make sure that any pending data on the socket is
9894 flushed ASAP and that the socket connection is reset. This makes sure that the
9895 socket can be reused immediately.
9898 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9901 README: add blurb about shared media factories
9903 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
9905 * gst/rtsp-server/rtsp-media.c:
9906 Add stdlib.h for atoi()
9908 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9910 * bindings/python/Makefile.am:
9911 * bindings/vala/Makefile.am:
9912 build: distcheck fixes
9913 Fix 'make distcheck', somewhat (it still fails because it tries to
9914 install files into /usr/share/vala/vapi/ irrespective of the
9917 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9920 configure: bump core/base requirements to released version
9921 Makes things less confusing for people.
9923 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9926 configure: fail if GStreamer core/base requirements are not met
9928 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9930 * gst/rtsp-server/rtsp-client.c:
9931 client: improve client cleanups
9932 Make sure the session does not timeout when using TCP. We need to do this
9933 because quicktime player does not send RTCP for some reason in tunneled
9935 Refactor some cleanup code.
9938 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9940 * gst/rtsp-server/rtsp-session.c:
9941 * gst/rtsp-server/rtsp-session.h:
9942 session: add support for prevent session timeouts
9943 Add an atomix counter to prevent session timeouts when we are, for example,
9946 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9948 * gst/rtsp-server/rtsp-client.c:
9949 client: fix unlink on session timeouts
9950 When our session times out, make sure we unlink all streams in this
9952 Remove the tunnelid when closing the connection.
9954 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9956 * gst/rtsp-server/rtsp-session.c:
9957 session: small cleanups
9959 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9961 * gst/rtsp-server/rtsp-client.c:
9962 client: handle lost_tunnel callbacks
9963 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
9964 hashtable so that we can reuse it for when the client reopens the POST
9966 Close the connection after a TEARDOWN.
9967 Make sure or watchid is cleared when the watch is removed.
9970 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9972 * gst/rtsp-server/rtsp-client.c:
9973 * gst/rtsp-server/rtsp-media.c:
9974 * gst/rtsp-server/rtsp-sdp.c:
9975 rtsp-server: add more support for multicast
9977 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9980 * gst/rtsp-server/rtsp-media.c:
9981 * gst/rtsp-server/rtsp-media.h:
9982 media: allow configuration of allowed lower transport
9984 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9986 * gst/rtsp-server/rtsp-client.h:
9987 * gst/rtsp-server/rtsp-media.c:
9988 * gst/rtsp-server/rtsp-media.h:
9989 * gst/rtsp-server/rtsp-sdp.c:
9990 * gst/rtsp-server/rtsp-sdp.h:
9991 * gst/rtsp-server/rtsp-server.c:
9992 rtsp: keep track of server ip and ipv6
9993 Keep track of how the client connected to the server and setup the udp ports
9994 with the same protocol.
9995 Copy the server ip address in the SDP so that clients can send RTCP back to
9998 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10000 * gst/rtsp-server/rtsp-session.c:
10003 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10005 * gst/rtsp-server/rtsp-client.c:
10006 client: use right size for malloc
10008 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10010 * gst/rtsp-server/rtsp-server.c:
10011 server: comment ipv6 server listening address
10013 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10015 * gst/rtsp-server/rtsp-media.c:
10016 media: allow for ipv6 sockets
10018 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10020 * gst/rtsp-server/rtsp-server.c:
10021 * gst/rtsp-server/rtsp-server.h:
10022 server: rework server part
10023 Allow setting a bind address, make sure we can deal with ipv6.
10024 Remove the port property and change with the service property.
10026 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10028 * gst/rtsp-server/rtsp-media.h:
10029 media: update comments a little
10031 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10033 * gst/rtsp-server/rtsp-client.c:
10034 client: make content-base better
10035 Use the URI formatting functions to make a content-base. Also make sure that
10036 there is a trailing / at the end.
10038 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10040 * gst/rtsp-server/rtsp-client.c:
10041 client: guard against invalid paths
10043 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10045 * examples/test-video.c:
10046 test: catch server bind errors
10048 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
10050 * gst/rtsp-server/rtsp-media.c:
10051 rtspmedia: emit "unprepared" if _prepare fails.
10052 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
10053 media object is removed from its factory's cache.
10055 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10057 * gst/rtsp-server/rtsp-media.c:
10058 media: collect media position when seek completes
10060 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
10062 * gst/rtsp-server/rtsp-client.c:
10063 client: call unlink_streams in client finalize
10066 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10068 * gst/rtsp-server/rtsp-media.c:
10069 media: limit the time to wait to something huge
10070 Avoid waiting forever but limit the timeout to 20 seconds.
10072 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10074 * gst/rtsp-server/rtsp-sdp.c:
10075 sdp: reindent and check for prepared status
10077 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10079 * gst/rtsp-server/rtsp-media.c:
10080 * gst/rtsp-server/rtsp-media.h:
10081 * gst/rtsp-server/rtsp-session.c:
10082 media: avoid doing _get_state() for state changes
10083 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
10084 until the media is prerolled or in error. This avoids doing a blocking call of
10085 gst_element_get_state() that can cause lockups when there is an error.
10088 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10090 * gst/rtsp-server/rtsp-media.c:
10093 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10095 * gst/rtsp-server/rtsp-media-factory.c:
10096 media-factory: better error handling
10097 Improve the error handling a bit.
10099 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10101 * gst/rtsp-server/rtsp-client.c:
10102 client: rework transport parsing
10103 Rework the transport parsing code so that we can ignore transports we don't
10104 support instead of just picking the first one we can parse.
10105 Configure a (for now hardcoded) destination for multicast transports.
10107 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10109 * gst/rtsp-server/rtsp-media.c:
10110 media: set multicast sink parameters
10111 Disable loop and automatic multicast join on the udpsink elements.
10112 Add some more debug info.
10113 Reset some state variables in the right place.
10114 Use the right port numbers for multicast.
10116 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10118 * gst/rtsp-server/rtsp-session.c:
10119 session: handle transport setup correctly
10120 Handle UDP, MCAST and TCP transport negotiation more correctly.
10121 Store the server session SSRC in the transport.
10123 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10125 * gst/rtsp-server/rtsp-client.c:
10126 rtsp-client: implement error_full
10127 Implement error_full to avoid some segfaults when the rtspconnection calls it.
10130 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10133 * gst/rtsp-server/rtsp-client.c:
10134 * gst/rtsp-server/rtsp-server.c:
10135 docs: update docs and comments
10137 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
10139 * gst/rtsp-server/rtsp-sdp.c:
10140 sdp: make server work better when behind a proxy
10142 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10144 * gst/rtsp-server/rtsp-client.c:
10145 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
10147 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10149 * gst/rtsp-server/rtsp-client.c:
10150 * gst/rtsp-server/rtsp-media-factory.c:
10151 * gst/rtsp-server/rtsp-media-mapping.c:
10152 * gst/rtsp-server/rtsp-media.c:
10153 * gst/rtsp-server/rtsp-server.c:
10154 * gst/rtsp-server/rtsp-session-pool.c:
10155 * gst/rtsp-server/rtsp-session.c:
10156 Use GStreamer's debugging subsystem
10158 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10160 * gst/rtsp-server/rtsp-media-factory.c:
10161 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
10163 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10166 back to development
10168 === release 0.10.5 ===
10170 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10175 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10178 configure: bump required versions
10180 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
10182 * gst/rtsp-server/rtsp-client.c:
10183 client: call weak-unref on client->sessions from finalize
10186 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10188 * gst/rtsp-server/rtsp-media.c:
10189 media: Fixed crasher where caps got unref'ed too often
10191 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10194 * pkgconfig/.gitignore:
10195 * pkgconfig/Makefile.am:
10196 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
10197 Added pkg-config file to use gst-rtsp-server uninstalled
10199 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10201 * gst/rtsp-server/rtsp-media.c:
10202 media: add some docs
10204 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
10206 * gst/rtsp-server/rtsp-client.c:
10207 rtsp: Use gst_rtsp_watch_send_message().
10208 Use gst_rtsp_watch_send_message() since the old API which used
10209 gst_rtsp_watch_queue_message() has been deprecated.
10211 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10214 back to development
10216 === release 0.10.4 ===
10218 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10223 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10225 * gst/rtsp-server/rtsp-client.c:
10226 * gst/rtsp-server/rtsp-session.c:
10227 * gst/rtsp-server/rtsp-session.h:
10228 rtsp: allocate channels in TCP mode
10229 When the client does not provide us with channels in TCP mode, allocate channels
10232 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10234 * gst/rtsp-server/rtsp-client.c:
10235 client: don't crash when tunnelid is missing
10236 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
10237 don't crash but return an error response to the client.
10240 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10242 * bindings/vala/gst-rtsp-server-0.10.vapi:
10243 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
10244 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
10245 bindings: update vala bindings with new method
10247 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10249 * gst/rtsp-server/rtsp-session-pool.c:
10250 * gst/rtsp-server/rtsp-session-pool.h:
10251 sessionpool: add function to filter sessions
10252 Add generic function to retrieve/remove sessions.
10254 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10257 configure: bump core/base requirements to release
10259 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10261 * gst/rtsp-server/rtsp-media.c:
10262 media: fix indentation
10264 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10266 * gst/rtsp-server/rtsp-media.c:
10267 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
10269 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10271 * gst/rtsp-server/rtsp-media.c:
10272 set state and remove elements of media in for loop
10274 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
10276 * bindings/vala/gst-rtsp-server-0.10.vapi:
10277 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
10278 Added gst_rtsp_media_remove_elements function to Vala bindings
10280 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
10282 * gst/rtsp-server/rtsp-media.c:
10283 * gst/rtsp-server/rtsp-media.h:
10284 Added gst_rtsp_media_remove_elements function
10286 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
10288 * gst/rtsp-server/rtsp-media.c:
10289 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
10291 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10293 * bindings/vala/gst-rtsp-server-0.10.vapi:
10294 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
10295 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
10296 Updated Vala bindings
10298 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10300 * gst/rtsp-server/rtsp-media.c:
10301 * gst/rtsp-server/rtsp-media.h:
10302 Added vmethod unprepare to GstRTSPMedia
10303 The default implementation sets the state of the pipeline to GST_STATE_NULL
10305 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10307 * gst/rtsp-server/rtsp-media-factory.c:
10308 * gst/rtsp-server/rtsp-media-factory.h:
10309 Made collect_streams function public
10311 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10313 * gst/rtsp-server/rtsp-media-factory.c:
10314 * gst/rtsp-server/rtsp-media-factory.h:
10315 * gst/rtsp-server/rtsp-media.c:
10316 Added vmethod create_pipeline to GstRTSPMediaFactory
10317 The pipeline is created in this method and the GstRTSPMedia's element is added to it
10319 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10321 * gst/rtsp-server/rtsp-client.c:
10322 client: use g_source_destroy()
10323 We need to use g_source_destroy() because we might have added the source to a
10324 different main context than the default one.
10326 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10328 * gst/rtsp-server/Makefile.am:
10329 * gst/rtsp-server/rtsp-client.c:
10330 * gst/rtsp-server/rtsp-params.c:
10331 * gst/rtsp-server/rtsp-params.h:
10332 rtsp: prepare for handling GET/SET_PARAMETER
10333 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
10335 Fix return codes of handlers.
10337 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10339 * gst/rtsp-server/rtsp-media.c:
10340 media: don't leak session pads
10342 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10344 * gst/rtsp-server/rtsp-media.c:
10345 media: clean up the messages a bit
10347 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10349 * gst/rtsp-server/rtsp-sdp.c:
10350 sdp: warn and skip streams without media
10352 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10354 * bindings/vala/gst-rtsp-server-0.10.vapi:
10355 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
10356 vala: Fixed typo in header file of RTSPMediaStream
10358 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10360 * gst/rtsp-server/rtsp-media.c:
10362 Fix a debug message
10363 Make dumping RTCP stats configurable
10365 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10367 * gst/rtsp-server/rtsp-media.c:
10368 media: be less verbose and leak less
10370 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10372 * gst/rtsp-server/rtsp-media.c:
10373 media: don't leak the destination address
10375 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10377 * gst/rtsp-server/rtsp-client.c:
10378 * gst/rtsp-server/rtsp-media.c:
10379 * gst/rtsp-server/rtsp-media.h:
10380 * gst/rtsp-server/rtsp-session.c:
10381 * gst/rtsp-server/rtsp-session.h:
10382 rtsp: use RTCP to keep the session alive
10383 Use the RTCP rtcp-from stats field to find the associated session and use this
10384 to keep the session alive.
10386 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10388 * gst/rtsp-server/rtsp-session.c:
10389 session: add 5sec to the real session timeout
10390 Allow the session to live 5sec longer before really timing out. This should give
10391 clients some extra time to keep the session active.
10393 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10395 * gst/rtsp-server/rtsp-client.c:
10396 client: replay OK to GET/SET_PARAMETER
10397 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
10398 so that we return OK for those requests.
10400 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10402 * gst/rtsp-server/rtsp-media.c:
10403 * gst/rtsp-server/rtsp-media.h:
10404 media: keep track of active transports
10405 Keep track of which transport is active to avoid closing the connection too
10407 Remove the destination transport also when going to NULL.
10408 Print some stats about the SDES and other RTCP messages we receive from the
10411 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10413 * examples/.gitignore:
10414 * examples/Makefile.am:
10415 * examples/test-sdp.c:
10416 example: add SDP relay example
10418 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10420 * gst/rtsp-server/rtsp-media.c:
10421 media: also count active TCP connections
10423 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10425 * gst/rtsp-server/rtsp-media-factory.c:
10426 * gst/rtsp-server/rtsp-media.c:
10427 * gst/rtsp-server/rtsp-media.h:
10428 rtsp: add support for dynamic elements
10429 Add support for dynamic elements.
10430 Don't set live pipelines back to paused.
10432 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10434 * gst/rtsp-server/rtsp-sdp.c:
10435 sdp: don't add encoding name when absent in caps
10437 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10439 * gst/rtsp-server/rtsp-client.c:
10440 client: warn when we can't do RTP-Info
10442 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10444 * gst/rtsp-server/rtsp-media-factory.c:
10445 factory: factor out the stream construction
10447 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10449 * gst/rtsp-server/rtsp-client.c:
10450 client: only add RTP-Info when we have the info
10451 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
10454 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10457 back to development
10459 === release 0.10.3 ===
10461 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10465 - Fixes a bug where it put the wrong verion in pkgconfig
10466 - Link RTP and RTCP sources
10468 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10470 * gst/rtsp-server/rtsp-media.c:
10471 * gst/rtsp-server/rtsp-media.h:
10472 media: link the RTP udpsrc to the session manager
10473 Link the RTP udpsrc and the appsrc to the session manager so that they don't
10474 shut down when the client sends a packet to open firewalls.
10476 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10478 * pkgconfig/gst-rtsp-server.pc.in:
10479 Don't use hard-coded version number in pkg-config file
10481 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10484 back to development
10486 === release 0.10.2 ===
10488 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10493 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10496 * common/m4/.gitignore:
10497 * examples/.gitignore:
10498 * pkgconfig/.gitignore:
10499 add some .gitignore files
10501 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10503 * gst/rtsp-server/rtsp-media.c:
10504 media: seek to key frames
10506 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10508 * gst/rtsp-server/rtsp-media.c:
10509 media: emit the unprepared signal by id
10510 Emit the unprepared signal by id instead of name and set the media as
10513 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10515 * gst/rtsp-server/rtsp-media.c:
10516 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
10518 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10520 * gst/rtsp-server/rtsp-server.c:
10521 Added finalize function to GstRTPSPServer to unref session pool and media mapping
10523 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10525 * bindings/vala/gst-rtsp-server-0.10.vapi:
10526 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
10527 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
10528 Updated vala bindings
10530 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10532 * gst/rtsp-server/Makefile.am:
10533 * gst/rtsp-server/rtsp-client.c:
10534 * gst/rtsp-server/rtsp-media.c:
10535 server: use appsink and appsrc with the API
10536 Use the appsink/appsrc API instead of the signals for higher
10539 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10541 * examples/test-ogg.c:
10542 tests: set the payload type correctly
10544 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10546 * gst/rtsp-server/rtsp-media-factory.c:
10547 factory: connect to the unprepare signal
10548 Connect to the unprepare signal for non-reusable media so that we can remove
10549 them from the cache.
10551 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10553 * gst/rtsp-server/rtsp-media.c:
10554 * gst/rtsp-server/rtsp-media.h:
10555 media: add signal to notify of unprepare
10557 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10559 * gst/rtsp-server/rtsp-media.c:
10560 * gst/rtsp-server/rtsp-media.h:
10561 media: more work on making the media shared
10562 Add a reusable flag to medias, indicating that they can be reused after a state
10566 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10568 * examples/test-readme.c:
10569 examples: mark the example as shared for testing
10571 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10573 * gst/rtsp-server/rtsp-media.c:
10574 * gst/rtsp-server/rtsp-media.h:
10575 client: support shared media
10576 Always perform the state actions even if the target state of the pipeline is
10577 already correct, we still want to add/remove the transports when we are dealing
10579 Keep a counter of the number of active transports for a media so that we can use
10580 this to perform a state change when needed.
10581 Perform a state change of the pipeline only when the first transport was added
10582 or when there are no active transports.
10584 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10586 * gst/rtsp-server/rtsp-client.c:
10587 client: fix refcounting crasher
10588 Don't need to remove the weak refs in the finalize methods, they are already
10589 removed in the dispose.
10590 Don't register the callback with a DestroyNofity.
10592 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10594 * gst/rtsp-server/rtsp-client.c:
10595 Fix rtsp client refcount management in TCP mode.
10596 Don't unref a client ref we never had. Fixes an unref
10597 of an already-free client object after a client
10598 teardown request for me.
10600 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10602 * gst/rtsp-server/rtsp-session.c:
10603 docs: fix typo in API docs
10605 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10607 * gst/rtsp-server/rtsp-media.c:
10608 More seeking fixes.
10609 Keep the udp sources in playing even if we go to paused. unlock the sources when
10611 Add some more debug info.
10612 Only seek when we need to.
10613 Keep track of the position when we go to paused.
10615 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10617 * gst/rtsp-server/rtsp-client.c:
10618 * gst/rtsp-server/rtsp-media.c:
10619 * gst/rtsp-server/rtsp-media.h:
10620 Add beginnings of seeking.
10621 Parse the Range header and perform a seek on the pipeline for the requested
10622 position. It's disabled currently until I figure out what's going wrong.
10624 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10626 * gst/rtsp-server/rtsp-client.c:
10627 allow pause requests for now.
10630 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10632 * gst/rtsp-server/rtsp-client.c:
10633 Remove weak ref on the session in teardown
10634 We need to remove our weakref from the session when we do a teardown because
10635 else we close the TCP connection prematurely.
10637 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10639 * gst/rtsp-server/rtsp-client.c:
10640 * gst/rtsp-server/rtsp-client.h:
10641 * gst/rtsp-server/rtsp-session-pool.c:
10642 Do some more session cleanup
10643 Make session timeout kill the TCP connection that currently watches the
10645 Remove the client timeout property.
10647 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10649 * gst/rtsp-server/rtsp-client.c:
10650 * gst/rtsp-server/rtsp-client.h:
10651 * gst/rtsp-server/rtsp-media.c:
10652 * gst/rtsp-server/rtsp-media.h:
10653 * gst/rtsp-server/rtsp-server.c:
10654 * gst/rtsp-server/rtsp-session.c:
10655 * gst/rtsp-server/rtsp-session.h:
10657 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
10660 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10662 * examples/Makefile.am:
10663 * examples/test-launch.c:
10664 Add example server that takes launch lines
10665 Add an example server that streams any -launch line.
10667 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10669 * examples/test-readme.c:
10670 * gst/rtsp-server/rtsp-client.c:
10671 * gst/rtsp-server/rtsp-media.c:
10672 * gst/rtsp-server/rtsp-media.h:
10673 Add support for live streams
10674 Add support for live streams and ranges
10675 Start on handling TCP data transfer.
10677 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10679 * gst/rtsp-server/rtsp-media.c:
10680 Free the pipeline before other things
10683 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10685 * gst/rtsp-server/rtsp-client.c:
10686 Only free the pending tunnel if there is one
10689 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10691 * gst/rtsp-server/rtsp-client.c:
10692 * gst/rtsp-server/rtsp-client.h:
10693 * gst/rtsp-server/rtsp-media.c:
10694 rtsp-server: Add support for tunneling
10695 Add support for tunneling over HTTP.
10696 Use new connection methods to retrieve the url.
10697 Dispatch messages based on the message type instead of blindly
10698 assuming it's always a request.
10699 Keep track of the watch id so that we can remove it later.
10700 Set the media pipeline to NULL before unreffing the pipeline.
10702 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10704 * gst/rtsp-server/rtsp-client.c:
10705 * gst/rtsp-server/rtsp-client.h:
10706 Fix for channel -> watch rename in gstreamer
10707 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
10709 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10711 * gst/rtsp-server/rtsp-client.c:
10712 * gst/rtsp-server/rtsp-client.h:
10714 Use the async RTSP channels instead of spawning a new thread for each client.
10715 If a sessionid is specified in a request, fail if we don't have the session.
10717 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10719 * gst/rtsp-server/rtsp-media.c:
10720 Add better debug info
10721 Add some better debug info.
10723 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10725 * examples/test-video.c:
10727 Add support for session timeouts in the example.
10729 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10731 * gst/rtsp-server/rtsp-session-pool.c:
10732 * gst/rtsp-server/rtsp-session-pool.h:
10733 Pass GTimeVal around for performance reasons
10734 Get the current time only once and pass it around so that sessions don't have to
10735 get the current time anymore.
10736 Add experimental support for a GSource that dispatches when the session needs to
10739 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10741 * gst/rtsp-server/rtsp-session.c:
10742 * gst/rtsp-server/rtsp-session.h:
10743 Add better support for session timeouts
10744 Add a method to request the number of milliseconds when a session will timeout.
10746 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10748 * gst/rtsp-server/rtsp-media.c:
10749 * gst/rtsp-server/rtsp-media.h:
10750 Add suport for RTP manager monitoring
10751 Add the first stage in monitoring the rtp manager.
10752 Make sure we don't update the state to something we don't want.
10754 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10756 * gst/rtsp-server/rtsp-client.c:
10757 Add support for session keepalive
10758 Get and update the session timeout for all requests. get the session as early as
10761 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10763 * gst/rtsp-server/rtsp-media-factory.h:
10764 * gst/rtsp-server/rtsp-media.c:
10765 * gst/rtsp-server/rtsp-media.h:
10766 Handle media bus messages
10767 Handle media bus messages in a custom mainloop and dispatch them to the
10768 RTSPMedia objects. Let the default implementation handle some common messages.
10770 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10772 * gst/rtsp-server/rtsp-client.c:
10773 * gst/rtsp-server/rtsp-session-pool.c:
10774 * gst/rtsp-server/rtsp-session.c:
10775 Some more session timeout handling
10776 Move the session header setting code to a central place so that we always add
10777 the timeout parameter too.
10778 Handle timeouts by running the session cleanup code.
10779 Stop media before cleaning up.
10781 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10783 * gst/rtsp-server/rtsp-client.c:
10784 * gst/rtsp-server/rtsp-client.h:
10785 Add timeout property
10786 Add a timeout property ot the client and make the other properties into GObject
10789 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10791 * gst/rtsp-server/rtsp-session-pool.c:
10792 Use getters and setters in property code
10793 Use the getters and setters for the timeout property instead of locking
10796 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10798 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
10800 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10802 * gst/rtsp-server/rtsp-session-pool.c:
10803 * gst/rtsp-server/rtsp-session-pool.h:
10804 * gst/rtsp-server/rtsp-session.c:
10805 * gst/rtsp-server/rtsp-session.h:
10806 Add more timeout stuff
10807 Add method to check if a session is expired.
10808 Add method to perform cleanup on a session pool.
10810 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10812 * gst/rtsp-server/rtsp-client.c:
10813 * gst/rtsp-server/rtsp-session-pool.c:
10814 * gst/rtsp-server/rtsp-session-pool.h:
10815 * gst/rtsp-server/rtsp-session.c:
10816 * gst/rtsp-server/rtsp-session.h:
10817 Add beginnings of session timeouts and limits
10818 Add the timeout value to the Session header for unusual timeout values.
10819 Allow us to configure a limit to the amount of active sessions in a pool. Set a
10820 limit on the amount of retry we do after a sessionid collision.
10821 Add properties to the sessionid and the timeout of a session. Keep track of
10822 creation time and last access time for sessions.
10824 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10826 * gst/rtsp-server/rtsp-client.c:
10827 * gst/rtsp-server/rtsp-media.c:
10828 * gst/rtsp-server/rtsp-media.h:
10829 * gst/rtsp-server/rtsp-sdp.c:
10830 * gst/rtsp-server/rtsp-session-pool.c:
10831 * gst/rtsp-server/rtsp-session.c:
10832 * gst/rtsp-server/rtsp-session.h:
10833 Cleanup of sessions and more
10834 Fix the refcounting of media and sessions in the client. Properly clean up the
10835 session data when the client performs a teardown.
10836 Add Server header to responses.
10837 Allow for multiple uri setups in one session.
10838 Add Range header to the PLAY response and add the range attribute to the SDP
10840 Fix the session pool remove method, it used the wrong key in the hashtable. Also
10841 give the ownership of the sessionid to the session object.
10843 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10845 * gst/rtsp-server/rtsp-server.c:
10846 * gst/rtsp-server/rtsp-server.h:
10848 Rename the 'server_port' variable to simply 'port'.
10850 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10853 * gst/rtsp-server/rtsp-client.c:
10854 * gst/rtsp-server/rtsp-media.c:
10855 * gst/rtsp-server/rtsp-media.h:
10856 * gst/rtsp-server/rtsp-session.c:
10857 * gst/rtsp-server/rtsp-session.h:
10858 Rework the way we handle transports for streams
10859 Make the media accept an array of transports for the streams that we have
10860 configured for the play/pause requests.
10861 Implement server states for a client and its media.
10862 Require 0.10.22.1 (git HEAD) of gstreamer.
10864 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10866 * gst/rtsp-server/rtsp-client.c:
10867 * gst/rtsp-server/rtsp-media-factory.c:
10868 Drop const from functions dealing with urls
10869 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
10870 have the right const in them.
10872 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10874 * gst/rtsp-server/rtsp-client.c:
10875 * gst/rtsp-server/rtsp-media.c:
10876 * gst/rtsp-server/rtsp-sdp.c:
10880 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10882 * gst/rtsp-server/rtsp-client.c:
10883 * gst/rtsp-server/rtsp-media-factory.c:
10884 * gst/rtsp-server/rtsp-media.c:
10885 * gst/rtsp-server/rtsp-media.h:
10887 Don't keep a reference to the GstRTSPMedia in the stream.
10888 Free more things when freeing the GstRTSPMedia.
10890 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10893 * gst/rtsp-server/rtsp-media-factory.c:
10894 * gst/rtsp-server/rtsp-media-factory.h:
10895 * gst/rtsp-server/rtsp-media.c:
10896 * gst/rtsp-server/rtsp-media.h:
10897 * gst/rtsp-server/rtsp-server.c:
10898 * gst/rtsp-server/rtsp-server.h:
10899 More docs and small cleanups
10900 Add some more docs and update the README
10901 Cleanup some method names.
10902 Remove an unneeded idx field in the GstRTSPMediaStream
10904 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10907 * examples/Makefile.am:
10908 * examples/test-readme.c:
10909 Add a README and more example code
10910 Add a README file that contains a small introduction on how to use the server
10911 along with the example code explained in the readme.
10913 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10915 * gst/rtsp-server/rtsp-media.c:
10916 * gst/rtsp-server/rtsp-server.c:
10917 Fix some leaks and change default port
10918 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
10919 we finished the initial preroll. If we keep them locked, setting the pipeline to
10920 NULL will not stop and clean up the sources correctly.
10921 Change the default RTSP port to 8554 aka the official alternative RTSP port.
10923 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10925 * gst/rtsp-server/rtsp-session.c:
10926 * gst/rtsp-server/rtsp-session.h:
10927 Cleanups to the session object
10928 Remove some unneeded variables in the session state of a stream such as the
10929 owner media and the server transport.
10930 Get the configuration of a media stream in a session based on the media_stream
10931 in the original object instead of our cached index.
10932 Free more data in the finalize method.
10934 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10936 * gst/rtsp-server/rtsp-client.c:
10937 * gst/rtsp-server/rtsp-client.h:
10938 Cleanups and reuse media from DESCRIBE
10939 Handle thread create errors.
10940 Rename some internal methods to better match what they actually do.
10941 Handle misconfiguration of session_pool and media_mapping gracefully.
10942 Cache the DESCRIBE media and uri in the client connection and reuse them when
10943 we receive a SETUP request in the same connection for the same uri.
10944 Cleanup the client connection object.
10946 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10948 * gst/rtsp-server/rtsp-media-factory.c:
10949 * gst/rtsp-server/rtsp-media-factory.h:
10950 * gst/rtsp-server/rtsp-media.c:
10951 * gst/rtsp-server/rtsp-media.h:
10952 Add shared properties to media and factory
10953 Add the shared property to media.
10954 Implement some simple caching in the factory depending on if the media is shared
10957 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10959 * gst/rtsp-server/rtsp-client.c:
10960 Add a little comment
10961 Add some comment about the content-base header.
10963 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10965 * examples/Makefile.am:
10966 * examples/test-mp4.c:
10967 * examples/test-ogg.c:
10968 * examples/test-video.c:
10969 * gst/rtsp-server/Makefile.am:
10970 * gst/rtsp-server/rtsp-client.c:
10971 * gst/rtsp-server/rtsp-client.h:
10972 * gst/rtsp-server/rtsp-media-factory.c:
10973 * gst/rtsp-server/rtsp-media-factory.h:
10974 * gst/rtsp-server/rtsp-media.c:
10975 * gst/rtsp-server/rtsp-media.h:
10976 * gst/rtsp-server/rtsp-sdp.c:
10977 * gst/rtsp-server/rtsp-sdp.h:
10978 * gst/rtsp-server/rtsp-server.c:
10979 * gst/rtsp-server/rtsp-server.h:
10980 * gst/rtsp-server/rtsp-session.c:
10981 * gst/rtsp-server/rtsp-session.h:
10982 Reorganize things, prepare for media sharing
10983 Added various other test server examples
10984 Move the SDP message generation to a separate helper.
10985 Refactor common code for finding the session.
10986 Add content-base for realplayer compatibility
10987 Clean up request uris before processing for better vlc compatibility.
10988 Move prerolling and pipeline construction to the RTSPMedia object.
10989 Use multiudpsink for future pipeline reuse.
10991 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10994 Back to development
10997 === release 0.10.1 ===
10999 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11002 Make 0.10.1 release
11005 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11007 * bindings/vala/Makefile.am:
11009 Add more directories and files to the dist.
11011 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11013 * bindings/python/Makefile.am:
11014 * bindings/python/rtspserver.override:
11015 Fixed compile error of python bindings
11017 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11019 * bindings/vala/gst-rtsp-server-0.10.vapi:
11020 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11021 Marked values as nullable accordingly
11023 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11025 * bindings/vala/gst-rtsp-server-0.10.vapi:
11026 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
11027 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11028 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11029 Updated Vala bindings
11031 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11033 * gst/rtsp-server/rtsp-client.c:
11034 * gst/rtsp-server/rtsp-media-mapping.c:
11035 * gst/rtsp-server/rtsp-media-mapping.h:
11036 * gst/rtsp-server/rtsp-media.h:
11037 * gst/rtsp-server/rtsp-session-pool.h:
11038 Cleanups and doc updates
11039 Add some more documentation and do some minor cleanups here and there.
11041 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11043 * gst/rtsp-server/rtsp-client.c:
11044 * gst/rtsp-server/rtsp-media-factory.c:
11045 * gst/rtsp-server/rtsp-media-factory.h:
11046 * gst/rtsp-server/rtsp-media.c:
11047 * gst/rtsp-server/rtsp-media.h:
11048 * gst/rtsp-server/rtsp-session.c:
11049 * gst/rtsp-server/rtsp-session.h:
11051 Rename GstRTSPMediaBin to GstRTSPMedia
11052 Parse the request url into a GstRTSPUri object and pass this object to the
11053 various handlers and methods that require the uri.
11055 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11059 Add some more docs and remove some old code from the example.
11061 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11063 * gst/rtsp-server/rtsp-client.c:
11064 Handle state change failures better
11065 Handle state change failures better when changing the state of the pipeline to
11068 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11070 * gst/rtsp-server/rtsp-media-factory.c:
11071 * gst/rtsp-server/rtsp-media-factory.h:
11072 Make element creation more extendible
11073 Add get_element vmethod to the default MediaFactory so that subclasses can just
11074 override that method and still use the default logic for making a MediaBin from
11077 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11080 * gst/rtsp-server/Makefile.am:
11081 * gst/rtsp-server/rtsp-client.c:
11082 * gst/rtsp-server/rtsp-client.h:
11083 * gst/rtsp-server/rtsp-media-factory.c:
11084 * gst/rtsp-server/rtsp-media-factory.h:
11085 * gst/rtsp-server/rtsp-media-mapping.c:
11086 * gst/rtsp-server/rtsp-media-mapping.h:
11087 * gst/rtsp-server/rtsp-media.c:
11088 * gst/rtsp-server/rtsp-media.h:
11089 * gst/rtsp-server/rtsp-server.c:
11090 * gst/rtsp-server/rtsp-server.h:
11091 * gst/rtsp-server/rtsp-session.c:
11092 * gst/rtsp-server/rtsp-session.h:
11093 Make the server handle arbitrary pipelines
11094 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
11095 The GstMediaBin object has a handle to a bin with elements and to a list of
11096 GstMediaStream objects that this bin produces.
11097 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
11098 with methods to register and remove those mappings.
11099 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
11100 used by the server instance.
11101 Modify the example application so that it shows how to create custom pipelines
11102 attached to a specific mount point.
11103 Various misc cleanps.
11105 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11107 * gst/rtsp-server/rtsp-server.c:
11108 * gst/rtsp-server/rtsp-server.h:
11109 Allow setting a custom media factory for a server
11111 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11113 * gst/rtsp-server/rtsp-client.c:
11114 * gst/rtsp-server/rtsp-client.h:
11115 Allow setting a custom media factory for a client.
11117 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11119 * gst/rtsp-server/Makefile.am:
11120 Add Makefile entry for the media factory
11122 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11124 * gst/rtsp-server/rtsp-media-factory.c:
11125 * gst/rtsp-server/rtsp-media-factory.h:
11126 Add media factory to map urls to media pipeline objects.
11128 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11130 * gst/rtsp-server/rtsp-media.c:
11131 * gst/rtsp-server/rtsp-media.h:
11132 Add comments. Remove unused field
11134 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11136 * gst/rtsp-server/rtsp-session-pool.c:
11137 * gst/rtsp-server/rtsp-session-pool.h:
11138 Allow custom session pools to override the session id allocation algorithms Add some comments.
11140 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11142 * gst/rtsp-server/rtsp-session.h:
11145 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11147 * gst/rtsp-server/rtsp-client.c:
11148 * gst/rtsp-server/rtsp-client.h:
11149 Move the connection code in one place Add some comments
11151 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11153 * gst/rtsp-server/rtsp-server.c:
11154 * gst/rtsp-server/rtsp-server.h:
11155 Make vmethod to create and accept new clients. Add some docs.
11157 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11159 * gst/rtsp-server/rtsp-server.c:
11160 * gst/rtsp-server/rtsp-server.h:
11161 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
11163 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11165 * gst/rtsp-server/rtsp-client.c:
11166 * gst/rtsp-server/rtsp-client.h:
11167 Name the parameters more appropriately.
11169 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11171 * gst/rtsp-server/rtsp-session-pool.c:
11172 Do some more cleanup of the session pool.
11174 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11176 * gst/rtsp-server/Makefile.am:
11177 * gst/rtsp-server/rtsp-client.c:
11178 Check if return value of gst_rtsp_session_get_media is not NULL
11180 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11182 * gst/rtsp-server/Makefile.am:
11183 Install rtsp-session and rtsp-session-pool headers
11185 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11190 * bindings/python/Makefile.am:
11191 * bindings/python/arg-types.py:
11192 * bindings/python/codegen/Makefile.am:
11193 * bindings/python/codegen/__init__.py:
11194 * bindings/python/codegen/argtypes.py:
11195 * bindings/python/codegen/code-coverage.py:
11196 * bindings/python/codegen/codegen.py:
11197 * bindings/python/codegen/definitions.py:
11198 * bindings/python/codegen/defsparser.py:
11199 * bindings/python/codegen/docextract.py:
11200 * bindings/python/codegen/docgen.py:
11201 * bindings/python/codegen/fileprefix.override:
11202 * bindings/python/codegen/fileprefixmodule.c:
11203 * bindings/python/codegen/h2def.py:
11204 * bindings/python/codegen/mergedefs.py:
11205 * bindings/python/codegen/mkskel.py:
11206 * bindings/python/codegen/override.py:
11207 * bindings/python/codegen/reversewrapper.py:
11208 * bindings/python/codegen/scmexpr.py:
11209 * bindings/python/rtspserver-types.defs:
11210 * bindings/python/rtspserver.defs:
11211 * bindings/python/rtspserver.override:
11212 * bindings/python/rtspservermodule.c:
11214 Add python bindings.
11216 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11218 * bindings/Makefile.am:
11220 Don't go into python dir when requirements for python bindings are missing
11222 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11224 * bindings/Makefile.am:
11225 * bindings/vala/Makefile.am:
11227 Install Vala bindings if vala is available
11229 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11231 * bindings/vala/gst-rtsp-server-0.10.deps:
11232 * bindings/vala/gst-rtsp-server-0.10.vapi:
11233 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
11234 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
11235 * bindings/vala/packages/gst-rtsp-server-0.10.files:
11236 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11237 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11238 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
11239 Regenerated Vala bindings
11241 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11243 * bindings/vala/gst-rtsp-server.vapi:
11244 * bindings/vala/packages/gst-rtsp-server.metadata:
11245 Fixed typo in included headers for vala bindings
11247 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11251 * pkgconfig/Makefile.am:
11252 * pkgconfig/gst-rtsp-server.pc.in:
11253 Added pkgconfig file
11255 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
11257 * bindings/vala/gst-rtsp-server.vapi:
11258 * bindings/vala/packages/gst-rtsp-server.excludes:
11259 * bindings/vala/packages/gst-rtsp-server.gi:
11260 * bindings/vala/packages/gst-rtsp-server.metadata:
11261 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
11263 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
11265 * bindings/vala/gst-rtsp-server.vapi:
11266 * bindings/vala/packages/gst-rtsp-server.deps:
11267 * bindings/vala/packages/gst-rtsp-server.files:
11268 * bindings/vala/packages/gst-rtsp-server.gi:
11269 * bindings/vala/packages/gst-rtsp-server.metadata:
11270 * bindings/vala/packages/gst-rtsp-server.namespace:
11271 Added Vala bindings
11273 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
11275 * gst/rtsp-server/rtsp-session.c:
11276 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
11278 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
11280 * examples/Makefile.am:
11281 * gst/rtsp-server/Makefile.am:
11282 Put GStreamer version in library name
11284 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11286 * examples/Makefile.am:
11287 * gst/rtsp-server/Makefile.am:
11288 Fix some issues to pass distcheck
11290 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11292 * gst/rtsp-server/rtsp-server.c:
11293 Added port property to GstRTSPServer class.
11295 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11300 * examples/Makefile.am:
11303 * gst/rtsp-server/Makefile.am:
11304 * gst/rtsp-server/rtsp-client.c:
11305 * gst/rtsp-server/rtsp-client.h:
11306 * gst/rtsp-server/rtsp-media.c:
11307 * gst/rtsp-server/rtsp-media.h:
11308 * gst/rtsp-server/rtsp-server.c:
11309 * gst/rtsp-server/rtsp-server.h:
11310 * gst/rtsp-server/rtsp-session-pool.c:
11311 * gst/rtsp-server/rtsp-session-pool.h:
11312 * gst/rtsp-server/rtsp-session.c:
11313 * gst/rtsp-server/rtsp-session.h:
11315 Split in library and example program
11317 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
11319 * src/rtsp-client.h:
11320 Removed obsolete variable
11322 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
11324 * src/rtsp-client.c:
11325 * src/rtsp-client.h:
11326 Removed pipeline variable GstRTSPClient, because it's only used in one function
11328 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11330 * src/rtsp-media.c:
11331 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
11333 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
11335 * src/rtsp-session.c:
11336 Initialize some more vars.
11338 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
11340 * src/rtsp-session.c:
11341 Initialize variable to avoid compiler warning.
11343 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
11346 Add a reasonable generic .gitignore
projects / platform / upstream / gstreamer.git / blob