1 === release 1.13.91 ===
3 2018-03-13 19:28:33 +0000 Tim-Philipp Müller <tim@centricular.com>
8 * gst-rtsp-server.doap:
12 2018-03-13 13:30:41 +0000 Tim-Philipp Müller <tim@centricular.com>
14 * gst/rtsp-server/Makefile.am:
15 * gst/rtsp-server/meson.build:
16 * gst/rtsp-server/rtsp-address-pool.h:
17 * gst/rtsp-server/rtsp-auth.h:
18 * gst/rtsp-server/rtsp-client.h:
19 * gst/rtsp-server/rtsp-context.h:
20 * gst/rtsp-server/rtsp-media-factory-uri.h:
21 * gst/rtsp-server/rtsp-media-factory.h:
22 * gst/rtsp-server/rtsp-media.h:
23 * gst/rtsp-server/rtsp-mount-points.h:
24 * gst/rtsp-server/rtsp-onvif-client.h:
25 * gst/rtsp-server/rtsp-onvif-media-factory.h:
26 * gst/rtsp-server/rtsp-onvif-media.h:
27 * gst/rtsp-server/rtsp-onvif-server.h:
28 * gst/rtsp-server/rtsp-params.h:
29 * gst/rtsp-server/rtsp-permissions.h:
30 * gst/rtsp-server/rtsp-sdp.h:
31 * gst/rtsp-server/rtsp-server-prelude.h:
32 * gst/rtsp-server/rtsp-server.h:
33 * gst/rtsp-server/rtsp-session-media.h:
34 * gst/rtsp-server/rtsp-session-pool.h:
35 * gst/rtsp-server/rtsp-session.h:
36 * gst/rtsp-server/rtsp-stream-transport.h:
37 * gst/rtsp-server/rtsp-stream.h:
38 * gst/rtsp-server/rtsp-thread-pool.h:
39 * gst/rtsp-server/rtsp-token.h:
40 rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API
41 We need different export decorators for the different libs.
42 For now no actual change though, just rename before the release,
43 and add prelude headers to define the new decorator to GST_EXPORT.
45 2018-03-07 12:20:05 +0200 Sebastian Dröge <sebastian@centricular.com>
47 * gst/rtsp-server/rtsp-onvif-media-factory.c:
48 rtsp-onvif-media-factory: Document that backchannel pipelines must end with async=false sinks
49 https://bugzilla.gnome.org/show_bug.cgi?id=794143
51 === release 1.13.90 ===
53 2018-03-03 22:49:34 +0000 Tim-Philipp Müller <tim@centricular.com>
59 * gst-rtsp-server.doap:
63 2018-03-02 16:24:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
65 * gst/rtsp-server/rtsp-media-factory.c:
66 * gst/rtsp-server/rtsp-permissions.c:
67 permissions: add Since tags and example for new API
69 2018-03-02 01:36:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
71 * docs/libs/gst-rtsp-server-sections.txt:
72 * gst/rtsp-server/rtsp-media-factory.c:
73 * gst/rtsp-server/rtsp-media-factory.h:
74 * gst/rtsp-server/rtsp-permissions.c:
75 * gst/rtsp-server/rtsp-permissions.h:
76 * tests/check/gst/permissions.c:
77 permissions: more bindings-friendly API
78 https://bugzilla.gnome.org/show_bug.cgi?id=793975
80 2018-03-01 19:28:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
83 meson: enable more warnings
85 2018-02-28 21:12:43 +0200 Sebastian Dröge <sebastian@centricular.com>
87 * gst/rtsp-server/rtsp-client.c:
88 rtsp-client: Place netaddress meta on packets received via TCP
89 This allows us to later map signals from rtpbin/rtpsource back to the
90 corresponding stream transport, and allows to do keep-alive based on
91 RTCP packets in case of TCP media transport.
92 https://bugzilla.gnome.org/show_bug.cgi?id=789646
94 2018-02-27 20:34:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
96 * gst/rtsp-sink/gstrtspclientsink.c:
97 rtspclientsink: if OPEN failed, unqueue next command
98 As READY_TO_PAUSED can no longer return async, the RECORD
99 command will be queued before the OPEN command fails
100 (for example in case the server could not be connected),
101 and record then waits for ever.
102 https://bugzilla.gnome.org/show_bug.cgi?id=793896
104 2018-02-26 22:59:17 +0100 Mathieu Duponchelle <mathieu@centricular.com>
106 * gst/rtsp-sink/gstrtspclientsink.c:
107 rtspclientsink: fix retrieval of custom payloader caps
108 If a bin is passed as the custom payloader, the caps of
109 its factory will be empty, the correct way to obtain the caps
110 is to query its sinkpad.
112 2018-02-26 22:59:00 +0100 Mathieu Duponchelle <mathieu@centricular.com>
114 * gst/rtsp-sink/gstrtspclientsink.c:
115 rtspclientsink: fix extra unref of custom payloader
117 2018-02-26 22:57:39 +0100 Mathieu Duponchelle <mathieu@centricular.com>
119 * gst/rtsp-sink/gstrtspclientsink.c:
120 rspclientsink: fix recent code indentation
122 2018-02-26 20:27:57 +0100 Mathieu Duponchelle <mathieu@centricular.com>
124 * gst/rtsp-sink/gstrtspclientsink.c:
125 rtspclientsink: add missing get_type prototype
127 2018-02-24 03:52:15 +0100 Mathieu Duponchelle <mathieu@centricular.com>
129 * gst/rtsp-sink/gstrtspclientsink.c:
130 rtspclientsink: allow setting payloader as pad property
131 This was a FIXME item, and can be quite useful, also
132 allowing to specify payloader properties from the command
133 line, which is always nice.
134 https://bugzilla.gnome.org/show_bug.cgi?id=793776
136 2018-02-26 14:16:54 +0100 Carlos Rafael Giani <dv@pseudoterminal.org>
138 * gst/rtsp-server/rtsp-media.c:
139 rtsp-media: Replace g_print() log line
140 https://bugzilla.gnome.org/show_bug.cgi?id=793838
142 2018-02-22 20:17:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
144 * gst/rtsp-server/rtsp-media.c:
145 * tests/check/gst/rtspclientsink.c:
146 rtsp-media: fix RECORD getting stuck
147 The test_record case was working because async=false had
148 been added in https://bugzilla.gnome.org/show_bug.cgi?id=757488
149 but that was incorrect, as it should not be needed.
150 Removing async=false made the test fail as expected, this is
151 fixed by not trying to preroll when preparing the media for
152 RECORD, as start_prepare is called upon receiving ANNOUNCE,
153 and our peer will not start sending media until it has received
154 a response to that request, and sent and received a response
155 to RECORD as well, thus obviously preventing preroll.
156 https://bugzilla.gnome.org/show_bug.cgi?id=793738
158 2018-02-23 03:26:21 +0100 Mathieu Duponchelle <mathieu@centricular.com>
160 * gst/rtsp-server/rtsp-auth.c:
161 rtsp-auth: fix set_tls_authentication_mode annotation
163 2018-02-19 11:57:29 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
165 * gst/rtsp-server/rtsp-onvif-media.c:
166 rtp-server: remove redefined variable
167 res is a boolean variable which is defined in the function scope and
168 redefined, with no reason, in the loop scope. This patch removes the
170 https://bugzilla.gnome.org/show_bug.cgi?id=793592
172 2018-02-05 11:49:07 +0100 Ognyan Tonchev <ognyan@axis.com>
174 * gst/rtsp-server/rtsp-media.c:
175 * gst/rtsp-server/rtsp-stream.c:
176 * gst/rtsp-server/rtsp-stream.h:
177 stream: Add functions for checking if stream is receiver or sender
178 ...and replace all checks for RECORD in GstRTSPMedia which are really
179 for "sender-only". This way the code becomes more generic and introducing
180 support for onvif-backchannel later on will require no changes in
183 2017-10-21 14:06:30 +0200 Ognyan Tonchev <ognyan@axis.com>
185 * gst/rtsp-server/rtsp-onvif-media-factory.c:
186 * gst/rtsp-server/rtsp-onvif-media-factory.h:
187 onvif: Make requires_backchannel() public
188 ...in order to let subclasses building the onvif part of the pipeline
189 check whether backchannel shall be included or not.
191 2018-01-22 12:46:34 +0200 Sebastian Dröge <sebastian@centricular.com>
193 * gst/rtsp-server/rtsp-onvif-media.c:
194 rtsp-server: Switch around sendonly/recvonly attributes
195 They are wrong in the ONVIF streaming spec. The backchannel should be
196 recvonly and the normal media should be sendonly: direction is always
197 from the point of view of the SDP offerer (the server) according to
200 2017-09-25 19:41:05 +0300 Sebastian Dröge <sebastian@centricular.com>
202 * docs/libs/gst-rtsp-server-docs.sgml:
203 * docs/libs/gst-rtsp-server-sections.txt:
204 * examples/.gitignore:
205 * examples/Makefile.am:
206 * examples/test-onvif-backchannel.c:
207 * gst/rtsp-server/Makefile.am:
208 * gst/rtsp-server/rtsp-media.h:
209 * gst/rtsp-server/rtsp-onvif-client.c:
210 * gst/rtsp-server/rtsp-onvif-client.h:
211 * gst/rtsp-server/rtsp-onvif-media-factory.c:
212 * gst/rtsp-server/rtsp-onvif-media-factory.h:
213 * gst/rtsp-server/rtsp-onvif-media.c:
214 * gst/rtsp-server/rtsp-onvif-media.h:
215 * gst/rtsp-server/rtsp-onvif-server.c:
216 * gst/rtsp-server/rtsp-onvif-server.h:
217 * gst/rtsp-server/rtsp-sdp.c:
218 * gst/rtsp-server/rtsp-sdp.h:
219 rtsp: Add support for ONVIF backchannel
220 This adds a new RTSP server, client, media-factory and media subclass
221 for handling the specifics of the backchannel. Ideally this later can be
222 extended with other ONVIF specific features.
224 2017-10-12 21:00:16 +0300 Sebastian Dröge <sebastian@centricular.com>
226 * gst/rtsp-server/rtsp-media.c:
227 rtsp-media: Add support for sending+receiving medias
228 We need to add an appsrc/appsink in that case because otherwise the
229 media bin will be a sink and a source for rtpbin, causing a pipeline
231 https://bugzilla.gnome.org/show_bug.cgi?id=788950
233 2018-02-15 19:44:28 +0000 Tim-Philipp Müller <tim@centricular.com>
239 === release 1.13.1 ===
241 2018-02-15 17:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
245 * gst-rtsp-server.doap:
249 2018-02-14 17:11:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
251 * gst/rtsp-server/rtsp-session-pool.c:
252 session-pool: remove nullable return annotation
253 create_watch can only return NULL from the API guards, no
256 2018-02-13 18:59:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
258 * gst/rtsp-server/rtsp-media-factory.c:
259 * gst/rtsp-server/rtsp-media.c:
260 set_clock functions: Add nullable annotations
262 2018-02-10 00:07:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
264 * gst/rtsp-server/rtsp-auth.c:
265 * gst/rtsp-server/rtsp-client.c:
266 * gst/rtsp-server/rtsp-media-factory.c:
267 * gst/rtsp-server/rtsp-media.c:
268 * gst/rtsp-server/rtsp-mount-points.c:
269 * gst/rtsp-server/rtsp-server.c:
270 * gst/rtsp-server/rtsp-session-media.c:
271 * gst/rtsp-server/rtsp-session-pool.c:
272 * gst/rtsp-server/rtsp-session.c:
273 * gst/rtsp-server/rtsp-stream-transport.c:
274 * gst/rtsp-server/rtsp-stream.c:
275 * gst/rtsp-server/rtsp-thread-pool.c:
276 All around: add annotations and API guards
278 2018-02-12 19:12:35 +0100 Mathieu Duponchelle <mathieu@centricular.com>
280 * tests/test-cleanup.c:
281 test-cleanup: bind any port
282 The meson test suite runs tests in parallel, trying to bind
283 a single port made the test fail.
285 2018-02-08 19:15:10 +0000 Tim-Philipp Müller <tim@centricular.com>
288 meson: make version numbers ints and fix int/string comparison
289 WARNING: Trying to compare values of different types (str, int).
290 The result of this is undefined and will become a hard error
291 in a future Meson release.
293 2018-02-06 18:00:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
295 * gst/rtsp-server/rtsp-context.c:
296 gst_rtsp_context_get_current: add (skip) annotation
297 The return value type is defined with G_DEFINE_POINTER_TYPE,
298 and gi emits the following warning:
299 Invalid non-constant return of bare structure or union; register as
302 2018-02-06 17:58:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
304 * gst/rtsp-server/rtsp-client.c:
305 rtsp-client: add type annotations
306 gi doesn't seem to be able to figure out the type of the
307 signal parameters when defined with G_DEFINE_POINTER_TYPE
309 2018-02-04 12:24:09 +0100 Tim-Philipp Müller <tim@centricular.com>
312 autotools: use -fno-strict-aliasing where supported
313 https://bugzilla.gnome.org/show_bug.cgi?id=769183
315 2018-01-30 20:35:21 +0000 Tim-Philipp Müller <tim@centricular.com>
318 meson: use -fno-strict-aliasing where supported
319 https://bugzilla.gnome.org/show_bug.cgi?id=769183
321 2018-01-25 12:09:03 +0000 Tim-Philipp Müller <tim@centricular.com>
323 * gst/rtsp-server/rtsp-mount-points.c:
324 mount-points: bail out of loop again when matching mount points
325 Previous patch led to us iterating the entire sequence. Bail out
326 of the loop again if we have a match but are moving away from it.
327 https://bugzilla.gnome.org/show_bug.cgi?id=771555
329 2018-01-25 12:06:57 +0000 Tim-Philipp Müller <tim@centricular.com>
331 * tests/check/gst/mountpoints.c:
332 tests: mountpoints: add more checks for mount point path matching
333 https://bugzilla.gnome.org/show_bug.cgi?id=771555
335 2016-09-16 20:41:19 +0000 Andrew Bott <andrew.bott@blackmoth.com>
337 * gst/rtsp-server/rtsp-mount-points.c:
338 mount-points: fix matching of paths where there's also an entry with a common prefix
339 e.g. with the following mount points
343 _match() would not match /raw/video and /raw/snapshot correctly.
344 https://bugzilla.gnome.org/show_bug.cgi?id=771555
346 2018-01-18 23:53:20 +0000 Tim-Philipp Müller <tim@centricular.com>
348 * docs/libs/gst-rtsp-server-sections.txt:
349 * gst/rtsp-server/rtsp-permissions.c:
350 * gst/rtsp-server/rtsp-permissions.h:
351 * tests/check/gst/permissions.c:
352 permissions: add some new API to make this usable from bindings
353 https://bugzilla.gnome.org/show_bug.cgi?id=787073
355 2018-01-18 11:32:32 +0000 Tim-Philipp Müller <tim@centricular.com>
357 * gst/rtsp-server/rtsp-token.c:
358 rtsp-token: annotate constructors for bindings
359 This maps _new_empty() to _new(), which also makes RTSPToken()
360 work properly now. Since this API wasn't usable from bindings
361 before, this should hopefully be fine.
362 https://bugzilla.gnome.org/show_bug.cgi?id=787073
364 2018-01-18 11:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
366 * docs/libs/gst-rtsp-server-sections.txt:
367 * gst/rtsp-server/rtsp-token.c:
368 * gst/rtsp-server/rtsp-token.h:
369 * tests/check/gst/token.c:
370 rtsp-token: add some API to set fields from bindings
371 The existing functions are all vararg-based and as such
372 not usable from bindings.
373 https://bugzilla.gnome.org/show_bug.cgi?id=787073
375 2018-01-13 15:02:28 +0000 Tim-Philipp Müller <tim@centricular.com>
377 * tests/check/gst/rtspclientsink.c:
378 * tests/check/gst/rtspserver.c:
379 * tests/check/gst/sessionpool.c:
380 * tests/check/gst/stream.c:
381 tests: fix indentation
384 2018-01-13 14:58:55 +0000 Tim-Philipp Müller <tim@centricular.com>
386 * tests/check/gst/rtspserver.c:
387 tests: rtspserver: fix another ref leak
388 Even if this didn't show up in valgrind.
390 2018-01-13 14:58:00 +0000 Tim-Philipp Müller <tim@centricular.com>
392 * tests/check/gst/rtspclientsink.c:
393 tests: rtspclientsink: fix leak
395 2018-01-02 14:19:31 +0100 Branko Subasic <branko@axis.com>
397 * tests/check/gst/rtspserver.c:
398 test: rtspserver: plug memory leak in test_no_session_timeout
399 In test_no_session_timeout, unref the rtsp session object when the
401 https://bugzilla.gnome.org/show_bug.cgi?id=792127
403 2017-12-20 14:17:02 +0100 Edward Hervey <edward@centricular.com>
405 * gst/rtsp-sink/gstrtspclientsink.c:
406 rtpsclientsink: Initialize and clear newly added mutex and cond
407 While it *did* work, glib would automatically create new mutex and cond
408 ... which never got freed
410 2017-12-19 11:34:37 +0200 Sebastian Dröge <sebastian@centricular.com>
412 * gst/rtsp-server/rtsp-stream.c:
413 rtsp-stream: Set multicast TTL on the multicast sockets
414 And not if we do unicast UDP.
415 https://bugzilla.gnome.org/show_bug.cgi?id=791743
417 2017-12-19 11:14:48 +0200 Sebastian Dröge <sebastian@centricular.com>
419 * gst/rtsp-server/rtsp-stream.c:
420 rtsp-stream: Decide based on the sockets, not the addresses if we already allocated a socket
421 In the multicast case (as in test-multicast, not test-multicast2), the
422 address could be allocated/reserved (and thus set) already without
423 allocating the actual socket. We need to allocate the socket here still
424 instead of just claiming that it was already allocated.
425 See https://bugzilla.gnome.org/show_bug.cgi?id=791743#c2
427 2017-12-16 21:46:53 +0100 Patricia Muscalu <patricia@dovakhiin.com>
429 * gst/rtsp-sink/gstrtspclientsink.c:
430 * gst/rtsp-sink/gstrtspclientsink.h:
431 rtspclientsink: Use the new rtsp-stream API
432 https://bugzilla.gnome.org/show_bug.cgi?id=790412
434 2017-12-16 21:01:43 +0100 Patricia Muscalu <patricia@dovakhiin.com>
436 * gst/rtsp-sink/gstrtspclientsink.c:
437 * gst/rtsp-sink/gstrtspclientsink.h:
438 rtspclientsink: Wait until OPEN has been scheduled
439 Make sure that the sink thread has started opening connection
440 to the server before continuing.
441 https://bugzilla.gnome.org/show_bug.cgi?id=790412
443 2017-12-14 14:53:35 +1100 Matthew Waters <matthew@centricular.com>
446 Automatic update of common submodule
447 From e8c7a71 to 3fa2c9e
449 2017-12-07 16:08:29 +0100 Edward Hervey <edward@centricular.com>
451 * gst/rtsp-server/rtsp-media.c:
452 * gst/rtsp-server/rtsp-session-media.c:
453 * gst/rtsp-server/rtsp-stream.c:
454 rtsp-server: Minor doc fixes
457 2017-12-06 20:47:22 +0000 Tim-Philipp Müller <tim@centricular.com>
461 tests: disable all tests when --disable-tests is used
462 Move conditional subdir include into top level.
463 Based on patch by: Joel Holdsworth
464 https://bugzilla.gnome.org/show_bug.cgi?id=757703
466 2017-12-06 20:42:39 +0000 Tim-Philipp Müller <tim@centricular.com>
471 meson: build more tests and add options to disable tests and examples
473 2017-11-26 13:26:39 -0300 Thibault Saunier <tsaunier@gnome.org>
475 * gst/rtsp-server/rtsp-session.c:
476 Fix build when -Werror=deprecated-declarations is on
477 As gst_rtsp_session_next_timeout is deprecated.
479 ../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]
480 res = (gst_rtsp_session_next_timeout (session, now) == 0);
482 ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-session.c:685:1: note: declared here
483 gst_rtsp_session_next_timeout (GstRTSPSession * session, GTimeVal * now)
484 ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~
487 2017-11-27 20:18:24 +1100 Matthew Waters <matthew@centricular.com>
490 Automatic update of common submodule
491 From 3f4aa96 to e8c7a71
493 2017-11-25 20:34:16 +0100 Patricia Muscalu <patricia@dovakhiin.com>
495 * tests/check/gst/media.c:
496 check/media: Add seekability test case: not all streams are active
497 Media contains two streams but only one is complete and prepared
499 https://bugzilla.gnome.org/show_bug.cgi?id=790674
501 2017-11-25 20:32:02 +0100 Patricia Muscalu <patricia@dovakhiin.com>
503 * gst/rtsp-server/rtsp-stream.c:
504 rtsp-stream: Do not reset 'blocking' if stream is already blocked
505 https://bugzilla.gnome.org/show_bug.cgi?id=790674
507 2017-11-25 20:45:44 +0100 Patricia Muscalu <patricia@dovakhiin.com>
509 * gst/rtsp-server/rtsp-media.c:
510 rtsp-media: Fix missing lock in gst_rtsp_media_seekable()
511 https://bugzilla.gnome.org/show_bug.cgi?id=790674
513 2017-11-26 16:29:49 +0000 Tim-Philipp Müller <tim@centricular.com>
516 meson: remove vs_module_defs_dir variable which is no longer needed
518 2017-11-26 14:46:05 +0000 Tim-Philipp Müller <tim@centricular.com>
520 * gst/rtsp-server/rtsp-session.h:
523 2017-11-26 12:53:42 +0000 Tim-Philipp Müller <tim@centricular.com>
526 * gst/rtsp-server/meson.build:
528 * win32/common/libgstrtspserver.def:
529 win32: remove .def file with exports
530 They're no longer needed, symbol exporting is now explicit
531 via GST_EXPORT in all cases (autotools, meson, incl. MSVC).
533 2017-11-26 12:28:40 +0000 Tim-Philipp Müller <tim@centricular.com>
536 autotools: stop controlling symbol visibility with -export-symbols-regex
537 Instead, use -fvisibility=hidden and explicit exports via GST_EXPORT.
538 This should result in consistent behaviour for the autotools and
541 2017-11-26 12:47:08 +0000 Tim-Philipp Müller <tim@centricular.com>
543 * gst/rtsp-server/rtsp-media.h:
544 * gst/rtsp-server/rtsp-server.h:
545 * gst/rtsp-server/rtsp-session.c:
546 * gst/rtsp-server/rtsp-session.h:
547 rtsp-server: add missing GST_EXPORT and export deprecated funcs
549 2017-11-25 07:53:30 +0100 Edward Hervey <edward@centricular.com>
551 * tests/check/gst/media.c:
552 check: Add seekability testing on medias
553 Make sure that once GstRTSPMedia are prepared they returned
554 the expected seekability results
555 https://bugzilla.gnome.org/show_bug.cgi?id=790674
557 2017-11-24 17:34:31 +0100 Edward Hervey <edward@centricular.com>
559 * docs/libs/gst-rtsp-server-sections.txt:
560 * gst/rtsp-server/rtsp-media.c:
561 * gst/rtsp-server/rtsp-stream.c:
562 * gst/rtsp-server/rtsp-stream.h:
563 * win32/common/libgstrtspserver.def:
564 rtsp-media: Enable seeking query before pipeline is complete
565 SDP are now provided *before* the pipeline is fully complete. In order
566 to know whether a media is seekable or not therefore requires asking
567 the invididual streams.
568 API: gst_rtsp_stream_seekable
569 https://bugzilla.gnome.org/show_bug.cgi?id=790674
571 2017-11-23 20:34:03 +0100 Patricia Muscalu <patricia@axis.com>
573 * gst/rtsp-server/rtsp-media.c:
574 rtsp-media: Fix handling in default_unsuspend()
575 Handle the case when streams are not blocked and media
576 is suspended from PAUSED.
577 Change-Id: I2f3d222ea7b9b20a0732ea5dc81a32d17ab75040
578 https://bugzilla.gnome.org/show_bug.cgi?id=790674
580 2017-11-23 18:51:21 +0100 Patricia Muscalu <patricia@axis.com>
582 * tests/check/gst/media.c:
583 check/media: Fix thread pool leak.
584 Change-Id: I0f92b1caca0ee518ae64a7dacfbd28a214c3eea1
585 https://bugzilla.gnome.org/show_bug.cgi?id=790674
587 2017-11-23 18:39:44 +0100 Patricia Muscalu <patricia@axis.com>
589 * gst/rtsp-server/rtsp-media.c:
590 rtsp-media: Removed fakesink elements
591 There is not need of adding fakesink elements to the media
592 pipeline in the dynamic-payloader case.
593 The media pipeline itself is dynamically updated with
594 the receiver and sender parts that are based on the client
595 transport information known after SETUP has been received.
596 Change-Id: I4e88c9b500c04030669822f0d03b1842913f6cb9
597 https://bugzilla.gnome.org/show_bug.cgi?id=790674
599 2017-11-23 09:10:54 +0100 Patricia Muscalu <patricia@axis.com>
601 * gst/rtsp-server/rtsp-media.c:
602 rtsp-media: Corrected ASYNC_DONE handling
603 Media is complete when all the transport based parts are
604 added to the media pipeline. At this point ASYNC_DONE is
605 posted by the media pipeline and media is ready to enter
607 Change-Id: I50fb8dfed88ebaf057d9a35fca2d7f0a70e9d1fa
608 https://bugzilla.gnome.org/show_bug.cgi?id=790674
610 2017-11-22 12:24:38 +0100 Edward Hervey <bilboed@bilboed.com>
612 * tests/check/gst/media.c:
613 check/media: Check that prepared media can provide a SDP
614 Whenever a RTSPMedia is prepared, it should be able to provide a SDP
616 2017-11-21 09:53:19 +0100 Edward Hervey <edward@centricular.com>
618 * gst/rtsp-server/rtsp-client.c:
619 rtsp-client: Don't leak addr
622 2017-11-21 09:53:08 +0100 Edward Hervey <bilboed@bilboed.com>
624 * gst/rtsp-server/rtsp-client.c:
625 * gst/rtsp-server/rtsp-session-media.c:
626 * gst/rtsp-server/rtsp-stream.c:
629 2017-11-20 18:30:19 +0100 Edward Hervey <bilboed@bilboed.com>
631 * gst/rtsp-server/rtsp-media.c:
632 rtsp-media: Don't unblock with remaining dynamic payloaders
633 If we still have some dynamic paylaoders which haven't posted
634 no-more-pads yet, don't go to PREPARED if one of the streams
636 The risk was that we would end up not exposing/using all specified
638 The downside is that if you have _multiple_ _live_ _dynamic_ payloaders
639 then it will take a bit more time to start. But only if those 3
640 conditions are present.
641 https://bugzilla.gnome.org/show_bug.cgi?id=769521
643 2017-11-20 16:49:29 +0100 Edward Hervey <edward@centricular.com>
645 * gst/rtsp-server/rtsp-media.c:
648 2017-11-20 16:48:55 +0100 Edward Hervey <edward@centricular.com>
650 * gst/rtsp-server/rtsp-media.c:
651 rtsp-media: Don't set float on a gint64 variable
652 Just use 0. Fixes 'undefined' behaviour from clang
654 2017-11-20 18:29:02 +0100 Edward Hervey <edward@centricular.com>
656 * gst/rtsp-server/rtsp-media.c:
657 rtsp-media: Fix previous commit
658 We only want to count dynamic payloaders
660 2017-11-20 09:32:07 +0100 Edward Hervey <edward@centricular.com>
662 * gst/rtsp-server/rtsp-media.c:
663 * tests/check/gst/media.c:
664 rtsp-media: Handle multiple dynamic elements
665 If we have more than one dynamic payloader in the pipeline, we need
666 to wait until the *last* one emits 'no-more-pads' before switching
668 Failure to do so would result in a race where some of the streams
669 wouldn't properly be prepared
670 https://bugzilla.gnome.org/show_bug.cgi?id=769521
672 2017-11-16 12:18:20 +0200 Sebastian Dröge <sebastian@centricular.com>
674 * win32/common/libgstrtspserver.def:
675 win32: Fix exported symbols list
677 2017-11-15 19:52:29 +0200 Sebastian Dröge <sebastian@centricular.com>
679 * gst/rtsp-server/rtsp-stream.c:
680 rtsp-stream: Only update the RTP udpsink if it actually exists
681 For send-only streams it does not exist, but the RTCP udpsink might.
683 2017-11-15 18:15:53 +0200 Sebastian Dröge <sebastian@centricular.com>
685 * win32/common/libgstrtspserver.def:
686 win32: Update exports
688 2017-10-23 09:49:09 +0200 Patricia Muscalu <patricia@axis.com>
690 * gst/rtsp-server/rtsp-media.c:
691 * gst/rtsp-server/rtsp-stream.c:
692 * gst/rtsp-server/rtsp-stream.h:
693 rtsp-media: seek on media pipelines that are complete
694 Make sure that a seek is performed on pipelines that
695 contain at least one sink element.
696 Change-Id: Icf398e10add3191d104b1289de612412da326819
697 https://bugzilla.gnome.org/show_bug.cgi?id=788340
699 2017-10-17 10:44:33 +0200 Patricia Muscalu <patricia@axis.com>
701 * gst/rtsp-server/rtsp-client.c:
702 * gst/rtsp-server/rtsp-media.c:
703 * gst/rtsp-server/rtsp-media.h:
704 * gst/rtsp-server/rtsp-stream.c:
705 * gst/rtsp-server/rtsp-stream.h:
706 * tests/check/gst/client.c:
707 * tests/check/gst/media.c:
708 * tests/check/gst/rtspserver.c:
709 * tests/check/gst/stream.c:
710 Dynamically reconfigure pipeline in PLAY based on transports
711 The initial pipeline does not contain specific transport
712 elements. The receiver and the sender parts are added
714 If the media is shared, the streams are dynamically
715 reconfigured after each PLAY.
716 https://bugzilla.gnome.org/show_bug.cgi?id=788340
718 2017-10-16 12:40:57 +0200 Patricia Muscalu <patricia@axis.com>
720 * gst/rtsp-server/rtsp-stream.c:
721 rtsp-stream: obtain stream position from pad
722 If no sinks have been added yet, obtain the current and
723 the stop position of the stream from the send_src pad.
724 Change-Id: Iacd4ab4bdc69f6b49370d06012880ce48a7d595a
725 https://bugzilla.gnome.org/show_bug.cgi?id=788340
727 2017-10-16 11:35:10 +0200 Patricia Muscalu <patricia@axis.com>
729 * gst/rtsp-server/rtsp-session-media.c:
730 * gst/rtsp-server/rtsp-session-media.h:
731 rtsp-session-media: add function to get a list of transports
732 Change-Id: I817e10624da0f3200f24d1b232cff481099278e3
733 https://bugzilla.gnome.org/show_bug.cgi?id=788340
735 2017-10-16 11:15:55 +0200 Patricia Muscalu <patricia@axis.com>
737 * gst/rtsp-server/rtsp-stream.c:
738 * gst/rtsp-server/rtsp-stream.h:
739 rtsp-stream: add functions to get rtp and rtcp multicast sockets
740 Change-Id: Iddfe6e0bd250cb0159096d5eba9e4202d22b56db
741 https://bugzilla.gnome.org/show_bug.cgi?id=788340
743 2017-10-20 12:21:48 +0200 Patricia Muscalu <patricia@axis.com>
745 * gst/rtsp-server/rtsp-stream.c:
746 stream: set async=sync=false only for RTCP appsink
747 Change-Id: I929a218a9adf4759f61322b6f2063aacc5595f90
748 https://bugzilla.gnome.org/show_bug.cgi?id=788340
750 2017-10-16 10:10:17 +0200 Patricia Muscalu <patricia@axis.com>
752 * gst/rtsp-server/rtsp-media.c:
753 rtsp-media: return minimum value in query position case
754 The minimum position should be returned as we are interested
755 in the whole interval.
756 Change-Id: I30e297fc040c995ae40c25dee8ff56321612fe2b
757 https://bugzilla.gnome.org/show_bug.cgi?id=788340
759 2017-08-09 11:52:38 +0200 Jonathan Karlsson <jonakn@axis.com>
761 * gst/rtsp-server/rtsp-session.c:
762 * tests/check/gst/rtspserver.c:
763 rtsp-session: Handle the case when timeout=0
764 According to the documentation, a timeout of value 0 means
765 that the session never timeouts. This adds handling of that.
766 If timeout=0 we just return with a -1 from
767 gst_rtsp_session_next_timeout_usec ().
768 https://bugzilla.gnome.org/show_bug.cgi?id=785058
770 2017-07-17 17:15:22 +0300 Sebastian Dröge <sebastian@centricular.com>
772 * gst/rtsp-sink/gstrtspclientsink.c:
773 rtspclientsink: Add "accept-certificate" signal for manually checking a TLS certificate for validity
774 https://bugzilla.gnome.org/show_bug.cgi?id=785024
776 2017-10-26 14:43:19 +0200 Mathieu Duponchelle <mathieu@centricular.com>
778 * docs/libs/gst-rtsp-server-sections.txt:
779 * gst/rtsp-server/rtsp-media-factory.c:
780 docs: add media factory transport mode accessors
781 and fix the documentation for the return value of the getter
783 2017-10-09 12:43:01 +0200 Branko Subasic <branko@axis.com>
785 * gst/rtsp-server/rtsp-client.c:
786 rtsp-client: unref 'pipelined_requests' in finalize
787 The hash table priv->pipelined_requests is not unref:ed in the
788 finalize funktion. Make sure it is.
789 https://bugzilla.gnome.org/show_bug.cgi?id=788704
791 2017-10-09 14:44:40 +0200 Thibault Saunier <thibault.saunier@osg.samsung.com>
793 * gst/rtsp-server/rtsp-media.c:
794 rtsp-media: Initialize scalar variable
797 2017-10-06 10:27:34 +0200 Edward Hervey <edward@centricular.com>
799 * win32/common/libgstrtspserver.def:
800 win32: Update export file
802 2017-04-22 09:26:07 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
804 * gst/rtsp-server/rtsp-client.c:
805 * gst/rtsp-server/rtsp-media.c:
806 * gst/rtsp-server/rtsp-media.h:
807 Start support for RTSP 2.0
808 This adds basic support for new 2.0 features, though the protocol is
809 subposdely backward incompatible, most semantics are the sames.
812 * version negotiation
813 * pipelined requests support
814 * Media-Properties support
815 * Accept-Ranges support
817 * gst_rtsp_media_seekable
818 The RTSP methods that have been removed when using 2.0 now return
820 https://bugzilla.gnome.org/show_bug.cgi?id=781446
822 2017-06-02 15:37:54 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
824 * gst/rtsp-server/rtsp-stream.c:
825 stream: Use stream duration as stream-stop if segment was not configured with a stop
826 Allowing client to know stream duration when no seeking happened.
827 https://bugzilla.gnome.org/show_bug.cgi?id=783435
829 2017-09-25 19:40:17 +0300 Sebastian Dröge <sebastian@centricular.com>
831 * gst/rtsp-server/rtsp-media-factory.c:
832 rtsp-media-factory: Don't cache any media if NULL was returned as key
833 The docs already mentioned this, but we actually stored it in the hash
834 table with key==NULL and leaked its reference forever.
836 2017-09-18 19:31:31 +0200 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
838 * gst/rtsp-sink/gstrtspclientsink.c:
839 * gst/rtsp-sink/gstrtspclientsink.h:
840 rtspclientsink: Use a mutex for protecting against concurrent send/receives
841 This is a simple port of:
842 * a722f6e8329032c6eda4865d6a07f4ba5981d7ea
843 * c438545dc9e2f14f657bc0ef261fff726449867b
844 * cd17c71dcea5c9310d21f1347c7520983e5869ac
847 2017-08-31 13:24:15 +0530 Satya Prakash Gupta <sp.gupta@samsung.com>
849 * gst/rtsp-server/rtsp-sdp.c:
850 sdp: fix Memory leak in error case
851 https://bugzilla.gnome.org/show_bug.cgi?id=787059
853 2017-08-18 17:37:01 +0100 Tim-Philipp Müller <tim@centricular.com>
855 * pkgconfig/meson.build:
856 meson: don't install -uninstalled.pc file
857 https://bugzilla.gnome.org/show_bug.cgi?id=786457
859 2017-08-17 12:26:17 +0100 Tim-Philipp Müller <tim@centricular.com>
862 Automatic update of common submodule
863 From 48a5d85 to 3f4aa96
865 2017-08-14 21:04:23 +0300 Sebastian Dröge <sebastian@centricular.com>
867 * gst/rtsp-server/rtsp-client.c:
868 rtsp-client: Fix typo in debug message
870 2017-08-11 14:14:32 +0100 Tim-Philipp Müller <tim@centricular.com>
873 meson: hide symbols by default unless explicitly exported
875 2017-08-10 14:20:12 +0100 Tim-Philipp Müller <tim@centricular.com>
877 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
878 pkgconfig: remove -I@srcdir@/.. which duplicates abs_top_srcdir
879 Fixes meson warning about undefined @srcdir@.
881 2017-07-21 13:36:00 +0100 Tim-Philipp Müller <tim@centricular.com>
884 meson: skip tests on windows for now
885 As we do in the other modules. As libgstcheck is currently not
886 built on windows. Fixes "Fallback variable 'gst_check_dep' in
887 the subproject 'gstreamer' does not exist"" Meson error.
889 2017-06-22 07:25:07 -0700 Julien Isorce <julien.isorce@gmail.com>
891 * gst/rtsp-server/rtsp-stream.c:
892 rtsp-stream: fix connection delay due to wrong assumption on last-sample
893 Commit 852cc09f542af5cadd79ffd7fe79d6475cf57e14 assumed that
894 multiudpsink's last-sample always comes from the payloader. Which
895 is wrong if auxiliary streams are multiplexed in the same stream.
896 So check the buffer's ssrc against the caps'ssrc before to use its
897 seqnum. If not the same ssrc just use the payloader as done prior
898 the commit above or when there is no last-sample yet.
899 https://bugzilla.gnome.org/show_bug.cgi?id=784094
901 2017-06-23 16:19:04 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
904 meson: Allow using glib as a subproject
906 2017-06-26 09:55:49 +0100 Tim-Philipp Müller <tim@centricular.com>
909 meson: fix with-package-name option
910 https://bugzilla.gnome.org/show_bug.cgi?id=784082
912 2017-06-09 20:16:28 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
915 Distribute meson_options.txt
917 2017-06-09 20:11:47 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
920 And config.h.meson is no longer dist either
922 2017-06-09 21:27:09 +0100 Tim-Philipp Müller <tim@centricular.com>
926 meson: config.h.meson is no longer needed
928 2017-06-07 13:04:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
930 * tests/check/meson.build:
932 meson: Fix building tests and activate them again
934 2017-06-07 12:55:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
936 * tests/check/meson.build:
937 meson: Do not use path separator in test names
938 Avoiding warnings like:
939 WARNING: Target "elements/audioamplify" has a path separator in its name.
941 2017-05-20 15:07:31 +0100 Tim-Philipp Müller <tim@centricular.com>
945 meson: add options to set package name and origin
946 https://bugzilla.gnome.org/show_bug.cgi?id=782172
948 2017-05-18 10:35:18 +0100 Tim-Philipp Müller <tim@centricular.com>
950 * gst/rtsp-server/rtsp-address-pool.h:
951 * gst/rtsp-server/rtsp-auth.h:
952 * gst/rtsp-server/rtsp-client.h:
953 * gst/rtsp-server/rtsp-context.h:
954 * gst/rtsp-server/rtsp-media-factory-uri.h:
955 * gst/rtsp-server/rtsp-media-factory.h:
956 * gst/rtsp-server/rtsp-media.h:
957 * gst/rtsp-server/rtsp-mount-points.h:
958 * gst/rtsp-server/rtsp-params.h:
959 * gst/rtsp-server/rtsp-permissions.h:
960 * gst/rtsp-server/rtsp-sdp.h:
961 * gst/rtsp-server/rtsp-server.h:
962 * gst/rtsp-server/rtsp-session-media.h:
963 * gst/rtsp-server/rtsp-session-pool.h:
964 * gst/rtsp-server/rtsp-session.h:
965 * gst/rtsp-server/rtsp-stream-transport.h:
966 * gst/rtsp-server/rtsp-stream.h:
967 * gst/rtsp-server/rtsp-thread-pool.h:
968 * gst/rtsp-server/rtsp-token.h:
969 Mark symbols explicitly for export with GST_EXPORT
971 2017-05-16 14:44:43 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
974 * gst/rtsp-sink/Makefile.am:
975 Remove plugin specific static build option
976 Static and dynamic plugins now have the same interface. The standard
977 --enable-static/--enable-shared toggle are sufficient.
979 2017-05-04 18:59:14 +0300 Sebastian Dröge <sebastian@centricular.com>
985 === release 1.12.0 ===
987 2017-05-04 15:40:46 +0300 Sebastian Dröge <sebastian@centricular.com>
993 * gst-rtsp-server.doap:
997 === release 1.11.91 ===
999 2017-04-27 17:42:02 +0300 Sebastian Dröge <sebastian@centricular.com>
1005 * gst-rtsp-server.doap:
1009 2017-04-24 20:30:37 +0100 Tim-Philipp Müller <tim@centricular.com>
1012 Automatic update of common submodule
1013 From 60aeef6 to 48a5d85
1015 2017-04-13 14:20:10 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
1017 * gst/rtsp-server/rtsp-media-factory.c:
1018 * gst/rtsp-server/rtsp-media.c:
1019 * gst/rtsp-server/rtsp-session.c:
1020 * gst/rtsp-server/rtsp-stream.c:
1021 gi: Fix some annotations and docstrings
1023 2017-04-13 13:52:26 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
1025 * gst/rtsp-server/meson.build:
1027 * meson_options.txt:
1030 2017-04-10 23:51:12 +0100 Tim-Philipp Müller <tim@centricular.com>
1034 Automatic update of common submodule
1035 From 39ac2f5 to 60aeef6
1037 === release 1.11.90 ===
1039 2017-04-07 16:35:03 +0300 Sebastian Dröge <sebastian@centricular.com>
1045 * gst-rtsp-server.doap:
1049 2017-03-27 18:19:33 +0100 Tim-Philipp Müller <tim@centricular.com>
1051 * examples/test-launch.c:
1052 examples: make test-launch pipeline shared by default as well
1054 2017-02-27 19:10:44 +0200 Sebastian Dröge <sebastian@centricular.com>
1056 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
1057 gstreamer-rtsp-server: Add both srcdir and builddir to the include path
1058 Just the build dir is not going to work for srcdir!=builddir.
1060 2017-02-24 15:59:54 +0200 Sebastian Dröge <sebastian@centricular.com>
1063 meson: Update version
1065 2017-02-24 15:37:49 +0200 Sebastian Dröge <sebastian@centricular.com>
1070 === release 1.11.2 ===
1072 2017-02-24 15:10:07 +0200 Sebastian Dröge <sebastian@centricular.com>
1078 * gst-rtsp-server.doap:
1081 2017-02-14 20:40:26 +0000 Tim-Philipp Müller <tim@centricular.com>
1084 meson: dist meson build files
1085 Ship meson build files in tarballs, so people who use tarballs
1086 in their builds can start playing with meson already.
1088 2017-02-07 23:39:37 +1100 Jan Schmidt <jan@centricular.com>
1090 * examples/test-record.c:
1091 examples/test-record: Add extra line to initial printout
1092 Add an example line of how to deliver a stream to the
1095 2017-01-19 14:57:19 +0200 Sebastian Dröge <sebastian@centricular.com>
1097 * gst/rtsp-server/rtsp-client.c:
1098 rtsp-client: Also handle the (S|G)ET_PARAMETER case of size==0 || !data as keep-alive
1099 If there is no Content-Length header, no body would be allocated and the
1100 '\0' would also not be appended to the body.
1102 2017-01-19 14:24:07 +0200 Sebastian Dröge <sebastian@centricular.com>
1104 * gst/rtsp-server/rtsp-client.c:
1105 rtsp-client: Fix handling of keep-alive GET_PARAMETER/SET_PARAMETER
1106 While they logically have 0 bytes length, GstRTSPConnection is appending
1107 a '\0' to everything making the size be 1 instead.
1109 2017-01-13 12:39:36 +0000 Tim-Philipp Müller <tim@centricular.com>
1114 2017-01-12 19:04:23 +0200 Sebastian Dröge <sebastian@centricular.com>
1116 * gst/rtsp-server/rtsp-session.c:
1117 rtsp-session: Only remove deprecated API if requested to do so, not just when disabling
1118 gst_rtsp_session_is_expired() and gst_rtsp_session_next_timeout() were
1121 2017-01-12 16:32:59 +0200 Sebastian Dröge <sebastian@centricular.com>
1126 === release 1.11.1 ===
1128 2017-01-12 16:14:46 +0200 Sebastian Dröge <sebastian@centricular.com>
1134 * gst-rtsp-server.doap:
1135 * win32/common/libgstrtspserver.def:
1138 2017-01-10 08:34:50 +0100 Patricia Muscalu <patricia@axis.com>
1140 * gst/rtsp-server/rtsp-stream.c:
1141 rtsp-stream: corrected if-statement in _get_server_port()
1142 This bug was accidentally introduced while fixing a segfault
1143 in _get_server_port() function.
1144 https://bugzilla.gnome.org/show_bug.cgi?id=776345
1146 2017-01-09 14:12:05 +0100 Patricia Muscalu <patricia@axis.com>
1148 * gst/rtsp-server/rtsp-stream.c:
1149 * tests/check/gst/stream.c:
1150 rtsp-stream: fixed segmenation fault in _get_server_port()
1151 Calling function gst_rtsp_stream_get_server_port() results in
1152 segmenation fault in the RTP/RTSP/TCP case.
1153 Port that the server will use to receive RTCP makes only
1154 sense in the UDP case, however the function should handle
1155 the TCP case in a nicer way.
1156 https://bugzilla.gnome.org/show_bug.cgi?id=776345
1158 2017-01-09 12:22:40 +0300 Aleksandr Slobodeniuk <alenuke@yandex.ru>
1160 * gst/rtsp-server/rtsp-media-factory.c:
1161 dosc: Fix a little typo
1162 https://bugzilla.gnome.org/show_bug.cgi?id=777037
1164 2017-01-04 16:20:54 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
1166 * pkgconfig/Makefile.am:
1167 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
1168 * pkgconfig/meson.build:
1169 meson: generate pkg-config -uninstalled pc files
1170 Generating those files is useful for users building the GStreamer stack
1171 using meson and having to link it to another project which is still
1172 using the autotools.
1173 https://bugzilla.gnome.org/show_bug.cgi?id=776810
1175 2017-01-04 16:11:08 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
1177 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
1178 pkgconfig: fix -uninstalled pc file
1179 pcfiledir was never defined so the paths were wrong.
1180 https://bugzilla.gnome.org/show_bug.cgi?id=776867
1182 2016-12-21 13:41:50 +0100 Patricia Muscalu <patricia@axis.com>
1184 * gst/rtsp-server/rtsp-stream.c:
1185 * tests/check/gst/rtspserver.c:
1186 rtsp-stream: Fixed TCP transport case
1187 Make sure that the appsink element is actually added to
1188 the bin before trying to link it with the elements in it.
1189 https://bugzilla.gnome.org/show_bug.cgi?id=776343
1191 2016-12-16 17:26:04 +0000 Tim-Philipp Müller <tim@centricular.com>
1197 Remove generated .spec file
1198 Likely extremely bitrotten, and we should not ship this anyway.
1200 2016-12-03 08:21:02 +0100 Edward Hervey <bilboed@bilboed.com>
1203 Automatic update of common submodule
1204 From f980fd9 to 39ac2f5
1206 2016-12-02 15:40:09 +0100 Edward Hervey <edward@centricular.com>
1208 * gst/rtsp-server/rtsp-media.c:
1209 media: Fix pt map caps
1210 Since decryption is handled within rtpbin, all outcoming stream
1211 caps will be application/x-rtp (i.e. regular rtp)
1212 Fixes RECORD with SRTP streams
1214 2016-12-02 15:38:04 +0100 Edward Hervey <edward@centricular.com>
1216 * gst/rtsp-server/rtsp-media-factory.c:
1217 media-factory: Create media objects with the proper transport mode
1218 The function called immediately afterwards (collect_streams()) will
1219 need it to work properly
1221 2016-12-02 14:36:50 +0200 Sebastian Dröge <sebastian@centricular.com>
1223 * gst/rtsp-server/rtsp-auth.c:
1224 rtsp-auth: Don't remove digest-auth nonces that already/still have a client connected
1226 2016-12-01 18:04:34 +0200 Sebastian Dröge <sebastian@centricular.com>
1228 * gst/rtsp-server/rtsp-media-factory.c:
1229 rtsp-media-factory: Don't create a pipeline for the media pipeline string
1230 We're going to put a pipeline into a pipeline otherwise, which is not
1233 2016-10-25 15:41:28 +0300 Kseniia Vasilchuk <vasilchukkseniia@gmail.com>
1235 * gst/rtsp-server/rtsp-media.c:
1236 media: Fix race condition around finish_unprepare() if called multiple time
1237 https://bugzilla.gnome.org/show_bug.cgi?id=755329
1239 2016-11-30 14:06:36 +1100 Jan Schmidt <jan@centricular.com>
1241 * gst/rtsp-sink/gstrtspclientsink.c:
1242 rtspclientsink: Don't leave stale pointer after unref
1243 Fix a warning on shutdown - don't keep a pointer to an
1244 alread-unreffed object.
1246 2016-11-26 11:24:50 +0000 Tim-Philipp Müller <tim@centricular.com>
1249 common: use https protocol for common submodule
1250 https://bugzilla.gnome.org/show_bug.cgi?id=775110
1252 2016-11-21 23:29:56 +1100 Matthew Waters <matthew@centricular.com>
1254 * gst/rtsp-server/rtsp-stream.c:
1255 stream: block the output of rtpbin instead of the source pipeline
1256 85c52e194bcb81928b96614be0ae47d59eccb1ce introduced a more correct
1257 detection of the srtp rollover counter to add to the SDP.
1258 Unfortunately, it was incomplete for live pipelines where the logic
1259 blocks the source bin before creating the SDP and thus would never have
1260 the necessary informaiton to create a correct SDP with srtp encryption.
1261 Move the pad blocks to rtpbin's output pads instead so that the
1262 necessary information can be created before we need the information for
1264 https://bugzilla.gnome.org/show_bug.cgi?id=770239
1266 2016-11-21 16:02:39 +0100 Dag Gullberg <dagg@axis.com>
1268 * gst/rtsp-server/rtsp-client.c:
1269 rtsp-client: add IDLE timeout, before session exists
1270 The RTSP server will not timeout an idle RTSP connection
1271 (note this is different from doing timeout on a RTSP
1273 At least for Apache this is a problem when running RTSP over
1274 HTTPS since it uses one of the threads (there is a rather
1275 limited number) that are available for handling requests.
1276 https://bugzilla.gnome.org/show_bug.cgi?id=771830
1278 2016-11-23 09:45:08 +0000 Tim-Philipp Müller <tim@centricular.com>
1283 2016-11-21 13:05:50 +0100 Göran Jönsson <goranjn@axis.com>
1285 * gst/rtsp-server/rtsp-stream.c:
1286 rtsp-stream: Set close-socket FALSE on UDP src:es
1287 With this RTSP server can use the sockets independent on the udpsrc
1289 When the udp src is finalized it will unref socket and when g_socket
1290 is finalized the socket will be closed.
1291 https://bugzilla.gnome.org/show_bug.cgi?id=765673
1293 2016-11-18 17:47:13 +0200 Sebastian Dröge <sebastian@centricular.com>
1295 * gst/rtsp-sink/gstrtspclientsink.c:
1296 rtspclientsink: Move to new helper function to parse authentication responses
1297 https://bugzilla.gnome.org/show_bug.cgi?id=774416
1299 2016-11-16 08:42:24 +0200 Sebastian Dröge <sebastian@centricular.com>
1301 * examples/Makefile.am:
1302 * examples/test-auth-digest.c:
1303 * gst/rtsp-server/rtsp-auth.c:
1304 * gst/rtsp-server/rtsp-auth.h:
1305 * win32/common/libgstrtspserver.def:
1306 rtsp-auth: Add support for Digest authentication
1307 https://bugzilla.gnome.org/show_bug.cgi?id=774416
1309 2016-11-17 09:41:53 -0800 Scott D Phillips <scott.d.phillips@intel.com>
1312 * gst/rtsp-server/meson.build:
1314 * tests/check/meson.build:
1316 * win32/common/libgstrtspserver.def:
1317 Enable building with MSVC
1318 https://bugzilla.gnome.org/show_bug.cgi?id=774640
1320 2016-11-18 20:23:14 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
1323 meson: gstreamer gst_check_dep does not exist on windows
1325 2016-11-17 09:43:37 -0800 Scott D Phillips <scott.d.phillips@intel.com>
1327 * gst/rtsp-server/rtsp-client.c:
1328 client: update do_send_message to match type GstRTSPClientSendFunc
1329 This type mismatch fails building with MSVC
1330 https://bugzilla.gnome.org/show_bug.cgi?id=774640
1332 2016-11-11 14:42:08 +0200 Sebastian Dröge <sebastian@centricular.com>
1334 * gst/rtsp-server/rtsp-sdp.c:
1335 rtsp-sdp: Fix indentation
1337 2016-11-10 05:16:00 +0000 Neha Arora <arora.neha@samsung.com>
1339 * gst/rtsp-server/rtsp-media.c:
1340 rtsp-media: Only signal "new-state" if the state has actually changed
1341 https://bugzilla.gnome.org/show_bug.cgi?id=774173
1343 2016-08-24 11:39:13 +0200 Branko Subasic <branko@axis.com>
1345 * gst/rtsp-server/rtsp-client.c:
1346 * gst/rtsp-server/rtsp-client.h:
1347 client: emit signal in the beginning of each rtsp request
1348 These signals let the application validate the requests, configure the
1349 media/stream in a certain way and also generate error status code in
1350 case of error or bad request.
1351 https://bugzilla.gnome.org/show_bug.cgi?id=758062
1353 2016-11-01 18:10:35 +0000 Tim-Philipp Müller <tim@centricular.com>
1356 meson: update version
1358 === release 1.11.0 ===
1360 2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com>
1365 === release 1.10.0 ===
1367 2016-11-01 18:06:46 +0200 Sebastian Dröge <sebastian@centricular.com>
1373 * gst-rtsp-server.doap:
1376 2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com>
1378 * tests/check/gst/rtspserver.c:
1379 * tests/check/gst/stream.c:
1380 tests: try to avoid using the same ports in different tests
1381 Causes problems with client multicast tests otherwise if
1382 tests are run in parallel.
1383 https://bugzilla.gnome.org/show_bug.cgi?id=773640
1385 2016-10-28 17:50:59 +0100 Tim-Philipp Müller <tim@centricular.com>
1387 * tests/check/gst/client.c:
1388 tests: client: use fail_unless_equals_foo() for better failure reporting
1390 2016-09-26 11:16:04 +0200 Göran Jönsson <goranjn@axis.com>
1392 * gst/rtsp-server/rtsp-client.c:
1393 rtsp-client: Session filter in unwatch session
1394 Call session filter with filter_session_media as paramer in
1395 client_unwatch_session if using drop_backlog = FALSE.
1396 In client_unwatch_session its allowed to grow the watchs backlog.
1397 If using drop_backlog = FALSE and the backlog is full it will cause
1398 a deadlock when setting session media state to NULL
1399 if the backlog is not allowed to grow.
1400 https://bugzilla.gnome.org/show_bug.cgi?id=771983
1402 2016-10-20 21:40:18 +0100 Tim-Philipp Müller <tim@centricular.com>
1405 meson: add fallbacks for gst modules
1408 2016-09-14 17:48:39 +0300 Nikita Bobkov <NikitaDBobkov@gmail.com>
1410 * gst/rtsp-server/rtsp-client.c:
1411 rtsp-client: Fix factory leaking in find_media() in error cases
1412 https://bugzilla.gnome.org/show_bug.cgi?id=771488
1414 2016-10-06 11:47:50 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1416 * gst/rtsp-server/rtsp-stream.c:
1417 stream: Fix randomly missing streams from SDP with dynamic elements
1418 When using dynamic elements, gst_rtsp_stream_join_bin() is called from
1419 "pad-added" signal. In that case priv->srcpad could already have its caps,
1420 and they'll be sent to priv->send_src[0] pad. That means that when it
1421 connects "notify::caps" signal, that pad could already have received its
1422 caps and the signal won't be emitted anymore.
1423 In that case priv->caps stay to NULL and when building the SDP that stream
1424 gets ignored. Leading to missing video or audio when playing in client side.
1425 https://bugzilla.gnome.org/show_bug.cgi?id=772478
1427 2016-09-30 11:42:08 +0100 Tim-Philipp Müller <tim@centricular.com>
1430 meson: update version
1432 === release 1.9.90 ===
1434 2016-09-30 13:04:12 +0300 Sebastian Dröge <sebastian@centricular.com>
1440 * gst-rtsp-server.doap:
1443 2016-09-17 13:17:19 +0100 Ian Jamison <ian.dev@arkver.com>
1445 * gst/rtsp-server/rtsp-media-factory.c:
1446 * gst/rtsp-server/rtsp-media.c:
1447 * gst/rtsp-server/rtsp-stream.c:
1448 rtsp-server: Hint that set_multicast_iface expects the name of the interface
1449 To prevent any possibly confusion with IPs or anything else.
1450 https://bugzilla.gnome.org/show_bug.cgi?id=771530
1452 2016-09-18 09:58:55 -0400 Sebastian Dröge <sebastian@centricular.com>
1454 * gst/rtsp-server/rtsp-media-factory.c:
1455 * gst/rtsp-server/rtsp-media.c:
1456 rtsp-media: Call g_free() instead of g_object_unref() on multicast-iface strings
1457 https://bugzilla.gnome.org/show_bug.cgi?id=763000#c5
1459 2016-09-14 11:31:15 +0200 Sebastian Dröge <sebastian@centricular.com>
1462 configure: Depend on gstreamer 1.9.2.1
1464 2016-09-10 20:52:31 +1000 Jan Schmidt <jan@centricular.com>
1468 Automatic update of common submodule
1469 From b18d820 to f980fd9
1471 2016-09-10 09:58:31 +1000 Jan Schmidt <jan@centricular.com>
1475 Automatic update of common submodule
1476 From 6f2d209 to b18d820
1478 2016-09-07 18:44:34 +0300 Sebastian Dröge <sebastian@centricular.com>
1480 * gst/rtsp-server/rtsp-stream.c:
1481 rtsp-stream: Remove unused _locked() variant of a function
1482 It was added during refactoring.
1484 2016-09-07 10:21:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1486 * gst/rtsp-server/rtsp-stream.c:
1487 stream: cosmetic cleanup
1488 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1490 2016-09-07 10:16:19 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1492 * gst/rtsp-server/rtsp-stream.c:
1493 stream: Compare IP addresses case insensitive in more places
1494 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1496 2016-09-07 10:12:18 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1499 * gst/rtsp-server/rtsp-stream.c:
1500 stream: Fix leaked joined_bin
1501 There is no need to keep a strong ref on it, and _leave_bin() was
1502 setting it to NULL before calling g_clear_object() so it was leaked.
1503 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1505 2016-09-06 19:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
1507 * gst/rtsp-server/rtsp-stream.c:
1508 rtsp-stream: Compare IP address strings case insensitive
1509 Otherwise IPv6 addresses might fail this comparision.
1511 2016-09-06 19:10:21 +0300 Sebastian Dröge <sebastian@centricular.com>
1513 * gst/rtsp-server/rtsp-stream.c:
1514 rtsp-stream: Bind multicast sockets to ANY as before
1515 https://bugzilla.gnome.org/show_bug.cgi?id=766612#c48
1517 2016-09-05 18:31:36 +0300 Kseniia <vasilchukkseniia@gmail.com>
1519 * gst/rtsp-server/rtsp-session.c:
1520 rtsp-session: Fix segfault when doing keep-alive after removing the session
1521 If keep-alive happens after removing the session but before finalizing the
1522 stream transport, we would segfault.
1523 https://bugzilla.gnome.org/show_bug.cgi?id=750544
1525 2016-09-05 18:04:50 +0300 Sebastian Dröge <sebastian@centricular.com>
1527 * gst/rtsp-server/rtsp-stream.c:
1528 rtsp-stream: Always create multicast UDP elements if the protocol flag is set
1529 Adding them later will cause deadlocks due to
1530 1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
1531 2) adding the multicast sink
1532 3) waiting for it to get data to preroll again
1533 3) never happens because the queues after the tee are full.
1535 2016-09-05 16:32:57 +0300 Sebastian Dröge <sebastian@centricular.com>
1537 * gst/rtsp-server/rtsp-stream.c:
1538 rtsp-stream: Fix up various multicast related issues
1540 2016-09-05 13:40:59 +0300 Sebastian Dröge <sebastian@centricular.com>
1542 * tests/check/gst/stream.c:
1543 tests: Fix compilation
1545 2016-07-28 15:33:05 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1547 * gst/rtsp-server/rtsp-client.c:
1548 * gst/rtsp-server/rtsp-stream.c:
1549 * tests/check/gst/stream.c:
1550 stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast
1551 This is basically reverting changes introduced in commit f62a9a7,
1552 because it was introducing various regressions:
1553 - It introduces a leak of udpsrc elements that got wrongly fixed by adding
1554 an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
1555 ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
1556 - If a mcast client connects, it creates a new socket in SETUP to try to respect
1557 the destination/port given by the client in the transport, and overrides the
1558 socket already set on the udpsink element. That means that if we already had a
1559 client connected, the source address on the udp packets it receives suddenly
1561 - If a 2nd mcast client connects, the destination/port in its transport is
1562 ignored but its transport wasn't updated.
1563 What this patch does:
1564 - Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
1565 - Always have a tee+queue when udp is enabled. This could be optimized
1566 again in a later patch, but is more complicated. If no unicast clients
1567 connects then those elements are useless, this could be also optimized
1569 - When mcast transport is added, it creates a new set of udpsrc/udpsink,
1570 seperated from those for unicast clients. Since we already support only
1571 one mcast address, we also create only one set of elements.
1572 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1574 2016-07-28 15:20:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1576 * gst/rtsp-server/rtsp-stream.c:
1577 stream: factor our plug_src function
1578 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1580 2016-07-21 21:46:16 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1582 * gst/rtsp-server/rtsp-stream.c:
1583 stream: factor out plug_sink function
1584 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1586 2016-07-20 23:05:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1588 * gst/rtsp-server/rtsp-stream.c:
1589 stream: small documentation clarification
1590 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1592 2016-07-20 15:35:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1594 * gst/rtsp-server/rtsp-stream.c:
1595 stream: rename addr_v4/6 to mcast_addr_v4/6 for clarity
1596 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1598 2016-07-14 11:10:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1600 * gst/rtsp-server/rtsp-stream.c:
1601 stream: Keep a ref on joined bin
1602 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1604 2016-07-20 15:11:32 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1606 * gst/rtsp-server/rtsp-stream.c:
1607 stream: code cleanup
1608 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1610 2016-07-20 23:18:23 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1612 * gst/rtsp-server/rtsp-stream.c:
1613 stream: small fix in error code path
1614 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1616 2016-07-20 20:09:57 -0400 Xavier Claessens <xavier.claessens@collabora.com>
1618 * gst/rtsp-server/rtsp-stream.c:
1619 Revert "rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc"
1620 This partly reverts commit cba045e1b19fad6e689e10206f57903e15f1229a,
1621 but keeps unit tests.
1622 https://bugzilla.gnome.org/show_bug.cgi?id=766612
1624 2016-09-01 12:33:00 +0300 Sebastian Dröge <sebastian@centricular.com>
1629 === release 1.9.2 ===
1631 2016-09-01 12:32:51 +0300 Sebastian Dröge <sebastian@centricular.com>
1637 * gst-rtsp-server.doap:
1640 2016-01-27 01:03:52 +0000 Tim-Philipp Müller <tim@centricular.com>
1643 * examples/meson.build:
1645 * gst/rtsp-server/meson.build:
1646 * gst/rtsp-sink/meson.build:
1648 * pkgconfig/meson.build:
1649 * tests/check/meson.build:
1650 * tests/meson.build:
1651 Add support for Meson as alternative/parallel build system
1652 https://github.com/mesonbuild/meson
1654 2016-08-26 21:56:13 +0200 Josep Torra <n770galaxy@gmail.com>
1657 * tests/check/Makefile.am:
1658 build: silence error about pthread for 'make check' in osx
1659 Fixes "clang: error: argument unused during compilation: '-pthread'"
1661 2015-09-25 15:04:00 +0000 Nikita Bobkov <NikitaDBobkov@gmail.com>
1663 * gst/rtsp-server/rtsp-client.c:
1664 rtsp-client: Fix leaking of media in error cases
1665 With additional fixes by Kseniya Vasilchuk <vasilchukkseniia@gmail.com>
1666 and myself to make the media refcounting a bit easier to follow.
1667 https://bugzilla.gnome.org/show_bug.cgi?id=755632
1669 2016-08-02 15:08:22 +0300 Sebastian Dröge <sebastian@centricular.com>
1671 * gst/rtsp-server/rtsp-client.c:
1672 rtsp-client: Fix leaking of session in error cases
1673 https://bugzilla.gnome.org/show_bug.cgi?id=755632
1675 2016-07-11 21:16:04 +0200 Stefan Sauer <ensonic@users.sf.net>
1678 Automatic update of common submodule
1679 From f363b32 to f49c55e
1681 2016-07-06 13:51:15 +0300 Sebastian Dröge <sebastian@centricular.com>
1686 === release 1.9.1 ===
1688 2016-07-06 13:28:12 +0300 Sebastian Dröge <sebastian@centricular.com>
1694 * gst-rtsp-server.doap:
1697 2016-06-24 02:02:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
1700 configure: Need to add -DGST_STATIC_COMPILATION when building only statically
1701 https://bugzilla.gnome.org/show_bug.cgi?id=767463
1703 2016-06-21 11:49:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
1706 Automatic update of common submodule
1707 From ac2f647 to f363b32
1709 2016-04-14 22:56:11 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1711 * gst/rtsp-server/rtsp-sdp.c:
1712 * gst/rtsp-server/rtsp-sdp.h:
1713 * gst/rtsp-server/rtsp-stream.c:
1714 * gst/rtsp-server/rtsp-stream.h:
1715 sdp: add rollover counters for all sender SSRC
1716 We add different crypto sessions in MIKEY, one for each sender
1717 SSRC. Currently, all of them will have the same security policy, 0.
1718 The rollover counters are obtained from the srtpenc element using the
1720 https://bugzilla.gnome.org/show_bug.cgi?id=730539
1722 2016-06-07 20:44:42 +0100 Tim-Philipp Müller <tim@centricular.com>
1724 * gst/rtsp-server/rtsp-media-factory.h:
1725 * gst/rtsp-server/rtsp-server.h:
1726 docs: fix some typos
1728 2016-05-25 10:28:43 +0100 Tim-Philipp Müller <tim@centricular.com>
1730 * gst/rtsp-server/Makefile.am:
1731 g-i: pass compiler env to g-ir-scanner
1732 It's what introspection.mak does as well. Should
1733 fix spurious build failures on gnome-continuous
1734 (caused by g-ir-scanner getting compiler details
1735 via python which is broken in some environments
1736 so passing the compiler details bypasses that).
1738 2016-05-18 16:48:44 +0100 Ian <ian.arkver.dev@gmail.com>
1740 * gst/rtsp-server/rtsp-session.c:
1741 rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header
1742 This works with rtspsrc and live555, but fails with e.g. ffmpeg.
1743 https://bugzilla.gnome.org/show_bug.cgi?id=766619
1745 2016-03-07 14:48:38 +0100 Edward Hervey <bilboed@bilboed.com>
1747 * gst/rtsp-sink/gstrtspclientsink.c:
1748 rtspclientsink: Check return value of sscanf
1749 And just make sure we always have 0/0 if we have an error
1752 2016-04-25 08:55:25 -0400 Jake Foytik <jake.foytik@ipconfigure.com>
1754 * gst/rtsp-server/rtsp-stream.c:
1755 * tests/check/gst/rtspserver.c:
1756 * tests/check/gst/stream.c:
1757 rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
1758 - Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
1759 - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
1760 - Create unit test for shared media.
1761 https://bugzilla.gnome.org/show_bug.cgi?id=764744
1763 2016-04-11 10:55:23 +0300 Sebastian Dröge <sebastian@centricular.com>
1765 * gst/rtsp-server/rtsp-stream.c:
1766 rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
1767 For IPv6 addresses, binding to a multicast group does not work on Linux
1768 either. Always bind to ANY and then later join the multicast group.
1769 https://bugzilla.gnome.org/show_bug.cgi?id=764679
1771 2016-04-14 10:05:02 +0100 Julien Isorce <j.isorce@samsung.com>
1774 Automatic update of common submodule
1775 From 6f2d209 to ac2f647
1777 2016-04-06 10:09:46 +0200 Patricia Muscalu <patricia@axis.com>
1779 * gst/rtsp-server/rtsp-thread-pool.c:
1780 rtsp-thread-pool: explained why GSource is a part of ThreadImpl
1781 Clarified why it is necessary to add source information to
1782 GstRTSPThreadImpl. See the reported bug in GLib:
1783 https://bugzilla.gnome.org/show_bug.cgi?id=720186
1784 for more information.
1785 https://bugzilla.gnome.org/show_bug.cgi?id=761702
1787 2016-04-04 12:58:38 +0300 Sebastian Dröge <sebastian@centricular.com>
1789 * examples/Makefile.am:
1790 examples: Clean up CFLAGS/LDADD even more
1791 The internal .la should come first and is part of LDADD, as is
1794 2016-04-04 12:39:39 +0300 Sebastian Dröge <sebastian@centricular.com>
1796 * examples/Makefile.am:
1797 examples: Clean up CFLAGS/LDADD to link with the correct versions of all libraries
1799 2016-04-03 12:06:29 +0300 Sebastian Dröge <sebastian@centricular.com>
1801 * gst/rtsp-server/Makefile.am:
1802 rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS)
1804 2015-12-30 18:39:05 +0200 Sebastian Dröge <sebastian@centricular.com>
1806 * gst/rtsp-server/rtsp-client.c:
1807 * gst/rtsp-server/rtsp-media-factory.c:
1808 * gst/rtsp-server/rtsp-media-factory.h:
1809 * gst/rtsp-server/rtsp-media.c:
1810 * gst/rtsp-server/rtsp-media.h:
1811 * gst/rtsp-server/rtsp-sdp.c:
1812 * gst/rtsp-server/rtsp-stream.c:
1813 * gst/rtsp-server/rtsp-stream.h:
1814 rtsp-server: Implement clock signalling according to RFC7273
1815 For NTP and PTP clocks we signal the actual clock that is used and signal
1816 the direct media clock offset.
1817 For all other clocks we at least signal that it's the local sender clock.
1818 This allows receivers to know which clock was used to generate the media and
1819 its RTP timestamps. Receivers can then implement network synchronization,
1820 either absolute or at least relative by getting the sender clock rate directly
1821 via NTP/PTP instead of estimating it from RTP timestamps and packet receive
1823 https://bugzilla.gnome.org/show_bug.cgi?id=760005
1825 2016-03-02 19:42:58 +0200 Sebastian Dröge <sebastian@centricular.com>
1827 * gst/rtsp-sink/gstrtspclientsink.c:
1828 rtspclientsink: Add support for setting the multicast interface
1829 https://bugzilla.gnome.org/show_bug.cgi?id=763000
1831 2016-03-02 19:42:13 +0200 Sebastian Dröge <sebastian@centricular.com>
1833 * gst/rtsp-server/rtsp-media-factory.c:
1834 * gst/rtsp-server/rtsp-media-factory.h:
1835 * gst/rtsp-server/rtsp-media.c:
1836 * gst/rtsp-server/rtsp-media.h:
1837 * gst/rtsp-server/rtsp-stream.c:
1838 * gst/rtsp-server/rtsp-stream.h:
1839 rtsp-media: Add support for setting the multicast interface
1840 https://bugzilla.gnome.org/show_bug.cgi?id=763000
1842 2016-03-07 08:50:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
1844 * gst/rtsp-sink/gstrtspclientsink.c:
1845 rtspclientsink: use new gst_element_class_add_static_pad_template()
1846 https://bugzilla.gnome.org/show_bug.cgi?id=763196
1848 2016-03-24 13:33:43 +0200 Sebastian Dröge <sebastian@centricular.com>
1853 === release 1.8.0 ===
1855 2016-03-24 13:00:35 +0200 Sebastian Dröge <sebastian@centricular.com>
1861 * gst-rtsp-server.doap:
1864 2016-03-16 23:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
1866 * gst/rtsp-server/rtsp-stream.c:
1867 rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
1868 This would get us NO_PREROLL in the bin again and break seeking.
1869 Thanks to Carlos Rafael Giani for helping to debug this!
1870 https://bugzilla.gnome.org/show_bug.cgi?id=740509
1872 === release 1.7.91 ===
1874 2016-03-15 12:26:13 +0200 Sebastian Dröge <sebastian@centricular.com>
1880 * gst-rtsp-server.doap:
1883 2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
1885 * gst/rtsp-server/rtsp-stream.c:
1886 rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
1887 Without this, RECORD pipelines are broken because
1888 a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
1889 added later. Previously it was there earlier and due to NO_PREROLL caused the
1890 pipeline to preroll immediately
1891 b) the udpsrc for the pipeline is added later and never set to PLAYING state,
1892 as the corresponding code previously was only for PLAY pipelines.
1893 https://bugzilla.gnome.org/show_bug.cgi?id=763281
1895 2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
1897 * gst/rtsp-server/rtsp-stream.c:
1898 rtsp-stream: Fix typo in the docstring
1899 gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
1901 2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
1903 * gst/rtsp-server/rtsp-stream.c:
1904 rtsp-stream: Disable multicast loopback for all our sockets
1905 On Windows this is a receiver-side setting, on Linux a sender-side setting. As
1906 we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
1907 loopback setting on the socket... while udpsink does which unfortunately has
1908 no effect here on Windows but on Linux.
1909 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1911 2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
1913 * tests/check/gst/stream.c:
1914 stream tests: added new tests
1915 Test a case when the address pool only contains multicast addresses
1916 and the client is requesting unicast udp.
1917 Added tests for multicast ports allocation.
1918 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1920 2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
1922 * gst/rtsp-server/rtsp-stream.c:
1923 rtsp-stream: Only bind multicast sockets to ANY on Windows
1924 On Linux it is still needed to bind to the multicast address
1925 to filter out random other packets, while on Windows binding
1926 to multicast addresses just fails.
1928 2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
1930 * gst/rtsp-server/rtsp-stream.c:
1931 rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
1932 Otherwise we fail to allocate UDP ports if the pool only contains multicast
1933 addresses, which is something that used to work before. For unicast addresses
1934 if the pool contains none, we just allocate them as if there is no pool at
1936 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1938 2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
1940 * gst/rtsp-server/rtsp-client.c:
1941 * gst/rtsp-server/rtsp-stream.c:
1942 rtsp-server: Fix indentation
1944 2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
1946 * gst/rtsp-server/rtsp-stream.c:
1947 rtsp-stream: Don't bind the sockets to multicast addresses
1948 This works on Linux but fails completely on Windows. You're supposed
1949 to bind to ANY and then join the multicast group.
1950 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1952 === release 1.7.90 ===
1954 2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
1960 * gst-rtsp-server.doap:
1963 2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
1966 Automatic update of common submodule
1967 From b64f03f to 6f2d209
1969 2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
1971 * gst/rtsp-sink/gstrtspclientsink.c:
1972 * tests/check/gst/rtspclientsink.c:
1973 rtspsink: Fix some leaks in rtspclientsink and the unit test.
1974 https://bugzilla.gnome.org/show_bug.cgi?id=762525
1976 2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
1978 * tests/check/gst/media.c:
1979 * tests/check/gst/rtspclientsink.c:
1980 * tests/check/gst/rtspserver.c:
1981 * tests/check/gst/stream.c:
1982 tests: unit test fixes
1983 Removed port allocation test from the media suite.
1984 The port allocation failure is now in the stream suite.
1986 Make sure that the media is suspended after the DESCRIBE request
1987 before reconfiguring the UDP sinks.
1989 In the RECORD case we have to set async property to false
1990 for the appsink element in the test in order to make sure
1991 that the media pipeline doesn't hang in start_preroll().
1992 https://bugzilla.gnome.org/show_bug.cgi?id=757488
1994 2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
1996 * gst/rtsp-server/rtsp-client.c:
1997 * gst/rtsp-server/rtsp-stream.c:
1998 * gst/rtsp-server/rtsp-stream.h:
1999 rtsp-stream: postpone UDP socket allocation until SETUP
2000 Postpone the allocation of the UDP sockets until we know
2001 what transport has been chosen by the client.
2002 Both unicast and multicast UDP sources are created in one
2004 https://bugzilla.gnome.org/show_bug.cgi?id=757488
2006 2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
2008 * gst/rtsp-server/rtsp-stream.c:
2009 rtsp-stream: postpone the creation of the UDP sources
2010 Code refactoring: allocate the UDP ports after the sender and
2011 the reciver parts have been created.
2012 We postpone the creation of the UDP sources until the UDP
2013 ports have been allocated.
2014 https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016 2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
2018 * gst/rtsp-server/rtsp-stream.c:
2019 rtsp-stream: added function for setting UDP sources to PLAYING state
2020 Code refactoring: Introduced a function for setting UDP sources
2022 https://bugzilla.gnome.org/show_bug.cgi?id=757488
2024 2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
2026 * gst/rtsp-server/rtsp-stream.c:
2027 rtsp-stream: added function for creating and configuring UDP sources
2028 Code refactoring: create and configure UDP sources in a separate function.
2029 https://bugzilla.gnome.org/show_bug.cgi?id=757488
2031 2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
2033 * gst/rtsp-server/rtsp-stream.c:
2034 rtsp-stream: added function for RTP/RTCP socket configuration
2035 Code refactoring: configure RTP and RTCP sockets for UDP sinks
2036 in a separate function.
2037 https://bugzilla.gnome.org/show_bug.cgi?id=757488
2039 2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
2041 * gst/rtsp-server/rtsp-stream.c:
2042 rtsp-stream: added function for creating and configuring UDP sinks
2043 Code refactoring: create and configure UDP sinks in a separate function.
2044 https://bugzilla.gnome.org/show_bug.cgi?id=757488
2046 2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
2048 * gst/rtsp-server/rtsp-stream.c:
2049 rtsp-stream: added helper function for creating the sender/receiver parts
2050 Code refactoring: introduced helper function for creating
2051 the receiver and the sender parts of the streaming pipeline.
2052 https://bugzilla.gnome.org/show_bug.cgi?id=757488
2054 2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
2059 === release 1.7.2 ===
2061 2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
2067 * gst-rtsp-server.doap:
2070 2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
2072 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
2073 uninstalled.pc: add support for non libtool build systems
2074 Currently the .la path is provided which requires to use libtool as
2075 mentioned in the GStreamer manual section-helloworld-compilerun.html.
2076 It is fine as long as the application is built using libtool.
2077 So currently it is not possible to compile a GStreamer application
2078 within gst-uninstalled with CMake or other build system different
2080 This patch allows to do the following in gst-uninstalled env:
2081 gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
2082 gstreamer-rtsp-server-1.0)
2083 Previously it required to prepend libtool --mode=link
2084 https://bugzilla.gnome.org/show_bug.cgi?id=720778
2086 2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
2088 * gst/rtsp-sink/gstrtspclientsink.c:
2089 rtspclientsink: remove check for impossible condition
2090 Goto error label checks stream to see if it needs to be unreferenced before
2091 returning, but this goto jumps happens before the stream is ever set, so it
2092 will always be NULL in this error label.
2095 2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
2097 * gst/rtsp-sink/gstrtspclientsink.c:
2098 rtspclientsink: clean switch statements
2099 Coverity demands for fallthrough statements to be clearly commented,
2100 to distinguish from accidental fall throughs. And it also needs all
2101 cases to finish with a break, even if the break is never going to be
2102 executed like in the case of a continue jump.
2106 2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
2108 * tests/check/Makefile.am:
2109 tests: extend the AM_TESTS_ENVIRONMENT from check.mak
2110 To get the CK_DEFAULT_TIMEOUT defined for all tests
2111 Also removes a 120 seconds timeout that was set as default
2112 explicitly in this module
2113 https://bugzilla.gnome.org/show_bug.cgi?id=761472
2115 2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
2119 Automatic update of common submodule
2120 From 86e4663 to b64f03f
2122 2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
2124 * gst/rtsp-server/rtsp-media.c:
2125 rtsp-media: fix state_lock not locked again when preroll fails
2126 https://bugzilla.gnome.org/show_bug.cgi?id=761399
2128 2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
2131 configure: Move plugin specific flags below all the others
2132 They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
2133 -no-undefined. And -no-undefined is required on Windows to build DLLs.
2135 2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
2137 * gst/rtsp-sink/gstrtspclientsink.c:
2138 rtspclientsink: Simplify slightly using new -base API
2139 Use the new Mikey and SDP API in the base plugins libs
2140 to simplify some code.
2141 https://bugzilla.gnome.org/show_bug.cgi?id=758180
2143 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
2148 * gst/rtsp-sink/Makefile.am:
2149 * gst/rtsp-sink/gstrtspclientsink.c:
2150 * gst/rtsp-sink/gstrtspclientsink.h:
2151 * gst/rtsp-sink/plugin.c:
2152 * tests/check/Makefile.am:
2153 * tests/check/gst/rtspclientsink.c:
2154 rtspsink: Add rtspclientsink element
2155 Add an rtspclientsink element that accepts streams for which
2156 there is a registered payloader and sends them to
2157 an RTSP server using RECORD.
2158 Sending is synchronised to the pipeline clock. Payload-types
2159 are automatically selected. The 'new-payloader' signal is fired
2160 for custom configuration of payloaders when they are created.
2161 Can now stream a movie like this:
2163 ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
2164 decodebin name=depay1 ! audioconvert ! autoaudiosink )"
2166 gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
2167 queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
2168 https://bugzilla.gnome.org/show_bug.cgi?id=758180
2170 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
2172 * gst/rtsp-server/rtsp-stream.c:
2173 * gst/rtsp-server/rtsp-stream.h:
2174 rtsp-stream: Add functions for using rtsp-stream from the client
2175 Add a boolean to indicate that the rtsp-stream is running on the
2176 'client' side of an RTSP connection, for sending streams via
2177 RECORD. In that case, the roles of the client/server ports
2178 in transport setup are swapped.
2179 https://bugzilla.gnome.org/show_bug.cgi?id=758180
2181 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
2183 * gst/rtsp-server/rtsp-sdp.c:
2184 * gst/rtsp-server/rtsp-sdp.h:
2185 rtsp-sdp: Add gst_rtsp_sdp_from_stream()
2186 A new function that adds info from a GstRTSPStream into an SDP message.
2187 https://bugzilla.gnome.org/show_bug.cgi?id=758180
2189 2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
2191 * gst/rtsp-server/rtsp-media.c:
2192 rtsp-media: Fix mutex beeing unlocked while they should be locked
2193 https://bugzilla.gnome.org/show_bug.cgi?id=761226
2195 2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
2197 * gst/rtsp-server/rtsp-media-factory.c:
2198 rtsp-media-factory: add missing break in "clock" property setter
2201 2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
2203 * gst/rtsp-server/rtsp-stream.c:
2204 rtsp-stream: fixed assert during update transport
2205 When RTSP server trying update transport during multicast, it throws an
2206 assert. The assert is thrown because it is trying to get the parent of
2207 an non-existing funnel element.
2208 https://bugzilla.gnome.org/show_bug.cgi?id=760150
2210 2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
2212 * gst/rtsp-server/rtsp-permissions.h:
2213 * gst/rtsp-server/rtsp-thread-pool.h:
2214 * gst/rtsp-server/rtsp-token.h:
2215 docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
2216 gtk-doc can handle static inline functions just fine these days,
2217 there's no need for this stuff any more.
2219 2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
2221 * gst/rtsp-server/rtsp-media.c:
2222 * gst/rtsp-server/rtsp-sdp.c:
2223 sdp: replace duplicated codes to call new base sdp apis
2224 https://bugzilla.gnome.org/show_bug.cgi?id=745880
2226 2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
2228 * examples/test-netclock.c:
2229 test-netclock: Use the new API to configure a clock directly
2231 2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
2233 * gst/rtsp-server/rtsp-media-factory.c:
2234 * gst/rtsp-server/rtsp-media-factory.h:
2235 * gst/rtsp-server/rtsp-media.c:
2236 * gst/rtsp-server/rtsp-media.h:
2237 rtsp-media: Add API to directly configure a clock on the media pipelines
2239 2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
2241 * gst/rtsp-server/rtsp-media.c:
2242 rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
2244 2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
2246 * gst/rtsp-server/rtsp-media-factory.c:
2247 rtsp-media-factory: Add FIXME for 2.0
2249 2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
2251 * gst/rtsp-server/rtsp-stream.c:
2252 rtsp-stream: Fix indentation
2254 2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2256 * gst/rtsp-server/rtsp-media.c:
2257 rtsp-media: Do not prepare media after media times out
2258 Deferred calls to start_prepare() can be deferred past the point until
2259 which wait_preroll() and by proxy gst_rtsp_media_get_status() is
2260 prepared to wait. Previously there was no lock and no check for this
2261 situation. This meant that a media could be prepared and unprepared
2262 simultaneously by two different threads. Now a lock is in place and a
2263 suitable check is done.
2264 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
2266 2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
2268 * gst/rtsp-server/rtsp-client.c:
2269 * gst/rtsp-server/rtsp-media-factory.c:
2270 * gst/rtsp-server/rtsp-media-factory.h:
2271 * gst/rtsp-server/rtsp-media.c:
2272 * gst/rtsp-server/rtsp-media.h:
2273 rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
2274 Without TEARDOWN it might be desireable to keep the media running and continue
2275 sending data to the client, even if the RTSP connection itself is
2277 Only do this for session medias that have only UDP transports. If there's at
2278 least on TCP transport, it will stop working and cause problems when the
2279 connection is disconnected.
2280 https://bugzilla.gnome.org/show_bug.cgi?id=758999
2282 2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
2287 === release 1.7.1 ===
2289 2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
2295 * gst-rtsp-server.doap:
2298 2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
2301 configure: Make -Bsymbolic check work with clang.
2302 Update the -Bsymbolic check with the version glib has. This version
2304 https://bugzilla.gnome.org/show_bug.cgi?id=759713
2306 2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
2308 * gst/rtsp-server/rtsp-session-pool.c:
2309 rtsp-session-pool: Avoid dollar sign ($) in session ids
2310 Live555 in VLC strips off dollar signs and then gets very confused,
2311 we don't loose too much entropy by just skipping it.
2313 2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
2315 * gst/rtsp-server/rtsp-address-pool.h:
2316 * gst/rtsp-server/rtsp-auth.h:
2317 * gst/rtsp-server/rtsp-client.h:
2318 * gst/rtsp-server/rtsp-media-factory-uri.h:
2319 * gst/rtsp-server/rtsp-media-factory.h:
2320 * gst/rtsp-server/rtsp-media.h:
2321 * gst/rtsp-server/rtsp-mount-points.h:
2322 * gst/rtsp-server/rtsp-permissions.h:
2323 * gst/rtsp-server/rtsp-server.h:
2324 * gst/rtsp-server/rtsp-session-media.h:
2325 * gst/rtsp-server/rtsp-session-pool.h:
2326 * gst/rtsp-server/rtsp-session.h:
2327 * gst/rtsp-server/rtsp-stream-transport.h:
2328 * gst/rtsp-server/rtsp-stream.h:
2329 * gst/rtsp-server/rtsp-thread-pool.h:
2330 * gst/rtsp-server/rtsp-token.h:
2331 rtsp-server: Add g_autoptr() support to all types
2332 https://bugzilla.gnome.org/show_bug.cgi?id=754464
2334 2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
2336 * gst/rtsp-server/rtsp-stream.c:
2337 rtsp-stream: fixed valgrind error
2338 Fixed the valgrind error in unit test. The UDP source created during
2339 gst_rtsp_stream_join_bin() was not released while destroying the rtp
2341 https://bugzilla.gnome.org/show_bug.cgi?id=759010
2343 2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
2347 Automatic update of common submodule
2348 From b319909 to 86e4663
2350 2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
2352 * gst/rtsp-server/rtsp-client.c:
2353 rtsp-client: suspend media during setup request
2354 SETUP request from clients needs to suspend the media to clear the
2355 prerolled buffers. Otherwise it will not affect the prerolled buffer
2356 and the prerolled buffers will be incorrect (for example block-size
2357 from setup request will not affect the prerolled buffer unless the
2358 media is suspended).
2359 https://bugzilla.gnome.org/show_bug.cgi?id=758268
2361 2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
2363 * gst/rtsp-server/rtsp-stream.c:
2364 rtsp-stream: create stream pipeline based on transport
2365 Based on the protocol, create the rtsp stream pipeline. If only TCP or
2366 only UDP is set as the transport protocol, it will not add the extra tee
2367 or queue element to the pipeline. Both these elements will be added, if
2368 it supports both TCP and UDP protocols. This improves the pipeline
2369 performance when one protocol is present.
2370 https://bugzilla.gnome.org/show_bug.cgi?id=758179
2372 2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
2374 * gst/rtsp-server/rtsp-stream.c:
2375 rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
2376 Adding them when not needed will start some logic inside rtpbin that might be
2377 problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
2378 would start up a rtpjitterbuffer and behave in weird ways.
2379 We still set up the UDP sources for RTP receiving for a sender media to be
2380 able to receive any packets sent by the client for NAT traversal. They will
2381 all go to a fakesink though.
2382 Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
2383 NO_PREROLL, which will cause deadlocks when seeking the media as it will never
2384 receive ASYNC_DONE after a seek.
2385 https://bugzilla.gnome.org/show_bug.cgi?id=758319
2387 2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
2389 * gst/rtsp-server/rtsp-stream.c:
2390 rtsp-stream: Disable multicast loopback for the multicast udp sources too
2391 On POSIX this setting is for sender sockets, on Windows for receiver sockets.
2392 Previously we were only setting this for sender sockets, which caused looped
2393 back packets to be received on Windows if a multicast transport was used.
2395 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
2397 * examples/test-record-auth.c:
2398 * examples/test-record.c:
2399 examples: Actually use the provided port in the record examples
2401 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
2403 * examples/test-record-auth.c:
2404 test-record-auth: Add the option to build in TLS support
2406 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
2408 * examples/test-auth.c:
2409 test-auth: Use an 'anonymous' user for unauthenticated default
2410 There's a comment on one of the resources that 'user' and 'admin'
2411 shouldn't even be able to see it, but they can if the default
2412 token is 'admin2', since that gives them access anyway.
2414 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
2416 * examples/.gitignore:
2417 * examples/Makefile.am:
2418 * examples/test-record-auth.c:
2419 Add test-record-auth example
2421 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
2423 * gst/rtsp-server/rtsp-client.c:
2424 * tests/check/gst/client.c:
2425 rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
2427 2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
2429 * gst/rtsp-server/rtsp-server.c:
2430 rtsp-server: Change the logic so we don't pop a NULL context
2431 When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
2432 will sometimes fail. This call is made before any context is pushed
2433 resulting in an attempt to pop a NULL context.
2434 https://bugzilla.gnome.org/show_bug.cgi?id=757949
2436 2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
2438 * tests/check/gst/rtspserver.c:
2439 rtspserver: Add udp-mcast transport SETUP test
2440 Refactor utility functions in the test file so they can handle
2441 more than UDP and TCP as lower transport.
2442 https://bugzilla.gnome.org/show_bug.cgi?id=756969
2444 2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
2446 * gst/rtsp-server/rtsp-stream.c:
2447 rtsp-stream: Always unref return value of gst_object_get_parent()
2448 Fixes a leak of a GstBin in the udp-mcast case.
2449 https://bugzilla.gnome.org/show_bug.cgi?id=756968
2451 2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
2454 Automatic update of common submodule
2455 From b99800a to b319909
2457 2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
2460 Use new GST_ENABLE_EXTRA_CHECKS #define
2461 https://bugzilla.gnome.org/show_bug.cgi?id=756870
2463 2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
2466 Automatic update of common submodule
2467 From 6babecd to b99800a
2469 2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
2472 Update GLib dependency to 2.40.0
2474 2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
2476 * examples/test-mp4.c:
2477 * gst/rtsp-server/rtsp-stream.c:
2478 stream: listen to sender ssrc signals
2479 https://bugzilla.gnome.org/show_bug.cgi?id=746747
2481 2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
2484 common: update for new suppression
2485 Makes check-valgrind pass with glib 2.46
2487 2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2489 * gst/rtsp-server/rtsp-media.c:
2490 rtsp-media: Take reference to media that will be prepared
2491 default_prepare() takes a transfer-none reference GstRTSPMedia object.
2492 Later on a g_idle_source_new() is created and a pointer to the media
2493 object is passed as user data. If the media is freed before the idle
2494 source is dispatched the media object pointer is invalid, but the idle
2495 source callback expects it to still be valid. To fix this a reference to
2496 the media object is taken when registering the source callback function
2497 and a corresponding release of the reference is done when the souce is
2499 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
2501 2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
2503 * examples/test-launch.c:
2504 * examples/test-mp4.c:
2505 * examples/test-ogg.c:
2506 * examples/test-record.c:
2507 * examples/test-uri.c:
2508 rtsp-server: Fix memory leaks when context parse fails
2509 When g_option_context_parse fails, context and error variables are not getting free'd
2510 which results in memory leaks. Free'ing the same.
2511 And replacing g_error_free with g_clear_error, which checks if the error being passed
2512 is not NULL and sets the variable to NULL on free'ing.
2513 https://bugzilla.gnome.org/show_bug.cgi?id=753863
2515 2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
2520 === release 1.6.0 ===
2522 2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
2528 * gst-rtsp-server.doap:
2531 === release 1.5.91 ===
2533 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
2539 * gst-rtsp-server.doap:
2542 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
2544 * docs/libs/gst-rtsp-server-sections.txt:
2545 * gst/rtsp-server/rtsp-stream.c:
2546 stream: fix docs for recently-added get/set_buffer_size API
2547 https://bugzilla.gnome.org/show_bug.cgi?id=749095
2549 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
2551 * gst/rtsp-server/rtsp-media.c:
2552 rtsp-media: Don't crash on encrypted RTX SDP
2553 In parse_keymgmt(), don't mutate the input string that's been passed
2554 as const, especially since we might need the original value again if
2555 the same key info applies to multiple streams (RTX, for example).
2556 https://bugzilla.gnome.org/show_bug.cgi?id=754753
2558 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
2560 * examples/test-mp4.c:
2561 test-mp4: Support filenames with spaces in them. Error out on too few arguments
2563 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
2565 * examples/test-record.c:
2566 test-record: Check parameter count and print out help
2567 If no launch pipeline was supplied, print out some help
2569 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
2571 * gst/rtsp-server/rtsp-media.c:
2572 * gst/rtsp-server/rtsp-stream.c:
2573 * gst/rtsp-server/rtsp-stream.h:
2574 rtsp-stream: Implement UDP buffer size setting.
2575 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
2577 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
2578 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
2580 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
2582 * gst/rtsp-server/rtsp-media.h:
2583 rtsp-media: Fix small typo causing gtk-doc to complain
2585 === release 1.5.90 ===
2587 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
2593 * gst-rtsp-server.doap:
2596 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
2598 * gst/rtsp-server/rtsp-media-factory.c:
2599 media-factory: get port number through gst_rtsp_url_get_port
2600 https://bugzilla.gnome.org/show_bug.cgi?id=753473
2602 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
2604 * tests/check/gst/media.c:
2605 media-test: Removing unnecessary assertion
2606 https://bugzilla.gnome.org/show_bug.cgi?id=753385
2608 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2610 * gst/rtsp-server/rtsp-server.c:
2611 Document that source keeps a ref on server until it's destroyed
2612 https://bugzilla.gnome.org/show_bug.cgi?id=749227
2614 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
2616 * tests/check/gst/media.c:
2617 media-test: Test for multiple dynamic payload
2618 https://bugzilla.gnome.org/show_bug.cgi?id=753385
2620 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
2622 * gst/rtsp-server/rtsp-media.c:
2623 media: Only add fakesink once per pipeline
2624 The intention is to prevent going PLAYING state before pads are created.
2625 If there was mutilple dynamic payload, it would leak few fakesink and
2626 actually prevent from ever reaching playing state.
2627 https://bugzilla.gnome.org/show_bug.cgi?id=753385
2629 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
2631 * gst/rtsp-server/rtsp-media.c:
2632 Revert "rtsp-media: Only add 1 fakesink per pipeline"
2633 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
2635 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
2637 * gst/rtsp-server/rtsp-media.c:
2638 rtsp-media: Only add 1 fakesink per pipeline
2639 There should be only one fakesink per pipeline, not per dynpay. This
2640 would lead to element naming clash.
2642 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
2644 * gst/rtsp-server/rtsp-media.c:
2645 rtsp-media: assertion error due to wrong condition check
2646 In media to caps function, reserved_keys array is being used for variable i,
2647 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
2648 changed it to variable j
2649 https://bugzilla.gnome.org/show_bug.cgi?id=753009
2651 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
2653 * gst/rtsp-server/rtsp-media.c:
2654 rtsp-media: Strip keys from the fmtp that we use internally in our caps
2655 Skip keys from the fmtp, which we already use ourselves for the
2656 caps. Some software is adding random things like clock-rate into
2657 the fmtp, and we would otherwise here set a string-typed clock-rate
2658 in the caps... and thus fail to create valid RTP caps
2659 https://bugzilla.gnome.org/show_bug.cgi?id=753009
2661 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2663 * gst/rtsp-server/rtsp-thread-pool.c:
2664 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
2665 https://bugzilla.gnome.org/show_bug.cgi?id=752640
2667 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
2670 Automatic update of common submodule
2671 From f74b2df to 9aed1d7
2673 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
2678 === release 1.5.2 ===
2680 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
2686 * gst-rtsp-server.doap:
2689 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
2691 * gst/rtsp-server/rtsp-client.c:
2692 * gst/rtsp-server/rtsp-client.h:
2693 * tests/check/gst/client.c:
2694 rtsp-client: allow application to decide what requirements are supported
2695 Add "check-requirements" signal and vfunc to allow application
2696 (and subclasses) to check the requirements.
2697 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
2698 https://bugzilla.gnome.org/show_bug.cgi?id=749417
2700 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
2703 Automatic update of common submodule
2704 From 6015d26 to f74b2df
2706 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
2708 * gst/rtsp-server/rtsp-media.c:
2709 rtsp-media: Always use real payloader when creating streams
2710 A bin that contains the real payloader might be used as payloader. In this
2711 case we have to get the real payloader for the various properties it provides.
2712 Example use cases for this are bins that payload some media and then have
2713 additional elements that add metadata or RTP extension headers to the stream.
2714 https://bugzilla.gnome.org/show_bug.cgi?id=750800
2716 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
2718 * examples/test-netclock-client.c:
2719 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
2721 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
2723 * examples/test-netclock-client.c:
2724 * examples/test-netclock.c:
2725 test-netclock: Use new ntp-time-source property on rtpbin
2726 Select the clock time to be used as NTP time source. This allows proper
2727 synchronization between receivers, independent of sharing base times, and just
2728 requires them to use the same clock.
2730 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
2732 * examples/test-netclock-client.c:
2733 * examples/test-netclock.c:
2734 test-netclock: Setting the same base time on sender and receiver is not necessary
2735 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
2737 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
2739 * gst/rtsp-server/rtsp-stream.c:
2740 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
2741 https://bugzilla.gnome.org/show_bug.cgi?id=750764
2743 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
2745 * docs/libs/gst-rtsp-server.types:
2746 docs: add missing types
2747 https://bugzilla.gnome.org/show_bug.cgi?id=750764
2749 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
2751 * docs/libs/gst-rtsp-server-sections.txt:
2752 docs: add missing apis
2753 https://bugzilla.gnome.org/show_bug.cgi?id=750764
2755 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
2757 * examples/test-netclock-client.c:
2758 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
2760 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2762 * docs/libs/gst-rtsp-server-sections.txt:
2763 * gst/rtsp-server/rtsp-auth.c:
2764 * gst/rtsp-server/rtsp-auth.h:
2765 GstRTSPAuth: Add client certificate authentication support
2766 https://bugzilla.gnome.org/show_bug.cgi?id=750471
2768 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
2770 * examples/test-netclock-client.c:
2771 test-netclock-client: Use new GstClock API to wait for clock synchronization
2773 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
2775 * examples/test-netclock-client.c:
2776 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
2777 A mainloop is needed to get glimagesink to display something on OSX, and
2778 the source-setup signal just makes things a little bit easier.
2780 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
2783 Automatic update of common submodule
2784 From d9a3353 to 6015d26
2786 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
2789 Automatic update of common submodule
2790 From d37af32 to d9a3353
2792 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
2795 Automatic update of common submodule
2796 From 21ba2e5 to d37af32
2798 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
2801 Automatic update of common submodule
2802 From c408583 to 21ba2e5
2804 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
2806 * docs/libs/Makefile.am:
2807 docs: remove variables that we define in the snippet from common
2808 This is syncing our Makefile.am with upstream gtkdoc.
2810 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
2813 Automatic update of common submodule
2814 From 44a3517 to c408583
2816 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
2821 === release 1.5.1 ===
2823 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
2829 * gst-rtsp-server.doap:
2832 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
2834 * gst/rtsp-server/rtsp-client.c:
2835 rtsp-client: No flush during Teardown.
2836 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
2837 backlog is empty it can happen that just a part of a message will be
2838 sent and rest is in backlog queue. If then flush during teardown
2839 just a part of message will be sent.This can lead to client miss
2840 teardown response since it expect to get the last part of message.
2841 The flushing during teardown was introduced to fix a deadlock that now
2842 is fixed more generally in handle_request by temporary setting backlog
2844 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
2846 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
2848 * tests/check/Makefile.am:
2849 tests: Use AM_TESTS_ENVIRONMENT
2850 Needed by the new automake test runner and the
2851 current version of the common submodule.
2853 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
2855 * gst/rtsp-server/rtsp-media.h:
2856 * gst/rtsp-server/rtsp-stream.h:
2857 rtsp-server: Use single-include rtsp header to make sure we get all definitions
2859 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
2861 * gst/rtsp-server/rtsp-media.c:
2862 rtsp-media: Mark some more functions static
2864 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
2866 * gst/rtsp-server/rtsp-media.c:
2867 rtsp-media: Only unblock the media in suspend() when actually changing the state
2868 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
2870 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
2872 * examples/test-video-rtx.c:
2873 examples: Use AVPF profile for the RTX example
2875 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
2877 * gst/rtsp-server/rtsp-sdp.c:
2878 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
2880 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
2882 * gst/rtsp-server/rtsp-stream.c:
2883 rtsp-stream: get valid clock-rate from last-sample
2884 clock-rate in last-sample's caps is integer, not unsigned.
2885 To get this value properly, variable needs to be type-casted to int.
2886 https://bugzilla.gnome.org/show_bug.cgi?id=747614
2888 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
2892 autogen.sh: only run autopoint if gettext requested in configure.ac
2893 Not just because there happens to be a po directory.
2894 https://bugzilla.gnome.org/show_bug.cgi?id=748058
2896 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
2899 Revert "configure.ac: uncomment gettext version setup"
2900 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
2901 We don't need a gettext setup here and there's no po
2902 directory either, so no reason why autopoint would be
2903 run in the first place.
2904 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
2906 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
2908 * examples/test-multicast.c:
2909 * examples/test-multicast2.c:
2910 * examples/test-sdp.c:
2911 * examples/test-video-rtx.c:
2912 * examples/test-video.c:
2913 * tests/test-cleanup.c:
2914 * tests/test-reuse.c:
2915 Fix timeout function signatures across tests and examples
2917 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
2919 * tests/check/Makefile.am:
2920 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
2921 Make sure the test environment is set up.
2922 https://bugzilla.gnome.org//show_bug.cgi?id=747624
2924 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
2927 configure: bump automake requirement to 1.14 and autoconf to 2.69
2928 This is only required for builds from git, people can still
2929 build tarballs if they only have older autotools.
2930 https://bugzilla.gnome.org//show_bug.cgi?id=747624
2932 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
2935 configure.ac: uncomment gettext version setup
2936 Fixes autogen.sh. It would run autopoint, which would complain
2937 that it could not find the gettext version in configure.ac.
2938 https://bugzilla.gnome.org/show_bug.cgi?id=748058
2940 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
2942 * examples/test-video-rtx.c:
2943 test-video-rtx: set exact payload type to PCMA payloader
2944 Setting wrong payload type causes failure to do retransmission through audio stream
2945 https://bugzilla.gnome.org/show_bug.cgi?id=747839
2947 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
2949 * gst/rtsp-server/rtsp-media.c:
2950 * gst/rtsp-server/rtsp-stream.c:
2951 * gst/rtsp-server/rtsp-stream.h:
2952 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
2953 Because of duplicated g_signal_connect for request-aux-sender signal,
2954 wrong stream pointer is passed to the signal handler.
2955 Instead of passing each stream, pass stream array and get the relevant stream.
2956 https://bugzilla.gnome.org/show_bug.cgi?id=747839
2958 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
2962 Update autogen.sh to latest version from common
2963 Fixes build after aclocal_check etc. helpers have been removed.
2965 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
2968 Automatic update of common submodule
2969 From bc76a8b to c8fb372
2971 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
2973 * gst/rtsp-server/rtsp-stream.c:
2974 rtsp-stream: Limit the queues to 1 buffer
2975 We only need them to be able to pre-roll, queueing up more data here
2976 is only going to harm latency and memory usage.
2978 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
2980 * gst/rtsp-server/rtsp-stream.c:
2981 rtsp-stream: Update comment and ASCII art to the latest code
2982 We have a queue in front of the udpsink too to prevent the pipeline from
2985 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
2987 * gst/rtsp-server/rtsp-stream.c:
2988 rtsp-media: Properly return first rtptime
2989 Instead we where returning first GstBuffer timestamp. This would result
2990 in clock skew and unwanted behaviour in RTSP playback.
2991 https://bugzilla.gnome.org/show_bug.cgi?id=746479
2993 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
2995 * gst/rtsp-server/rtsp-stream.c:
2996 rtsp-stream: Don't leave buffer mapped
2997 If the seq is NULL, the RTP buffer was left mapped. We should always
3000 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
3005 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
3007 * gst/rtsp-server/rtsp-media-factory.c:
3008 * tests/check/gst/client.c:
3009 Fix double semicolons
3011 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
3013 * gst/rtsp-server/rtsp-stream.c:
3014 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
3015 This gives more accurate values than asking the payloader. There might be
3016 queueing happening between the payloader and the sink.
3017 https://bugzilla.gnome.org/show_bug.cgi?id=745704
3019 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
3021 * gst/rtsp-server/rtsp-media.c:
3022 rtsp-media: Don't seek for PLAY if the position will not change
3023 https://bugzilla.gnome.org/show_bug.cgi?id=745704
3025 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
3027 * gst/rtsp-server/rtsp-media.c:
3028 rtsp-media: Don't include payload type in the caps for framesize
3029 When the sdp media attribute framesize are converted to caps
3030 the <payload> should not be included.
3031 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
3032 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
3034 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
3036 * gst/rtsp-server/rtsp-sdp.c:
3037 rtsp-sdp: add payload type to the sdp framesize attribute
3038 The sdp framesize attribute is desribed in RFC6064. It is specified
3039 for payloading of H263 and has the following form
3040 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
3041 should be added to the caps in a payloader and the <payload type> should
3042 be added by the rtsp-server.
3043 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
3045 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
3047 * examples/test-uri.c:
3048 examples: test-uri: fix tainted variable
3049 Insignificant but this keeps Coverity happy.
3052 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
3054 * examples/.gitignore:
3055 * examples/Makefile.am:
3056 * examples/test-netclock-client.c:
3057 * examples/test-netclock.c:
3058 examples: Add a simple example of network synch for live streams.
3059 An example server and client that works for synchronising live streams
3060 only - as it can't support pause/play.
3062 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
3064 * gst/rtsp-server/rtsp-media-factory.c:
3065 * gst/rtsp-server/rtsp-media-factory.h:
3066 rtsp-media-factory: Add functions to set/get the media gtype
3067 Allow specifying the GType of a GstRtspMedia subclass to create
3068 as a simpler way to get the factory to create a custom
3069 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
3071 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
3073 * gst/rtsp-server/rtsp-media.c:
3074 rtsp-media: fix double unlock in _get_buffer_size()
3075 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
3076 because of double g_mutex_unlock () usage.
3077 https://bugzilla.gnome.org/show_bug.cgi?id=745434
3079 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
3081 * gst/rtsp-server/rtsp-session-pool.c:
3082 * gst/rtsp-server/rtsp-session.c:
3083 * gst/rtsp-server/rtsp-session.h:
3084 rtsp-session: Use monotonic time for RTSP session timeout
3085 Changed RTSP session timeout handling to monotonic time
3086 and deprecating the API for current system time.
3087 This fixes timeouts when the system time changes.
3088 https://bugzilla.gnome.org/show_bug.cgi?id=743346
3090 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
3092 * gst/rtsp-server/rtsp-client.c:
3093 * gst/rtsp-server/rtsp-media.c:
3094 rtsp-client: Only error out in PLAY if seeking actually failed
3095 If the media was just not seekable, we continue from whatever position we are
3096 and let the client decide if that is what is wanted or not.
3097 Only if the actual seek failed, we can't really recover and should error out.
3099 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
3101 * gst/rtsp-server/rtsp-stream.c:
3102 rtsp-stream: Add necessary queues between tee and multiudpsink
3103 https://bugzilla.gnome.org/show_bug.cgi?id=744379
3105 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
3107 * gst/rtsp-server/rtsp-client.c:
3108 * gst/rtsp-server/rtsp-media.c:
3109 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
3110 Instead error out properly the same way as if the SEEKING query already
3113 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
3115 * gst/rtsp-server/rtsp-stream.h:
3116 rtsp-stream: minor code formatting fix
3118 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
3120 * gst/rtsp-server/rtsp-media.c:
3121 rtsp-media: fix logic for collect_streams
3122 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
3123 all streams it knows if it got any, and can check if the transport mode is OK.
3126 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
3128 * gst/rtsp-server/rtsp-media.c:
3129 rtsp-media: Don't set the transport mode based on what elements we find
3130 Just print a warning if the one that was set before disagrees with what
3131 elements we found. It must already be set to something before as this
3132 function is called after we received the SDP from ANNOUNCE in RECORD mode,
3133 and we would reject ANNOUNCE if the RECORD flag was not set.
3135 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
3137 * tests/check/gst/rtspserver.c:
3138 tests: rtspserver: rename shadowed variable
3139 We have two different 'sink' variables here,
3140 rename one of them for clarity.
3142 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
3144 * gst/rtsp-server/rtsp-client.c:
3145 rtsp-client: fix awkward if clause
3147 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
3149 * examples/test-uri.c:
3150 examples: test-uri: improve uri argument handling and accept file names
3151 Print an error if the argument passed is not a URI and can't
3152 be converted into one, or no arguments have been provided.
3154 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
3156 * examples/test-uri.c:
3157 examples: test-uri: don't remove mount point after 10 seconds
3158 It's very irritating when trying to test stuff repeatedly
3159 and serves no real purpose other than showing that it can
3162 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
3164 * examples/.gitignore:
3165 examples: add new test-record to .gitignore
3167 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
3169 * examples/test-record.c:
3170 * gst/rtsp-server/rtsp-client.c:
3171 * gst/rtsp-server/rtsp-media-factory.c:
3172 * gst/rtsp-server/rtsp-media-factory.h:
3173 * gst/rtsp-server/rtsp-media.c:
3174 * gst/rtsp-server/rtsp-media.h:
3175 * tests/check/gst/rtspserver.c:
3176 rtsp-media: Use flags to distinguish between PLAY and RECORD media
3178 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
3180 * examples/test-record.c:
3181 test-record: Set latency for playback-style example to 2s instead of 200ms
3183 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
3185 * tests/check/gst/rtspserver.c:
3186 tests: add some unit tests for ANNOUNCE and RECORD
3187 https://bugzilla.gnome.org/show_bug.cgi?id=743175
3189 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
3191 * gst/rtsp-server/rtsp-client.c:
3192 rtsp-client: fix a couple of leaks in handle_announce
3194 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
3196 * gst/rtsp-server/rtsp-media-factory.c:
3197 * gst/rtsp-server/rtsp-media-factory.h:
3198 * gst/rtsp-server/rtsp-media.c:
3199 * gst/rtsp-server/rtsp-media.h:
3200 rtsp-media: Expose latency setting for setting the rtpbin latency
3202 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
3204 * examples/test-record.c:
3205 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
3207 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
3209 * gst/rtsp-server/rtsp-stream.c:
3210 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
3212 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
3214 * examples/Makefile.am:
3215 * examples/test-record.c:
3216 * gst/rtsp-server/rtsp-client.c:
3217 * gst/rtsp-server/rtsp-client.h:
3218 * gst/rtsp-server/rtsp-media-factory.c:
3219 * gst/rtsp-server/rtsp-media-factory.h:
3220 * gst/rtsp-server/rtsp-media.c:
3221 * gst/rtsp-server/rtsp-media.h:
3222 * gst/rtsp-server/rtsp-session-media.c:
3223 * gst/rtsp-server/rtsp-stream.c:
3224 * gst/rtsp-server/rtsp-stream.h:
3225 Add initial support for RECORD
3226 We currently only support media that is RECORD or PLAY only, not both at once.
3227 https://bugzilla.gnome.org/show_bug.cgi?id=743175
3229 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
3231 * gst/rtsp-server/rtsp-stream.c:
3232 rtsp-stream: RTCP and RTP transport cache cookies seperated
3233 RTCP packets were not sent because the same tr_cache_cookie was used for
3234 both RTP and RTCP. So only one of the tr_cache lists were populated
3235 depending on which one was sent first. If the tr_cache list is not
3236 populated then no packets can be sent. Most often this happened to be
3237 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
3238 resulted in both the tr_cache_lists to be populated regardless of which
3240 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
3242 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
3244 * gst/rtsp-server/rtsp-stream.c:
3245 rtsp-stream: fix false compiler warning
3246 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
3248 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
3250 * gst/rtsp-server/rtsp-client.c:
3251 rtsp-client: log interleaved data received
3253 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
3255 * gst/rtsp-server/rtsp-client.c:
3256 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
3258 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
3260 * gst/rtsp-server/rtsp-client.c:
3261 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
3263 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
3265 * gst/rtsp-server/rtsp-client.c:
3266 rtsp-client: Use a random session ID in the SDP
3267 RFC4566 Section 5.2 says that it should make the username, session id,
3268 nettype, addrtype and unicast address tuple globally unique. Always using
3269 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
3270 Instead let's create a 64 bit random number, which at least brings us
3271 closer to the goal of global uniqueness.
3272 https://tools.ietf.org/html/rfc4566#section-5.2
3274 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
3276 * examples/test-launch.c:
3277 * examples/test-mp4.c:
3278 * examples/test-ogg.c:
3279 * examples/test-uri.c:
3280 examples: Don't call gst_init() and gst_get_option_group()
3281 The latter calls the former at the appropriate time.
3283 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
3285 * gst/rtsp-server/rtsp-client.c:
3286 rtsp-client: Drop trailing \0 of RTSP DATA messages
3287 We add a trailing \0 in GstRTSPConnection to make parsing of
3288 string message bodies easier (e.g. the SDP from DESCRIBE) but
3289 for actual data this means we have to drop it or otherwise
3290 create invalid data.
3292 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
3294 * gst/rtsp-server/rtsp-stream.c:
3295 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
3296 Fixes crash when two threads access handle_new_sample() at the same
3297 time, one for RTP, one for RTCP.
3298 Otherwise, when iterating over the transports cache, it might be modified by
3299 another thread at the same time if the transports cookie has changed.
3300 https://bugzilla.gnome.org/show_bug.cgi?id=742954
3302 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
3304 * gst/rtsp-server/rtsp-stream.c:
3305 rtsp-stream: Set format=TIME on our app sources for TCP
3307 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
3309 * gst/rtsp-server/rtsp-session-pool.c:
3310 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
3311 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
3312 RFC 2326 states that session IDs may consist of alphanumeric as well as
3313 the safe characters $-_.+ -- N.B. the percent character is not allowed.
3314 Previously the session ID was URI-escaped, this meant that any character
3315 which was not alphanumeric or any of the characters +-._~ would be
3316 percent encoded. While the RFC (surprisingly) mentions that linear white
3317 space in session IDs should be URI-escaped, it does not say anything
3318 about other characters. Moreover no white space is allowed in the
3319 session ID. Finally the percent character which is the result of
3320 URI-escaping is not allowed in a session ID.
3321 So there is no reason to do any URI-escaping, and now it is removed.
3322 https://bugzilla.gnome.org/show_bug.cgi?id=742869
3324 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
3327 Automatic update of common submodule
3328 From f2c6b95 to bc76a8b
3330 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
3333 Fix 'make check' from top-level directory
3335 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
3337 * examples/test-launch.c:
3338 * examples/test-mp4.c:
3339 * examples/test-ogg.c:
3340 * examples/test-uri.c:
3341 examples: Add command-line parsing and take a 'port' argument
3342 This allows users to run multiple servers on different ports for testing.
3343 Only done for examples that actually take arguments and hence are capable of
3344 outputting different streams for each instance on each port.
3345 https://bugzilla.gnome.org/show_bug.cgi?id=742115
3347 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
3349 * gst/rtsp-server/rtsp-client.c:
3350 * gst/rtsp-server/rtsp-client.h:
3351 rtsp-client: Add a send_message default signal handler
3352 This allows subclasses to easily hook into the response sending
3353 mechanism without doing everything from a signal, which seems
3354 awkward from subclasses.
3356 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
3359 Automatic update of common submodule
3360 From ef1ffdc to f2c6b95
3362 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3366 configure: add --disable-examples switch
3367 https://bugzilla.gnome.org/show_bug.cgi?id=741678
3369 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
3371 * examples/.gitignore:
3372 * examples/Makefile.am:
3373 * examples/test-video-rtx.c:
3374 examples: add a retransmisison example implementing RFC4588
3375 Currently only SSRC-multiplexed rtx streams are supported
3377 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
3379 * gst/rtsp-server/rtsp-stream.c:
3380 rtsp-stream: Fix some minor memory leaks
3382 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
3384 * gst/rtsp-server/rtsp-media.c:
3385 rtsp-media: Some minor cleanup
3387 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
3389 * gst/rtsp-server/rtsp-stream.c:
3390 rtsp-stream: Fix compiler warnings
3391 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
3392 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
3394 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
3395 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
3398 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
3400 * docs/libs/gst-rtsp-server-sections.txt:
3401 * gst/rtsp-server/rtsp-media-factory.c:
3402 * gst/rtsp-server/rtsp-media-factory.h:
3403 * gst/rtsp-server/rtsp-media.c:
3404 * gst/rtsp-server/rtsp-media.h:
3405 * gst/rtsp-server/rtsp-sdp.c:
3406 * gst/rtsp-server/rtsp-stream.c:
3407 * gst/rtsp-server/rtsp-stream.h:
3408 media: implement ssrc-multiplexed retransmission support
3409 based off RFC 4588 and the server-rtpaux example in -good
3411 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
3413 * gst/rtsp-server/rtsp-client.c:
3414 * gst/rtsp-server/rtsp-stream-transport.c:
3415 * gst/rtsp-server/rtsp-stream.c:
3416 rtsp: Ref transports in hash table.
3417 Also ref streams for transports.
3418 This solves a crash when reciving a rtcp after teardown but before
3420 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
3422 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
3425 Automatic update of common submodule
3426 From 7bb2bce to ef1ffdc
3428 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
3430 * gst/rtsp-server/rtsp-client.c:
3431 client: refactor cleanup of cached media
3433 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
3435 * tests/check/gst/client.c:
3437 The session leak is now fixed, lets remove those FIXME comments.
3439 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
3441 * tests/check/gst/rtspserver.c:
3442 tests: Test to setup two sessions on one connection
3443 https://bugzilla.gnome.org/show_bug.cgi?id=739112
3445 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
3447 * tests/check/gst/rtspserver.c:
3448 tests: Test setup with tcp transport
3449 https://bugzilla.gnome.org/show_bug.cgi?id=739112
3451 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
3453 * gst/rtsp-server/rtsp-client.c:
3454 client: Configure transport after creating session media
3455 The default implementation of configure_client_transport() in
3456 rtsp-client uses the session media when it chooses channels for
3457 interleaved traffic.
3458 https://bugzilla.gnome.org/show_bug.cgi?id=739112
3460 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
3462 * gst/rtsp-server/rtsp-client.c:
3463 * gst/rtsp-server/rtsp-session-media.c:
3464 client: Stop caching media in client when doing setup
3465 If the media has been managed by a session media, it should not be
3466 cached in the client any longer. The GstRTSPSessionMedia object is now
3467 responsible for unpreparing the GstRTSPMedia object using
3468 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
3470 https://bugzilla.gnome.org/show_bug.cgi?id=739112
3472 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
3474 * gst/rtsp-server/rtsp-stream.c:
3475 rtsp-stream: unref srtp decoder when leaving bin
3476 https://bugzilla.gnome.org/show_bug.cgi?id=739481
3478 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
3480 * gst/rtsp-server/rtsp-client.c:
3481 rtsp-client: mikey memory leaks
3482 https://bugzilla.gnome.org/show_bug.cgi?id=739383
3484 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
3487 Automatic update of common submodule
3488 From 84d06cd to 7bb2bce
3490 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
3493 Parallelise 'make check-valgrind'
3495 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
3498 Automatic update of common submodule
3499 From a8c8939 to 84d06cd
3501 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
3504 Automatic update of common submodule
3505 From 36388a1 to a8c8939
3507 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
3509 * gst/rtsp-server/rtsp-media.c:
3510 rtsp-media: deactivate media when shutting down from paused
3511 This was only done when going directly from playing.
3512 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
3514 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
3516 * gst/rtsp-server/rtsp-client.c:
3517 * gst/rtsp-server/rtsp-context.h:
3518 rtsp-client: add stream transport to context
3519 We add the stream transport to the context so we can get the configured
3520 client stream transport in the setup request signal.
3521 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
3523 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
3525 * gst/rtsp-server/rtsp-stream.c:
3526 stream: release lock even not all transports have been removed
3527 We don't want to keep the lock even we return FALSE because not all the
3528 transports have been removed. This could lead into a deadlock.
3529 https://bugzilla.gnome.org/show_bug.cgi?id=737797
3531 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
3533 * gst/rtsp-server/rtsp-sdp.c:
3534 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
3535 These were renamed in GstRTPBasePayload in 1.0
3537 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
3539 * gst/rtsp-server/rtsp-client.c:
3540 client: set session media to NULL without the lock
3541 We need to set session medias to NULL without the client lock otherwise
3542 we can end up in a deadlock if another thread is waiting for the lock
3543 and media unprepare is also waiting for that thread to end.
3544 https://bugzilla.gnome.org/show_bug.cgi?id=737690
3546 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
3548 * gst/rtsp-server/rtsp-media.c:
3549 rtsp-media: Set state to UNPREPARING in all cases
3551 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
3553 * gst/rtsp-server/rtsp-media.c:
3554 media: set state to unpreparing when unprepare is initiated
3555 https://bugzilla.gnome.org/show_bug.cgi?id=737675
3557 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
3559 * gst/rtsp-server/rtsp-client.c:
3560 rtsp-client: Remove backlog limit while processings requests
3561 If the backlog limit is kept two cases of deadlocks may be
3562 encountered when streaming over TCP. Without the backlog
3563 limit this deadlocks can not happen, at the expence of
3565 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
3567 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
3569 * gst/rtsp-server/rtsp-client.c:
3570 rtsp-client: do not free main context before rtsp watch
3571 https://bugzilla.gnome.org/show_bug.cgi?id=737110
3573 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
3575 * tests/check/gst/rtspserver.c:
3576 tests: Extend unit test timeout to accomodate for valgrind
3577 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
3579 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
3581 * gst/rtsp-server/rtsp-client.c:
3582 * gst/rtsp-server/rtsp-session.c:
3583 * gst/rtsp-server/rtsp-stream-transport.c:
3584 rtsp-*: Treat sending packets to clients as keepalive
3585 As long as gst-rtsp-server can successfully send RTP/RTCP data to
3586 clients then the client must be reading. This change makes the server
3587 timeout the connection if the client stops reading.
3588 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
3590 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
3592 * gst/rtsp-server/rtsp-client.c:
3593 rtsp-client: Allow backlog to grow while expiring session
3594 Allow the send backlog in the RTSP watch to grow to unlimited size while
3595 attempting to bring the media pipeline to NULL due to a session
3596 expiring. Without this change the appsink element cannot change state
3597 because it is blocked while rendering data in the new_sample callback.
3598 This callback will block until it has successfully put the data into the
3599 send backlog. There is a chance that the send backlog is full at this
3600 point which means that the callback may block for a long time, possibly
3601 forever. Therefore the media pipeline may also be prevented from
3602 changing state for a long time.
3603 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
3605 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
3607 * gst/rtsp-server/rtsp-client.c:
3608 rtsp-client: Make old compilers happy
3609 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
3610 Just in case that guint8 doesn't fit in a pointer. Just in case ...
3612 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
3614 * gst/rtsp-server/rtsp-client.c:
3615 client: raise the backlog limits before pausing
3616 We need to raise the backlog limits before pausing the pipeline or else
3617 the appsink might be blocking in the render method in wait_backlog() and
3618 we would deadlock waiting for paused.
3619 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
3621 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
3623 * gst/rtsp-server/rtsp-client.c:
3624 client: make define for the WATCH_BACKLOG
3625 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
3627 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
3629 * gst/rtsp-server/rtsp-client.c:
3630 client: simplify session transport handling
3631 link/unlink of the transport in a session was done to keep track of all
3632 TCP transports and to send RTP/RTCP data to the streams. We can simplify
3633 that by putting all the TCP transports in a hashtable indexed with the
3635 We also don't need to link/unlink the transports when we pause/resume
3636 the streams. The same effect is already achieved when we pause/play the
3637 media. Indeed, when we pause the media, the transport is removed from
3638 the media and the callbacks will not be called anymore.
3639 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
3641 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
3643 * gst/rtsp-server/rtsp-stream-transport.c:
3644 * gst/rtsp-server/rtsp-stream-transport.h:
3645 stream-transport: make method to handle received data
3646 Make a method to handle the data received on a channel. It sends the
3647 data to the stream of the transport on the RTP or RTCP pads based on
3650 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
3652 * examples/test-mp4.c:
3653 test: add example of dumping RTCP reports
3655 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
3657 * gst/rtsp-server/rtsp-media.c:
3658 * gst/rtsp-server/rtsp-stream.c:
3659 * gst/rtsp-server/rtsp-stream.h:
3660 rtsp-media: Make sure that sequence numbers are monotonic after pause
3661 The sequence number is not monotonic for RTP packets after pause. The
3662 reason is basepayloader generates a randon sequence number when the
3663 pipeline goes from ready to pause. With this fix generation of sequence
3664 number will be monotonic when going from pause to play request.
3665 https://bugzilla.gnome.org/show_bug.cgi?id=736017
3667 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
3669 * gst/rtsp-server/rtsp-client.c:
3670 rtsp-client: Protect saved clients watch with a mutex
3671 Fixes a crash when close() is called while merging clients
3672 in handle_tunnel(). In that case close() would destroy the
3673 watch while it is still being used in handle_tunnel().
3674 https://bugzilla.gnome.org/show_bug.cgi?id=735570
3676 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
3678 * gst/rtsp-server/rtsp-stream.c:
3679 rtsp-stream: Remove the multicast group udp sources when removing from the bin
3681 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
3683 * gst/rtsp-server/rtsp-media.c:
3684 * gst/rtsp-server/rtsp-stream.c:
3685 * gst/rtsp-server/rtsp-stream.h:
3686 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
3687 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
3688 seeking and will always continue counting the time. This leads to
3689 the NPT after a backwards seek to be something completely different
3690 to the actual seek position.
3691 https://bugzilla.gnome.org/show_bug.cgi?id=732644
3693 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
3695 * examples/test-appsrc.c:
3696 examples: fix another reference leak
3697 gst_rtsp_media_get_element() returns a new ref.
3699 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3701 * examples/test-appsrc.c:
3702 examples: unref element after usage
3703 gst_bin_get_by_name_recurse_up() returns an element
3704 reference that must be unreffed after usage.
3705 https://bugzilla.gnome.org/show_bug.cgi?id=734546
3707 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
3709 * gst/rtsp-server/rtsp-media.c:
3710 signals: Fix copy-pasto in target-state signal offset
3712 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
3716 Makefile: Add usage of build-checks step
3717 Allows building checks without running them
3719 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
3721 * gst/rtsp-server/rtsp-stream.c:
3722 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
3723 When a UDP multicast transport is used it is expected that the server listens
3724 for RTP and RTCP packets on the multicast group with the corresponding port.
3725 Without this we will never get RTCP packets from clients in multicast mode.
3726 https://bugzilla.gnome.org/show_bug.cgi?id=732238
3728 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
3733 === release 1.4.0 ===
3735 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
3741 * gst-rtsp-server.doap:
3744 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
3746 * gst/rtsp-server/rtsp-media.h:
3747 media: correct misspelled words in description
3748 https://bugzilla.gnome.org/show_bug.cgi?id=733244
3750 === release 1.3.91 ===
3752 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
3758 * gst-rtsp-server.doap:
3761 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
3763 * docs/libs/gst-rtsp-server-sections.txt:
3766 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
3768 * gst/rtsp-server/rtsp-server.c:
3769 server: implement client REMOVE filter
3771 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
3773 * gst/rtsp-server/rtsp-client.c:
3774 * gst/rtsp-server/rtsp-client.h:
3775 client: expose _close() method
3776 Expose a previously internal close method to close the client
3779 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
3781 * gst/rtsp-server/rtsp-session-pool.c:
3782 session-pool: signal session-removed outside of the lock
3783 Release the lock before emiting the session-removed signal.
3785 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
3787 * gst/rtsp-server/rtsp-client.c:
3788 * gst/rtsp-server/rtsp-server.c:
3789 * gst/rtsp-server/rtsp-session-pool.c:
3790 * gst/rtsp-server/rtsp-session.c:
3791 * gst/rtsp-server/rtsp-stream.c:
3792 filter: Release lock in filter functions
3793 Release the object lock before calling the filter functions. We need to
3794 keep a cookie to detect when the list changed during the filter
3795 callback. We also keep a hashtable to make sure we only call the filter
3796 function once for each object in case of concurrent modification.
3797 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
3799 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
3801 * gst/rtsp-server/rtsp-client.c:
3802 client: check if watch is set in handle_teardown()
3803 The unit tests run without a watch
3805 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
3807 * tests/check/gst/client.c:
3808 client tests: send teardown to cleanup session
3810 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
3812 * tests/check/gst/rtspserver.c:
3813 server tests: send teardown to cleanup session
3815 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
3817 * gst/rtsp-server/rtsp-client.c:
3818 client: keep ref to client for the session removed handler
3819 This extra ref will be dropped when all client sessions have been
3820 removed. A session is removed when a client sends teardown, closes its
3821 endpoint of the TCP connection or the sessions expires.
3822 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
3824 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
3826 * gst/rtsp-server/rtsp-client.c:
3827 * gst/rtsp-server/rtsp-session.c:
3828 * tests/check/gst/client.c:
3829 client: manage media in session as a last step
3830 Once we manage a media in a session, we can't unmanage it anymore
3831 without destroying it. Therefore, first check everything before we
3832 manage the media, otherwise if something is wrong we have no way to
3834 If we created a new session and something went wrong, remove the session
3835 again. Fixes a leak in the unit test.
3837 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
3839 * examples/test-mp4.c:
3840 * examples/test-ogg.c:
3841 examples: print 'stream ready at url' for mp4 and ogg example
3843 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
3845 * gst/rtsp-server/rtsp-client.c:
3846 * gst/rtsp-server/rtsp-sdp.c:
3847 rtsp: fix for MIKEY api change
3849 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
3851 * gst/rtsp-server/rtsp-client.c:
3852 client: free watch context only once
3853 The watch context is freed when the source is destroyed. Avoids
3854 a CRITICAL when we try to unref the context twice.
3856 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
3858 * gst/rtsp-server/rtsp-client.c:
3861 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
3863 * gst/rtsp-server/rtsp-client.c:
3864 client: protect sessions with lock
3865 Protect the list of sessions with the lock.
3866 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
3868 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
3870 * gst/rtsp-server/rtsp-client.c:
3871 Client: keep a ref to the session
3872 Don't just keep a weak ref to the session objects but use a hard ref. We
3873 will be notified when a session is removed from the pool (expired) with
3874 the new session-removed signal.
3875 Don't automatically close the RTSP connection when all the sessions of
3876 a client are removed, a client can continue to operate and it can create
3877 a new session if it wants. If you want to remove the client from the
3878 server, you have to use gst_rtsp_server_client_filter() now.
3879 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
3880 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
3882 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
3884 * gst/rtsp-server/rtsp-session-pool.c:
3885 * gst/rtsp-server/rtsp-session-pool.h:
3886 session-pool: add session-removed signal
3887 Add a signal to be notified when a session is removed from the pool.
3889 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
3891 * gst/rtsp-server/Makefile.am:
3892 * gst/rtsp-server/rtsp-server.h:
3893 Make rtsp-server.h a single-include header, use it for G-I
3894 https://bugzilla.gnome.org/show_bug.cgi?id=732411
3896 === release 1.3.90 ===
3898 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
3904 * gst-rtsp-server.doap:
3907 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
3909 * gst/rtsp-server/rtsp-stream.c:
3910 stream: crypto can be NULL
3912 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
3914 * gst/rtsp-server/rtsp-client.c:
3915 * gst/rtsp-server/rtsp-media.c:
3916 * gst/rtsp-server/rtsp-mount-points.c:
3917 introspection: add missing allow-none annotations
3918 https://bugzilla.gnome.org/show_bug.cgi?id=730952
3920 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
3922 * gst/rtsp-server/rtsp-address-pool.c:
3923 * gst/rtsp-server/rtsp-media.c:
3924 * gst/rtsp-server/rtsp-session-media.c:
3925 * gst/rtsp-server/rtsp-session-pool.c:
3926 * gst/rtsp-server/rtsp-stream-transport.c:
3927 * gst/rtsp-server/rtsp-stream.c:
3928 * gst/rtsp-server/rtsp-token.c:
3929 introspection: add (nullable) annotations to return values
3930 https://bugzilla.gnome.org/show_bug.cgi?id=730952
3932 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
3934 * gst/rtsp-server/rtsp-client.c:
3935 * gst/rtsp-server/rtsp-stream.c:
3936 gi: improve annotations
3937 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
3939 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
3941 * gst/rtsp-server/rtsp-client.c:
3942 * gst/rtsp-server/rtsp-media-factory.c:
3943 * gst/rtsp-server/rtsp-media.c:
3944 * gst/rtsp-server/rtsp-server.c:
3945 signals: use generic marshal function
3946 Use the generic C marshal function.
3947 Use more explicit type instead of G_TYPE_POINTER
3949 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
3951 * gst/rtsp-server/rtsp-context.h:
3952 context: add type macro
3954 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
3956 * gst/rtsp-server/rtsp-client.c:
3957 * gst/rtsp-server/rtsp-sdp.c:
3958 * gst/rtsp-server/rtsp-sdp.h:
3959 sdp: hide key length defines
3960 They don't have a namespace.
3962 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
3967 === release 1.3.3 ===
3969 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
3975 * gst-rtsp-server.doap:
3978 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
3980 * gst/rtsp-server/rtsp-client.c:
3981 * gst/rtsp-server/rtsp-sdp.c:
3982 * gst/rtsp-server/rtsp-sdp.h:
3983 mikey: add different key length parameters
3984 Add encryption and authentication key length parameters to MIKEY. For
3985 the encoders, the key lengths are obtained from the cipher and auth
3986 algorithms set in the caps. For the decoders, they are obtained while
3987 parsing the key management from the client.
3988 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
3990 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
3992 * tests/check/gst/stream.c:
3993 stream tests: Make sure we get right multicast address from stream
3994 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
3996 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
3998 * gst/rtsp-server/rtsp-client.c:
3999 client: ref the context until rtsp watch is alive
4000 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
4002 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
4004 * gst/rtsp-server/rtsp-client.c:
4005 client: Destroy the rtsp watch after connection close
4007 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
4009 * gst/rtsp-server/rtsp-media.c:
4010 media: fix confusing comment
4012 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
4014 * gst/rtsp-server/rtsp-session.c:
4015 rtsp-session: Timeout in header.
4016 Adding the possbilty to always have timout in header.
4017 This is configurabe with setting "timeout-always-visible".
4018 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
4020 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
4025 === release 1.3.2 ===
4027 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
4034 * gst-rtsp-server.doap:
4037 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
4040 Automatic update of common submodule
4041 From 211fa5f to 1f5d3c3
4043 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
4045 * gst/rtsp-server/rtsp-client.c:
4046 client: store TCP ports in transport
4047 Store the TCP ports in the transport when we are doing RTSP over TCP.
4048 This way, we can easily get to the ports from the transport.
4049 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
4051 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4053 * gst/rtsp-server/rtsp-stream.c:
4054 stream: add signals for new RTP/RTCP encoders
4055 New signals to allow the user to configure the dynamically created
4057 https://bugzilla.gnome.org/show_bug.cgi?id=730228
4059 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
4061 * gst/rtsp-server/rtsp-media.c:
4062 * gst/rtsp-server/rtsp-media.h:
4063 media: Make suspend()/unsuspend() virtual
4064 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
4066 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4068 * gst/rtsp-server/rtsp-client.c:
4069 client: fix send-message signal marshaller
4070 Use generic marshalling for the send-message signal. It has
4071 two POINTER arguments, not just one.
4072 https://bugzilla.gnome.org/show_bug.cgi?id=729900
4074 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
4076 * tests/check/gst/media.c:
4077 tests: add and remove pads only once
4078 In this test we simulate a dynamic pad by watching the caps event.
4079 Because of renegotiation in the base payloader now, this caps is sent
4080 multiple times but we can only deal with 1 invocation, use a variable to
4081 only 'add and remove' the pad once.
4083 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
4085 * tests/check/gst/rtspserver.c:
4086 tests: add unit test for correct handling of Require headers
4087 https://bugzilla.gnome.org/show_bug.cgi?id=729426
4089 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
4091 * gst/rtsp-server/rtsp-client.c:
4092 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
4093 Servers must handle Require headers and must report a failure
4094 if they don't handle any of the Required options, see RFC 2326,
4095 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
4096 https://bugzilla.gnome.org/show_bug.cgi?id=729426
4098 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
4103 === release 1.3.1 ===
4105 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
4111 * gst-rtsp-server.doap:
4114 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
4117 Automatic update of common submodule
4118 From bcb1518 to 211fa5f
4120 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
4125 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
4127 * tests/check/gst/sessionmedia.c:
4128 tests: fix memory leak in sessionmedia unit test
4130 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
4132 * gst/rtsp-server/rtsp-client.c:
4133 client: emit a signal before sending a message
4134 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
4136 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
4138 * gst/rtsp-server/rtsp-client.c:
4139 client: pass context to send_message
4140 Pass the current context to send_message, we will need it later.
4142 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
4144 * gst/rtsp-server/rtsp-client.c:
4145 client: fix typo in comment
4147 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
4149 * gst/rtsp-server/rtsp-media.c:
4150 media: Do not stop thread twice if default_prepare() fails
4152 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
4154 * gst/rtsp-server/rtsp-client.c:
4155 client: set the watch to flushing before going to NULL
4156 First set the watch to flushing so that we unblock any current and
4157 future attempt to send data on the watch, Then set the pipeline to
4159 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
4161 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
4163 * gst/rtsp-server/rtsp-session-pool.c:
4164 * tests/check/gst/sessionpool.c:
4165 rtsp-session-pool: Fixes annotation
4166 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
4167 in the sessionpool test.
4168 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
4170 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
4172 * gst/rtsp-server/rtsp-media.c:
4173 * gst/rtsp-server/rtsp-media.h:
4174 media: make media_prepare virtual
4175 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
4177 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
4179 * gst/rtsp-server/rtsp-media.c:
4180 * tests/check/gst/media.c:
4181 media: stop the thread in more error cases
4183 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
4185 * gst/rtsp-server/rtsp-media.c:
4186 * tests/check/gst/media.c:
4187 media: allow NULL as the thread
4188 Use the default context whan passing a NULL thread.
4190 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
4192 * gst/rtsp-server/rtsp-client.c:
4193 rtsp-client: indent cleanup
4194 Coverity was moaning about unreachable code, and I think it was just
4195 confused by { being before the label. We'll see if it pops up again.
4198 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
4200 * gst/rtsp-server/rtsp-client.c:
4201 * gst/rtsp-server/rtsp-media.c:
4202 client: Add drop-backlog property
4203 When we have too many messages queued for a client (currently hardcoded
4204 to 100) we overflow and drop the messages. Add a drop-backlog property
4205 to control this behaviour. Setting this property to FALSE will retry
4206 to send the messages to the client by waiting for more room in the
4208 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
4210 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
4212 * gst/rtsp-server/rtsp-client.c:
4213 client: support for POST before GET when setting up a tunnel
4215 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
4217 * gst/rtsp-server/rtsp-client.c:
4218 client: remove watch of the second client after http tunnel setup
4219 The second client will be freed after the HTTP tunnel has been set up.
4220 Make sure it's RTSP watch is never dispatched again.
4221 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
4223 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
4225 * gst/rtsp-server/rtsp-media.c:
4226 * tests/check/gst/media.c:
4227 media: Make media_prepare() fail if port allocation fails
4228 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
4230 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
4232 * tests/check/gst/media.c:
4233 media test: cleanup the thread pool in tests
4235 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
4237 * gst/rtsp-server/rtsp-media.c:
4238 * tests/check/gst/media.c:
4239 rtsp-media: Unblock blocked streams in unprepare
4240 The streams will be blocked when a live media is prepared.
4241 The streams should be unblocked in gst_rtsp_media_unprepare.
4242 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
4244 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
4246 * gst/rtsp-server/rtsp-media.c:
4247 media: release the state lock when going to NULL
4248 Set our state to UNPREPARING and release the state-lock before
4249 setting the pipeline to the NULL state. This way, any pad-added
4250 callback will be able to take the state-lock and check that we are now
4251 unpreparing instead of deadlocking.
4252 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
4254 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
4256 * gst/rtsp-server/rtsp-media.c:
4257 media: protect status with lock
4258 Make sure we only update the status with the lock.
4260 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
4262 * gst/rtsp-server/rtsp-client.c:
4263 * gst/rtsp-server/rtsp-sdp.c:
4264 rtsp: update for MIKEY API changes
4266 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
4268 * gst/rtsp-server/rtsp-client.c:
4269 client: parse the mikey response from the client
4270 Parse the mikey response from the client and update the policy for
4273 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
4275 * gst/rtsp-server/rtsp-stream.c:
4276 * gst/rtsp-server/rtsp-stream.h:
4277 stream: add method to set crypto info
4278 Make a method to configure the crypto information of a stream.
4279 Set udpsrc in READY instead of PAUSED so that we can configure caps
4282 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
4284 * gst/rtsp-server/rtsp-client.c:
4285 client: cleanup error paths
4287 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
4289 * gst/rtsp-server/rtsp-media.c:
4292 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
4294 * examples/test-video.c:
4295 test: enable SRTP only on RTSPS
4296 We only want to enable SRTP when doing rtsp over TLS so that we can
4297 exchange the keys in a secure way.
4299 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
4301 * examples/test-video.c:
4302 test: print an error on failure
4304 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
4307 * examples/test-video.c:
4308 * gst/rtsp-server/rtsp-sdp.c:
4309 * gst/rtsp-server/rtsp-stream.c:
4310 * tests/check/Makefile.am:
4311 stream: add SRTP support
4312 Install srtp encoder and decoder elements in rtpbin
4315 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4317 * tests/check/Makefile.am:
4318 * tests/check/gst/sessionpool.c:
4319 tests: Add unit tests for sessionpool
4320 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
4322 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4324 * tests/check/gst/threadpool.c:
4325 tests: Improve code coverage of rtsp-threadpool tests
4326 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
4328 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4330 * tests/check/gst/sessionmedia.c:
4331 tests: Improve code coverage for rtsp-session-media
4332 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
4334 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4336 gobject-introspection: Add annotations to support language bindings
4337 In addition a few cosmetic changes:
4338 * Adjust the order of arguments
4339 * Fix typo: occured -> occurred
4340 * Fix indentation after Return:-clauses
4341 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
4343 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4345 * gst/rtsp-server/rtsp-stream.c:
4346 rtsp-stream: Don't mix IPv4 and IPv6 addresses
4347 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
4349 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
4351 * gst/rtsp-server/rtsp-stream.c:
4352 stream: take caps after the session manager
4353 Take the caps for the SDP after they leave the rtpbin so that we can
4354 also get the properties added by rtpbin elements.
4356 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
4358 * gst/rtsp-server/rtsp-stream.c:
4359 stream: release lock while pushing out packets
4360 Keep a cache of the transports and use this to iterate the transport
4361 while pushing packets. This allows us to release the lock early.
4362 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
4364 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
4366 * gst/rtsp-server/rtsp-client.c:
4367 * gst/rtsp-server/rtsp-client.h:
4368 rtsp-client: vmethod for modifying tunnel GET response
4369 Add a vmethod tunnel_http_response where the response to the HTTP GET
4370 for tunneled connections can be modified.
4371 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
4373 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
4375 * gst/rtsp-server/rtsp-sdp.c:
4376 sdp: make 1 media line per profile
4377 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
4378 line in the SDP for each profile. The client is then supposed to pick
4379 one of the profiles in the SETUP request. Because the m= lines have the
4380 same pt, the client also knows that only 1 option is possible.
4382 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
4384 * gst/rtsp-server/rtsp-media-factory.c:
4385 * gst/rtsp-server/rtsp-media-factory.h:
4386 * gst/rtsp-server/rtsp-media.c:
4387 factory: add profile property and pass to media and streams
4389 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
4391 * examples/test-multicast.c:
4392 * gst/rtsp-server/rtsp-sdp.c:
4393 sdp: pass multicast connection for multicast-only stream
4394 Pass the multicast address of the stream in the connection info in the
4395 SDP so that clients try a multicast connection first.
4396 Only allow multicast connections in the test-multicast example. Also
4397 increase the TTL a little.
4399 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4402 .gitignore: Ignore gcov intermediate files
4403 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
4405 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
4407 * gst/rtsp-server/rtsp-stream.c:
4408 stream: release some locks in error cases
4410 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4412 docs: Enable and fix gtk-doc warnings
4413 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
4414 * addresspool/mediafactory: Add missing annotation colon
4415 * stream: Annotate return value
4416 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
4418 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
4421 Automatic update of common submodule
4422 From fe1672e to bcb1518
4424 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
4427 Automatic update of common submodule
4428 From 1a07da9 to fe1672e
4430 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
4432 * examples/Makefile.am:
4433 examples: use LDADD for libs instead of LDFLAGS
4435 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
4438 configure: make sure releases are in .doap file
4440 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
4442 * examples/test-cgroups.c:
4443 examples: test-cgroups: don't put code with side effects into g_assert()
4444 The g_assert() might get compiled out with the right
4445 compiler/preprocessor flags.
4447 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
4449 * examples/.gitignore:
4450 examples: add cgroup test binary to .gitignore
4452 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
4454 * examples/test-cgroups.c:
4455 examples: fix cgroup test build
4456 Fixes build failure caused by compiler warning:
4457 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
4459 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
4462 .gitignore: ignore temp files created in the course of 'make check'
4464 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
4466 * gst/rtsp-server/rtsp-media.c:
4467 rtsp-media: don't loose frames handling new PLAY request
4468 If client supplied a range check if the range specifies the start point.
4469 If not, then do an accurate seek to the current position. If a start
4470 point was specified do do a key unit seek to make sure the streaming
4471 starts with decodeable frames.
4472 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
4474 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
4476 * gst/rtsp-server/rtsp-media.c:
4477 Revert "media: only flush when setting a new start position"
4478 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
4479 We need to do the flush in all cases, demuxer block currently for
4482 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
4484 * gst/rtsp-server/rtsp-media.c:
4485 media: only flush when setting a new start position
4486 Only flush the pipeline when we change the start position with
4488 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
4490 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
4492 * gst/rtsp-server/rtsp-stream.c:
4493 stream: set ttl-mc before adding the socket
4494 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
4495 never be set on socket.
4496 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
4498 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
4500 * gst/rtsp-server/rtsp-media.c:
4501 media: stop thread if media is already prepared
4502 in gst_rtsp_media_prepare() the thread is not used if media is already
4503 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
4505 https://bugzilla.gnome.org/show_bug.cgi?id=724182
4507 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
4510 build: Ship gst-rtsp-server.doap file
4512 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
4514 * tests/check/gst/rtspserver.c:
4515 tests: Fix another compiler warning with gcc
4517 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
4519 * gst/rtsp-server/rtsp-client.c:
4520 * gst/rtsp-server/rtsp-mount-points.c:
4521 * gst/rtsp-server/rtsp-stream.c:
4522 * tests/check/gst/client.c:
4523 rtsp-server: Fix lots of compiler warnings with clang
4525 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
4528 * gst-rtsp-server.doap:
4529 * tests/Makefile.am:
4530 configure: Synchronise with the configure scripts of the other modules
4532 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
4535 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
4537 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
4539 * gst/rtsp-server/rtsp-media.c:
4540 * gst/rtsp-server/rtsp-stream.c:
4541 Revert "rtsp-server: support build against last stable release"
4542 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
4543 Let us require 1.2.3 now, which is going to be released in a few
4546 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
4548 * gst/rtsp-server/rtsp-session-media.c:
4549 * gst/rtsp-server/rtsp-stream-transport.c:
4550 session: improve RTP-Info
4551 Ignore streams that can't generate RTP-Info instead of failing.
4552 Don't return the empty string when all streams are unconfigured but
4553 return NULL so that we don't generate and empty RTP-Info header.
4554 Improve docs a little.
4556 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
4558 * gst/rtsp-server/rtsp-session-media.c:
4559 Don't free rtpinfo GString when it is NULL
4560 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
4562 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
4564 * gst/rtsp-server/rtsp-media.c:
4565 media: only set keyframe flag when modifying start
4566 Only set the keyframe flag when we modify the start position. The
4567 keyframe flag should probably be ignored when no change is requested but
4568 until we can claim this is all documented properly and all demuxer
4569 implement this, avoid setting the flag.
4570 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
4572 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
4574 * gst/rtsp-server/rtsp-thread-pool.c:
4575 thread-pool: Unref source after mainloop has quit to avoid races in GLib
4576 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
4578 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
4580 * gst/rtsp-server/rtsp-stream.c:
4581 stream: handle NULL seqnum and rtptime arguments
4583 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
4585 * gst/rtsp-server/rtsp-thread-pool.c:
4586 * tests/check/gst/threadpool.c:
4587 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
4588 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
4590 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
4592 * gst/rtsp-server/rtsp-stream.c:
4593 stream: add fallback for missing stats property
4594 Use a fallback when the payloader does not have a stats property
4595 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
4597 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
4600 Automatic update of common submodule
4601 From f7bc1c3 to 1a07da9
4603 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
4605 * gst/rtsp-server/rtsp-stream.c:
4606 stream: don't leak stats structure
4607 Don't leak the stats structure and deal with NULL stats.
4609 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
4611 * gst/rtsp-server/rtsp-stream.c:
4612 stream: Get rtpinfo properties atomically from payloader
4613 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
4615 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
4617 * gst/rtsp-server/rtsp-media.c:
4618 media: refactor state change functions and signals
4619 Make functions to set the target state and the pipeline state and emit
4620 the signals from those functions.
4622 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
4624 * gst/rtsp-server/rtsp-media.c:
4625 * gst/rtsp-server/rtsp-media.h:
4626 media: add signal to notify of pending state changes
4628 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
4630 * gst/rtsp-server/rtsp-media.c:
4631 * gst/rtsp-server/rtsp-stream.c:
4632 rtsp-server: support build against last stable release
4633 Until 1.2.3 is out with the new get_type function and we
4636 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
4638 * gst/rtsp-server/rtsp-stream.c:
4639 stream: fix compilation
4641 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
4643 * gst/rtsp-server/rtsp-media.c:
4644 * gst/rtsp-server/rtsp-media.h:
4645 * gst/rtsp-server/rtsp-stream.c:
4646 * gst/rtsp-server/rtsp-stream.h:
4647 stream: add property to configure profiles
4649 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
4651 * gst/rtsp-server/rtsp-client.c:
4652 client: let stream check supported transport
4653 Delegate the check if a transport is allowed to the stream.
4654 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
4656 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
4658 * gst/rtsp-server/rtsp-stream.c:
4659 * gst/rtsp-server/rtsp-stream.h:
4660 stream: add method to check supported transport
4661 Add a method to check if a transport is supported
4663 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
4666 configure.ac: Only check for gstreamer-check, not check
4667 We include check in gstreamer-check since quite some time now.
4669 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
4671 * gst/rtsp-server/rtsp-session-media.c:
4672 * gst/rtsp-server/rtsp-stream-transport.c:
4673 * gst/rtsp-server/rtsp-stream.c:
4674 * gst/rtsp-server/rtsp-stream.h:
4675 stream: return clock-rate from get_rtpinfo
4676 And use it to correct the rtptime to the requested start-time.
4677 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
4679 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
4681 * gst/rtsp-server/rtsp-session-media.c:
4682 * gst/rtsp-server/rtsp-stream-transport.c:
4683 * gst/rtsp-server/rtsp-stream-transport.h:
4684 session-media: calculate start-time
4686 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
4688 * gst/rtsp-server/rtsp-stream-transport.c:
4689 * gst/rtsp-server/rtsp-stream.c:
4690 * gst/rtsp-server/rtsp-stream.h:
4691 stream: also return the running-time
4692 Return the running-time in the rtpinfo as well.
4694 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
4696 * gst/rtsp-server/rtsp-client.c:
4697 * gst/rtsp-server/rtsp-session-media.c:
4698 * gst/rtsp-server/rtsp-session-media.h:
4699 * gst/rtsp-server/rtsp-stream-transport.c:
4700 * gst/rtsp-server/rtsp-stream-transport.h:
4701 session-media: let the session-media make the RTPInfo
4702 Add method to create the RTPInfo for a stream-transport.
4703 Add method to create the RTPInfo for all stream-transports in a
4705 Use the session-media RTPInfo code in client. This allows us to refactor
4706 another method to link the TCP callbacks.
4708 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
4710 mount-points: sort sequence before g_sequence_lookup
4711 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
4712 sort sequence if dirty, otherwise lookup will fail.
4713 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
4715 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
4718 configure: rename package from gst-rtsp to gst-rtsp-server
4719 To match git module name and avoid confusion with the
4720 rtsp lib in gst-plugins-base and rtsp plugin in -good.
4722 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
4725 configure: bump core/base/good requirement to 1.2.0
4726 Bump to released stable version and make implicit
4727 requirements explicit.
4729 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
4734 Fix broken gettext setup which is not used anyway
4736 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
4739 Automatic update of common submodule
4740 From dbedaa0 to d48bed3
4742 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
4744 * gst/rtsp-server/rtsp-client.c:
4745 * gst/rtsp-server/rtsp-media.c:
4746 * gst/rtsp-server/rtsp-media.h:
4747 media: add setup_sdp vmethod
4748 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
4749 gst_rtsp_media_setup_sdp.
4750 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
4752 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
4754 * gst/rtsp-server/rtsp-stream.c:
4755 rtsp-stream: Check return value of sscanf
4756 streamid is only valid if sscanf matched something.
4758 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
4760 * gst/rtsp-server/rtsp-client.c:
4761 rtsp-client: Fix iteration
4762 Wouldn't even enter the code block otherwise (i++ was used as the check
4763 and not the postfix).
4765 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
4767 * gst/rtsp-server/rtsp-client.c:
4768 * gst/rtsp-server/rtsp-client.h:
4769 client: add vmethod to configure media and streams
4770 Implement a vmethod that can be used to configure the media and the
4771 streams based on the current context. Handle the blocksize handling in
4772 the default handler.
4773 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
4775 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
4778 Make git ignore more unit test binaries
4780 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
4782 * gst/rtsp-server/rtsp-address-pool.h:
4783 * gst/rtsp-server/rtsp-auth.h:
4784 * gst/rtsp-server/rtsp-client.h:
4785 * gst/rtsp-server/rtsp-context.h:
4786 * gst/rtsp-server/rtsp-media-factory-uri.h:
4787 * gst/rtsp-server/rtsp-media-factory.h:
4788 * gst/rtsp-server/rtsp-media.h:
4789 * gst/rtsp-server/rtsp-mount-points.h:
4790 * gst/rtsp-server/rtsp-server.h:
4791 * gst/rtsp-server/rtsp-session-media.h:
4792 * gst/rtsp-server/rtsp-session-pool.h:
4793 * gst/rtsp-server/rtsp-session.h:
4794 * gst/rtsp-server/rtsp-stream-transport.h:
4795 * gst/rtsp-server/rtsp-stream.h:
4796 * gst/rtsp-server/rtsp-thread-pool.h:
4797 * gst/rtsp-server/rtsp-token.h:
4798 rtsp-server: add padding to many public structures
4799 Not mini objects though, since they are not subclassable
4800 anyway, nor kept on the stack or inlined in a structure.
4802 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
4804 media: add new create_rtpbin vmethod
4805 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
4806 https://bugzilla.gnome.org/show_bug.cgi?id=719734
4808 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
4810 * tests/check/gst/media.c:
4811 tests: fix memory leak, free test's thread pool
4812 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
4814 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
4816 * gst/rtsp-server/rtsp-stream-transport.c:
4817 stream-transport: free url in finalize
4819 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
4821 * gst/rtsp-server/rtsp-media.c:
4822 media: also do state change in suspended state
4824 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
4826 * gst/rtsp-server/rtsp-client.c:
4827 * gst/rtsp-server/rtsp-media.c:
4828 media: also handle prepare and range in suspended state
4829 When we are suspended, we are already prepared.
4830 We can get the range in the suspended state.
4832 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
4834 * tests/check/Makefile.am:
4835 * tests/check/gst/sessionmedia.c:
4836 check: add test for uri in setup
4837 Added unit tests for the new functionality in GstRTSPStreamTransport.
4838 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
4840 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
4842 * gst/rtsp-server/rtsp-client.c:
4843 client: store setup uri and use in PLAY response
4844 Store the uri used when doing the setup and use that in the PLAY
4846 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
4848 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
4850 * gst/rtsp-server/rtsp-stream-transport.c:
4851 * gst/rtsp-server/rtsp-stream-transport.h:
4852 stream-transport: add method to get/set url
4854 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
4856 * gst/rtsp-server/rtsp-client.c:
4857 client: suspend after SDP and unsuspend before PLAYING
4858 Based on patches by Ognyan Tonchev <ognyan@axis.com>
4859 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
4861 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
4863 * gst/rtsp-server/rtsp-media-factory.c:
4864 * gst/rtsp-server/rtsp-media-factory.h:
4865 * gst/rtsp-server/rtsp-media.c:
4866 * gst/rtsp-server/rtsp-media.h:
4867 * gst/rtsp-server/rtsp-session-media.c:
4868 * gst/rtsp-server/rtsp-session.c:
4869 * tests/check/gst/media.c:
4870 * tests/check/gst/mediafactory.c:
4871 media: add suspend modes
4872 Add support for different suspend modes. The stream is suspended right after
4873 producing the SDP and after PAUSE. Different suspend modes are available that
4874 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
4875 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
4876 state and RESET will bring the pipeline to the NULL state.
4877 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
4878 this means that the pipeline needs to be prerolled again.
4879 Base on patches by Ognyan Tonchev <ognyan@axis.com>
4880 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
4882 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
4884 * gst/rtsp-server/rtsp-media.c:
4885 media: start live streams in blocked state
4886 Start live streams in the blocked state and make them preroll using the
4887 messages. This ensure that no data is played by the sink until we explicitly
4888 unblock the stream right before going to PLAYING.
4889 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
4891 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
4893 * gst/rtsp-server/rtsp-media.c:
4894 media: refactor starting and waiting for preroll
4895 Based on patches from Ognyan Tonchev <ognyan@axis.com>
4896 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
4898 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
4900 * gst/rtsp-server/rtsp-stream.c:
4901 * gst/rtsp-server/rtsp-stream.h:
4902 stream: add API to block streams
4903 Add an API to block on the streams and make it post a message.
4904 Based on patch by Ognyan Tonchev <ognyan@axis.com>
4905 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
4907 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
4909 * docs/libs/Makefile.am:
4910 docs: Specify the override file
4911 Even if it's empty (for now) it avoids make distcheck complaining
4913 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
4915 * gst/rtsp-server/rtsp-media.c:
4916 media: move default implementations to where they are used
4918 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
4920 * gst/rtsp-server/rtsp-media.c:
4921 media: take the right lock in gst_rtsp_media_set_pipeline_state()
4922 We need to take the state_lock when calling this method.
4924 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
4926 * gst/rtsp-server/rtsp-media.c:
4927 media: handle add-added on non-bins too
4928 Handle dynamic payloaders that are not bins, as used in the unit-test.
4930 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4932 * gst/rtsp-server/rtsp-media-factory.c:
4933 * gst/rtsp-server/rtsp-media-factory.h:
4934 * gst/rtsp-server/rtsp-media.c:
4935 rtsp-media/-factory: Fix request pad name comments
4936 These must be escaped for gtk-doc to parse the comments without warnings.
4938 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
4940 rtsp-media: remove transports if media is in error status
4941 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
4942 trying to change to GST_STATE_NULL and media is in error status, we
4943 remove all transports.
4944 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
4946 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
4948 * gst/rtsp-server/rtsp-media.c:
4949 rtsp-media: use element metadata to find payloader
4950 Use the element metadata to find the payloader instead of checking
4952 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
4954 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
4956 rtsp-stream: add getter for payload type
4957 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
4958 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
4959 element and create the stream with this one instead of the dynpay%d
4961 https://bugzilla.gnome.org/show_bug.cgi?id=712396
4963 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4965 * gst/rtsp-server/rtsp-client.c:
4966 * gst/rtsp-server/rtsp-context.h:
4967 * gst/rtsp-server/rtsp-media.c:
4968 * gst/rtsp-server/rtsp-mount-points.c:
4969 * gst/rtsp-server/rtsp-server.c:
4970 * gst/rtsp-server/rtsp-token.c:
4971 rtsp-*: Refer to NULL as a constant in comments
4973 https://bugzilla.gnome.org/show_bug.cgi?id=714988
4975 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4977 rtsp-*: Fix type name typos in comments
4978 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
4979 * rtsp-auth: Refer to part of constant name as text
4980 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
4981 * rtsp-session-media: Fix GstRTSPSessionMedia typo
4982 * rtsp-stream: Fix typo when refering to GstBin
4983 https://bugzilla.gnome.org/show_bug.cgi?id=714988
4985 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4988 * docs/libs/gst-rtsp-server-docs.sgml:
4989 * docs/libs/gst-rtsp-server-sections.txt:
4990 docs: Improve documentation
4991 * Include annotation-glossary to quiet gtk-doc
4992 * Rename remaining ClientState -> Context
4993 * Rename object hierarchy file
4994 * Remove stale chapter references
4995 * Add missing function and object references
4996 * Include missing GstRTSPAddressPoolResult
4997 https://bugzilla.gnome.org/show_bug.cgi?id=714988
4999 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
5001 * gst/rtsp-server/rtsp-client.c:
5002 * gst/rtsp-server/rtsp-server.c:
5003 * gst/rtsp-server/rtsp-session-pool.c:
5004 * gst/rtsp-server/rtsp-session.c:
5005 * gst/rtsp-server/rtsp-stream.c:
5006 rtsp-server: sprinkle some allow-none annotations for g-i
5008 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
5010 * gst/rtsp-server/rtsp-stream.c:
5011 * gst/rtsp-server/rtsp-stream.h:
5012 stream: add method to filter transports
5013 Add a method to safely iterate and collect the stream transports
5014 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
5016 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
5018 * gst/rtsp-server/rtsp-client.c:
5019 * gst/rtsp-server/rtsp-server.c:
5020 * gst/rtsp-server/rtsp-session-pool.c:
5021 * gst/rtsp-server/rtsp-session.c:
5022 rtsp: allow NULL func in filters
5023 Passing a null function make the filters return a list of
5026 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
5028 * gst/rtsp-server/rtsp-address-pool.c:
5029 * tests/check/gst/addresspool.c:
5030 address-pool: fix address increment
5031 Use a guint instead of guint8 to increment the address. It's still not
5032 completely correct because a guint might not be able to hold the complete
5033 address range, but that's an enhacement for later.
5034 Add unit test to test improved behaviour.
5035 https://bugzilla.gnome.org/show_bug.cgi?id=708237
5037 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
5039 * gst/rtsp-server/rtsp-client.c:
5040 * tests/check/gst/client.c:
5041 client: allow absolute path in requests
5042 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
5044 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
5046 * gst/rtsp-server/rtsp-client.c:
5047 * gst/rtsp-server/rtsp-client.h:
5048 client: make make_path_from_uri a vmethod
5050 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
5052 * docs/libs/gst-rtsp-server-sections.txt:
5053 * gst/rtsp-server/rtsp-stream.c:
5054 * gst/rtsp-server/rtsp-stream.h:
5055 * tests/check/Makefile.am:
5056 * tests/check/gst/stream.c:
5057 stream: Add functions to get rtp and rtcp sockets
5058 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
5060 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
5062 * gst/rtsp-server/rtsp-context.c:
5063 * gst/rtsp-server/rtsp-context.h:
5064 context: defing a GType for the context
5065 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
5067 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
5069 * gst/rtsp-server/Makefile.am:
5070 * gst/rtsp-server/rtsp-auth.c:
5071 * gst/rtsp-server/rtsp-context.c:
5072 * gst/rtsp-server/rtsp-media.c:
5073 * gst/rtsp-server/rtsp-mount-points.c:
5074 * gst/rtsp-server/rtsp-server.h:
5075 * gst/rtsp-server/rtsp-session-media.c:
5076 * gst/rtsp-server/rtsp-session.c:
5077 * gst/rtsp-server/rtsp-stream.c:
5078 Fixed several GIR warnings
5080 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
5082 * gst/rtsp-server/rtsp-auth.c:
5085 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5087 * tests/check/Makefile.am:
5088 * tests/check/gst/token.c:
5089 tests: Add unit tests for token
5090 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
5092 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5094 * gst/rtsp-server/rtsp-token.c:
5095 token: Validate args for gst_rtsp_token_is_allowed
5096 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
5098 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5100 * gst/rtsp-server/rtsp-token.c:
5101 token: Fix bug when creating empty token
5102 We always want to have a valid GstStructure in the token.
5103 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
5105 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
5107 * gst/rtsp-server/rtsp-thread-pool.c:
5108 thread-pool: avoid race in shutdown
5109 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
5110 don't actually stop the mainloop ever. Solve this race by adding an idle source
5111 to the mainloop that calls the _quit. This way we immediately exit the mainloop
5112 if quit was called before we started it.
5114 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5116 * tests/check/Makefile.am:
5117 * tests/check/gst/permissions.c:
5118 tests: Add unit tests for permissions
5119 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
5121 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5123 * tests/check/gst/mediafactory.c:
5124 tests: Test mediafactory permissions
5125 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
5127 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5129 * gst/rtsp-server/rtsp-permissions.c:
5130 permissions: Fix refcounting when adding/removing roles
5131 Previously a role that was removed was unreffed twice, and when
5132 replacing an existing role the replaced role was freed while still being
5133 referenced. Both bugs are now fixed.
5134 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
5136 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5138 * tests/check/gst/media.c:
5139 * tests/check/gst/mediafactory.c:
5140 * tests/check/gst/rtspserver.c:
5141 tests: Check gst_rtsp_url_parse return value
5142 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
5144 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
5147 Automatic update of common submodule
5148 From 865aa20 to dbedaa0
5150 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
5152 * gst/rtsp-server/rtsp-server.c:
5153 rtsp-server: Fix socket leak
5154 https://bugzilla.gnome.org/show_bug.cgi?id=710088
5156 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
5158 * gst/rtsp-server/rtsp-session-pool.c:
5159 rtsp-session-pool: Make sure session IDs are properly URI-escaped
5160 https://bugzilla.gnome.org/show_bug.cgi?id=643812
5162 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
5164 * examples/.gitignore:
5165 * examples/test-video.c:
5166 examples: fix compilation when WITH_AUTH is defined
5167 https://bugzilla.gnome.org/show_bug.cgi?id=710228
5169 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
5172 gitignore: Add new test binary
5174 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
5176 * tests/check/Makefile.am:
5177 * tests/check/gst/threadpool.c:
5178 thread-pool: Add unit test for the thread pools
5179 https://bugzilla.gnome.org/show_bug.cgi?id=710228
5181 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
5183 * gst/rtsp-server/rtsp-thread-pool.c:
5184 thread-pool: Fix thread leak when reusing threads
5185 https://bugzilla.gnome.org/show_bug.cgi?id=709730
5187 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
5189 * gst/rtsp-server/rtsp-server.c:
5190 * tests/check/gst/rtspserver.c:
5191 tests: fixed racy behavior in rtspserver tests
5192 https://bugzilla.gnome.org/show_bug.cgi?id=710078
5194 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5196 * tests/check/gst/addresspool.c:
5197 tests: Improve address pool unit tests
5198 Add a range with mixed IPV4 and IPV6 addresses to pool.
5199 Get an IPV4 address from an IPV6-only pool.
5200 Get an IPV6 address from an IPV4-only pool.
5201 Reserve a IPV6 address from an IPV4-only pool.
5202 Check for unicast addresses in multicast-only pool.
5203 Check for unicast addresses in uni-/multicast-mixed pool.
5204 https://bugzilla.gnome.org/show_bug.cgi?id=710128
5206 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5208 * gst/rtsp-server/rtsp-client.c:
5209 client: append query string in PAUSE/PLAY/TEARDOWN as well
5211 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
5213 * gst/rtsp-server/rtsp-client.c:
5214 client: Add query to control path
5215 If the SETUP url contains a query it must be appended to the control
5216 path so that it matches any already created stream in the media. The
5217 query will also be appended to the session media path.
5219 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5221 * gst/rtsp-server/rtsp-media.c:
5222 rtsp-media: remove old line
5224 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
5226 * gst/rtsp-server/rtsp-stream.c:
5227 stream: Correct control comparison
5228 https://bugzilla.gnome.org/show_bug.cgi?id=709176
5230 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5232 * gst/rtsp-server/rtsp-media.c:
5233 media: Check dynamically if the pipeline supports seeking
5234 We should not depend on whether or not the pipeline state change
5235 returned NO_PREROLL or not. A media could dynamically change its
5236 element and switch from seekable to non seekable so it's best to test
5237 the seekable nature of the pipeline dynamically when we try to do a seek.
5239 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5241 * gst/rtsp-server/rtsp-media.c:
5242 media: Return FALSE if seeking is not supported
5244 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5246 * gst/rtsp-server/rtsp-media.c:
5247 rtsp-media: don't seek accurate by default
5248 Accurate seeking is perhaps a little overkill in the most common situation and
5249 causes some formats (mp3) over slow media to seek extremely slowly.
5251 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
5253 * tests/check/gst/rtspserver.c:
5254 tests: fix unit test
5255 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
5257 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
5259 * gst/rtsp-server/rtsp-client.c:
5260 client: Reply 400 if media cannot be constructed
5261 Reply 400 Bad Request instead of 503 Service Unavailable if media
5262 cannot be constructed in SETUP.
5263 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
5265 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
5267 * gst/rtsp-server/rtsp-client.c:
5268 client: Send setup reply once only
5269 If find_media() failed in handle_setup_request() two replies was sent.
5270 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
5272 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
5275 Automatic update of common submodule
5276 From 6b03ba7 to 865aa20
5278 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
5280 * gst/rtsp-server/rtsp-server.c:
5281 server: Emit client-connected signal earlier
5282 Emit client-connected before the client ref is given to a GSource,
5283 otherwise client-connected can be emitted after the client object has
5286 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
5288 * gst/rtsp-server/rtsp-address-pool.c:
5289 * gst/rtsp-server/rtsp-address-pool.h:
5290 * gst/rtsp-server/rtsp-stream.c:
5291 * tests/check/gst/addresspool.c:
5292 addresspool: return reason of failure
5293 Let gst_rtsp_address_pool_reserve_address() return the reason why
5294 the address could not be reserved.
5295 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
5297 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
5300 autogen.sh: Sync behaviour with other GStreamer modules
5301 Allows building from outside of tree amongst other things
5303 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
5306 Automatic update of common submodule
5307 From b613661 to 6b03ba7
5309 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
5312 Automatic update of common submodule
5313 From 74a6857 to b613661
5315 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
5318 Automatic update of common submodule
5319 From 01a7a46 to 74a6857
5321 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
5323 * gst/rtsp-server/rtsp-client.c:
5324 client: Do not read beyond end of path string
5325 If the setup was done without a control url, make sure we don't try to read the
5326 non-existing control string and crash.
5328 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5330 * gst/rtsp-server/rtsp-client.c:
5331 client: Fix RTPInfo header
5332 Refactor the method to make the content_base.
5333 Use the content-base and the control url to construct the RTPInfo
5336 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5338 * gst/rtsp-server/rtsp-client.c:
5339 client: map url to path only in describe
5340 Only map the request url to a path in the DESCRIBE method. The SDP then
5341 contains the base and control urls that should be used to SETUP/PAUSE/
5342 PLAY/TEARDOWN the media.
5344 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5346 * gst/rtsp-server/rtsp-client.c:
5347 Revert "client: map URL to path in requests"
5348 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
5349 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
5350 contains the base and control urls which are used in the SETUP, PLAY,
5351 PAUSE and TEARDOWN requests.
5353 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5355 * gst/rtsp-server/rtsp-client.c:
5356 client: map URL to path in requests
5358 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5360 * gst/rtsp-server/rtsp-client.c:
5361 * gst/rtsp-server/rtsp-mount-points.c:
5362 * gst/rtsp-server/rtsp-mount-points.h:
5363 mount-points: make vmethod to make path from uri
5364 Make a vmethod to transform an url into a path. The path is then used to lookup
5365 the factory. This makes it possible to also use other bits of the url, such as
5366 the query parameters, to locate the factory.
5368 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
5370 * gst/rtsp-server/rtsp-thread-pool.c:
5371 * gst/rtsp-server/rtsp-thread-pool.h:
5372 thread-pool: Add cleanup to wait for the threadpool to finish
5373 Also fix race condition if two threads are asking for the first
5374 thread from the thread pool at once. This would case two internal
5375 GThreadPools to be created.
5376 https://bugzilla.gnome.org/show_bug.cgi?id=707753
5378 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
5380 * gst/rtsp-server/rtsp-client.c:
5381 * tests/check/gst/client.c:
5382 client: free threadpool
5383 https://bugzilla.gnome.org/show_bug.cgi?id=707638
5385 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
5387 * tests/check/gst/mountpoints.c:
5388 mountpoints tests: unref matched factories
5389 https://bugzilla.gnome.org/show_bug.cgi?id=707638
5391 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
5393 * tests/check/gst/media.c:
5394 media tests: unref thread pool and caps
5395 https://bugzilla.gnome.org/show_bug.cgi?id=707638
5397 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
5399 * gst/rtsp-server/rtsp-auth.c:
5400 * gst/rtsp-server/rtsp-media-factory.c:
5401 * gst/rtsp-server/rtsp-media.c:
5402 auth, media, media-factory: unref permissions
5403 https://bugzilla.gnome.org/show_bug.cgi?id=707638
5405 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5407 * examples/Makefile.am:
5408 Makefile: add rule for appsrc example
5410 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5412 * examples/test-appsrc.c:
5413 tests: add appsrc example
5414 Add an example on how to use appsrc to feed the server pipeline with data.
5416 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
5418 * gst/rtsp-server/rtsp-client.c:
5419 rtsp-client: remove query part from content-base string
5420 Make sure that after the control url has been resolved, it's
5421 not a part of the query-string.
5422 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
5424 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5426 * gst/rtsp-server/rtsp-client.c:
5427 client: don't check url in response
5428 There is no url or method in the response to check
5430 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5432 * gst/rtsp-server/rtsp-client.c:
5433 * gst/rtsp-server/rtsp-client.h:
5434 Add handle-response signal for when we receive a GET_PARAMETER response
5436 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5438 * gst/rtsp-server/rtsp-server.c:
5439 Fix gst_rtsp_server_client_filter, using wrong variable type
5441 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
5443 * gst/rtsp-server/rtsp-media-factory-uri.c:
5444 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
5445 For AAC we need to check for framed=true instead of parsed=true.
5446 https://bugzilla.gnome.org/show_bug.cgi?id=701384
5448 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5450 * gst/rtsp-server/rtsp-stream.c:
5451 stream: optimize pipeline for protocols
5452 When TCP is not an allowed protocol for the stream, avoid creating the
5453 appsrc/appsink/queue and tee elements.
5455 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5457 * gst/rtsp-server/rtsp-media.c:
5458 media: set protocols on streams
5460 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5462 * gst/rtsp-server/rtsp-client.c:
5463 client: use protocols supported by stream
5465 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5467 * gst/rtsp-server/rtsp-media-factory.c:
5468 * gst/rtsp-server/rtsp-media.c:
5469 * gst/rtsp-server/rtsp-stream.c:
5470 media-factory: allow all protocols
5472 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5474 * gst/rtsp-server/rtsp-media.c:
5475 media: configure protocols in new streams
5477 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5479 * gst/rtsp-server/rtsp-stream.c:
5480 * gst/rtsp-server/rtsp-stream.h:
5481 stream: add protocols property
5483 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5485 * gst/rtsp-server/rtsp-media.c:
5486 rtsp-media: send state in "new-state" signal
5487 https://bugzilla.gnome.org/show_bug.cgi?id=705110
5489 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
5492 build: add subdir-objects to AM_INIT_AUTOMAKE
5493 Fixes warnings with automake 1.14
5494 https://bugzilla.gnome.org/show_bug.cgi?id=705350
5496 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5498 * docs/libs/gst-rtsp-server-sections.txt:
5499 * gst/rtsp-server/rtsp-client.c:
5500 * gst/rtsp-server/rtsp-server.c:
5501 * gst/rtsp-server/rtsp-server.h:
5502 server: add method to iterate clients of server
5504 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5506 * gst/rtsp-server/rtsp-media.c:
5507 * gst/rtsp-server/rtsp-media.h:
5508 Add vmethod for rtsp-media subclass to access rtpbin
5510 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5512 * gst/rtsp-server/rtsp-client.h:
5513 small documentation fix
5515 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5517 * gst/rtsp-server/rtsp-client.c:
5518 Do not take range header if range is invalid
5520 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5522 * docs/libs/gst-rtsp-server-sections.txt:
5523 * gst/rtsp-server/rtsp-media.c:
5524 media: add docs for new method
5526 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5528 * gst/rtsp-server/rtsp-media.c:
5529 * gst/rtsp-server/rtsp-media.h:
5530 Add API to rtsp-media set the pipeline's state
5532 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
5534 * gst/rtsp-server/rtsp-media.c:
5535 Update current position/duration when gst_rtsp_media_get_range_string is called
5537 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5539 * examples/test-cgroups.c:
5540 tests: add some more docs
5542 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5544 * examples/test-cgroups.c:
5545 * gst/rtsp-server/Makefile.am:
5546 * gst/rtsp-server/rtsp-auth.c:
5547 * gst/rtsp-server/rtsp-auth.h:
5548 * gst/rtsp-server/rtsp-client.c:
5549 * gst/rtsp-server/rtsp-client.h:
5550 * gst/rtsp-server/rtsp-context.c:
5551 * gst/rtsp-server/rtsp-context.h:
5552 * gst/rtsp-server/rtsp-params.c:
5553 * gst/rtsp-server/rtsp-params.h:
5554 * gst/rtsp-server/rtsp-server.c:
5555 * gst/rtsp-server/rtsp-thread-pool.c:
5556 * gst/rtsp-server/rtsp-thread-pool.h:
5557 * tests/check/gst/client.c:
5558 ClientState -> Context
5559 Rename the clientstate to context and put the code in a separate file.
5561 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5563 * examples/test-auth.c:
5564 * gst/rtsp-server/rtsp-auth.c:
5565 * gst/rtsp-server/rtsp-auth.h:
5566 auth: add support for default token
5567 The default token is used when the user is not authenticated and can be used to
5568 give minimal permissions.
5570 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5572 * examples/test-auth.c:
5573 * gst/rtsp-server/rtsp-auth.c:
5574 auth: use defines when possible
5576 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5578 * gst/rtsp-server/rtsp-address-pool.c:
5579 address-pool: improve docs
5581 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5583 * gst/rtsp-server/rtsp-permissions.c:
5584 permissions: add the role to the copy
5586 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
5588 * gst/rtsp-server/rtsp-permissions.c:
5589 permissions: Also copy the roles
5591 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
5593 * gst/rtsp-server/rtsp-permissions.c:
5594 permissions: Make it build
5596 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5598 * gst/rtsp-server/rtsp-address-pool.h:
5601 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5603 * docs/libs/gst-rtsp-server-sections.txt:
5604 * gst/rtsp-server/rtsp-auth.c:
5605 * gst/rtsp-server/rtsp-auth.h:
5606 * gst/rtsp-server/rtsp-media.c:
5607 * gst/rtsp-server/rtsp-session-media.c:
5608 * gst/rtsp-server/rtsp-stream-transport.c:
5609 * gst/rtsp-server/rtsp-stream-transport.h:
5610 * gst/rtsp-server/rtsp-stream.c:
5611 * tests/check/gst/client.c:
5614 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5616 * docs/libs/gst-rtsp-server-sections.txt:
5617 * gst/rtsp-server/rtsp-address-pool.c:
5618 * gst/rtsp-server/rtsp-address-pool.h:
5619 * tests/check/gst/addresspool.c:
5620 * tests/check/gst/rtspserver.c:
5621 address-pool: cleanups
5622 Remove redundant method, improve docs.
5624 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5626 * docs/libs/gst-rtsp-server-sections.txt:
5627 * gst/rtsp-server/rtsp-auth.h:
5628 * gst/rtsp-server/rtsp-permissions.c:
5629 * gst/rtsp-server/rtsp-permissions.h:
5630 * gst/rtsp-server/rtsp-token.c:
5633 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5635 * gst/rtsp-server/rtsp-permissions.c:
5636 permissions: implement _remove_role
5638 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5640 * gst/rtsp-server/rtsp-permissions.c:
5641 permissions: update docs
5643 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5645 * tests/check/gst/client.c:
5646 tests: simplify tests
5647 Client settings are now disabled by default so we don't need an auth
5648 module to disable them.
5650 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5652 * gst/rtsp-server/rtsp-auth.c:
5653 auth: add default authorizations
5654 When no auth module is specified, use our table of defaults to look up the
5655 default value of the check instead of always allowing everything. This was
5656 we can disallow client settings by default.
5658 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5661 README: update readme
5663 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5665 * gst/rtsp-server/rtsp-thread-pool.c:
5666 * gst/rtsp-server/rtsp-thread-pool.h:
5667 thread-pool: add more docs
5669 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5671 * gst/rtsp-server/rtsp-thread-pool.c:
5672 * gst/rtsp-server/rtsp-thread-pool.h:
5673 thread-pool: fix race in thread reuse
5674 If we try to reuse a thread right after we made it stop, we end up using a
5675 stopped thread. Catch this case and only reuse threads that are not stopping.
5677 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5679 * gst/rtsp-server/rtsp-server.c:
5680 server: add small debug
5682 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5684 * tests/check/gst/client.c:
5686 Add some permissions to media so we can use the auth and enable
5689 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5691 * gst/rtsp-server/rtsp-client.c:
5692 client: support pushed context in handle_request
5693 If we already have a pushed state, reuse it and add our own things. This makes
5694 it easier to write tests.
5696 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5698 * gst/rtsp-server/rtsp-auth.c:
5699 auth: don't auth on methods
5700 Don't authorize on methods anymore but on the resources that we
5701 try to access, this is more flexible.
5702 Move the authorization checks to where they are needed and let the
5703 check return the response on error.
5705 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5707 * gst/rtsp-server/rtsp-mount-points.c:
5708 mount-points: add some debug
5710 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5712 * tests/check/gst/client.c:
5713 tests: almost fix test
5715 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5717 * gst/rtsp-server/rtsp-auth.c:
5718 * gst/rtsp-server/rtsp-auth.h:
5719 * gst/rtsp-server/rtsp-client.c:
5720 * gst/rtsp-server/rtsp-client.h:
5721 * gst/rtsp-server/rtsp-server.c:
5722 * gst/rtsp-server/rtsp-server.h:
5723 auth: let the auth module check client_settings
5724 Let the auth module decide if client settings are allowed for the
5727 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5729 * gst/rtsp-server/rtsp-token.c:
5730 * gst/rtsp-server/rtsp-token.h:
5731 token: add method to check boolean permission
5733 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5735 * examples/test-auth.c:
5736 * examples/test-cgroups.c:
5737 * gst/rtsp-server/rtsp-token.c:
5738 * gst/rtsp-server/rtsp-token.h:
5739 token: simplify token constructor
5740 Use variable arguments to make easier API.
5742 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5744 * examples/test-auth.c:
5745 * examples/test-cgroups.c:
5746 * gst/rtsp-server/rtsp-media-factory.c:
5747 * gst/rtsp-server/rtsp-media-factory.h:
5748 media-factory: add convenience API for factory
5750 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5752 * examples/test-auth.c:
5753 * examples/test-cgroups.c:
5754 * gst/rtsp-server/rtsp-permissions.c:
5755 * gst/rtsp-server/rtsp-permissions.h:
5756 permissions: simplify API a little
5757 Avoid passing GstStructure in the add_role method, use varargs instead
5758 to construct the structure behind the scenes. We can then also use the
5759 structure name as the role and simplify some more logic.
5761 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5763 * gst/rtsp-server/rtsp-auth.c:
5766 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5768 * gst/rtsp-server/rtsp-auth.c:
5769 * gst/rtsp-server/rtsp-auth.h:
5770 * gst/rtsp-server/rtsp-client.c:
5771 auth: handle unauthorized response
5772 Move handling of the unauthorized response to the auth module, it can add
5773 the appropriate headers to request authorization for the required method
5774 much better than the client.
5776 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5778 * gst/rtsp-server/rtsp-client.c:
5779 * gst/rtsp-server/rtsp-client.h:
5780 client: allow for sending any message, not only requests
5781 Change the _send_request() method to _send_message() so that we
5782 can both send requests and replies.
5784 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5786 * docs/libs/gst-rtsp-server-sections.txt:
5787 * gst/rtsp-server/rtsp-server.h:
5790 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5792 * examples/test-video.c:
5793 * gst/rtsp-server/rtsp-auth.c:
5794 * gst/rtsp-server/rtsp-auth.h:
5795 * gst/rtsp-server/rtsp-server.c:
5796 * gst/rtsp-server/rtsp-server.h:
5797 auth: move TLS handling to auth module
5798 Remove the TLS settings on the server and move it to the auth module because
5799 that is where security related bits go.
5801 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5803 * gst/rtsp-server/rtsp-client.c:
5804 * gst/rtsp-server/rtsp-client.h:
5805 client: add state push/pop
5807 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5809 * gst/rtsp-server/rtsp-client.c:
5810 * gst/rtsp-server/rtsp-client.h:
5811 client: add connection to state
5813 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5815 * gst/rtsp-server/rtsp-mount-points.c:
5816 mount-points: fix debug
5818 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5820 * tests/check/gst/media.c:
5821 tests: fix media test
5823 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5825 * gst/rtsp-server/rtsp-thread-pool.c:
5826 thread-pool: we don't require a state
5828 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5830 * gst/rtsp-server/rtsp-server.c:
5831 server: let context ref the server
5832 So that we don't risk losing the server object early anc crash.
5834 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5836 * tests/check/gst/client.c:
5837 tests: fix client test
5839 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5842 * docs/libs/gst-rtsp-server-docs.sgml:
5843 * docs/libs/gst-rtsp-server-sections.txt:
5844 * gst/rtsp-server/rtsp-address-pool.c:
5845 * gst/rtsp-server/rtsp-auth.c:
5846 * gst/rtsp-server/rtsp-client.c:
5847 * gst/rtsp-server/rtsp-client.h:
5848 * gst/rtsp-server/rtsp-media-factory-uri.c:
5849 * gst/rtsp-server/rtsp-media-factory.c:
5850 * gst/rtsp-server/rtsp-media-factory.h:
5851 * gst/rtsp-server/rtsp-media.c:
5852 * gst/rtsp-server/rtsp-mount-points.c:
5853 * gst/rtsp-server/rtsp-params.c:
5854 * gst/rtsp-server/rtsp-permissions.c:
5855 * gst/rtsp-server/rtsp-sdp.c:
5856 * gst/rtsp-server/rtsp-server.c:
5857 * gst/rtsp-server/rtsp-server.h:
5858 * gst/rtsp-server/rtsp-session-media.c:
5859 * gst/rtsp-server/rtsp-session-pool.c:
5860 * gst/rtsp-server/rtsp-session.c:
5861 * gst/rtsp-server/rtsp-stream-transport.c:
5862 * gst/rtsp-server/rtsp-stream.c:
5863 * gst/rtsp-server/rtsp-thread-pool.c:
5864 * gst/rtsp-server/rtsp-token.c:
5867 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5869 * gst/rtsp-server/rtsp-session-pool.c:
5870 * gst/rtsp-server/rtsp-session-pool.h:
5871 session-pool: make vmethod to create a session
5872 Make a vmethod to create a sessions so that subclasses can create
5873 custom session objects
5875 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5877 * gst/rtsp-server/rtsp-auth.c:
5878 * gst/rtsp-server/rtsp-media-factory.h:
5879 * gst/rtsp-server/rtsp-media.h:
5880 * gst/rtsp-server/rtsp-mount-points.h:
5881 * gst/rtsp-server/rtsp-session-pool.h:
5882 * gst/rtsp-server/rtsp-stream.h:
5885 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5887 * docs/libs/gst-rtsp-server-docs.sgml:
5888 * docs/libs/gst-rtsp-server-sections.txt:
5889 * gst/rtsp-server/rtsp-address-pool.c:
5890 * gst/rtsp-server/rtsp-address-pool.h:
5891 * gst/rtsp-server/rtsp-auth.c:
5892 * gst/rtsp-server/rtsp-client.h:
5893 * gst/rtsp-server/rtsp-media-factory.h:
5894 * gst/rtsp-server/rtsp-media.c:
5895 * gst/rtsp-server/rtsp-media.h:
5896 * gst/rtsp-server/rtsp-permissions.c:
5897 * gst/rtsp-server/rtsp-permissions.h:
5898 * gst/rtsp-server/rtsp-server.h:
5899 * gst/rtsp-server/rtsp-session-media.c:
5900 * gst/rtsp-server/rtsp-session-media.h:
5901 * gst/rtsp-server/rtsp-session-pool.h:
5902 * gst/rtsp-server/rtsp-session.h:
5903 * gst/rtsp-server/rtsp-stream-transport.h:
5904 * gst/rtsp-server/rtsp-stream.c:
5905 * gst/rtsp-server/rtsp-thread-pool.h:
5908 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5911 * examples/Makefile.am:
5912 configure: compile cgroup example conditionally
5913 Only compile the cgroup example when we have libcgroup
5915 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5918 * examples/Makefile.am:
5919 * examples/test-cgroups.c:
5920 examples: add cgroups example
5922 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5924 * tests/check/gst/rtspserver.c:
5925 tests: fix compilation
5927 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5929 * gst/rtsp-server/rtsp-thread-pool.c:
5930 thread-pool: fix vmethod invocation
5932 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5934 * gst/rtsp-server/rtsp-thread-pool.c:
5935 * gst/rtsp-server/rtsp-thread-pool.h:
5936 thread-pool: store thread type in thread
5938 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5940 * gst/rtsp-server/rtsp-client.c:
5941 client: pass thread from pool to media _prepare
5942 Get a thread from the configured threadpool and pass it to the prepare method of
5945 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5947 * gst/rtsp-server/rtsp-media.c:
5948 * gst/rtsp-server/rtsp-media.h:
5949 media: Accept a thread in _prepare
5950 Remove out own threadpool handling and use the provided thread and
5951 maincontext for the bus messages and the state changes.
5953 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5955 * gst/rtsp-server/rtsp-server.c:
5956 server: configure client thread pool
5958 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5960 * gst/rtsp-server/rtsp-client.c:
5961 * gst/rtsp-server/rtsp-client.h:
5962 client: add method to configure thread pool
5964 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5966 * gst/rtsp-server/rtsp-client.h:
5967 * gst/rtsp-server/rtsp-server.c:
5968 * gst/rtsp-server/rtsp-server.h:
5969 server: use thread pool
5970 Use the thread pool instead of doing our own thing.
5972 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5974 * gst/rtsp-server/Makefile.am:
5975 * gst/rtsp-server/rtsp-thread-pool.c:
5976 * gst/rtsp-server/rtsp-thread-pool.h:
5977 thread-pool: add object to manage threads
5978 Add an object to manage the client and media threads.
5980 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5982 * gst/rtsp-server/rtsp-auth.c:
5983 auth: debug authorization check
5985 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5987 * gst/rtsp-server/rtsp-media.c:
5988 media: start media pipeline in context
5989 Start the media pipeline in the provided context (or our default one
5990 when NULL). This makes sure that we run the bus thread in this context and that
5991 all media threads are children of this context.
5993 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5995 * gst/rtsp-server/rtsp-media-factory.c:
5996 factory: pass permissions to media by default
5998 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6000 * examples/test-auth.c:
6001 test: add permissions to auth test
6002 Ass some permissions to the media factory in the test.
6004 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6006 * gst/rtsp-server/rtsp-auth.c:
6007 * gst/rtsp-server/rtsp-auth.h:
6008 * gst/rtsp-server/rtsp-client.c:
6009 auth: simplify auth checks
6010 Remove client from methods, it's now in the state
6011 Perform the check specified by the string, use the information from the
6012 thread local context.
6014 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6016 * gst/rtsp-server/rtsp-client.c:
6017 * gst/rtsp-server/rtsp-client.h:
6018 client: add state to current thread
6019 Add the client to the ClientState object.
6020 Place the ClientState on the current thread.
6022 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6024 * gst/rtsp-server/rtsp-media-factory.c:
6025 * gst/rtsp-server/rtsp-media-factory.h:
6026 * gst/rtsp-server/rtsp-media.c:
6027 * gst/rtsp-server/rtsp-media.h:
6028 media: make it possible to set permissions
6029 Make it possible to set permissions on media and media factory objects
6031 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6033 * gst/rtsp-server/Makefile.am:
6034 * gst/rtsp-server/rtsp-permissions.c:
6035 * gst/rtsp-server/rtsp-permissions.h:
6036 permissions: add permissions object
6037 Add a mini object to store permissions based on a role.
6039 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6041 * examples/test-auth.c:
6042 * gst/rtsp-server/rtsp-auth.c:
6043 * gst/rtsp-server/rtsp-auth.h:
6044 * gst/rtsp-server/rtsp-client.c:
6045 auth: add auth checks
6046 Add an enum with auth checks and implement the checks in the auth object.
6047 Perform the checks from the client.
6049 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6051 * examples/test-auth.c:
6052 * gst/rtsp-server/rtsp-auth.c:
6053 * gst/rtsp-server/rtsp-auth.h:
6054 * gst/rtsp-server/rtsp-client.h:
6055 auth: use the token after authentication
6056 After we authenticated a user, keep the Token around in the state.
6058 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6060 * gst/rtsp-server/rtsp-client.c:
6061 * gst/rtsp-server/rtsp-media.c:
6062 * gst/rtsp-server/rtsp-media.h:
6063 * tests/check/gst/media.c:
6064 media: add optional context for bus messages
6065 Add an optional mainloop to _prepare that will handle the bus messages instead
6066 of always using the shared mainloop.
6068 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6070 * gst/rtsp-server/Makefile.am:
6071 * gst/rtsp-server/rtsp-token.c:
6072 * gst/rtsp-server/rtsp-token.h:
6073 token: add authorization token
6074 Add a simply miniobject that contains the authorizations. The object contains a
6075 GstStructure that hold all authorization fields. When a user is authenticated,
6076 the auth module will create a Token for the user. The token is then used to
6077 check what operations the user is allowed to do and various other configuration
6080 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6082 * examples/test-auth.c:
6083 * gst/rtsp-server/rtsp-auth.c:
6084 * gst/rtsp-server/rtsp-auth.h:
6085 * gst/rtsp-server/rtsp-client.c:
6086 * gst/rtsp-server/rtsp-client.h:
6087 * gst/rtsp-server/rtsp-media-factory.c:
6088 * gst/rtsp-server/rtsp-media-factory.h:
6089 * gst/rtsp-server/rtsp-media.c:
6090 * gst/rtsp-server/rtsp-media.h:
6091 auth: remove auth from media and factory
6092 Remove the auth object from media and factory. We want to have the RTSPClient
6093 authenticate and authorize resources, there is no need to place another auth
6094 manager on the media/factory.
6096 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6098 * examples/test-auth.c:
6099 * gst/rtsp-server/rtsp-auth.c:
6100 * gst/rtsp-server/rtsp-auth.h:
6101 * gst/rtsp-server/rtsp-client.h:
6102 auth: add support for multiple basic auth tokens
6103 Make it possible to add multiple basic authorisation tokens to one authorization
6104 object. Associate with each token an authorization group that will define what
6105 capabilities are allowed.
6107 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6109 * gst/rtsp-server/rtsp-client.c:
6110 client: error out on non-aggregate control
6111 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
6113 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6115 * gst/rtsp-server/rtsp-client.c:
6116 client: rework setup request a little
6117 Cache the media in DESCRIBE based on the longest matching path with the uri
6118 that we can find in the mount points.
6119 Rework the setup request a little to get the media from the session or from
6120 the longest matching path, this way we can derive the control string as
6121 everything after the path instead of hardcoding it.
6122 Find the stream based on the control string and only open a session when all
6125 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6127 * gst/rtsp-server/rtsp-media.c:
6128 * gst/rtsp-server/rtsp-media.h:
6129 media: add method to find a stream by control url
6131 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6133 * gst/rtsp-server/rtsp-stream.c:
6134 * gst/rtsp-server/rtsp-stream.h:
6135 stream: add method to check control url of stream
6137 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6139 * gst/rtsp-server/rtsp-client.c:
6140 * gst/rtsp-server/rtsp-session-media.c:
6141 * gst/rtsp-server/rtsp-session-media.h:
6142 * gst/rtsp-server/rtsp-session.c:
6143 * gst/rtsp-server/rtsp-session.h:
6144 session: use path matching for session media
6145 Use a path string instead of a uri to lookup session media in the sessions. Also
6146 use path matching to find the largest possible path that matches.
6148 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6150 * gst/rtsp-server/rtsp-client.c:
6151 * gst/rtsp-server/rtsp-mount-points.c:
6152 * gst/rtsp-server/rtsp-mount-points.h:
6153 * tests/check/gst/mountpoints.c:
6154 mount-points: remove useless vmethod
6155 Making lookups in the mount points should not be done with a URL, if there is a
6156 mapping to be done from URL to mount points, we'll need to do it somewhere
6159 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6161 * gst/rtsp-server/rtsp-mount-points.c:
6162 * gst/rtsp-server/rtsp-mount-points.h:
6163 * tests/check/gst/mountpoints.c:
6164 mount-points: improve mount point searching
6165 Use a GSequence to keep track of the mount points.
6166 Match a URL to the longest matching registered mount point. This should be the
6167 URL to perform aggreagate control and the remainder is the stream specific
6169 Add some unit tests for this.
6171 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
6173 * gst/rtsp-server/Makefile.am:
6174 rtsp-server: Allow building of static library
6176 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6178 * tests/check/gst/mediafactory.c:
6179 tests: fix compilation
6181 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6183 * gst/rtsp-server/rtsp-sdp.c:
6184 sdp: get control string from stream
6185 Use the control string as configured in the stream.
6187 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6189 * gst/rtsp-server/rtsp-stream.c:
6190 * gst/rtsp-server/rtsp-stream.h:
6191 stream: add methods and property to set control string
6193 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6195 * gst/rtsp-server/rtsp-client.c:
6197 Rename variables for clarity
6198 Keep media in state when we can
6200 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6202 * gst/rtsp-server/rtsp-client.c:
6203 * gst/rtsp-server/rtsp-stream.c:
6204 * gst/rtsp-server/rtsp-stream.h:
6205 stream: add more support for IPv6
6206 Rename _get_address to _get_multicast_address in GstRTSPStream to
6207 make it clear that this function only deals with multicast.
6208 Make it possible to have both an IPv4 and IPv6 multicast address on
6209 a stream. Give the client an IPv4 or IPv6 address depending on the
6210 address it used to connect to the server.
6211 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
6213 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6215 * gst/rtsp-server/rtsp-client.c:
6218 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6220 * gst/rtsp-server/rtsp-stream.c:
6221 stream: handle failed port allocation
6222 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
6223 can't allocate any family at all. Also keep track of what port families we
6225 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
6227 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6229 * gst/rtsp-server/rtsp-stream.c:
6230 stream: improve docs
6232 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6234 * gst/rtsp-server/rtsp-stream-transport.c:
6235 stream-transport: remove old if 0 block
6237 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
6239 * tests/check/gst/client.c:
6241 gst_rtsp_client_get_uri() has been removed
6242 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
6244 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6246 * gst/rtsp-server/rtsp-client.c:
6247 * gst/rtsp-server/rtsp-client.h:
6248 client: add method to filter managed sessions
6249 Add a method to filter the sessions managed by this client connection.
6250 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
6252 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6254 * gst/rtsp-server/rtsp-client.c:
6255 * gst/rtsp-server/rtsp-client.h:
6256 client: remove _get_uri() method
6257 Remove the get_uri() method on the client. A client has no uri, the uri
6258 property is an internal property to manage the last cached media for
6261 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6263 * gst/rtsp-server/rtsp-media-factory.h:
6264 media-factory: fix typo
6266 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
6268 * gst/rtsp-server/rtsp-media.c:
6269 rtsp-media: Do not leak the query in default_query_stop
6270 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
6272 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6274 * gst/rtsp-server/rtsp-media.c:
6275 media: don't unlock when conversion fails
6276 Don't unlock the state lock when conversion fails because it was not locked.
6278 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6280 * gst/rtsp-server/rtsp-media.c:
6281 * gst/rtsp-server/rtsp-media.h:
6282 Add query_position and query_stop vmethods to rtsp-media
6284 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6286 * gst/rtsp-server/rtsp-media.c:
6287 Fix typo in property install for rtsp-media's time-provider
6289 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6291 * gst/rtsp-server/rtsp-client.c:
6292 * gst/rtsp-server/rtsp-client.h:
6293 client: clean some variables
6294 Clean some variables and add some guards to _send_request()
6296 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6298 * gst/rtsp-server/rtsp-client.c:
6299 * gst/rtsp-server/rtsp-client.h:
6300 Add gst_rtsp_client_send_request API
6301 This makes it possible to send arbitrary messages to a client, such as
6302 SET_PARAMETER or GET_PARAMETER
6304 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6306 * gst/rtsp-server/rtsp-media.c:
6307 * gst/rtsp-server/rtsp-media.h:
6308 media: add _get_element() method
6309 Add method to get the element used when creating the media.
6310 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
6312 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6314 * gst/rtsp-server/rtsp-media.c:
6317 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
6319 * gst/rtsp-server/rtsp-stream.c:
6320 * gst/rtsp-server/rtsp-stream.h:
6321 stream: allow access to the rtp session
6322 https://bugzilla.gnome.org/show_bug.cgi?id=703004
6324 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
6326 * gst/rtsp-server/rtsp-stream.c:
6327 * gst/rtsp-server/rtsp-stream.h:
6328 dscp qos support in gst-rtsp-stream
6329 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
6331 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6333 * tests/check/gst/rtspserver.c:
6335 Actually do what the comment says. Also keep the old code around, not sure what
6336 should happen when you get a 454 from a TEARDOWN, does it close the connection?
6337 it currently doesn't.
6339 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6341 * gst/rtsp-server/rtsp-client.c:
6342 client: also watch newly created session
6343 When we newly created a session, start watching it immediately instead of
6344 on the next request.
6346 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
6348 * tests/check/gst/client.c:
6349 tests: add unit test for new-session
6350 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
6352 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6354 * gst/rtsp-server/rtsp-client.c:
6355 client: emit new-session when new session is created
6356 Only emit new-session when we created a new session for a client, not when a
6357 client picked up a previous session.
6358 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
6360 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
6362 * gst/rtsp-server/rtsp-client.c:
6363 client: handle asterisk as path in requests
6364 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
6366 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6368 * gst/rtsp-server/rtsp-media.c:
6369 media: handle segment query format mismatch
6370 It's possible that the segment query returns with a different format than what
6371 we asked for, handle this case also.
6373 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
6375 * gst/rtsp-server/rtsp-media.c:
6376 media: use segment stop in collect_media_stats
6377 Use segment stop instead of duration as range end point.
6378 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
6380 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
6382 * gst/rtsp-server/rtsp-media.c:
6383 * tests/check/gst/media.c:
6384 rtsp-media: Do not leak the element in take_pipeline
6385 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
6387 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
6389 * gst/rtsp-server/rtsp-client.c:
6390 * gst/rtsp-server/rtsp-client.h:
6391 rtsp-client: Make configure_client_transport virtual
6392 This patch makes configure_client_transport virtual. The functionality is
6393 needed to handle some weird clients sending multicast transport settings as url
6395 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
6397 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
6399 * gst/rtsp-server/rtsp-client.c:
6400 * gst/rtsp-server/rtsp-client.h:
6401 rtsp-client: Make param_set and param_get virtual
6402 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
6404 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
6406 * gst/rtsp-server/rtsp-client.c:
6407 * gst/rtsp-server/rtsp-media.c:
6408 * gst/rtsp-server/rtsp-media.h:
6409 media: convert_range replaces get_range_times
6410 get_range_times worked for handling UTC ranges for seeks, but we also
6411 need to convert back from NPT to the requested unit in
6412 get_range_string. convert_range is now used for both.
6413 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
6415 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6417 * gst/rtsp-server/rtsp-client.c:
6418 * gst/rtsp-server/rtsp-sdp.c:
6419 * gst/rtsp-server/rtsp-sdp.h:
6420 sdp: cleanup sdp info
6421 We don't need to pass the proto, we can more easily check a boolean.
6422 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
6424 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
6426 * gst/rtsp-server/rtsp-sdp.c:
6427 use 0.0.0.0 or :: for c= line instead of server address
6429 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
6431 * gst/rtsp-server/rtsp-client.c:
6432 use local address, not remote, in SDP
6433 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
6435 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6438 Automatic update of common submodule
6439 From 098c0d7 to 01a7a46
6441 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
6443 * gst/rtsp-server/rtsp-media.c:
6444 * gst/rtsp-server/rtsp-media.h:
6445 media: possibility to override range time conversion
6446 Make it possible to override the conversion from GstRTSPTimeRange to
6447 GstClockTimes, that is done before seeking on the media
6448 pipeline. Overriding can be useful for UTC ranges, where the default
6449 conversion gives nanoseconds since 1900.
6450 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
6452 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
6454 * gst/rtsp-server/rtsp-server.c:
6455 * gst/rtsp-server/rtsp-server.h:
6456 rtsp-server: Expose the use_client_settings API
6457 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
6459 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
6461 * gst/rtsp-server/rtsp-client.c:
6462 * gst/rtsp-server/rtsp-stream.c:
6463 * gst/rtsp-server/rtsp-stream.h:
6464 rtspstream: handle both ipv4 and ipv6 clients
6465 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
6467 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6469 * gst/rtsp-server/rtsp-sdp.c:
6470 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
6471 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
6472 We already have a way to place extra attributes in the SDP by using a string
6473 property with prefix x- or a- in the caps.
6475 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6477 * gst/rtsp-server/rtsp-sdp.c:
6478 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
6479 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
6480 We already have a way to place extra attributes in the SDP, just make a string
6481 property in the payloader with a- or x- prefix.
6483 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6485 * gst/rtsp-server/rtsp-sdp.c:
6486 rtsp: place a- and x- properties as attributes
6487 application/x-rtp has properties with a- and x- prefixes that should be
6488 placed as attributes in the SDP for the media instead of being added to the
6491 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6493 * examples/Makefile.am:
6494 * examples/test-video.c:
6495 example: add TLS example
6497 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6499 * gst/rtsp-server/rtsp-server.c:
6500 * gst/rtsp-server/rtsp-server.h:
6501 server: add support for TLS
6502 Add methods to set and get a TLS certificate.
6503 Add vmethod to configure a new connection. By default, configure the TLS
6504 certificate in a new connection if needed.
6506 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6508 * gst/rtsp-server/rtsp-server.c:
6509 * gst/rtsp-server/rtsp-server.h:
6510 server: remove accept_client vmethod
6511 This vmethod is not very useful so remove it.
6513 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6515 * gst/rtsp-server/rtsp-server.c:
6516 server: don't crash on NULL GError
6518 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
6520 * gst/rtsp-server/rtsp-session-pool.c:
6521 rtsp-session-pool: corrected session timeout detection
6522 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
6524 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6526 * gst/rtsp-server/rtsp-client.c:
6527 client: improve debug
6529 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6531 * gst/rtsp-server/rtsp-client.c:
6532 * gst/rtsp-server/rtsp-client.h:
6533 * gst/rtsp-server/rtsp-server.c:
6534 server: refactor connection setup
6535 Let the server accept the socket connection and construct a GstRTSPConnection
6536 from it. Remove the code from the client and let the client only deal with
6537 a fully configure GstRTSPConnection object.
6538 We will need this later when the server will configure the connection for
6541 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6543 * gst/rtsp-server/rtsp-stream.c:
6544 stream: keep the transport object alive
6545 Keep the transport object alive while we have it as qdata on the
6548 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
6550 * gst/rtsp-server/rtsp-client.c:
6551 * gst/rtsp-server/rtsp-server.c:
6552 rtsp-server: Do not crash on nmapping of server
6553 * generate error when gst_rtsp_connection_accept fails
6554 * do not stop accepting incoming connections because
6555 accepting a client fails
6556 https://bugzilla.gnome.org/show_bug.cgi?id=701072
6558 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
6560 * gst/rtsp-server/rtsp-client.c:
6561 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
6562 https://bugzilla.gnome.org/show_bug.cgi?id=700953
6564 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
6566 * gst/rtsp-server/rtsp-sdp.c:
6567 rtsp-sdp: Parse framerate caps field and set SDP attribute
6568 The SDP attribute and its format is described in RFC4566.
6569 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
6571 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
6573 * gst/rtsp-server/rtsp-sdp.c:
6574 rtsp-sdp: Parse width/height from caps and set SDP attribute
6575 The SDP attribute and its format is described in RFC6064.
6576 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
6578 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
6580 * gst/rtsp-server/rtsp-sdp.c:
6581 * tests/check/gst/client.c:
6582 rtsp-sdp: add bandwidth line
6583 https://bugzilla.gnome.org/show_bug.cgi?id=699220
6585 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6588 Automatic update of common submodule
6589 From 5edcd85 to 098c0d7
6591 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
6593 * tests/check/gst/media.c:
6594 tests: add dynamic payloader prepare/unprepare check
6596 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6598 * gst/rtsp-server/rtsp-media.c:
6599 media: release lock when removing fakesink
6601 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6603 * gst/rtsp-server/rtsp-stream.c:
6604 stream: set elements to NULL before removing
6605 When removing a stream, set the elements to NULL first. This avoids
6606 element-is-not-in-NULL-state errors when we dispose the elements.
6608 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
6611 Automatic update of common submodule
6612 From 3cb3d3c to 5edcd85
6614 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6616 * gst/rtsp-server/rtsp-media.c:
6617 * gst/rtsp-server/rtsp-media.h:
6618 media: listen to pad-removed signals
6619 Listen to the pad-removed signal and remove the stream associated with the
6621 Add signal to be notified of the removed pad.
6622 Remove the fakesink in unprepare()
6623 Fix signatures of the signal methods
6625 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6627 * examples/test-sdp.c:
6628 tests: add example of reusable pipelines
6630 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
6632 * gst/rtsp-server/rtsp-stream.c:
6633 * gst/rtsp-server/rtsp-stream.h:
6634 stream: add method to get the srcpad
6636 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
6638 * tests/check/gst/media.c:
6639 check: add media prepare/unprepare test
6640 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
6642 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
6644 * gst/rtsp-server/rtsp-media.c:
6645 media: disconnect from signal handlers in unprepare()
6646 We connected to the pad-added and no-more-pads signals in prepare() so
6647 we need to disconnect from them in unprepare().
6648 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
6650 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
6652 * gst/rtsp-server/rtsp-media.c:
6653 media: don't free streams array
6654 Don't free the streams array in the unprepare() method, they were not
6656 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
6658 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
6660 * gst/rtsp-server/rtsp-media.c:
6661 media: don't unref the pipeline in unprepare
6662 Unprepare() should undo what prepare() does. Because the pipeline is
6663 not created in prepare(), we should not unref it in unprepare()
6665 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
6667 * gst/rtsp-server/rtsp-stream.c:
6668 stream: clear session and caps for reuse
6669 Set the session and caps to NULL after unref otherwise we might unref
6671 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
6673 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
6675 * gst/rtsp-server/rtsp-client.c:
6676 client: send out teardown signal before tearing down
6677 The advantage is that in the signal handler you get direct access to
6678 information about what streams are about to get torn down (in the
6679 GstRTSPClientState).
6680 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
6682 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
6684 * gst/rtsp-server/rtsp-client.c:
6685 * gst/rtsp-server/rtsp-client.h:
6686 client: expose connection
6687 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
6689 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
6692 Automatic update of common submodule
6693 From aed87ae to 3cb3d3c
6695 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6697 * gst/rtsp-server/rtsp-media.c:
6698 * gst/rtsp-server/rtsp-media.h:
6699 * gst/rtsp-server/rtsp-session-media.c:
6700 * gst/rtsp-server/rtsp-session-media.h:
6701 media: add method to get the base_time of the pipeline
6702 Together with a shared clock, this base-time could eventually be sent to
6703 the client so that it can reconstruct the exact running-time of the clock
6706 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6708 * gst/rtsp-server/Makefile.am:
6709 * gst/rtsp-server/rtsp-media.c:
6710 * gst/rtsp-server/rtsp-media.h:
6711 * gst/rtsp-server/rtsp-sdp.c:
6712 media: add GstNetTimeProvider support
6713 Add a property to let the media provide a GstNetTimeProvider for its clock.
6714 Make methods to get the clock and nettimeprovider
6715 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
6716 provider and also the current time of the clock. This should make it possible
6717 for (GStreamer) clients to slave their clock to the server clock.
6719 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
6722 Automatic update of common submodule
6723 From 04c7a1e to aed87ae
6725 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6727 * gst/rtsp-server/rtsp-media.c:
6728 media: wait for buffering to complete
6729 Wait for buffering to complete before changing the state to the target state.
6731 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6733 * gst/rtsp-server/rtsp-media.c:
6734 media: small cleanup
6736 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
6738 * tests/check/gst/rtspserver.c:
6739 tests: remove extra unref in test_setup_non_existing_stream
6740 The unref is not needed anymore, teardown runs without it.
6741 https://bugzilla.gnome.org/show_bug.cgi?id=696542
6743 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
6745 * tests/check/gst/rtspserver.c:
6746 tests: GSocketService cleanup in test_bind_already_in_use
6747 Use g_socket_service_stop so the rtspserver test stops listening for
6748 incoming connections in test_bind_already_in_use.
6749 https://bugzilla.gnome.org/show_bug.cgi?id=696541
6751 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
6753 * gst/rtsp-server/rtsp-media-factory.c:
6754 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
6755 Instead use a GWeakRef which is safe to use
6756 This is a known GLib bug, see:
6757 https://bugzilla.gnome.org/show_bug.cgi?id=667145
6759 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
6761 * gst/rtsp-server/rtsp-client.c:
6762 * gst/rtsp-server/rtsp-media.c:
6763 * gst/rtsp-server/rtsp-media.h:
6764 * gst/rtsp-server/rtsp-sdp.c:
6765 * tests/check/gst/media.c:
6766 * tests/check/gst/rtspserver.c:
6767 rtsp-media/client: Reply to PLAY request with same type of Range
6768 Remember the type of Range from the PLAY request and use the same type for
6771 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
6773 * gst/rtsp-server/rtsp-client.c:
6774 * gst/rtsp-server/rtsp-client.h:
6775 * tests/check/gst/client.c:
6776 rtsp-client: expose uri
6778 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
6780 * tests/check/gst/mediafactory.c:
6781 tests: Hold ref while creating second media
6782 To test if the media aren't shared, make sure we keep the first one while creating a second
6783 otherwise the same memory address may be reused.
6785 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
6788 configure: remove out-of-date comment
6790 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
6793 .gitignore: ignore more build files
6795 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
6797 * tests/check/Makefile.am:
6798 tests: use right _LIBS variable for gst-plugins-base libs
6800 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6802 * tests/check/Makefile.am:
6803 check: add librtp to libs
6805 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
6807 * tests/check/gst/rtspserver.c:
6808 tests: Add test to check selecting a port the server will send from
6810 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
6812 * tests/check/gst/rtspserver.c:
6813 tests: Make sure packets are actually received
6815 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
6817 * gst/rtsp-server/rtsp-stream.c:
6818 stream: Select unicast address from pool if appropriate
6820 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
6822 * gst/rtsp-server/rtsp-stream.c:
6823 stream: Properties are always there in Gst 1.0
6825 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
6827 * tests/check/gst/addresspool.c:
6828 tests: Add tests for unicast addresses in pool
6830 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
6832 * gst/rtsp-server/rtsp-address-pool.c:
6833 * tests/check/gst/addresspool.c:
6834 address-pool: Verify that multicast addresses are used for multicast and vice-versa
6836 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
6838 * docs/libs/gst-rtsp-server-sections.txt:
6839 * gst/rtsp-server/rtsp-address-pool.c:
6840 * gst/rtsp-server/rtsp-address-pool.h:
6841 * gst/rtsp-server/rtsp-stream.c:
6842 * tests/check/gst/addresspool.c:
6843 address-pool: Add unicast addresses
6845 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
6848 * gst/rtsp-server/rtsp-server.c:
6849 * tests/check/gst/rtspserver.c:
6850 rtsp-server: Limit the number of threads per server instance
6851 If we exceed the maximum, just round robin the clients over the existing
6854 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
6856 * gst/rtsp-server/rtsp-server.c:
6857 rtsp-server: No need to store the GMainContext in the client context
6859 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
6861 * tests/check/gst/rtspserver.c:
6862 tests: Add test for client disconnection
6864 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
6866 * tests/check/gst/rtspserver.c:
6867 tests: Test client and session timeouts with multiple threads
6869 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
6871 * gst/rtsp-server/rtsp-address-pool.c:
6872 * gst/rtsp-server/rtsp-auth.c:
6873 * gst/rtsp-server/rtsp-client.c:
6874 * gst/rtsp-server/rtsp-media-factory-uri.c:
6875 * gst/rtsp-server/rtsp-media-factory.c:
6876 * gst/rtsp-server/rtsp-media.c:
6877 * gst/rtsp-server/rtsp-mount-points.c:
6878 * gst/rtsp-server/rtsp-server.c:
6879 * gst/rtsp-server/rtsp-session-media.c:
6880 * gst/rtsp-server/rtsp-session-pool.c:
6881 * gst/rtsp-server/rtsp-session.c:
6882 Document locking and its order
6884 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
6886 * tests/check/gst/rtspserver.c:
6887 tests: Test that slow DESCRIBE don't block other clients
6889 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
6891 * tests/check/gst/client.c:
6892 tests: Add tests for client-requested multicast address
6894 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
6896 * docs/libs/gst-rtsp-server-sections.txt:
6897 docs: Put the various functions in the right sections
6899 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
6901 * docs/libs/gst-rtsp-server-docs.sgml:
6902 * docs/libs/gst-rtsp-server-sections.txt:
6903 * gst/rtsp-server/rtsp-address-pool.c:
6904 * gst/rtsp-server/rtsp-address-pool.h:
6905 docs: Generate docs for GstRTSPAddressPool
6907 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
6909 * gst/rtsp-server/rtsp-client.c:
6910 * gst/rtsp-server/rtsp-stream.c:
6911 * gst/rtsp-server/rtsp-stream.h:
6912 client: Check client provided addresses against the address pool
6914 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
6916 * gst/rtsp-server/rtsp-address-pool.c:
6917 * gst/rtsp-server/rtsp-address-pool.h:
6918 * tests/check/gst/addresspool.c:
6919 address-pool: Add API to request a specific address from the pool
6920 Also add relevant unit tests.
6922 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
6924 * tests/check/gst/mediafactory.c:
6925 tests: Check the passing around of a RTSPAddressPool
6926 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
6927 way down to the stream.
6929 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
6931 * tests/check/gst/addresspool.c:
6932 tests: Add more tests for the address pool
6934 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
6936 * gst/rtsp-server/rtsp-address-pool.c:
6937 address-pool: Fix off by one error
6938 When splitting a port range, the port after a skip is not part of range.
6940 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
6943 Automatic update of common submodule
6944 From 2de221c to 04c7a1e
6946 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
6949 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
6950 AM_CONFIG_HEADER was removed in automake 1.13
6951 https://bugzilla.gnome.org/show_bug.cgi?id=693368
6953 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
6956 Automatic update of common submodule
6957 From a942293 to 2de221c
6959 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6961 * gst/rtsp-server/rtsp-client.c:
6962 client: make sure the watch exists while sending data
6963 Protect the send_func with a lock. This allows us to wait for sending
6964 to complete before changing the send_func and user_data. We add an
6965 extra ref to the watch to make sure that it remains valid during
6967 When closing the connection, set the send_func to NULL
6968 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
6970 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6972 * tests/check/Makefile.am:
6973 tests: use GST_*_1_0 environment variables everywhere
6974 The _1_0 suffixed environment variables override the
6975 non-suffixed ones, so if we're in an environment that
6976 sets the _1_0 suffixed ones, such as jhbuild, we need
6977 to set those to make sure ours actually always get
6980 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6983 Automatic update of common submodule
6984 From acb04d9 to a942293
6986 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6988 * gst/rtsp-server/rtsp-client.c:
6989 rtsp-client: set the client backlog
6990 Set the client backlog to a reasonable default
6992 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
6994 * gst/rtsp-server/rtsp-media.c:
6995 rtsp-media: Make the element a constructor parameter
6996 https://bugzilla.gnome.org/show_bug.cgi?id=689594
6998 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
7000 * docs/libs/Makefile.am:
7001 docs: Link with gcov library when gcov is enabled
7002 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
7004 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7006 * gst/rtsp-server/rtsp-media.c:
7007 media: match prepare with unprepare
7008 Really unprepare when there were an equal amount of prepare calls.
7010 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7012 * gst/rtsp-server/rtsp-media.c:
7013 media: media has to be unprepared in finalize
7014 Because unprepare takes away the last ref on the media.
7016 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7018 * gst/rtsp-server/rtsp-client.c:
7019 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
7020 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
7021 We can't use the refcount to trigger unprepare because it is the unprepare call
7022 that removes the last refcount after all messages are consumed. What we should
7023 probably do is make a prepared refcount and only unprepare when the refcount
7026 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7028 * gst/rtsp-server/rtsp-media.c:
7029 media: let the source unref the last media ref
7030 the last ref to the media is held by the source so we don't need to add more ref
7031 and unrefs, we simply destroy the media when the source is gone.
7033 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7035 * gst/rtsp-server/rtsp-media.c:
7036 media: improve debug
7038 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7040 * gst/rtsp-server/rtsp-media.c:
7042 Make sure we are in the right state when collecting the position and duration.
7043 Only make ourselves PREPARED when we were previously PREPARING.
7045 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7047 * gst/rtsp-server/rtsp-media.c:
7048 media: use g_object_ref/unref for GObjects
7050 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
7052 * gst/rtsp-server/rtsp-client.c:
7053 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
7054 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
7055 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
7056 isn't being used anymore.
7058 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
7060 * gst/rtsp-server/rtsp-media.c:
7061 Fix compiler warning
7063 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
7065 * gst/rtsp-server/rtsp-media-factory-uri.c:
7066 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
7068 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7070 * gst/rtsp-server/rtsp-session-media.h:
7073 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7075 * gst/rtsp-server/rtsp-media.c:
7076 * tests/check/gst/media.c:
7077 media: avoid element leak
7079 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7081 * gst/rtsp-server/rtsp-media.c:
7082 media: require an element in media constructor
7084 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7086 * gst/rtsp-server/rtsp-client.c:
7087 Revert "client: TEARDOWN brings that state to Init again"
7088 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
7089 The object is already disposed, there is no point in setting the state.
7091 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7093 * gst/rtsp-server/rtsp-client.c:
7094 client: TEARDOWN brings that state to Init again
7096 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7098 * docs/libs/gst-rtsp-server-sections.txt:
7099 * examples/test-auth.c:
7100 * gst/rtsp-server/rtsp-auth.c:
7101 * gst/rtsp-server/rtsp-auth.h:
7102 * gst/rtsp-server/rtsp-client.c:
7103 * gst/rtsp-server/rtsp-client.h:
7104 * gst/rtsp-server/rtsp-media-factory-uri.c:
7105 * gst/rtsp-server/rtsp-media-factory-uri.h:
7106 * gst/rtsp-server/rtsp-media-factory.c:
7107 * gst/rtsp-server/rtsp-media-factory.h:
7108 * gst/rtsp-server/rtsp-media.c:
7109 * gst/rtsp-server/rtsp-media.h:
7110 * gst/rtsp-server/rtsp-mount-points.c:
7111 * gst/rtsp-server/rtsp-mount-points.h:
7112 * gst/rtsp-server/rtsp-sdp.c:
7113 * gst/rtsp-server/rtsp-server.c:
7114 * gst/rtsp-server/rtsp-server.h:
7115 * gst/rtsp-server/rtsp-session-media.c:
7116 * gst/rtsp-server/rtsp-session-media.h:
7117 * gst/rtsp-server/rtsp-session-pool.c:
7118 * gst/rtsp-server/rtsp-session-pool.h:
7119 * gst/rtsp-server/rtsp-session.c:
7120 * gst/rtsp-server/rtsp-session.h:
7121 * gst/rtsp-server/rtsp-stream-transport.c:
7122 * gst/rtsp-server/rtsp-stream-transport.h:
7123 * gst/rtsp-server/rtsp-stream.c:
7124 * gst/rtsp-server/rtsp-stream.h:
7125 * tests/check/gst/media.c:
7126 rtsp: make object details private
7127 Make all object details private
7128 Add methods to access private bits
7130 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7132 * tests/check/Makefile.am:
7133 * tests/check/gst/media.c:
7134 tests: add media tests
7136 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7138 * gst/rtsp-server/rtsp-media.c:
7139 media: check if prepared for some methods
7140 Check that the media object is prepared before doing seek and getting the
7141 current position etc.
7142 Add some g_return checks.
7144 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7146 * tests/check/Makefile.am:
7147 * tests/check/gst/mediafactory.c:
7148 tests: add mediafactory test
7150 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7152 * gst/rtsp-server/rtsp-stream.c:
7153 stream: improve debug
7155 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7157 * gst/rtsp-server/rtsp-media.c:
7158 * gst/rtsp-server/rtsp-media.h:
7159 media: unref pipeline in finalize to avoid leaking it
7161 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7163 * gst/rtsp-server/rtsp-media-factory-uri.c:
7164 * gst/rtsp-server/rtsp-media.c:
7165 rtsp: use gst_object_unref on GstObjects
7167 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7169 * gst/rtsp-server/rtsp-media-factory.c:
7170 media-factory: require an url
7172 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7174 * examples/test-uri.c:
7175 examples: fix include
7177 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7179 * gst/rtsp-server/rtsp-server.h:
7180 server: remove unused include
7182 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7184 * tests/check/Makefile.am:
7185 * tests/check/gst/mountpoints.c:
7186 tests: add test for mountpoints
7188 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7190 * gst/rtsp-server/rtsp-client.c:
7191 client: fix factory leak
7192 Keep the factory in the state object only for authorization checks and make
7193 sure we unref it on failure. Also don't keep invalid objects in the state
7196 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7198 * gst/rtsp-server/rtsp-mount-points.c:
7199 mounts: add g_return_if guards
7201 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7203 * tests/check/gst/client.c:
7204 tests: add more tests
7206 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7208 * gst/rtsp-server/rtsp-client.c:
7209 client: improve debug
7211 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7213 * gst/rtsp-server/rtsp-client.c:
7214 client: improve debug and fix leaks
7215 Cleanup the uri and session when there is a bad request.
7217 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7222 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7224 * tests/check/gst/client.c:
7225 test: add test for session in options request
7227 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7229 * gst/rtsp-server/rtsp-client.c:
7230 client: use 454 when session can't be found
7231 We should use 454 when a session can't be found because there was no session
7232 pool configured in the server. This is not a server configuration problem
7233 because the server on which the request is done might not be the same one that
7234 will keep the sessions for us and so it does not need to support sessions.
7236 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7238 * gst/rtsp-server/rtsp-client.c:
7239 client: only free connection when there is one
7240 It's possible that the client doesn't have a connection when we try to free it.
7242 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7244 * tests/check/Makefile.am:
7245 * tests/check/gst/client.c:
7246 tests: add unit test for the client object
7248 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7250 * gst/rtsp-server/rtsp-client.c:
7251 client: small cleanup
7253 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7255 * gst/rtsp-server/rtsp-client.h:
7256 client: remove unused include
7258 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7260 * gst/rtsp-server/rtsp-client.c:
7261 client: fix compilation
7263 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7265 * gst/rtsp-server/rtsp-client.c:
7266 client: call destroy without the lock
7268 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7270 * gst/rtsp-server/rtsp-client.c:
7271 * gst/rtsp-server/rtsp-client.h:
7272 client: make the client usable without a socket
7273 Make a method to let the client handle a message and a callback when the client
7274 wants us to send a response message back. This makes it possible to also use the
7275 client object without the sockets, which should make it easier to test.
7277 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7279 * gst/rtsp-server/rtsp-client.c:
7280 * gst/rtsp-server/rtsp-client.h:
7281 client: small cleanup
7283 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7285 * docs/libs/gst-rtsp-server-sections.txt:
7286 * gst/rtsp-server/rtsp-client.c:
7287 * gst/rtsp-server/rtsp-client.h:
7288 * gst/rtsp-server/rtsp-server.c:
7289 client: remove reference to server
7290 We don't need to keep a ref to the server
7292 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7294 * gst/rtsp-server/rtsp-client.c:
7295 * gst/rtsp-server/rtsp-client.h:
7297 Also add some g_return_if()
7299 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7301 * gst/rtsp-server/rtsp-client.c:
7302 client: log more errors
7304 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7306 * gst/rtsp-server/rtsp-client.c:
7307 client: fix compilation
7309 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7311 * gst/rtsp-server/rtsp-client.c:
7312 * gst/rtsp-server/rtsp-client.h:
7313 client: add generic close-after-send support
7314 Add a property to send_response() to close the connection after the response has
7315 been sent to the client.
7317 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7320 * docs/libs/gst-rtsp-server-docs.sgml:
7321 * docs/libs/gst-rtsp-server-sections.txt:
7322 * docs/libs/gst-rtsp-server.types:
7323 * examples/test-auth.c:
7324 * examples/test-launch.c:
7325 * examples/test-mp4.c:
7326 * examples/test-multicast.c:
7327 * examples/test-multicast2.c:
7328 * examples/test-ogg.c:
7329 * examples/test-readme.c:
7330 * examples/test-sdp.c:
7331 * examples/test-uri.c:
7332 * examples/test-video.c:
7333 * gst/rtsp-server/Makefile.am:
7334 * gst/rtsp-server/rtsp-auth.h:
7335 * gst/rtsp-server/rtsp-client.c:
7336 * gst/rtsp-server/rtsp-client.h:
7337 * gst/rtsp-server/rtsp-media-mapping.c:
7338 * gst/rtsp-server/rtsp-media-mapping.h:
7339 * gst/rtsp-server/rtsp-mount-points.c:
7340 * gst/rtsp-server/rtsp-mount-points.h:
7341 * gst/rtsp-server/rtsp-server.c:
7342 * gst/rtsp-server/rtsp-server.h:
7343 * gst/rtsp-server/rtsp-session-media.c:
7344 * gst/rtsp-server/rtsp-session-pool.c:
7345 * gst/rtsp-server/rtsp-session-pool.h:
7346 * tests/check/gst/rtspserver.c:
7347 MediaMapping -> MountPoints
7348 Describes better what the object manages.
7350 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7353 configure: bump required version of -base
7355 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7357 * gst/rtsp-server/rtsp-media.c:
7360 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7362 * gst/rtsp-server/rtsp-media.c:
7363 * gst/rtsp-server/rtsp-media.h:
7364 media: support more Range formats
7365 Use the new -base methods to convert the Range string into a seek start and stop
7368 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7370 * examples/test-launch.c:
7371 examples: fix whitespace
7373 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7375 * examples/test-auth.c:
7376 test-auth: add example of how to remove sessions
7377 Add an example of the session filter api.
7379 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7381 * examples/test-uri.c:
7382 test-uri: remove mapping example
7384 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7386 * examples/test-uri.c:
7387 test-uri: fix callback signature
7389 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7391 * gst/rtsp-server/rtsp-media-factory.c:
7392 factory: keep ref to factory while media active
7393 While the media from a factory is alive, keep a ref to the factory.
7394 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
7396 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7398 * gst/rtsp-server/rtsp-media-factory-uri.c:
7399 factory-uri: add some debug
7401 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7403 * gst/rtsp-server/rtsp-stream.c:
7404 stream: set udp sources to PLAYING
7405 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
7406 so that it doesn't cause our pipeline to produce ASYNC-DONE.
7408 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7410 * gst/rtsp-server/rtsp-media-factory-uri.c:
7411 factory-uri: take ref to factory
7412 Take a ref to the factory that we place in our list.
7414 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7416 * tests/Makefile.am:
7417 * tests/test-reuse.c:
7418 test: add test for server reuse
7419 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
7421 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
7423 * gst/rtsp-server/rtsp-server.c:
7424 server: start and stop multiple times
7425 Stop listening on the RTSP port when the GSource is removed, so clients
7426 can't connect and the server can be started again.
7427 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
7429 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7431 * gst/rtsp-server/rtsp-server.c:
7432 server: fix small leak
7434 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7436 * gst/rtsp-server/rtsp-media.c:
7437 media: unref source in finish_unprepare
7438 The source is created in prepare, unref it in finish_unprepare.
7439 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
7441 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
7443 * gst/rtsp-server/rtsp-client.c:
7444 * gst/rtsp-server/rtsp-media.c:
7445 rtsp-media: remove bus watch before finalizing
7446 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
7447 * An extra media ref is added for the bus watch. This extra ref is unreffed by
7448 the GDestroyNotify function.
7449 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
7450 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
7451 gst_rtsp_media_unprepare before unreffing the media.
7452 This way, the bus watch will be removed before the media is finalized.
7453 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
7455 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
7457 * gst/rtsp-server/rtsp-client.c:
7458 * gst/rtsp-server/rtsp-client.h:
7459 client: wait until the TEARDOWN response is sent to close the connection
7460 Responses can be sent async so we need to wait until the TEARDOWN response has
7461 been written before we close the connection to the client. This avoids the risk
7462 of writing/polling closed sockets.
7463 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
7465 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
7467 * gst/rtsp-server/rtsp-stream.c:
7468 rtsp-stream: plug socket leak
7469 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
7471 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
7474 Automatic update of common submodule
7475 From 6bb6951 to a72faea
7477 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
7479 * gst/rtsp-server/rtsp-media-factory-uri.c:
7480 rtsp-server: don't use deprecated API
7482 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
7484 * gst/rtsp-server/rtsp-client.c:
7485 rtsp-client: fix unused-but-set-variable compiler warning
7486 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
7488 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7491 * docs/libs/gst-rtsp-server-sections.txt:
7492 * gst/rtsp-server/rtsp-client.c:
7495 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7497 * examples/Makefile.am:
7498 * examples/test-multicast2.c:
7499 examples: add another multicast example
7500 Add an example for how to configure separate multicast ranges for each media
7503 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7505 * examples/test-multicast.c:
7508 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7510 * gst/rtsp-server/rtsp-client.c:
7511 * gst/rtsp-server/rtsp-media.c:
7512 * gst/rtsp-server/rtsp-session-media.c:
7513 * gst/rtsp-server/rtsp-session-media.h:
7514 * gst/rtsp-server/rtsp-stream-transport.c:
7515 * gst/rtsp-server/rtsp-stream-transport.h:
7516 stream: use the address managed by the stream
7517 Use the address managed by the stream for multicast. This allows us to have 1
7518 multicast address for each stream.
7519 Because the address is now managed by the stream we don't have to pass it around
7521 Set the address pool on the streams.
7523 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7525 * gst/rtsp-server/rtsp-client.c:
7526 * gst/rtsp-server/rtsp-media.c:
7527 * gst/rtsp-server/rtsp-stream.c:
7530 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7532 * gst/rtsp-server/rtsp-media.c:
7533 * gst/rtsp-server/rtsp-media.h:
7534 media: add signal for new streams
7535 This allows applications to listen for new streams and configure properties on
7536 them, like the address pool.
7538 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7540 * gst/rtsp-server/rtsp-media.c:
7541 media: configure address pool in new streams
7543 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7545 * gst/rtsp-server/rtsp-stream.c:
7546 * gst/rtsp-server/rtsp-stream.h:
7547 stream: add methods to deal with address pool
7548 Add methods to get and set the address pool for the stream
7549 Add method to allocate and get the multicast addresses for this stream.
7551 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7553 * docs/libs/gst-rtsp-server-sections.txt:
7554 * gst/rtsp-server/rtsp-media.c:
7555 * gst/rtsp-server/rtsp-media.h:
7556 media: remove MTU property
7557 It is a stream property
7559 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7561 * gst/rtsp-server/rtsp-client.c:
7562 client: set blocksize only on stream
7563 Set the blocksize only on the current stream.
7565 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7567 * gst/rtsp-server/rtsp-stream.c:
7568 stream: share src and sink sockets
7569 the allocated socket is in the used-socket property, not socket.
7571 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7573 * gst/rtsp-server/rtsp-address-pool.c:
7574 * gst/rtsp-server/rtsp-address-pool.h:
7575 * gst/rtsp-server/rtsp-client.c:
7576 * gst/rtsp-server/rtsp-session-media.c:
7577 * gst/rtsp-server/rtsp-session-media.h:
7578 * gst/rtsp-server/rtsp-stream-transport.c:
7579 * gst/rtsp-server/rtsp-stream-transport.h:
7580 * tests/check/gst/addresspool.c:
7581 rtsp: make address-pool return an address object
7582 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
7583 store more info in the structure and allows us to more easily return the address
7584 to the right pool when no longer needed.
7585 Pass the address to the StreamTransport so that we can return it to the pool
7586 when the stream transport is freed or changed.
7588 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7590 * examples/Makefile.am:
7591 * examples/test-multicast.c:
7592 examples: add multicast example
7593 Show how to set up the multicast address pool so that media can be
7594 server with multicast.
7596 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7598 * gst/rtsp-server/rtsp-client.c:
7599 * gst/rtsp-server/rtsp-media-factory.c:
7600 * gst/rtsp-server/rtsp-media-factory.h:
7601 * gst/rtsp-server/rtsp-media.c:
7602 * gst/rtsp-server/rtsp-media.h:
7603 rtsp: use AddressPool
7604 Remove the multicast_group property.
7605 Use the configured addresspool to allocate multicast addresses.
7607 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7609 * gst/rtsp-server/rtsp-address-pool.c:
7610 * gst/rtsp-server/rtsp-address-pool.h:
7611 address-pool: add clear method
7613 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7615 * gst/rtsp-server/rtsp-address-pool.c:
7616 address-pool: small cleanups
7618 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7620 * tests/check/Makefile.am:
7621 * tests/check/gst/addresspool.c:
7622 tests: add addresspool unit test
7624 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7626 * gst/rtsp-server/Makefile.am:
7627 * gst/rtsp-server/rtsp-address-pool.c:
7628 * gst/rtsp-server/rtsp-address-pool.h:
7629 address-pool: add object to manage multicast addresses
7630 Make an object that can manage a rage of multicast addresses and ports.
7632 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7634 * gst/rtsp-server/rtsp-server.c:
7635 server: set default max-threads property
7637 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7639 * gst/rtsp-server/rtsp-media.c:
7640 media: wait for concurrent _prepare
7641 If a prepare is busy, wait for the result.
7643 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7645 * gst/rtsp-server/rtsp-media.c:
7646 media: add lock around message handler
7647 We don't want to dispatch messages while we are still processing the result of
7650 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7652 * gst/rtsp-server/rtsp-media.c:
7653 * gst/rtsp-server/rtsp-media.h:
7654 media: add lock to protect state changes
7656 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7658 * gst/rtsp-server/rtsp-stream.c:
7659 * gst/rtsp-server/rtsp-stream.h:
7662 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7664 * gst/rtsp-server/rtsp-stream-transport.c:
7665 * gst/rtsp-server/rtsp-stream-transport.h:
7666 * gst/rtsp-server/rtsp-stream.c:
7667 stream-transport: add keep-alive method
7669 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7671 * gst/rtsp-server/rtsp-stream-transport.c:
7672 * gst/rtsp-server/rtsp-stream-transport.h:
7673 * gst/rtsp-server/rtsp-stream.c:
7674 stream-transport: add method to handle RTP/RTCP
7675 Call new methods instead of poking into the structures directly.
7677 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7679 * gst/rtsp-server/rtsp-session-media.c:
7680 * gst/rtsp-server/rtsp-session-media.h:
7681 session-media: add locking
7683 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7685 * gst/rtsp-server/rtsp-session.c:
7686 * gst/rtsp-server/rtsp-session.h:
7687 session: add locking
7689 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7691 * gst/rtsp-server/rtsp-server.c:
7692 server: free old socket
7694 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7696 * gst/rtsp-server/rtsp-media-mapping.c:
7697 * gst/rtsp-server/rtsp-media-mapping.h:
7698 mapping: add locking
7700 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7702 * gst/rtsp-server/rtsp-media-factory.c:
7703 media-factory: add locking
7705 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7707 * gst/rtsp-server/rtsp-auth.c:
7708 * gst/rtsp-server/rtsp-auth.h:
7711 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7713 * gst/rtsp-server/rtsp-server.c:
7714 * gst/rtsp-server/rtsp-server.h:
7715 server: add max-thread property
7717 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7719 * gst/rtsp-server/rtsp-server.c:
7720 * gst/rtsp-server/rtsp-server.h:
7721 server: use a threadpool for the mainloops
7723 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7725 * gst/rtsp-server/rtsp-client.c:
7726 * gst/rtsp-server/rtsp-client.h:
7727 client: rename method
7728 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
7729 don't really create the client from the socket, we use the socket for the
7732 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7734 * gst/rtsp-server/rtsp-client.c:
7735 * gst/rtsp-server/rtsp-client.h:
7736 * gst/rtsp-server/rtsp-server.c:
7737 server: rework maincontext handling in clients
7738 Make a separate method to attach a client to a MainContext.
7739 Let the server decide in what GMainContext the client will operate and give this
7740 context to the client in attach. Then the server can later decide to use a
7741 separate thread for each client or just use the mainthread.
7743 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7745 * gst/rtsp-server/rtsp-client.c:
7746 * gst/rtsp-server/rtsp-session.c:
7747 * gst/rtsp-server/rtsp-session.h:
7748 session: move session header code in session object
7750 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
7754 * examples/test-auth.c:
7755 * examples/test-launch.c:
7756 * examples/test-mp4.c:
7757 * examples/test-ogg.c:
7758 * examples/test-readme.c:
7759 * examples/test-sdp.c:
7760 * examples/test-uri.c:
7761 * examples/test-video.c:
7762 * gst/rtsp-server/rtsp-auth.c:
7763 * gst/rtsp-server/rtsp-auth.h:
7764 * gst/rtsp-server/rtsp-client.c:
7765 * gst/rtsp-server/rtsp-client.h:
7766 * gst/rtsp-server/rtsp-media-factory-uri.c:
7767 * gst/rtsp-server/rtsp-media-factory-uri.h:
7768 * gst/rtsp-server/rtsp-media-factory.c:
7769 * gst/rtsp-server/rtsp-media-factory.h:
7770 * gst/rtsp-server/rtsp-media-mapping.c:
7771 * gst/rtsp-server/rtsp-media-mapping.h:
7772 * gst/rtsp-server/rtsp-media.c:
7773 * gst/rtsp-server/rtsp-media.h:
7774 * gst/rtsp-server/rtsp-params.c:
7775 * gst/rtsp-server/rtsp-params.h:
7776 * gst/rtsp-server/rtsp-sdp.c:
7777 * gst/rtsp-server/rtsp-sdp.h:
7778 * gst/rtsp-server/rtsp-server.c:
7779 * gst/rtsp-server/rtsp-server.h:
7780 * gst/rtsp-server/rtsp-session-media.c:
7781 * gst/rtsp-server/rtsp-session-media.h:
7782 * gst/rtsp-server/rtsp-session-pool.c:
7783 * gst/rtsp-server/rtsp-session-pool.h:
7784 * gst/rtsp-server/rtsp-session.c:
7785 * gst/rtsp-server/rtsp-session.h:
7786 * gst/rtsp-server/rtsp-stream-transport.c:
7787 * gst/rtsp-server/rtsp-stream-transport.h:
7788 * gst/rtsp-server/rtsp-stream.c:
7789 * gst/rtsp-server/rtsp-stream.h:
7790 * tests/check/gst/rtspserver.c:
7791 * tests/test-cleanup.c:
7794 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7796 * gst/rtsp-server/rtsp-media.c:
7797 * gst/rtsp-server/rtsp-session-media.c:
7798 * gst/rtsp-server/rtsp-session.c:
7799 rtsp-server: added annotations to indicate type of ownership transfer of return values
7800 https://bugzilla.gnome.org/show_bug.cgi?id=680777
7802 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
7805 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
7807 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
7810 * bindings/Makefile.am:
7811 * bindings/vala/Makefile.am:
7812 * bindings/vala/gst-rtsp-server-0.10.deps:
7813 * bindings/vala/gst-rtsp-server-0.10.vapi:
7814 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
7815 * bindings/vala/packages/gst-rtsp-server-0.10.files:
7816 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7817 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7818 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
7820 bindings: remove vala bindings
7821 They'll be reunited with the other GStreamer bindings
7822 https://bugzilla.gnome.org/show_bug.cgi?id=680777
7824 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7826 * gst/rtsp-server/rtsp-client.c:
7827 * gst/rtsp-server/rtsp-session-media.c:
7828 * gst/rtsp-server/rtsp-session-media.h:
7829 * gst/rtsp-server/rtsp-stream-transport.c:
7830 * gst/rtsp-server/rtsp-stream-transport.h:
7831 rtsp: only create transport when needed
7832 Only create the StreamTransport when configured.
7834 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7836 * gst/rtsp-server/rtsp-client.c:
7837 client: small cleanup
7839 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7841 * gst/rtsp-server/rtsp-client.c:
7842 * gst/rtsp-server/rtsp-client.h:
7843 * gst/rtsp-server/rtsp-stream-transport.c:
7844 * gst/rtsp-server/rtsp-stream-transport.h:
7845 rtsp: refactor configuration of transport
7846 Move the configuration of the transport to a place where it makes
7849 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7851 * gst/rtsp-server/rtsp-client.c:
7852 client: refactor transport parsing
7854 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7856 * gst/rtsp-server/rtsp-client.c:
7857 client: refuse to change the MTU on shared media
7858 If we change the MTU of chared media, it changes for all clients.
7859 We don't want to set the MTU to something large for clients that
7862 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7864 * examples/test-mp4.c:
7865 * gst/rtsp-server/rtsp-media.c:
7866 small fixes to docs and debug
7868 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7870 * gst/rtsp-server/rtsp-stream.c:
7871 stream: transports must already have been removed
7873 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7875 * gst/rtsp-server/rtsp-media.c:
7876 * gst/rtsp-server/rtsp-stream.c:
7877 * gst/rtsp-server/rtsp-stream.h:
7878 stream: improve join and leave of the pipeline
7880 Do the cleanup properly
7883 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7885 * gst/rtsp-server/rtsp-media.c:
7886 media: move unprepare below default implementation
7887 Makes it easier to find the default implementation
7889 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7891 * gst/rtsp-server/rtsp-media.c:
7892 media: signal unprepared when we actually finish
7894 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7896 * gst/rtsp-server/rtsp-media.c:
7897 media: no need to unlock, unprepare does that when needed
7899 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7901 * docs/libs/gst-rtsp-server-sections.txt:
7902 * gst/rtsp-server/rtsp-media-factory.h:
7903 * gst/rtsp-server/rtsp-media-mapping.c:
7904 * gst/rtsp-server/rtsp-media.h:
7905 * gst/rtsp-server/rtsp-params.c:
7906 * gst/rtsp-server/rtsp-server.c:
7907 * gst/rtsp-server/rtsp-session-pool.h:
7908 * gst/rtsp-server/rtsp-session.c:
7909 * gst/rtsp-server/rtsp-session.h:
7910 * gst/rtsp-server/rtsp-stream-transport.h:
7911 * gst/rtsp-server/rtsp-stream.h:
7914 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7916 * gst/rtsp-server/rtsp-client.c:
7917 * gst/rtsp-server/rtsp-media-mapping.h:
7918 * gst/rtsp-server/rtsp-media.c:
7919 * gst/rtsp-server/rtsp-media.h:
7920 * gst/rtsp-server/rtsp-server.h:
7921 * gst/rtsp-server/rtsp-stream.c:
7922 * gst/rtsp-server/rtsp-stream.h:
7923 rtsp: fix MTU setting
7924 Fix setting of the MTU. There is no need for a vmethod.
7926 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7931 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7934 configure: bump version number after refactoring
7936 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7938 * gst/rtsp-server/Makefile.am:
7939 * gst/rtsp-server/rtsp-client.c:
7940 * gst/rtsp-server/rtsp-client.h:
7941 * gst/rtsp-server/rtsp-media-factory-uri.c:
7942 * gst/rtsp-server/rtsp-media-factory.c:
7943 * gst/rtsp-server/rtsp-media-factory.h:
7944 * gst/rtsp-server/rtsp-media.c:
7945 * gst/rtsp-server/rtsp-media.h:
7946 * gst/rtsp-server/rtsp-sdp.c:
7947 * gst/rtsp-server/rtsp-session-media.c:
7948 * gst/rtsp-server/rtsp-session-media.h:
7949 * gst/rtsp-server/rtsp-session.c:
7950 * gst/rtsp-server/rtsp-session.h:
7951 * gst/rtsp-server/rtsp-stream-transport.c:
7952 * gst/rtsp-server/rtsp-stream-transport.h:
7953 * gst/rtsp-server/rtsp-stream.c:
7954 * gst/rtsp-server/rtsp-stream.h:
7955 rtsp: massive refactoring
7956 Make GObjects from the remaining simple structures.
7957 Remove GstRTSPSessionStream, it's not needed.
7958 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
7959 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
7960 a GstRTSPStream should be transported to a client.
7961 Rename GstRTSPMediaFactory::get_element -> create_element because that
7962 more accurately describes what it does.
7963 Make nice methods instead of poking in the structures.
7964 Move some methods inside the relevant object source code.
7965 Use GPtrArray to store objects instead of plain arrays, it is more
7966 natural and allows us to more easily clean up.
7967 Move the allocation of udp ports to the Stream object. The Stream object
7968 contains the elements needed to stream the media to a client.
7969 Improve the prepare and unprepare methods. Unprepare should now undo
7970 everything prepare did. Improve also async unprepare when doing EOS on
7971 shutdown. Make sure we always unprepare correctly.
7973 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
7975 * gst/rtsp-server/rtsp-client.c:
7976 rtsp-client: Unref server address clients connected to
7977 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
7979 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
7981 * gst/rtsp-server/rtsp-server.c:
7982 rtsp-server: don't ref server socket if it is NULL
7983 Fixes test_bind_already_in_use unit test again after commit 6a497440.
7984 https://bugzilla.gnome.org/show_bug.cgi?id=686644
7986 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
7988 * tests/check/Makefile.am:
7989 tests: Add libgio link dependency
7990 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
7992 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7994 * gst/rtsp-server/rtsp-media-mapping.c:
7995 * gst/rtsp-server/rtsp-media-mapping.h:
7996 rtsp-media-mapping: rename find_media vfunc to find_factory
7997 The virtual method and class method should have the same name
7998 so it is correctly represented in GIR file
7999 https://bugzilla.gnome.org/show_bug.cgi?id=680777
8001 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8003 * gst/rtsp-server/rtsp-auth.c:
8004 * gst/rtsp-server/rtsp-client.c:
8005 * gst/rtsp-server/rtsp-media-factory-uri.c:
8006 * gst/rtsp-server/rtsp-media-factory.c:
8007 * gst/rtsp-server/rtsp-media-mapping.c:
8008 * gst/rtsp-server/rtsp-media.c:
8009 * gst/rtsp-server/rtsp-server.c:
8010 * gst/rtsp-server/rtsp-session-pool.c:
8011 * gst/rtsp-server/rtsp-session.c:
8012 rtsp-server: fixed comments and GIR annotations
8013 https://bugzilla.gnome.org/show_bug.cgi?id=680777
8015 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
8017 * gst/rtsp-server/rtsp-media-mapping.c:
8018 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
8020 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
8022 * gst/rtsp-server/rtsp-server.c:
8023 rtsp-server: allow binding on port 0 (binds on a random port)
8025 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
8027 * gst/rtsp-server/rtsp-server.c:
8028 * gst/rtsp-server/rtsp-server.h:
8029 rtsp-server: add bound-port property
8030 bound-port can be used to retrieve the port number when the server is bound on
8031 port 0, which binds on a random port.
8033 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
8035 * gst/rtsp-server/rtsp-media-factory.c:
8036 * gst/rtsp-server/rtsp-media-factory.h:
8037 rtsp-media-factory: make ::get_element overridable by GI bindings
8038 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
8039 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
8040 as the invoker for ::get_element(), making it overridable by GI generated
8043 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
8045 * gst/rtsp-server/rtsp-media-factory-uri.c:
8046 rtsp-media-factory-uri: don't autoplug parsers in a loop
8047 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
8050 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
8052 * gst/rtsp-server/Makefile.am:
8053 Explicitly link against gio. Fix link error on mac.
8055 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
8057 * gst/rtsp-server/rtsp-session.c:
8058 session: add ttl to the transport header in SETUP
8059 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
8061 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
8063 * gst/rtsp-server/rtsp-client.c:
8064 * gst/rtsp-server/rtsp-client.h:
8065 * gst/rtsp-server/rtsp-media.c:
8066 client: Use client transport settings for multicast if allowed.
8067 This patch makes it possible for the client to send transport settings for
8068 multicast (destination && ttl). Client settings must be explicitly allowed or
8069 the server will use its own settings.
8070 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
8072 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
8075 Automatic update of common submodule
8076 From 6c0b52c to 6bb6951
8078 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
8080 * gst/rtsp-server/rtsp-client.c:
8081 rtsp-client: do not destroy the rtsp watch
8082 Don't destroy the client watch while dispatching. The rtsp watch is
8083 automatically destroyed after the rtsp watch function closed() has
8085 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
8087 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
8090 Automatic update of common submodule
8091 From 4f962f7 to 6c0b52c
8093 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
8095 * gst/rtsp-server/rtsp-media.c:
8096 media: fix check for seekability
8098 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8100 * gst/rtsp-server/rtsp-client.c:
8101 client: use more GIO
8102 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
8104 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8106 * gst/rtsp-server/rtsp-server.c:
8107 server: remove obsolete includes
8109 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
8111 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
8112 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
8113 be available in "on_new_ssrc". The transports are added in
8114 gst_rtsp_media_set_state when going to PLAYING state. However,
8115 "on_new_ssrc" might be called before this happens.
8116 https://bugzilla.gnome.org/show_bug.cgi?id=683304
8118 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
8120 * gst/rtsp-server/rtsp-client.c:
8121 * gst/rtsp-server/rtsp-client.h:
8122 rtsp-client: add signals for rtsp requests (fixes #683287)
8124 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
8126 * gst/rtsp-server/rtsp-client.c:
8127 * gst/rtsp-server/rtsp-client.h:
8128 add new-session signal to rtsp-client (fixes #683058)
8130 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
8133 Automatic update of common submodule
8134 From 668acee to 4f962f7
8136 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
8138 * gst/rtsp-server/rtsp-server.c:
8139 * tests/check/gst/rtspserver.c:
8140 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
8141 Do not assume that *error is set in g_socket_address_enumerator_next.
8142 Added test_bind_already_in_use unit-test.
8143 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
8145 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
8148 Automatic update of common submodule
8149 From 94ccf4c to 668acee
8151 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
8153 * gst/rtsp-server/rtsp-client.c:
8154 * gst/rtsp-server/rtsp-client.h:
8155 rtsp-client: make create_sdp virtual method
8156 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
8158 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8161 Automatic update of common submodule
8162 From 98e386f to 94ccf4c
8164 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8166 * gst/rtsp-server/rtsp-client.c:
8169 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
8171 * gst/rtsp-server/rtsp-client.c:
8172 * gst/rtsp-server/rtsp-client.h:
8173 * gst/rtsp-server/rtsp-server.c:
8174 * gst/rtsp-server/rtsp-server.h:
8175 rtsp-server: use an existing socket to establish HTTP tunnel
8176 Make it possible to transfer a socket from an HTTP server to be used as
8177 an RTSP over HTTP tunnel.
8179 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
8181 * gst/rtsp-server/rtsp-client.c:
8182 * gst/rtsp-server/rtsp-media.c:
8183 * gst/rtsp-server/rtsp-media.h:
8184 rtsp: Handle the blocksize parameter
8185 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
8187 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
8189 * tests/check/Makefile.am:
8190 * tests/check/gst/rtspserver.c:
8191 Have unit test get header from source dir, not installed dir
8192 This makes compilation of unit tests work in a build directory other
8193 than the source directory.
8194 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
8196 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
8198 * gst/rtsp-server/rtsp-media.c:
8199 rtsp-media: update for gst_element_make_from_uri() changes
8201 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
8204 * tests/Makefile.am:
8205 * tests/check/Makefile.am:
8206 * tests/check/gst/rtspserver.c:
8208 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
8210 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
8212 * gst/rtsp-server/rtsp-media.c:
8213 rtsp-media: don't collect media stats when going to NULL
8214 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
8216 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8218 * gst/rtsp-server/rtsp-client.c:
8219 client: don't leak transports
8221 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
8223 * gst/rtsp-server/rtsp-client.c:
8224 rtsp-client: free transport on no_stream in SETUP handler
8226 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
8228 * gst/rtsp-server/rtsp-client.c:
8229 rtsp-client: changed session media iteration
8230 In client_unlink_session: now don't iterate in session->medias
8231 list where items are removed by gst_rtsp_session_release_media.
8232 Instead, repeatedly remove the first item.
8234 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
8236 * gst/rtsp-server/rtsp-client.c:
8237 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
8238 GstRTSPSessionMedia is not a GObject type. When the
8239 GstRTSPSession is freed, it will free the media.
8241 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
8243 * gst/rtsp-server/rtsp-media-factory.c:
8244 factory: plug pad leak in collect_streams
8245 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
8246 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
8247 will take one reference, and the other reference will otherwise
8250 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
8253 configure: suppress some warnings when debug is disabled
8254 Warnings about unused variables should be suppressed if core has the
8255 debug system disabled.
8256 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
8258 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8260 * docs/libs/Makefile.am:
8261 docs: fix build in uninstalled setup
8262 Include gst-plugins-base libs properly.
8264 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
8266 * docs/libs/gst-rtsp-server.types:
8267 docs: include headers defining rtsp-server object types
8268 Fixes compiler warnings during docs build.
8269 https://bugzilla.gnome.org/show_bug.cgi?id=676824
8271 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
8274 configure: Add warning flags for compiler when configuring
8275 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
8277 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
8280 Automatic update of common submodule
8281 From 03a0e57 to 98e386f
8283 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
8286 Automatic update of common submodule
8287 From 1fab359 to 03a0e57
8289 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
8291 * gst/rtsp-server/rtsp-client.c:
8292 client: fix GSocketAddress leak in gst_rtsp_client_accept
8293 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
8295 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
8298 Automatic update of common submodule
8299 From f1b5a96 to 1fab359
8301 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8304 Automatic update of common submodule
8305 From 92b7266 to f1b5a96
8307 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8310 Automatic update of common submodule
8311 From ec1c4a8 to 92b7266
8313 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8316 Automatic update of common submodule
8317 From 3429ba6 to ec1c4a8
8319 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
8321 * gst/rtsp-server/rtsp-auth.c:
8322 * gst/rtsp-server/rtsp-client.c:
8323 * gst/rtsp-server/rtsp-media-factory-uri.c:
8324 * gst/rtsp-server/rtsp-server.c:
8325 rtsp: fix compiler warnings
8326 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
8328 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8331 Automatic update of common submodule
8332 From dc70203 to 3429ba6
8334 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8336 * gst/rtsp-server/rtsp-client.c:
8337 * gst/rtsp-server/rtsp-media-factory.c:
8338 * gst/rtsp-server/rtsp-media-factory.h:
8339 * gst/rtsp-server/rtsp-media.c:
8340 * gst/rtsp-server/rtsp-media.h:
8341 * gst/rtsp-server/rtsp-server.c:
8342 * gst/rtsp-server/rtsp-server.h:
8343 * gst/rtsp-server/rtsp-session-pool.c:
8344 * gst/rtsp-server/rtsp-session-pool.h:
8345 rtsp-server: port to new thread API
8347 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8350 Automatic update of common submodule
8351 From 6db25be to dc70203
8353 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8355 * gst/rtsp-server/rtsp-auth.c:
8356 * gst/rtsp-server/rtsp-auth.h:
8357 * gst/rtsp-server/rtsp-client.c:
8358 rtsp-server: Fix compilation and compiler warnings
8360 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8364 * gst/rtsp-server/Makefile.am:
8365 configure: Modernize autotools setup a bit
8366 Also we now only create tar.bz2 and tar.xz tarballs.
8368 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8371 Automatic update of common submodule
8372 From 464fe15 to 6db25be
8374 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8377 Automatic update of common submodule
8378 From 7fda524 to 464fe15
8380 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8383 * docs/libs/Makefile.am:
8384 * docs/version.entities.in:
8386 * gst/rtsp-server/Makefile.am:
8387 * pkgconfig/Makefile.am:
8388 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
8389 * pkgconfig/gstreamer-rtsp-server.pc.in:
8390 * tests/Makefile.am:
8391 rtsp-server: Update versioning
8393 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8395 Merge remote-tracking branch 'origin/0.10'
8397 gst/rtsp-server/rtsp-session-pool.c
8399 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8401 * gst/rtsp-server/rtsp-session-pool.c:
8402 rtsp-server: Don't use deprecated GLib API
8404 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8406 Replace master with 0.11
8408 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8410 Merge branch 'master' into 0.11
8412 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8414 Merge branch 'master' into 0.11
8416 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
8419 A couple minor typo fixes
8421 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8423 * gst/rtsp-server/rtsp-media.c:
8424 media: fix state of the appqueue
8426 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8428 * gst/rtsp-server/rtsp-media-factory-uri.c:
8429 factory: use videoconvert
8431 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8433 * gst/rtsp-server/rtsp-media-factory-uri.c:
8434 factory: change to new style caps
8436 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8438 * gst/rtsp-server/rtsp-client.c:
8439 * gst/rtsp-server/rtsp-client.h:
8440 * gst/rtsp-server/rtsp-media-factory-uri.c:
8441 * gst/rtsp-server/rtsp-media.c:
8442 * gst/rtsp-server/rtsp-server.c:
8443 * gst/rtsp-server/rtsp-server.h:
8444 * gst/rtsp-server/rtsp-session-pool.c:
8445 rtsp-server: port to GIO
8448 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8451 configure: fix build
8453 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8456 docs: fix for gst_rtsp_server_set_port() -> _set_service()
8457 https://bugzilla.gnome.org/show_bug.cgi?id=666548
8459 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8462 * examples/Makefile.am:
8463 First rule of gst-rtsp-server club: don't talk about gst-phonon
8465 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8468 * pkgconfig/Makefile.am:
8469 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
8470 * pkgconfig/gstreamer-rtsp-server.pc.in:
8471 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
8472 For consistency with all other modules.
8474 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8476 * gst/rtsp-server/rtsp-client.c:
8477 rtsp-client: update for new map API
8479 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8482 * bindings/Makefile.am:
8483 * bindings/python/Makefile.am:
8484 * bindings/python/arg-types.py:
8485 * bindings/python/codegen/Makefile.am:
8486 * bindings/python/codegen/__init__.py:
8487 * bindings/python/codegen/argtypes.py:
8488 * bindings/python/codegen/code-coverage.py:
8489 * bindings/python/codegen/codegen.py:
8490 * bindings/python/codegen/definitions.py:
8491 * bindings/python/codegen/defsparser.py:
8492 * bindings/python/codegen/docextract.py:
8493 * bindings/python/codegen/docgen.py:
8494 * bindings/python/codegen/fileprefix.override:
8495 * bindings/python/codegen/fileprefixmodule.c:
8496 * bindings/python/codegen/h2def.py:
8497 * bindings/python/codegen/mergedefs.py:
8498 * bindings/python/codegen/mkskel.py:
8499 * bindings/python/codegen/override.py:
8500 * bindings/python/codegen/reversewrapper.py:
8501 * bindings/python/codegen/scmexpr.py:
8502 * bindings/python/rtspserver-types.defs:
8503 * bindings/python/rtspserver.defs:
8504 * bindings/python/rtspserver.override:
8505 * bindings/python/rtspservermodule.c:
8506 * bindings/python/test.py:
8508 python: remove pygst-based python bindings
8509 pygi is the future, apparently.
8511 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
8514 Automatic update of common submodule
8515 From c463bc0 to 7fda524
8517 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8520 Automatic update of common submodule
8521 From 2a59016 to c463bc0
8523 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8526 Automatic update of common submodule
8527 From 0807187 to 2a59016
8529 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8532 Automatic update of common submodule
8533 From 11f0cd5 to 0807187
8535 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8537 * examples/test-auth.c:
8538 example: update for new caps
8540 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8542 * examples/test-video.c:
8543 * gst/rtsp-server/rtsp-client.c:
8544 * gst/rtsp-server/rtsp-media-factory-uri.c:
8545 * gst/rtsp-server/rtsp-media.c:
8546 * gst/rtsp-server/rtsp-media.h:
8547 * gst/rtsp-server/rtsp-session.c:
8548 * gst/rtsp-server/rtsp-session.h:
8549 rtsp-server: port some more to 0.11
8551 Remove bufferlist stuff
8553 Add queue before appsink now that preroll-queue-len is gone.
8554 Update for request pad changes.
8556 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8558 Merge branch 'master' into 0.11
8560 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
8562 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8563 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
8564 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
8566 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
8568 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8569 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
8570 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
8572 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8574 Merge branch 'master' into 0.11
8576 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8578 * gst/rtsp-server/rtsp-media.c:
8579 * gst/rtsp-server/rtsp-media.h:
8580 media: add a seekable boolean
8581 Maintain the seekable state with a new variable instead of reusing the
8584 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
8586 * gst/rtsp-server/rtsp-media.c:
8587 Disallow seek in live media
8589 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8591 Merge branch 'master' into 0.11
8593 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
8595 * gst/rtsp-server/rtsp-server.c:
8596 #ifdef statements for windows socket creation were missing
8598 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
8601 Automatic update of common submodule
8602 From a39eb83 to 11f0cd5
8604 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
8607 Automatic update of common submodule
8608 From 605cd9a to a39eb83
8610 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8612 Merge branch 'master' into 0.11
8614 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8616 * gst/rtsp-server/rtsp-client.c:
8617 client: use method to access property
8619 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8621 * gst/rtsp-server/rtsp-media-factory.c:
8622 * gst/rtsp-server/rtsp-media-factory.h:
8623 media-factory: add protocols property
8624 Add a property to configure the allowed protocols in the media created from the
8627 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8629 * gst/rtsp-server/rtsp-media-factory.c:
8630 * gst/rtsp-server/rtsp-media-factory.h:
8631 media-factory: add media-configure signal
8632 Add signal to allow the application to configure the media after it was created
8635 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8637 * gst/rtsp-server/rtsp-client.c:
8638 client: use method to access property
8640 2011-08-16 15:15:19 +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 protocols property
8645 Add a property to configure the allowed protocols in the media created from the
8648 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8650 * gst/rtsp-server/rtsp-media-factory.c:
8651 * gst/rtsp-server/rtsp-media-factory.h:
8652 media-factory: add media-configure signal
8653 Add signal to allow the application to configure the media after it was created
8656 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8658 Merge branch 'master' into 0.11
8660 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8662 * gst/rtsp-server/rtsp-client.c:
8663 client: use media multicast group
8665 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8667 * gst/rtsp-server/rtsp-media-factory.h:
8668 * gst/rtsp-server/rtsp-server.h:
8669 * gst/rtsp-server/rtsp-session-pool.h:
8670 * gst/rtsp-server/rtsp-session.h:
8673 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
8675 * gst/rtsp-server/rtsp-client.c:
8676 * gst/rtsp-server/rtsp-sdp.h:
8677 sdp: copy and free the server ip address
8678 Copy and free the server ip address to make memory management easier later.
8680 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8682 * gst/rtsp-server/rtsp-media-factory.c:
8683 media-factory: configure multicast in media
8685 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8687 * gst/rtsp-server/rtsp-media.c:
8688 * gst/rtsp-server/rtsp-media.h:
8689 media: add property for multicast group
8690 Add a property to configure the multicast group in the media.
8691 Based on patches from Marc Leeman and Robert Krakora.
8693 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8695 * gst/rtsp-server/rtsp-media-factory.c:
8696 * gst/rtsp-server/rtsp-media-factory.h:
8697 media-factory: add property for multicast group
8698 Add a property to configure the multicast group in the media factory.
8699 Based on patches from Marc Leeman and Robert Krakora.
8701 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8703 * gst/rtsp-server/rtsp-client.c:
8704 client: do configuration of transport in one place
8705 Move the configuration of the transport destination address to where we also
8706 configure the other bits.
8708 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8710 * gst/rtsp-server/rtsp-client.c:
8711 client: use media multicast group
8713 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8715 * gst/rtsp-server/rtsp-media-factory.h:
8716 * gst/rtsp-server/rtsp-server.h:
8717 * gst/rtsp-server/rtsp-session-pool.h:
8718 * gst/rtsp-server/rtsp-session.h:
8721 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
8723 * gst/rtsp-server/rtsp-client.c:
8724 * gst/rtsp-server/rtsp-sdp.h:
8725 sdp: copy and free the server ip address
8726 Copy and free the server ip address to make memory management easier later.
8728 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8730 * gst/rtsp-server/rtsp-media-factory.c:
8731 media-factory: configure multicast in media
8733 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8735 * gst/rtsp-server/rtsp-media.c:
8736 * gst/rtsp-server/rtsp-media.h:
8737 media: add property for multicast group
8738 Add a property to configure the multicast group in the media.
8739 Based on patches from Marc Leeman and Robert Krakora.
8741 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8743 * gst/rtsp-server/rtsp-media-factory.c:
8744 * gst/rtsp-server/rtsp-media-factory.h:
8745 media-factory: add property for multicast group
8746 Add a property to configure the multicast group in the media factory.
8747 Based on patches from Marc Leeman and Robert Krakora.
8749 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8751 * gst/rtsp-server/rtsp-client.c:
8752 client: do configuration of transport in one place
8753 Move the configuration of the transport destination address to where we also
8754 configure the other bits.
8756 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8758 Merge branch 'master' into 0.11
8760 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
8762 * gst/rtsp-server/rtsp-client.c:
8763 client: destroy pipeline on client disconnect with no prior TEARDOWN.
8764 The problem occurs when the client abruptly closes the connection without
8765 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
8766 server is where the pipeline gets torn down. Since this handler is not called,
8767 the pipeline remains and is up and running. Subsequent clients get their own
8768 pipelines and if the do not issue TEARDOWNs then those pipelines will also
8769 remain up and running. This is a resource leak.
8771 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8773 Merge branch 'master' into 0.11
8775 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
8777 * gst/rtsp-server/rtsp-media-factory.c:
8778 * gst/rtsp-server/rtsp-media-factory.h:
8779 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
8780 For example, it can be used to retrieve source elements like appsrc, in a more
8781 convenient way than subclassing get_element.
8783 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8785 Merge branch 'master' into 0.11
8787 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
8789 * gst/rtsp-server/rtsp-server.c:
8790 rtsp-server: hold on to reference while using object
8792 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8794 * gst/rtsp-server/rtsp-media.c:
8797 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8800 configure: use unstable api
8802 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
8804 * gst/rtsp-server/rtsp-client.c:
8805 client: fix reference counting
8807 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
8809 * gst/rtsp-server/rtsp-client.c:
8810 * gst/rtsp-server/rtsp-media.c:
8811 fix compiler warnings about unused variables
8813 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
8815 * examples/test-launch.c:
8816 * examples/test-readme.c:
8817 * examples/test-uri.c:
8818 * examples/test-video.c:
8819 examples: tell rtsp uri when ready
8821 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
8824 Automatic update of common submodule
8825 From 69b981f to 605cd9a
8827 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8829 * gst/rtsp-server/rtsp-client.c:
8830 client: update for buffer API change
8832 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
8834 * gst/rtsp-server/Makefile.am:
8835 Makefile.am: 0.10 => @GST_MAJORMINOR@
8837 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
8839 * gst/rtsp-server/rtsp-media-factory-uri.c:
8840 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
8842 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
8844 * gst/rtsp-server/.gitignore:
8845 .gitignore: 0.10 => 0.11
8847 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
8849 * gst/rtsp-server/Makefile.am:
8850 Makefile.am: 0.10 => @GST_MAJORMINOR@
8852 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8854 Merge branch 'master' into 0.11
8856 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
8859 Automatic update of common submodule
8860 From 9e5bbd5 to 69b981f
8862 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
8865 Automatic update of common submodule
8866 From fd35073 to 9e5bbd5
8868 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
8871 Automatic update of common submodule
8872 From 46dfcea to fd35073
8874 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8876 * gst/rtsp-server/rtsp-media-factory-uri.c:
8877 * gst/rtsp-server/rtsp-media.c:
8878 media: port to new caps API
8880 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8882 Merge branch 'master' into 0.11
8884 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
8886 * bindings/vala/gst-rtsp-server-0.10.vapi:
8887 Updated Vala bindings.
8888 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
8890 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
8892 * gst/rtsp-server/rtsp-server.c:
8893 * gst/rtsp-server/rtsp-server.h:
8894 Add a signal for newly connected clients.
8895 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
8897 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
8899 * bindings/python/rtspserver.override:
8900 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
8902 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8904 * gst/rtsp-server/Makefile.am:
8905 * gst/rtsp-server/rtsp-client.c:
8906 * gst/rtsp-server/rtsp-funnel.c:
8907 * gst/rtsp-server/rtsp-funnel.h:
8908 * gst/rtsp-server/rtsp-media.c:
8909 rtsp-server: port to 0.11
8911 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8916 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8918 Merge branch 'master' into 0.11
8923 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8926 Automatic update of common submodule
8927 From c3cafe1 to 46dfcea
8929 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
8931 * bindings/python/Makefile.am:
8932 * bindings/python/rtspserver.defs:
8933 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
8935 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
8937 * bindings/python/arg-types.py:
8938 python bindings: add GstRTSPUrlParam
8939 Needed to implement MediaFactory virtual proxies
8941 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
8943 * bindings/python/arg-types.py:
8944 python bindings: fix returning GstRTSPUrl types
8946 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
8948 * bindings/python/arg-types.py:
8949 python bindings: add arg type for GstRTSPUrl
8951 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
8953 * bindings/python/rtspserver.defs:
8954 python bindings: fix the definition of MediaFactory.collect_stream
8956 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
8959 Automatic update of common submodule
8960 From 1ccbe09 to c3cafe1
8962 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8965 Automatic update of common submodule
8966 From 193b717 to 1ccbe09
8968 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
8971 Automatic update of common submodule
8972 From b77e2bf to 193b717
8974 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8977 build: Include lcov.mak to allow test coverage report generation
8979 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8982 Automatic update of common submodule
8983 From d8814b6 to b77e2bf
8985 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8988 Automatic update of common submodule
8989 From 6aaa286 to d8814b6
8991 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
8994 Automatic update of common submodule
8995 From 6aec6b9 to 6aaa286
8997 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
9000 autogen: wingo signed comment
9002 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
9004 * gst/rtsp-server/rtsp-session-pool.c:
9005 session: use full charset for RTSP session ID
9006 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
9007 session ID more difficult.
9008 https://bugzilla.gnome.org/show_bug.cgi?id=643812
9010 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9012 * gst/rtsp-server/Makefile.am:
9013 rtsp-server: Don't install the funnel header
9015 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
9018 Automatic update of common submodule
9019 From 1de7f6a to 6aec6b9
9021 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9024 configure: require core/base 0.10.31
9025 Needed at least for gst_plugin_feature_rank_compare_func().
9027 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
9030 Automatic update of common submodule
9031 From f94d739 to 1de7f6a
9033 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9035 * gst/rtsp-server/rtsp-media.c:
9036 media: remove more unused code
9038 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9040 * gst/rtsp-server/rtsp-media.c:
9041 * gst/rtsp-server/rtsp-media.h:
9042 media: remove duplicate filtering
9043 Remove the duplicate filtering code now that we have a released -good version.
9044 Give a warning instead.
9046 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9048 * gst/rtsp-server/rtsp-media-factory.c:
9049 * gst/rtsp-server/rtsp-media.c:
9050 media: fix default buffer size
9052 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9054 * gst/rtsp-server/rtsp-media-factory.c:
9055 * gst/rtsp-server/rtsp-media-factory.h:
9056 media-factory: add property to configure the buffer-size
9057 Add a property to configure the kernel UDP buffer size.
9059 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9061 * gst/rtsp-server/rtsp-media.c:
9062 * gst/rtsp-server/rtsp-media.h:
9063 media: add property to configure kernel buffer sizes
9064 Add a property to configure the kernel UDP buffer size.
9066 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9069 configure: set PYGOBJECT_REQ before using it
9070 https://bugzilla.gnome.org/show_bug.cgi?id=640641
9072 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9075 docs: recursive into sub-directories on 'make upload'
9077 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9079 * docs/libs/gst-rtsp-server-docs.sgml:
9080 * docs/version.entities.in:
9081 docs: mention full version these docs are for, not just major-minor
9083 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9088 === release 0.10.8 ===
9090 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9095 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9097 * gst/rtsp-server/rtsp-server.c:
9098 rtsp-server: clarify docs a little
9100 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9102 * gst/rtsp-server/rtsp-media.c:
9103 media: init debug category before starting thread
9105 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9107 * gst/rtsp-server/rtsp-auth.c:
9108 auth: add realm to make it more spec compliant
9110 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9112 * gst/rtsp-server/rtsp-server.c:
9113 * gst/rtsp-server/rtsp-server.h:
9116 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9118 * examples/test-video.c:
9119 example: improve example docs a little
9121 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9123 * gst/rtsp-server/rtsp-server.c:
9124 server: ensure the watch has a ref to the server
9126 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9128 * gst/rtsp-server/rtsp-server.c:
9129 server: simpify channel function
9131 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9133 * gst/rtsp-server/rtsp-server.c:
9134 * gst/rtsp-server/rtsp-server.h:
9135 server: simplify management of channel and source
9136 We don't need to keep around the channel and source objects. Let the mainloop
9137 and the source manage the source and channel respectively.
9139 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9145 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9148 * tests/Makefile.am:
9149 * tests/test-cleanup.c:
9150 tests: add tests directory and cleanup test
9152 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9154 * gst/rtsp-server/rtsp-media-factory-uri.c:
9155 * gst/rtsp-server/rtsp-media-factory.c:
9156 * gst/rtsp-server/rtsp-media-mapping.c:
9157 * gst/rtsp-server/rtsp-media.c:
9158 * gst/rtsp-server/rtsp-session-pool.c:
9159 * gst/rtsp-server/rtsp-session.c:
9160 server: improve debugging in various objects
9162 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9164 * gst/rtsp-server/rtsp-server.c:
9165 server: chain up to the parent finalize
9167 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
9169 * bindings/python/rtspserver-types.defs:
9170 * bindings/python/rtspserver.defs:
9171 * bindings/python/rtspserver.override:
9172 * bindings/python/test.py:
9173 gst-rtsp-server: update python bindings
9175 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9177 * gst/rtsp-server/rtsp-client.c:
9178 client: use the response from the clientstate
9179 Create the response object only once and store in the client state.
9180 Make all methods use the state response,
9182 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9184 * gst/rtsp-server/rtsp-server.c:
9185 server: use signal to keep track of clients
9186 Keep track of all the clients that the server creates and remove them when they
9187 fire the 'closed' signal.
9189 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9191 * gst/rtsp-server/rtsp-client.c:
9192 * gst/rtsp-server/rtsp-client.h:
9193 client: emit signal when closing
9195 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9197 * examples/.gitignore:
9198 * examples/Makefile.am:
9199 * examples/test-auth.c:
9200 * examples/test-video.c:
9201 * gst/rtsp-server/rtsp-auth.c:
9202 * gst/rtsp-server/rtsp-auth.h:
9203 * gst/rtsp-server/rtsp-client.c:
9204 * gst/rtsp-server/rtsp-media-factory.c:
9205 * gst/rtsp-server/rtsp-media.c:
9206 * gst/rtsp-server/rtsp-media.h:
9207 * gst/rtsp-server/rtsp-session-pool.h:
9208 * gst/rtsp-server/rtsp-session.h:
9209 media: enable per factory authorisations
9210 Allow for adding a GstRTSPAuth on the factory and media level and check
9211 permissions when accessing the factory.
9212 Add hints to the auth methods for future more fine grained authorisation.
9213 Add example application for per factory authentication.
9215 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9217 * gst/rtsp-server/rtsp-auth.c:
9218 * gst/rtsp-server/rtsp-auth.h:
9219 * gst/rtsp-server/rtsp-client.c:
9220 * gst/rtsp-server/rtsp-client.h:
9221 * gst/rtsp-server/rtsp-params.c:
9222 * gst/rtsp-server/rtsp-params.h:
9223 rtsp-server: Pass ClientState structure arround
9224 Pass the collected information for the ongoing request in a GstRTSPClientState
9225 structure that we can then pass around to simplify the method arguments. This
9226 will also be handy when we implement logging functionality.
9228 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9230 * gst/rtsp-server/rtsp-media-factory.c:
9231 * gst/rtsp-server/rtsp-media-factory.h:
9232 media-factory: add methods to configure authorisation
9234 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9236 * gst/rtsp-server/rtsp-client.c:
9237 client: unref auth in finalize
9239 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9241 * gst/rtsp-server/rtsp-server.c:
9242 server: unref auth in finalize
9244 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9246 * docs/libs/gst-rtsp-server-docs.sgml:
9247 * docs/libs/gst-rtsp-server-sections.txt:
9248 * docs/libs/gst-rtsp-server.types:
9251 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9253 * gst/rtsp-server/rtsp-server.c:
9254 * gst/rtsp-server/rtsp-server.h:
9255 server: separate create and accept
9256 Create separate create and accept methods so that subclasses can create custom
9258 Configure the server in the client object and prepare for keeping track of
9261 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9263 * gst/rtsp-server/rtsp-client.c:
9264 * gst/rtsp-server/rtsp-client.h:
9265 client: add support for setting the server.
9266 Add support for keeping a ref to the server that started this client
9269 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9271 * gst/rtsp-server/rtsp-auth.c:
9272 auth: fix memleak and add some docs
9273 Fix a memleak of the basic auth token.
9274 Add docs for the helper function
9276 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9278 * gst/rtsp-server/rtsp-auth.c:
9279 * gst/rtsp-server/rtsp-auth.h:
9280 * gst/rtsp-server/rtsp-client.c:
9281 client: delegate setup of auth to the manager
9282 Delegate the configuration of the authentication tokens to the manager object
9285 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9287 * examples/test-video.c:
9288 * gst/rtsp-server/Makefile.am:
9289 * gst/rtsp-server/rtsp-auth.c:
9290 * gst/rtsp-server/rtsp-auth.h:
9291 * gst/rtsp-server/rtsp-client.c:
9292 * gst/rtsp-server/rtsp-client.h:
9293 * gst/rtsp-server/rtsp-server.c:
9294 * gst/rtsp-server/rtsp-server.h:
9295 auth: add authentication object
9296 Add an object that can check the authorization of requests.
9297 Implement basic authentication.
9298 Add example authentication to test-video
9300 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9302 * gst/rtsp-server/rtsp-server.c:
9303 * gst/rtsp-server/rtsp-server.h:
9304 server: move includes back
9305 the includes are needed for sockaddr_in.
9307 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9309 * gst/rtsp-server/rtsp-client.c:
9310 * gst/rtsp-server/rtsp-client.h:
9311 * gst/rtsp-server/rtsp-server.c:
9312 * gst/rtsp-server/rtsp-server.h:
9313 rtsp: move network includes where they are needed
9315 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
9317 * gst/rtsp-server/rtsp-media.h:
9318 rtsp-media.h: Minor corrections in comments.
9321 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
9324 Automatic update of common submodule
9325 From e572c87 to f94d739
9327 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
9331 * docs/libs/.gitignore:
9332 * examples/.gitignore:
9333 * gst/rtsp-server/.gitignore:
9336 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
9338 * docs/libs/Makefile.am:
9339 docs: We don't build ps/pdf for API reference docs
9341 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9344 Automatic update of common submodule
9345 From ccbaa85 to e572c87
9347 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9350 Automatic update of common submodule
9351 From 46445ad to ccbaa85
9353 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9355 * gst/rtsp-server/Makefile.am:
9356 * gst/rtsp-server/rtsp-funnel.c:
9357 * gst/rtsp-server/rtsp-funnel.h:
9358 * gst/rtsp-server/rtsp-media.c:
9359 funnel: rename fsfunnel to rtspfunnel
9360 Rename the funnel to avoid conflicts with the farsight one.
9362 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9364 * gst/rtsp-server/Makefile.am:
9365 * gst/rtsp-server/fs-funnel.c:
9366 * gst/rtsp-server/fs-funnel.h:
9367 * gst/rtsp-server/rtsp-media.c:
9368 rtsp-media: add and use fsfunnel
9369 Add a copy of fsfunnel to the build because input-selector removed the (broken)
9370 select-all property that we need.
9372 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9374 * gst/rtsp-server/Makefile.am:
9375 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
9376 Use PKG_CONFIG_PATH specified at configure time (if any) as well
9377 for the g-ir-compiler, rather than just assuming the env var has
9380 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9387 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
9389 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9392 * gst/rtsp-server/Makefile.am:
9393 gobject-introspection: fix g-i build for uninstalled setup
9394 Requires gst-plugins-base git (> 0.10.31.2).
9396 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9398 * examples/test-uri.c:
9399 examples: add some more options and comments
9401 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9403 * gst/rtsp-server/rtsp-media-factory-uri.c:
9404 factory-uri: use right property type
9406 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9408 * gst/rtsp-server/rtsp-media-factory-uri.c:
9409 factory-uri: attempt to configure buffer-lists
9410 Attempt to configure buffer lists in the payloader for improved performance.
9412 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9414 * gst/rtsp-server/rtsp-media.c:
9415 media: attempt to configure bigger UDP buffers
9416 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
9417 send buffers with high bitrate streams.
9419 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
9421 * gst/rtsp-server/rtsp-client.c:
9422 client: use the socket length from getsockname
9423 Use the length returned by getsockname to perform the getnameinfo call because
9424 the size can depend on the socket type and platform.
9427 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9429 * docs/libs/gst-rtsp-server-docs.sgml:
9430 * docs/libs/gst-rtsp-server-sections.txt:
9431 docs: add uri factory to the docs
9433 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9435 * gst/rtsp-server/rtsp-client.c:
9436 * gst/rtsp-server/rtsp-media.h:
9439 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9441 * gst/rtsp-server/rtsp-client.c:
9442 * gst/rtsp-server/rtsp-media.c:
9443 * gst/rtsp-server/rtsp-media.h:
9444 * gst/rtsp-server/rtsp-session.c:
9445 * gst/rtsp-server/rtsp-session.h:
9446 rtsp-server: add support for buffer lists
9447 Add support for sending bufferlists received from appsink.
9450 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9452 * gst/rtsp-server/rtsp-client.c:
9453 * gst/rtsp-server/rtsp-media.c:
9454 * gst/rtsp-server/rtsp-media.h:
9455 * gst/rtsp-server/rtsp-sdp.c:
9456 media: make method to retrieve the play range
9457 Make a method to retrieve the playback range so that we can conditionally create
9458 a different range for the SDP and the PLAY requests.
9460 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9462 * gst/rtsp-server/rtsp-media.c:
9463 * gst/rtsp-server/rtsp-media.h:
9464 media: add signal to notify of state changes
9466 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9468 * gst/rtsp-server/rtsp-client.h:
9469 client: cleanup headers
9471 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9473 * gst/rtsp-server/rtsp-client.c:
9476 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9478 * gst/rtsp-server/rtsp-media-factory-uri.c:
9479 * gst/rtsp-server/rtsp-media-factory-uri.h:
9480 factory-uri: add support for gstpay
9481 Add an option to prefer gstpay over decoder + raw payloader.
9483 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9485 * gst/rtsp-server/rtsp-media-factory-uri.c:
9486 * gst/rtsp-server/rtsp-media-factory-uri.h:
9487 factory-uri: rework the autoplugger.
9488 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
9491 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9493 * gst/rtsp-server/rtsp-media-factory-uri.c:
9494 factory-uri: use better factory filter
9495 Make better payloader filter based on autoplug rank and RTP use case.
9497 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
9500 Automatic update of common submodule
9501 From 169462a to 46445ad
9503 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9505 * gst/rtsp-server/rtsp-server.c:
9506 server: set SO_REUSEADDR before bind
9507 Set the SO_REUSEADDR _before_ bind() to make it actually work.
9509 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9511 * gst/rtsp-server/rtsp-media.c:
9512 * gst/rtsp-server/rtsp-media.h:
9513 media: emit prepared signal when prepared
9514 Make a 'prepared' signal and emit it when we successfully prepared the element.
9515 This signal can be used to configure the media object after it has been prepared
9518 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
9521 Automatic update of common submodule
9522 From 011bcc8 to 169462a
9524 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
9526 python an optional dependency
9527 * configure.ac: Move up valgrind and g-i checks. Make the python
9528 dependency optional, as it was before.
9530 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9532 Merge branch 'master' into 0.11
9537 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9539 * gst/rtsp-server/rtsp-media.c:
9540 media: update range when active clients changed
9541 When we changed the number of active clients, update the current range
9542 information because we want the second client connecting to a shared resource
9543 continue from where the stream currently.
9545 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9547 * gst/rtsp-server/rtsp-media-factory-uri.c:
9548 * gst/rtsp-server/rtsp-media-factory-uri.h:
9549 factory-uri: add colorspace and fix pt
9550 Rework the way we pass data to the autoplugger.
9551 When we have raw caps, plug a converter element to make pluggin to raw
9552 payloaders more successful.
9553 Make sure all dynamically plugged payloaders have a unique payload types.
9555 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9557 * examples/Makefile.am:
9558 * examples/test-uri.c:
9559 example: add example of the uri factory
9561 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9563 * gst/rtsp-server/Makefile.am:
9564 * gst/rtsp-server/rtsp-media-factory-uri.c:
9565 * gst/rtsp-server/rtsp-media-factory-uri.h:
9566 * gst/rtsp-server/rtsp-server.h:
9567 factory-uri: add a factory to stream any URI
9568 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
9571 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9573 * gst/rtsp-server/rtsp-media.c:
9574 * gst/rtsp-server/rtsp-media.h:
9575 media: ignore spurious ASYNC_DONE messages
9576 When we are dynamically adding pads, the addition of the udpsrc elements will
9577 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
9578 the real ASYNC_DONE when everything is prerolled.
9580 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9582 * gst/rtsp-server/rtsp-media-factory.c:
9583 * gst/rtsp-server/rtsp-media-factory.h:
9584 media-factory: make lock macro
9586 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
9588 * gst/rtsp-server/rtsp-client.c:
9589 rtsp-server: Remove unused variable and dead assignment
9591 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
9593 * examples/test-launch.c:
9594 * examples/test-mp4.c:
9595 * examples/test-ogg.c:
9596 * examples/test-readme.c:
9597 * examples/test-sdp.c:
9598 * examples/test-video.c:
9599 examples: Run gst-indent
9601 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
9603 * gst/rtsp-server/rtsp-client.c:
9604 * gst/rtsp-server/rtsp-media-factory.c:
9605 * gst/rtsp-server/rtsp-media-mapping.c:
9606 * gst/rtsp-server/rtsp-media.c:
9607 * gst/rtsp-server/rtsp-params.c:
9608 * gst/rtsp-server/rtsp-sdp.c:
9609 * gst/rtsp-server/rtsp-server.c:
9610 * gst/rtsp-server/rtsp-session-pool.c:
9611 * gst/rtsp-server/rtsp-session.c:
9612 rtsp-server: Run gst-indent
9613 Since it wasn't using the upstream common previously, there was no
9614 indentation check before commiting.
9616 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
9618 * gst/rtsp-server/rtsp-media-mapping.h:
9619 * gst/rtsp-server/rtsp-media.c:
9620 * gst/rtsp-server/rtsp-media.h:
9621 * gst/rtsp-server/rtsp-sdp.c:
9622 * gst/rtsp-server/rtsp-session-pool.h:
9623 * gst/rtsp-server/rtsp-session.c:
9624 * gst/rtsp-server/rtsp-session.h:
9625 rtsp-server: Some more doc fixups
9627 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
9630 Makefile: Add cruft-cleaning support
9632 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
9637 * docs/libs/Makefile.am:
9638 * docs/libs/gst-rtsp-server-docs.sgml:
9639 * docs/libs/gst-rtsp-server-sections.txt:
9640 * docs/libs/gst-rtsp-server.types:
9641 * docs/version.entities.in:
9642 docs: Add gtk-doc build system
9644 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
9646 * gst/rtsp-server/Makefile.am:
9647 Makefile.am: Use standard GIR make behaviour
9649 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
9653 autogen/configure: Bring more in sync to standard gst module behaviour
9655 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9657 * gst/rtsp-server/rtsp-media.c:
9658 media: warn and fail when gstrtpbin is not found
9660 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9663 configure: open 0.11 branch
9665 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
9669 Add common submodule
9671 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
9674 * common/Makefile.am:
9675 * common/c-to-xml.py:
9677 * common/coverage/coverage-report-entry.pl:
9678 * common/coverage/coverage-report.pl:
9679 * common/coverage/coverage-report.xsl:
9680 * common/coverage/lcov.mak:
9681 * common/gettext.patch:
9682 * common/glib-gen.mak:
9683 * common/gst-autogen.sh:
9684 * common/gst-xmlinspect.py:
9686 * common/gstdoc-scangobj:
9687 * common/gtk-doc-plugins.mak:
9688 * common/gtk-doc.mak:
9689 * common/m4/.gitignore:
9690 * common/m4/Makefile.am:
9692 * common/m4/as-ac-expand.m4:
9693 * common/m4/as-auto-alt.m4:
9694 * common/m4/as-compiler-flag.m4:
9695 * common/m4/as-compiler.m4:
9696 * common/m4/as-docbook.m4:
9697 * common/m4/as-libtool-tags.m4:
9698 * common/m4/as-libtool.m4:
9699 * common/m4/as-python.m4:
9700 * common/m4/as-scrub-include.m4:
9701 * common/m4/as-version.m4:
9702 * common/m4/ax_create_stdint_h.m4:
9703 * common/m4/check.m4:
9704 * common/m4/glib-gettext.m4:
9705 * common/m4/gst-arch.m4:
9706 * common/m4/gst-args.m4:
9707 * common/m4/gst-check.m4:
9708 * common/m4/gst-debuginfo.m4:
9709 * common/m4/gst-default.m4:
9710 * common/m4/gst-doc.m4:
9711 * common/m4/gst-error.m4:
9712 * common/m4/gst-feature.m4:
9713 * common/m4/gst-function.m4:
9714 * common/m4/gst-gettext.m4:
9715 * common/m4/gst-glib2.m4:
9716 * common/m4/gst-libxml2.m4:
9717 * common/m4/gst-plugindir.m4:
9718 * common/m4/gst-valgrind.m4:
9719 * common/m4/gtk-doc.m4:
9720 * common/m4/introspection.m4:
9722 * common/mangle-tmpl.py:
9723 * common/plugins.xsl:
9725 * common/release.mak:
9726 * common/scangobj-merge.py:
9727 * common/upload.mak:
9728 common: Remove static version
9730 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
9732 * common/m4/introspection.m4:
9733 Update introspection.m4 to match usage
9735 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9739 Remove old stuff from the README
9741 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9746 === release 0.10.7 ===
9748 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9753 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9755 * examples/test-ogg.c:
9756 test-ogg: remove parsers
9757 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
9758 buffers with timestamps. Using the parsers also seems to break things.
9760 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9762 * bindings/vala/gst-rtsp-server-0.10.vapi:
9763 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9764 Updated Vala bindings
9766 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9768 * common/m4/introspection.m4:
9770 * gst/rtsp-server/Makefile.am:
9771 Added initial gobject-introspection support
9773 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9775 * gst/rtsp-server/rtsp-media-factory.c:
9776 media-factory: don't use host for shared hash key
9777 When we generate the key to share made between connections, don't include the
9778 host used to connect so that we can share media even if between clients that
9779 connected with localhost and ones with the ip address.
9781 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9783 * bindings/vala/Makefile.am:
9784 build: fix distcheck
9786 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9788 * bindings/vala/gst-rtsp-server-0.10.vapi:
9789 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9790 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9791 Update Vala bindings
9793 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9795 * bindings/vala/Makefile.am:
9797 Fix configure checks and installation location for Vala bindings
9800 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9805 === release 0.10.6 ===
9807 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9810 configure: release 0.10.6
9812 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9814 * gst/rtsp-server/rtsp-media.c:
9815 media: help the compiler a little
9817 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9819 * gst/rtsp-server/rtsp-media.c:
9820 * gst/rtsp-server/rtsp-media.h:
9821 * gst/rtsp-server/rtsp-session.c:
9822 media: cleanup media transport before freeing
9823 Cleanup the media transport data before freeing. In particular, remove the qdata
9824 from the rtpsource object.
9826 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9828 * gst/rtsp-server/rtsp-media-factory.c:
9829 * gst/rtsp-server/rtsp-media-factory.h:
9830 * gst/rtsp-server/rtsp-media.c:
9831 * gst/rtsp-server/rtsp-media.h:
9832 media-factory: add eos-shutdown property
9833 Add an eos-shutdown property that will send an EOS to the pipeline before
9834 shutting it down. This allows for nice cleanup in case of a muxer.
9837 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9839 * gst/rtsp-server/rtsp-media.c:
9840 * gst/rtsp-server/rtsp-media.h:
9841 media: use multiudpsink send-duplicates when we can
9842 If we have a new enough multiudpsink with the send-duplicates property, use this
9843 instead of doing our own filtering. Our custom filtering code should eventually
9844 be removed when we can depend on a released -good.
9846 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9848 * gst/rtsp-server/rtsp-media.c:
9849 media: don't leak destinations
9850 Refactor and cleanup the destinations array when the stream is destroyed.
9852 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9854 * gst/rtsp-server/rtsp-media.c:
9855 * gst/rtsp-server/rtsp-media.h:
9856 media: don't add udp addresses multiple times
9857 Keep track of the udp addresses we added to udpsink and never add the same udp
9858 destination twice. This avoids duplicate packets when using multicast.
9860 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9862 * gst/rtsp-server/rtsp-server.c:
9863 server: disable use of SO_LINGER
9864 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
9865 server close()s the connection.
9867 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9869 * gst/rtsp-server/rtsp-server.c:
9870 server: use 5 second linger period in SO_LINGER
9871 Wait 5 seconds before clearing the send buffers and reseting the connection with
9872 the client when we do a close. This should be enough time to get the message to
9876 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
9878 * gst/rtsp-server/rtsp-server.c:
9879 server: use SO_LINGER
9880 SO_LINGER on the socket will make sure that any pending data on the socket is
9881 flushed ASAP and that the socket connection is reset. This makes sure that the
9882 socket can be reused immediately.
9885 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9888 README: add blurb about shared media factories
9890 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
9892 * gst/rtsp-server/rtsp-media.c:
9893 Add stdlib.h for atoi()
9895 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9897 * bindings/python/Makefile.am:
9898 * bindings/vala/Makefile.am:
9899 build: distcheck fixes
9900 Fix 'make distcheck', somewhat (it still fails because it tries to
9901 install files into /usr/share/vala/vapi/ irrespective of the
9904 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9907 configure: bump core/base requirements to released version
9908 Makes things less confusing for people.
9910 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9913 configure: fail if GStreamer core/base requirements are not met
9915 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9917 * gst/rtsp-server/rtsp-client.c:
9918 client: improve client cleanups
9919 Make sure the session does not timeout when using TCP. We need to do this
9920 because quicktime player does not send RTCP for some reason in tunneled
9922 Refactor some cleanup code.
9925 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9927 * gst/rtsp-server/rtsp-session.c:
9928 * gst/rtsp-server/rtsp-session.h:
9929 session: add support for prevent session timeouts
9930 Add an atomix counter to prevent session timeouts when we are, for example,
9933 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9935 * gst/rtsp-server/rtsp-client.c:
9936 client: fix unlink on session timeouts
9937 When our session times out, make sure we unlink all streams in this
9939 Remove the tunnelid when closing the connection.
9941 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9943 * gst/rtsp-server/rtsp-session.c:
9944 session: small cleanups
9946 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9948 * gst/rtsp-server/rtsp-client.c:
9949 client: handle lost_tunnel callbacks
9950 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
9951 hashtable so that we can reuse it for when the client reopens the POST
9953 Close the connection after a TEARDOWN.
9954 Make sure or watchid is cleared when the watch is removed.
9957 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9959 * gst/rtsp-server/rtsp-client.c:
9960 * gst/rtsp-server/rtsp-media.c:
9961 * gst/rtsp-server/rtsp-sdp.c:
9962 rtsp-server: add more support for multicast
9964 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9967 * gst/rtsp-server/rtsp-media.c:
9968 * gst/rtsp-server/rtsp-media.h:
9969 media: allow configuration of allowed lower transport
9971 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9973 * gst/rtsp-server/rtsp-client.h:
9974 * gst/rtsp-server/rtsp-media.c:
9975 * gst/rtsp-server/rtsp-media.h:
9976 * gst/rtsp-server/rtsp-sdp.c:
9977 * gst/rtsp-server/rtsp-sdp.h:
9978 * gst/rtsp-server/rtsp-server.c:
9979 rtsp: keep track of server ip and ipv6
9980 Keep track of how the client connected to the server and setup the udp ports
9981 with the same protocol.
9982 Copy the server ip address in the SDP so that clients can send RTCP back to
9985 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9987 * gst/rtsp-server/rtsp-session.c:
9990 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9992 * gst/rtsp-server/rtsp-client.c:
9993 client: use right size for malloc
9995 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9997 * gst/rtsp-server/rtsp-server.c:
9998 server: comment ipv6 server listening address
10000 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10002 * gst/rtsp-server/rtsp-media.c:
10003 media: allow for ipv6 sockets
10005 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10007 * gst/rtsp-server/rtsp-server.c:
10008 * gst/rtsp-server/rtsp-server.h:
10009 server: rework server part
10010 Allow setting a bind address, make sure we can deal with ipv6.
10011 Remove the port property and change with the service property.
10013 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10015 * gst/rtsp-server/rtsp-media.h:
10016 media: update comments a little
10018 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10020 * gst/rtsp-server/rtsp-client.c:
10021 client: make content-base better
10022 Use the URI formatting functions to make a content-base. Also make sure that
10023 there is a trailing / at the end.
10025 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10027 * gst/rtsp-server/rtsp-client.c:
10028 client: guard against invalid paths
10030 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10032 * examples/test-video.c:
10033 test: catch server bind errors
10035 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
10037 * gst/rtsp-server/rtsp-media.c:
10038 rtspmedia: emit "unprepared" if _prepare fails.
10039 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
10040 media object is removed from its factory's cache.
10042 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10044 * gst/rtsp-server/rtsp-media.c:
10045 media: collect media position when seek completes
10047 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
10049 * gst/rtsp-server/rtsp-client.c:
10050 client: call unlink_streams in client finalize
10053 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10055 * gst/rtsp-server/rtsp-media.c:
10056 media: limit the time to wait to something huge
10057 Avoid waiting forever but limit the timeout to 20 seconds.
10059 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10061 * gst/rtsp-server/rtsp-sdp.c:
10062 sdp: reindent and check for prepared status
10064 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10066 * gst/rtsp-server/rtsp-media.c:
10067 * gst/rtsp-server/rtsp-media.h:
10068 * gst/rtsp-server/rtsp-session.c:
10069 media: avoid doing _get_state() for state changes
10070 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
10071 until the media is prerolled or in error. This avoids doing a blocking call of
10072 gst_element_get_state() that can cause lockups when there is an error.
10075 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10077 * gst/rtsp-server/rtsp-media.c:
10080 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10082 * gst/rtsp-server/rtsp-media-factory.c:
10083 media-factory: better error handling
10084 Improve the error handling a bit.
10086 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10088 * gst/rtsp-server/rtsp-client.c:
10089 client: rework transport parsing
10090 Rework the transport parsing code so that we can ignore transports we don't
10091 support instead of just picking the first one we can parse.
10092 Configure a (for now hardcoded) destination for multicast transports.
10094 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10096 * gst/rtsp-server/rtsp-media.c:
10097 media: set multicast sink parameters
10098 Disable loop and automatic multicast join on the udpsink elements.
10099 Add some more debug info.
10100 Reset some state variables in the right place.
10101 Use the right port numbers for multicast.
10103 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10105 * gst/rtsp-server/rtsp-session.c:
10106 session: handle transport setup correctly
10107 Handle UDP, MCAST and TCP transport negotiation more correctly.
10108 Store the server session SSRC in the transport.
10110 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10112 * gst/rtsp-server/rtsp-client.c:
10113 rtsp-client: implement error_full
10114 Implement error_full to avoid some segfaults when the rtspconnection calls it.
10117 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10120 * gst/rtsp-server/rtsp-client.c:
10121 * gst/rtsp-server/rtsp-server.c:
10122 docs: update docs and comments
10124 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
10126 * gst/rtsp-server/rtsp-sdp.c:
10127 sdp: make server work better when behind a proxy
10129 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10131 * gst/rtsp-server/rtsp-client.c:
10132 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
10134 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10136 * gst/rtsp-server/rtsp-client.c:
10137 * gst/rtsp-server/rtsp-media-factory.c:
10138 * gst/rtsp-server/rtsp-media-mapping.c:
10139 * gst/rtsp-server/rtsp-media.c:
10140 * gst/rtsp-server/rtsp-server.c:
10141 * gst/rtsp-server/rtsp-session-pool.c:
10142 * gst/rtsp-server/rtsp-session.c:
10143 Use GStreamer's debugging subsystem
10145 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
10147 * gst/rtsp-server/rtsp-media-factory.c:
10148 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
10150 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10153 back to development
10155 === release 0.10.5 ===
10157 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10162 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10165 configure: bump required versions
10167 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
10169 * gst/rtsp-server/rtsp-client.c:
10170 client: call weak-unref on client->sessions from finalize
10173 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10175 * gst/rtsp-server/rtsp-media.c:
10176 media: Fixed crasher where caps got unref'ed too often
10178 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10181 * pkgconfig/.gitignore:
10182 * pkgconfig/Makefile.am:
10183 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
10184 Added pkg-config file to use gst-rtsp-server uninstalled
10186 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10188 * gst/rtsp-server/rtsp-media.c:
10189 media: add some docs
10191 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
10193 * gst/rtsp-server/rtsp-client.c:
10194 rtsp: Use gst_rtsp_watch_send_message().
10195 Use gst_rtsp_watch_send_message() since the old API which used
10196 gst_rtsp_watch_queue_message() has been deprecated.
10198 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10201 back to development
10203 === release 0.10.4 ===
10205 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10210 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10212 * gst/rtsp-server/rtsp-client.c:
10213 * gst/rtsp-server/rtsp-session.c:
10214 * gst/rtsp-server/rtsp-session.h:
10215 rtsp: allocate channels in TCP mode
10216 When the client does not provide us with channels in TCP mode, allocate channels
10219 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10221 * gst/rtsp-server/rtsp-client.c:
10222 client: don't crash when tunnelid is missing
10223 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
10224 don't crash but return an error response to the client.
10227 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10229 * bindings/vala/gst-rtsp-server-0.10.vapi:
10230 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
10231 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
10232 bindings: update vala bindings with new method
10234 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10236 * gst/rtsp-server/rtsp-session-pool.c:
10237 * gst/rtsp-server/rtsp-session-pool.h:
10238 sessionpool: add function to filter sessions
10239 Add generic function to retrieve/remove sessions.
10241 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10244 configure: bump core/base requirements to release
10246 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10248 * gst/rtsp-server/rtsp-media.c:
10249 media: fix indentation
10251 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10253 * gst/rtsp-server/rtsp-media.c:
10254 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
10256 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10258 * gst/rtsp-server/rtsp-media.c:
10259 set state and remove elements of media in for loop
10261 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
10263 * bindings/vala/gst-rtsp-server-0.10.vapi:
10264 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
10265 Added gst_rtsp_media_remove_elements function to Vala bindings
10267 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
10269 * gst/rtsp-server/rtsp-media.c:
10270 * gst/rtsp-server/rtsp-media.h:
10271 Added gst_rtsp_media_remove_elements function
10273 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
10275 * gst/rtsp-server/rtsp-media.c:
10276 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
10278 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10280 * bindings/vala/gst-rtsp-server-0.10.vapi:
10281 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
10282 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
10283 Updated Vala bindings
10285 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10287 * gst/rtsp-server/rtsp-media.c:
10288 * gst/rtsp-server/rtsp-media.h:
10289 Added vmethod unprepare to GstRTSPMedia
10290 The default implementation sets the state of the pipeline to GST_STATE_NULL
10292 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10294 * gst/rtsp-server/rtsp-media-factory.c:
10295 * gst/rtsp-server/rtsp-media-factory.h:
10296 Made collect_streams function public
10298 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10300 * gst/rtsp-server/rtsp-media-factory.c:
10301 * gst/rtsp-server/rtsp-media-factory.h:
10302 * gst/rtsp-server/rtsp-media.c:
10303 Added vmethod create_pipeline to GstRTSPMediaFactory
10304 The pipeline is created in this method and the GstRTSPMedia's element is added to it
10306 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10308 * gst/rtsp-server/rtsp-client.c:
10309 client: use g_source_destroy()
10310 We need to use g_source_destroy() because we might have added the source to a
10311 different main context than the default one.
10313 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10315 * gst/rtsp-server/Makefile.am:
10316 * gst/rtsp-server/rtsp-client.c:
10317 * gst/rtsp-server/rtsp-params.c:
10318 * gst/rtsp-server/rtsp-params.h:
10319 rtsp: prepare for handling GET/SET_PARAMETER
10320 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
10322 Fix return codes of handlers.
10324 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10326 * gst/rtsp-server/rtsp-media.c:
10327 media: don't leak session pads
10329 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10331 * gst/rtsp-server/rtsp-media.c:
10332 media: clean up the messages a bit
10334 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10336 * gst/rtsp-server/rtsp-sdp.c:
10337 sdp: warn and skip streams without media
10339 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10341 * bindings/vala/gst-rtsp-server-0.10.vapi:
10342 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
10343 vala: Fixed typo in header file of RTSPMediaStream
10345 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10347 * gst/rtsp-server/rtsp-media.c:
10349 Fix a debug message
10350 Make dumping RTCP stats configurable
10352 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10354 * gst/rtsp-server/rtsp-media.c:
10355 media: be less verbose and leak less
10357 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10359 * gst/rtsp-server/rtsp-media.c:
10360 media: don't leak the destination address
10362 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10364 * gst/rtsp-server/rtsp-client.c:
10365 * gst/rtsp-server/rtsp-media.c:
10366 * gst/rtsp-server/rtsp-media.h:
10367 * gst/rtsp-server/rtsp-session.c:
10368 * gst/rtsp-server/rtsp-session.h:
10369 rtsp: use RTCP to keep the session alive
10370 Use the RTCP rtcp-from stats field to find the associated session and use this
10371 to keep the session alive.
10373 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10375 * gst/rtsp-server/rtsp-session.c:
10376 session: add 5sec to the real session timeout
10377 Allow the session to live 5sec longer before really timing out. This should give
10378 clients some extra time to keep the session active.
10380 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10382 * gst/rtsp-server/rtsp-client.c:
10383 client: replay OK to GET/SET_PARAMETER
10384 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
10385 so that we return OK for those requests.
10387 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10389 * gst/rtsp-server/rtsp-media.c:
10390 * gst/rtsp-server/rtsp-media.h:
10391 media: keep track of active transports
10392 Keep track of which transport is active to avoid closing the connection too
10394 Remove the destination transport also when going to NULL.
10395 Print some stats about the SDES and other RTCP messages we receive from the
10398 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10400 * examples/.gitignore:
10401 * examples/Makefile.am:
10402 * examples/test-sdp.c:
10403 example: add SDP relay example
10405 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10407 * gst/rtsp-server/rtsp-media.c:
10408 media: also count active TCP connections
10410 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10412 * gst/rtsp-server/rtsp-media-factory.c:
10413 * gst/rtsp-server/rtsp-media.c:
10414 * gst/rtsp-server/rtsp-media.h:
10415 rtsp: add support for dynamic elements
10416 Add support for dynamic elements.
10417 Don't set live pipelines back to paused.
10419 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10421 * gst/rtsp-server/rtsp-sdp.c:
10422 sdp: don't add encoding name when absent in caps
10424 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10426 * gst/rtsp-server/rtsp-client.c:
10427 client: warn when we can't do RTP-Info
10429 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10431 * gst/rtsp-server/rtsp-media-factory.c:
10432 factory: factor out the stream construction
10434 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10436 * gst/rtsp-server/rtsp-client.c:
10437 client: only add RTP-Info when we have the info
10438 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
10441 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10444 back to development
10446 === release 0.10.3 ===
10448 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10452 - Fixes a bug where it put the wrong verion in pkgconfig
10453 - Link RTP and RTCP sources
10455 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10457 * gst/rtsp-server/rtsp-media.c:
10458 * gst/rtsp-server/rtsp-media.h:
10459 media: link the RTP udpsrc to the session manager
10460 Link the RTP udpsrc and the appsrc to the session manager so that they don't
10461 shut down when the client sends a packet to open firewalls.
10463 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10465 * pkgconfig/gst-rtsp-server.pc.in:
10466 Don't use hard-coded version number in pkg-config file
10468 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10471 back to development
10473 === release 0.10.2 ===
10475 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10480 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10483 * common/m4/.gitignore:
10484 * examples/.gitignore:
10485 * pkgconfig/.gitignore:
10486 add some .gitignore files
10488 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10490 * gst/rtsp-server/rtsp-media.c:
10491 media: seek to key frames
10493 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10495 * gst/rtsp-server/rtsp-media.c:
10496 media: emit the unprepared signal by id
10497 Emit the unprepared signal by id instead of name and set the media as
10500 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10502 * gst/rtsp-server/rtsp-media.c:
10503 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
10505 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10507 * gst/rtsp-server/rtsp-server.c:
10508 Added finalize function to GstRTPSPServer to unref session pool and media mapping
10510 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
10512 * bindings/vala/gst-rtsp-server-0.10.vapi:
10513 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
10514 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
10515 Updated vala bindings
10517 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10519 * gst/rtsp-server/Makefile.am:
10520 * gst/rtsp-server/rtsp-client.c:
10521 * gst/rtsp-server/rtsp-media.c:
10522 server: use appsink and appsrc with the API
10523 Use the appsink/appsrc API instead of the signals for higher
10526 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10528 * examples/test-ogg.c:
10529 tests: set the payload type correctly
10531 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10533 * gst/rtsp-server/rtsp-media-factory.c:
10534 factory: connect to the unprepare signal
10535 Connect to the unprepare signal for non-reusable media so that we can remove
10536 them from the cache.
10538 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10540 * gst/rtsp-server/rtsp-media.c:
10541 * gst/rtsp-server/rtsp-media.h:
10542 media: add signal to notify of unprepare
10544 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10546 * gst/rtsp-server/rtsp-media.c:
10547 * gst/rtsp-server/rtsp-media.h:
10548 media: more work on making the media shared
10549 Add a reusable flag to medias, indicating that they can be reused after a state
10553 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10555 * examples/test-readme.c:
10556 examples: mark the example as shared for testing
10558 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10560 * gst/rtsp-server/rtsp-media.c:
10561 * gst/rtsp-server/rtsp-media.h:
10562 client: support shared media
10563 Always perform the state actions even if the target state of the pipeline is
10564 already correct, we still want to add/remove the transports when we are dealing
10566 Keep a counter of the number of active transports for a media so that we can use
10567 this to perform a state change when needed.
10568 Perform a state change of the pipeline only when the first transport was added
10569 or when there are no active transports.
10571 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10573 * gst/rtsp-server/rtsp-client.c:
10574 client: fix refcounting crasher
10575 Don't need to remove the weak refs in the finalize methods, they are already
10576 removed in the dispose.
10577 Don't register the callback with a DestroyNofity.
10579 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10581 * gst/rtsp-server/rtsp-client.c:
10582 Fix rtsp client refcount management in TCP mode.
10583 Don't unref a client ref we never had. Fixes an unref
10584 of an already-free client object after a client
10585 teardown request for me.
10587 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10589 * gst/rtsp-server/rtsp-session.c:
10590 docs: fix typo in API docs
10592 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10594 * gst/rtsp-server/rtsp-media.c:
10595 More seeking fixes.
10596 Keep the udp sources in playing even if we go to paused. unlock the sources when
10598 Add some more debug info.
10599 Only seek when we need to.
10600 Keep track of the position when we go to paused.
10602 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10604 * gst/rtsp-server/rtsp-client.c:
10605 * gst/rtsp-server/rtsp-media.c:
10606 * gst/rtsp-server/rtsp-media.h:
10607 Add beginnings of seeking.
10608 Parse the Range header and perform a seek on the pipeline for the requested
10609 position. It's disabled currently until I figure out what's going wrong.
10611 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10613 * gst/rtsp-server/rtsp-client.c:
10614 allow pause requests for now.
10617 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10619 * gst/rtsp-server/rtsp-client.c:
10620 Remove weak ref on the session in teardown
10621 We need to remove our weakref from the session when we do a teardown because
10622 else we close the TCP connection prematurely.
10624 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10626 * gst/rtsp-server/rtsp-client.c:
10627 * gst/rtsp-server/rtsp-client.h:
10628 * gst/rtsp-server/rtsp-session-pool.c:
10629 Do some more session cleanup
10630 Make session timeout kill the TCP connection that currently watches the
10632 Remove the client timeout property.
10634 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10636 * gst/rtsp-server/rtsp-client.c:
10637 * gst/rtsp-server/rtsp-client.h:
10638 * gst/rtsp-server/rtsp-media.c:
10639 * gst/rtsp-server/rtsp-media.h:
10640 * gst/rtsp-server/rtsp-server.c:
10641 * gst/rtsp-server/rtsp-session.c:
10642 * gst/rtsp-server/rtsp-session.h:
10644 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
10647 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10649 * examples/Makefile.am:
10650 * examples/test-launch.c:
10651 Add example server that takes launch lines
10652 Add an example server that streams any -launch line.
10654 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10656 * examples/test-readme.c:
10657 * gst/rtsp-server/rtsp-client.c:
10658 * gst/rtsp-server/rtsp-media.c:
10659 * gst/rtsp-server/rtsp-media.h:
10660 Add support for live streams
10661 Add support for live streams and ranges
10662 Start on handling TCP data transfer.
10664 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10666 * gst/rtsp-server/rtsp-media.c:
10667 Free the pipeline before other things
10670 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10672 * gst/rtsp-server/rtsp-client.c:
10673 Only free the pending tunnel if there is one
10676 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10678 * gst/rtsp-server/rtsp-client.c:
10679 * gst/rtsp-server/rtsp-client.h:
10680 * gst/rtsp-server/rtsp-media.c:
10681 rtsp-server: Add support for tunneling
10682 Add support for tunneling over HTTP.
10683 Use new connection methods to retrieve the url.
10684 Dispatch messages based on the message type instead of blindly
10685 assuming it's always a request.
10686 Keep track of the watch id so that we can remove it later.
10687 Set the media pipeline to NULL before unreffing the pipeline.
10689 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10691 * gst/rtsp-server/rtsp-client.c:
10692 * gst/rtsp-server/rtsp-client.h:
10693 Fix for channel -> watch rename in gstreamer
10694 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
10696 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10698 * gst/rtsp-server/rtsp-client.c:
10699 * gst/rtsp-server/rtsp-client.h:
10701 Use the async RTSP channels instead of spawning a new thread for each client.
10702 If a sessionid is specified in a request, fail if we don't have the session.
10704 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10706 * gst/rtsp-server/rtsp-media.c:
10707 Add better debug info
10708 Add some better debug info.
10710 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10712 * examples/test-video.c:
10714 Add support for session timeouts in the example.
10716 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10718 * gst/rtsp-server/rtsp-session-pool.c:
10719 * gst/rtsp-server/rtsp-session-pool.h:
10720 Pass GTimeVal around for performance reasons
10721 Get the current time only once and pass it around so that sessions don't have to
10722 get the current time anymore.
10723 Add experimental support for a GSource that dispatches when the session needs to
10726 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10728 * gst/rtsp-server/rtsp-session.c:
10729 * gst/rtsp-server/rtsp-session.h:
10730 Add better support for session timeouts
10731 Add a method to request the number of milliseconds when a session will timeout.
10733 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10735 * gst/rtsp-server/rtsp-media.c:
10736 * gst/rtsp-server/rtsp-media.h:
10737 Add suport for RTP manager monitoring
10738 Add the first stage in monitoring the rtp manager.
10739 Make sure we don't update the state to something we don't want.
10741 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10743 * gst/rtsp-server/rtsp-client.c:
10744 Add support for session keepalive
10745 Get and update the session timeout for all requests. get the session as early as
10748 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10750 * gst/rtsp-server/rtsp-media-factory.h:
10751 * gst/rtsp-server/rtsp-media.c:
10752 * gst/rtsp-server/rtsp-media.h:
10753 Handle media bus messages
10754 Handle media bus messages in a custom mainloop and dispatch them to the
10755 RTSPMedia objects. Let the default implementation handle some common messages.
10757 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10759 * gst/rtsp-server/rtsp-client.c:
10760 * gst/rtsp-server/rtsp-session-pool.c:
10761 * gst/rtsp-server/rtsp-session.c:
10762 Some more session timeout handling
10763 Move the session header setting code to a central place so that we always add
10764 the timeout parameter too.
10765 Handle timeouts by running the session cleanup code.
10766 Stop media before cleaning up.
10768 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10770 * gst/rtsp-server/rtsp-client.c:
10771 * gst/rtsp-server/rtsp-client.h:
10772 Add timeout property
10773 Add a timeout property ot the client and make the other properties into GObject
10776 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10778 * gst/rtsp-server/rtsp-session-pool.c:
10779 Use getters and setters in property code
10780 Use the getters and setters for the timeout property instead of locking
10783 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10785 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
10787 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10789 * gst/rtsp-server/rtsp-session-pool.c:
10790 * gst/rtsp-server/rtsp-session-pool.h:
10791 * gst/rtsp-server/rtsp-session.c:
10792 * gst/rtsp-server/rtsp-session.h:
10793 Add more timeout stuff
10794 Add method to check if a session is expired.
10795 Add method to perform cleanup on a session pool.
10797 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10799 * gst/rtsp-server/rtsp-client.c:
10800 * gst/rtsp-server/rtsp-session-pool.c:
10801 * gst/rtsp-server/rtsp-session-pool.h:
10802 * gst/rtsp-server/rtsp-session.c:
10803 * gst/rtsp-server/rtsp-session.h:
10804 Add beginnings of session timeouts and limits
10805 Add the timeout value to the Session header for unusual timeout values.
10806 Allow us to configure a limit to the amount of active sessions in a pool. Set a
10807 limit on the amount of retry we do after a sessionid collision.
10808 Add properties to the sessionid and the timeout of a session. Keep track of
10809 creation time and last access time for sessions.
10811 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10813 * gst/rtsp-server/rtsp-client.c:
10814 * gst/rtsp-server/rtsp-media.c:
10815 * gst/rtsp-server/rtsp-media.h:
10816 * gst/rtsp-server/rtsp-sdp.c:
10817 * gst/rtsp-server/rtsp-session-pool.c:
10818 * gst/rtsp-server/rtsp-session.c:
10819 * gst/rtsp-server/rtsp-session.h:
10820 Cleanup of sessions and more
10821 Fix the refcounting of media and sessions in the client. Properly clean up the
10822 session data when the client performs a teardown.
10823 Add Server header to responses.
10824 Allow for multiple uri setups in one session.
10825 Add Range header to the PLAY response and add the range attribute to the SDP
10827 Fix the session pool remove method, it used the wrong key in the hashtable. Also
10828 give the ownership of the sessionid to the session object.
10830 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10832 * gst/rtsp-server/rtsp-server.c:
10833 * gst/rtsp-server/rtsp-server.h:
10835 Rename the 'server_port' variable to simply 'port'.
10837 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10840 * gst/rtsp-server/rtsp-client.c:
10841 * gst/rtsp-server/rtsp-media.c:
10842 * gst/rtsp-server/rtsp-media.h:
10843 * gst/rtsp-server/rtsp-session.c:
10844 * gst/rtsp-server/rtsp-session.h:
10845 Rework the way we handle transports for streams
10846 Make the media accept an array of transports for the streams that we have
10847 configured for the play/pause requests.
10848 Implement server states for a client and its media.
10849 Require 0.10.22.1 (git HEAD) of gstreamer.
10851 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10853 * gst/rtsp-server/rtsp-client.c:
10854 * gst/rtsp-server/rtsp-media-factory.c:
10855 Drop const from functions dealing with urls
10856 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
10857 have the right const in them.
10859 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10861 * gst/rtsp-server/rtsp-client.c:
10862 * gst/rtsp-server/rtsp-media.c:
10863 * gst/rtsp-server/rtsp-sdp.c:
10867 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10869 * gst/rtsp-server/rtsp-client.c:
10870 * gst/rtsp-server/rtsp-media-factory.c:
10871 * gst/rtsp-server/rtsp-media.c:
10872 * gst/rtsp-server/rtsp-media.h:
10874 Don't keep a reference to the GstRTSPMedia in the stream.
10875 Free more things when freeing the GstRTSPMedia.
10877 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10880 * gst/rtsp-server/rtsp-media-factory.c:
10881 * gst/rtsp-server/rtsp-media-factory.h:
10882 * gst/rtsp-server/rtsp-media.c:
10883 * gst/rtsp-server/rtsp-media.h:
10884 * gst/rtsp-server/rtsp-server.c:
10885 * gst/rtsp-server/rtsp-server.h:
10886 More docs and small cleanups
10887 Add some more docs and update the README
10888 Cleanup some method names.
10889 Remove an unneeded idx field in the GstRTSPMediaStream
10891 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10894 * examples/Makefile.am:
10895 * examples/test-readme.c:
10896 Add a README and more example code
10897 Add a README file that contains a small introduction on how to use the server
10898 along with the example code explained in the readme.
10900 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10902 * gst/rtsp-server/rtsp-media.c:
10903 * gst/rtsp-server/rtsp-server.c:
10904 Fix some leaks and change default port
10905 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
10906 we finished the initial preroll. If we keep them locked, setting the pipeline to
10907 NULL will not stop and clean up the sources correctly.
10908 Change the default RTSP port to 8554 aka the official alternative RTSP port.
10910 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10912 * gst/rtsp-server/rtsp-session.c:
10913 * gst/rtsp-server/rtsp-session.h:
10914 Cleanups to the session object
10915 Remove some unneeded variables in the session state of a stream such as the
10916 owner media and the server transport.
10917 Get the configuration of a media stream in a session based on the media_stream
10918 in the original object instead of our cached index.
10919 Free more data in the finalize method.
10921 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10923 * gst/rtsp-server/rtsp-client.c:
10924 * gst/rtsp-server/rtsp-client.h:
10925 Cleanups and reuse media from DESCRIBE
10926 Handle thread create errors.
10927 Rename some internal methods to better match what they actually do.
10928 Handle misconfiguration of session_pool and media_mapping gracefully.
10929 Cache the DESCRIBE media and uri in the client connection and reuse them when
10930 we receive a SETUP request in the same connection for the same uri.
10931 Cleanup the client connection object.
10933 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10935 * gst/rtsp-server/rtsp-media-factory.c:
10936 * gst/rtsp-server/rtsp-media-factory.h:
10937 * gst/rtsp-server/rtsp-media.c:
10938 * gst/rtsp-server/rtsp-media.h:
10939 Add shared properties to media and factory
10940 Add the shared property to media.
10941 Implement some simple caching in the factory depending on if the media is shared
10944 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10946 * gst/rtsp-server/rtsp-client.c:
10947 Add a little comment
10948 Add some comment about the content-base header.
10950 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10952 * examples/Makefile.am:
10953 * examples/test-mp4.c:
10954 * examples/test-ogg.c:
10955 * examples/test-video.c:
10956 * gst/rtsp-server/Makefile.am:
10957 * gst/rtsp-server/rtsp-client.c:
10958 * gst/rtsp-server/rtsp-client.h:
10959 * gst/rtsp-server/rtsp-media-factory.c:
10960 * gst/rtsp-server/rtsp-media-factory.h:
10961 * gst/rtsp-server/rtsp-media.c:
10962 * gst/rtsp-server/rtsp-media.h:
10963 * gst/rtsp-server/rtsp-sdp.c:
10964 * gst/rtsp-server/rtsp-sdp.h:
10965 * gst/rtsp-server/rtsp-server.c:
10966 * gst/rtsp-server/rtsp-server.h:
10967 * gst/rtsp-server/rtsp-session.c:
10968 * gst/rtsp-server/rtsp-session.h:
10969 Reorganize things, prepare for media sharing
10970 Added various other test server examples
10971 Move the SDP message generation to a separate helper.
10972 Refactor common code for finding the session.
10973 Add content-base for realplayer compatibility
10974 Clean up request uris before processing for better vlc compatibility.
10975 Move prerolling and pipeline construction to the RTSPMedia object.
10976 Use multiudpsink for future pipeline reuse.
10978 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10981 Back to development
10984 === release 0.10.1 ===
10986 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10989 Make 0.10.1 release
10992 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10994 * bindings/vala/Makefile.am:
10996 Add more directories and files to the dist.
10998 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11000 * bindings/python/Makefile.am:
11001 * bindings/python/rtspserver.override:
11002 Fixed compile error of python bindings
11004 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11006 * bindings/vala/gst-rtsp-server-0.10.vapi:
11007 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11008 Marked values as nullable accordingly
11010 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11012 * bindings/vala/gst-rtsp-server-0.10.vapi:
11013 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
11014 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11015 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11016 Updated Vala bindings
11018 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11020 * gst/rtsp-server/rtsp-client.c:
11021 * gst/rtsp-server/rtsp-media-mapping.c:
11022 * gst/rtsp-server/rtsp-media-mapping.h:
11023 * gst/rtsp-server/rtsp-media.h:
11024 * gst/rtsp-server/rtsp-session-pool.h:
11025 Cleanups and doc updates
11026 Add some more documentation and do some minor cleanups here and there.
11028 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11030 * gst/rtsp-server/rtsp-client.c:
11031 * gst/rtsp-server/rtsp-media-factory.c:
11032 * gst/rtsp-server/rtsp-media-factory.h:
11033 * gst/rtsp-server/rtsp-media.c:
11034 * gst/rtsp-server/rtsp-media.h:
11035 * gst/rtsp-server/rtsp-session.c:
11036 * gst/rtsp-server/rtsp-session.h:
11038 Rename GstRTSPMediaBin to GstRTSPMedia
11039 Parse the request url into a GstRTSPUri object and pass this object to the
11040 various handlers and methods that require the uri.
11042 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11046 Add some more docs and remove some old code from the example.
11048 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11050 * gst/rtsp-server/rtsp-client.c:
11051 Handle state change failures better
11052 Handle state change failures better when changing the state of the pipeline to
11055 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11057 * gst/rtsp-server/rtsp-media-factory.c:
11058 * gst/rtsp-server/rtsp-media-factory.h:
11059 Make element creation more extendible
11060 Add get_element vmethod to the default MediaFactory so that subclasses can just
11061 override that method and still use the default logic for making a MediaBin from
11064 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11067 * gst/rtsp-server/Makefile.am:
11068 * gst/rtsp-server/rtsp-client.c:
11069 * gst/rtsp-server/rtsp-client.h:
11070 * gst/rtsp-server/rtsp-media-factory.c:
11071 * gst/rtsp-server/rtsp-media-factory.h:
11072 * gst/rtsp-server/rtsp-media-mapping.c:
11073 * gst/rtsp-server/rtsp-media-mapping.h:
11074 * gst/rtsp-server/rtsp-media.c:
11075 * gst/rtsp-server/rtsp-media.h:
11076 * gst/rtsp-server/rtsp-server.c:
11077 * gst/rtsp-server/rtsp-server.h:
11078 * gst/rtsp-server/rtsp-session.c:
11079 * gst/rtsp-server/rtsp-session.h:
11080 Make the server handle arbitrary pipelines
11081 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
11082 The GstMediaBin object has a handle to a bin with elements and to a list of
11083 GstMediaStream objects that this bin produces.
11084 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
11085 with methods to register and remove those mappings.
11086 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
11087 used by the server instance.
11088 Modify the example application so that it shows how to create custom pipelines
11089 attached to a specific mount point.
11090 Various misc cleanps.
11092 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11094 * gst/rtsp-server/rtsp-server.c:
11095 * gst/rtsp-server/rtsp-server.h:
11096 Allow setting a custom media factory for a server
11098 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11100 * gst/rtsp-server/rtsp-client.c:
11101 * gst/rtsp-server/rtsp-client.h:
11102 Allow setting a custom media factory for a client.
11104 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11106 * gst/rtsp-server/Makefile.am:
11107 Add Makefile entry for the media factory
11109 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11111 * gst/rtsp-server/rtsp-media-factory.c:
11112 * gst/rtsp-server/rtsp-media-factory.h:
11113 Add media factory to map urls to media pipeline objects.
11115 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11117 * gst/rtsp-server/rtsp-media.c:
11118 * gst/rtsp-server/rtsp-media.h:
11119 Add comments. Remove unused field
11121 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11123 * gst/rtsp-server/rtsp-session-pool.c:
11124 * gst/rtsp-server/rtsp-session-pool.h:
11125 Allow custom session pools to override the session id allocation algorithms Add some comments.
11127 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11129 * gst/rtsp-server/rtsp-session.h:
11132 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11134 * gst/rtsp-server/rtsp-client.c:
11135 * gst/rtsp-server/rtsp-client.h:
11136 Move the connection code in one place Add some comments
11138 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11140 * gst/rtsp-server/rtsp-server.c:
11141 * gst/rtsp-server/rtsp-server.h:
11142 Make vmethod to create and accept new clients. Add some docs.
11144 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11146 * gst/rtsp-server/rtsp-server.c:
11147 * gst/rtsp-server/rtsp-server.h:
11148 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
11150 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11152 * gst/rtsp-server/rtsp-client.c:
11153 * gst/rtsp-server/rtsp-client.h:
11154 Name the parameters more appropriately.
11156 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11158 * gst/rtsp-server/rtsp-session-pool.c:
11159 Do some more cleanup of the session pool.
11161 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11163 * gst/rtsp-server/Makefile.am:
11164 * gst/rtsp-server/rtsp-client.c:
11165 Check if return value of gst_rtsp_session_get_media is not NULL
11167 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11169 * gst/rtsp-server/Makefile.am:
11170 Install rtsp-session and rtsp-session-pool headers
11172 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11177 * bindings/python/Makefile.am:
11178 * bindings/python/arg-types.py:
11179 * bindings/python/codegen/Makefile.am:
11180 * bindings/python/codegen/__init__.py:
11181 * bindings/python/codegen/argtypes.py:
11182 * bindings/python/codegen/code-coverage.py:
11183 * bindings/python/codegen/codegen.py:
11184 * bindings/python/codegen/definitions.py:
11185 * bindings/python/codegen/defsparser.py:
11186 * bindings/python/codegen/docextract.py:
11187 * bindings/python/codegen/docgen.py:
11188 * bindings/python/codegen/fileprefix.override:
11189 * bindings/python/codegen/fileprefixmodule.c:
11190 * bindings/python/codegen/h2def.py:
11191 * bindings/python/codegen/mergedefs.py:
11192 * bindings/python/codegen/mkskel.py:
11193 * bindings/python/codegen/override.py:
11194 * bindings/python/codegen/reversewrapper.py:
11195 * bindings/python/codegen/scmexpr.py:
11196 * bindings/python/rtspserver-types.defs:
11197 * bindings/python/rtspserver.defs:
11198 * bindings/python/rtspserver.override:
11199 * bindings/python/rtspservermodule.c:
11201 Add python bindings.
11203 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11205 * bindings/Makefile.am:
11207 Don't go into python dir when requirements for python bindings are missing
11209 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11211 * bindings/Makefile.am:
11212 * bindings/vala/Makefile.am:
11214 Install Vala bindings if vala is available
11216 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11218 * bindings/vala/gst-rtsp-server-0.10.deps:
11219 * bindings/vala/gst-rtsp-server-0.10.vapi:
11220 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
11221 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
11222 * bindings/vala/packages/gst-rtsp-server-0.10.files:
11223 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11224 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11225 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
11226 Regenerated Vala bindings
11228 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11230 * bindings/vala/gst-rtsp-server.vapi:
11231 * bindings/vala/packages/gst-rtsp-server.metadata:
11232 Fixed typo in included headers for vala bindings
11234 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11238 * pkgconfig/Makefile.am:
11239 * pkgconfig/gst-rtsp-server.pc.in:
11240 Added pkgconfig file
11242 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
11244 * bindings/vala/gst-rtsp-server.vapi:
11245 * bindings/vala/packages/gst-rtsp-server.excludes:
11246 * bindings/vala/packages/gst-rtsp-server.gi:
11247 * bindings/vala/packages/gst-rtsp-server.metadata:
11248 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
11250 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
11252 * bindings/vala/gst-rtsp-server.vapi:
11253 * bindings/vala/packages/gst-rtsp-server.deps:
11254 * bindings/vala/packages/gst-rtsp-server.files:
11255 * bindings/vala/packages/gst-rtsp-server.gi:
11256 * bindings/vala/packages/gst-rtsp-server.metadata:
11257 * bindings/vala/packages/gst-rtsp-server.namespace:
11258 Added Vala bindings
11260 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
11262 * gst/rtsp-server/rtsp-session.c:
11263 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
11265 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
11267 * examples/Makefile.am:
11268 * gst/rtsp-server/Makefile.am:
11269 Put GStreamer version in library name
11271 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11273 * examples/Makefile.am:
11274 * gst/rtsp-server/Makefile.am:
11275 Fix some issues to pass distcheck
11277 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11279 * gst/rtsp-server/rtsp-server.c:
11280 Added port property to GstRTSPServer class.
11282 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11287 * examples/Makefile.am:
11290 * gst/rtsp-server/Makefile.am:
11291 * gst/rtsp-server/rtsp-client.c:
11292 * gst/rtsp-server/rtsp-client.h:
11293 * gst/rtsp-server/rtsp-media.c:
11294 * gst/rtsp-server/rtsp-media.h:
11295 * gst/rtsp-server/rtsp-server.c:
11296 * gst/rtsp-server/rtsp-server.h:
11297 * gst/rtsp-server/rtsp-session-pool.c:
11298 * gst/rtsp-server/rtsp-session-pool.h:
11299 * gst/rtsp-server/rtsp-session.c:
11300 * gst/rtsp-server/rtsp-session.h:
11302 Split in library and example program
11304 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
11306 * src/rtsp-client.h:
11307 Removed obsolete variable
11309 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
11311 * src/rtsp-client.c:
11312 * src/rtsp-client.h:
11313 Removed pipeline variable GstRTSPClient, because it's only used in one function
11315 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11317 * src/rtsp-media.c:
11318 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
11320 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
11322 * src/rtsp-session.c:
11323 Initialize some more vars.
11325 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
11327 * src/rtsp-session.c:
11328 Initialize variable to avoid compiler warning.
11330 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
11333 Add a reasonable generic .gitignore