3 2019-12-03 11:16:06 +0000 Tim-Philipp Müller <tim@centricular.com>
9 * gst-rtsp-server.doap:
13 2019-09-17 13:45:57 +0200 Adam x Nilsson <adamni@axis.com>
15 * gst/rtsp-server/rtsp-media.c:
16 rtsp-media: Use lock in gst_rtsp_media_is_receive_only
17 (cherry picked from commit f1d2a0cae9a791ce07c753faaf82a6cdefecd373)
19 2019-11-04 12:56:13 +0100 Kristofer Bjorkstrom <kristofer.bjorkstrom@axis.com>
21 * gst/rtsp-server/rtsp-client.c:
22 * gst/rtsp-server/rtsp-media.c:
23 * tests/check/gst/client.c:
24 rtsp-client: RTP Info when completed_sender
25 Change condition that should be fulfilled regarding RTPInfo.
26 Replace !gst_rtsp_media_is_receive_only with
27 gst_rtsp_media_has_completed_sender. It is more correct to actually look
28 for a sender pipeline that is complete. Only then a RTPInfo should
30 gst_rtsp_media_is_receive_only gives different answears depending on
32 If Describe is called wth URL+options for backchannel SDP will give only
33 audio and only backchannel a=sendonly
34 If Describe is called on URL+options that gives both audio and video
35 direction from server to client, pipelines are created. Thus
36 receive_only will return false, even though Setup only would setup
38 RTP-Info is only for outgoing streams. Thus one should look if outgoing
41 2019-10-17 12:15:42 +0200 Muhammet Ilendemli <mi@tailored-apps.com>
43 * gst/rtsp-server/rtsp-client.c:
44 rtsp-client: Generate correct URI for MIKEY in ANNOUNCE responses
45 Instead of hardcoding the URI, take the actual URI (and especially the correct port)
46 from the RTSP context.
49 === release 1.16.1 ===
51 2019-09-23 11:17:41 +0100 Tim-Philipp Müller <tim@centricular.com>
57 * gst-rtsp-server.doap:
61 2019-08-30 14:00:52 +0200 Kristofer Björkström <kristofb@axis.com>
63 * gst/rtsp-server/rtsp-client.c:
64 rtsp-client: RTP Info must exist in PLAY response
65 If RTP Info is missing. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR
68 2019-06-25 13:19:44 +0100 Tim-Philipp Müller <tim@centricular.com>
70 * gst/rtsp-server/rtsp-onvif-media-factory.c:
71 * gst/rtsp-server/rtsp-onvif-media.c:
72 onvif-media: fix "void function returning a value" compiler warning
74 2019-04-23 14:38:05 +0300 Sebastian Dröge <sebastian@centricular.com>
76 * gst/rtsp-server/rtsp-media-factory.c:
77 * gst/rtsp-server/rtsp-media.c:
78 * gst/rtsp-server/rtsp-stream-transport.c:
79 * gst/rtsp-server/rtsp-stream.c:
80 rtsp-server: Add various missing Since: 1.16 markers
82 2019-04-23 15:01:32 +0300 Sebastian Dröge <sebastian@centricular.com>
84 * gst/rtsp-server/rtsp-media.c:
85 * gst/rtsp-server/rtsp-sdp.c:
86 * gst/rtsp-server/rtsp-session-media.c:
87 * gst/rtsp-server/rtsp-stream.c:
88 rtsp-server: Add various Since: 1.14 markers
90 2019-04-23 15:09:34 +0300 Sebastian Dröge <sebastian@centricular.com>
92 * gst/rtsp-server/rtsp-auth.c:
93 * gst/rtsp-server/rtsp-client.h:
94 rtsp-server: Fix various Since markers
96 2019-05-02 12:35:34 +0100 Tim-Philipp Müller <tim@centricular.com>
99 ci: use template from 1.16 branch
101 === release 1.16.0 ===
103 2019-04-19 00:34:54 +0100 Tim-Philipp Müller <tim@centricular.com>
109 * gst-rtsp-server.doap:
113 2019-04-15 20:33:01 +0300 Sebastian Dröge <sebastian@centricular.com>
115 * gst/rtsp-sink/gstrtspclientsink.c:
116 rtspclientsink: Notify the stream transport about each written message
117 Otherwise it will never try to send us the next one: it tries to keep
118 exactly one message in-flight all the time.
119 In gst-rtsp-server this is done asynchronously via the GstRTSPWatch but
120 in the client sink we always write data out synchronously.
122 2019-04-02 08:05:03 +0200 Göran Jönsson <goranjn@axis.com>
124 * gst/rtsp-server/rtsp-stream.c:
125 rtsp_server: Free thread pool before clean transport cache
126 If not waiting for free thread pool before clean transport caches, there
127 can be a crash if a thread is executing in transport list loop in
128 function send_tcp_message.
129 Also add a check if priv->send_pool in on_message_sent to avoid that a
130 new thread is pushed during wait of free thread pool. This is possible
131 since when waiting for free thread pool mutex have to be unlocked.
133 === release 1.15.90 ===
135 2019-04-11 00:35:55 +0100 Tim-Philipp Müller <tim@centricular.com>
141 * gst-rtsp-server.doap:
145 2019-04-10 10:32:53 +0200 Ulf Olsson <ulfo@axis.com>
147 * gst/rtsp-server/rtsp-stream.c:
148 rtsp-stream: Add support for GCM (RFC 7714)
151 2019-03-28 00:27:37 +0100 Erlend Eriksen <erlend_ne@hotmail.com>
153 * gst/rtsp-server/rtsp-session-pool.c:
154 session pool: fix missing klass-> in klass->create_session
156 2019-03-23 19:16:17 +0000 Tim-Philipp Müller <tim@centricular.com>
159 g-i: pass --quiet to g-ir-scanner
160 This suppresses the annoying 'g-ir-scanner: link: cc ..' output
161 that we get even if everything works just fine.
162 We still get g-ir-scanner warnings and compiler warnings if
165 2019-03-23 19:15:48 +0000 Tim-Philipp Müller <tim@centricular.com>
168 g-i: silence 'nested extern' compiler warnings when building scanner binary
169 We need a nested extern in our init section for the scanner binary
170 so we can call gst_init to make sure GStreamer types are initialised
171 (they are not all lazy init via get_type functions, but some are in
172 exported variables). There doesn't seem to be any other mechanism to
173 achieve this, so just remove that warning, it's not important at all.
175 2019-03-21 11:49:10 +0000 Tim-Philipp Müller <tim@centricular.com>
178 meson: pass -Wno-unused to compiler if gstreamer debug system is disabled
180 2019-03-14 07:37:26 +0100 Göran Jönsson <goranjn@axis.com>
182 * gst/rtsp-server/rtsp-media.c:
183 * tests/check/gst/media.c:
184 rtsp-media: Handle set state when preparing.
185 Handle the situation when a call to gst_rtsp_media_set_state is done
186 when media status is preparing.
187 Also add unit test for this scenario.
188 The unit test simulate on a media level when two clients share a (live)
190 Both clients have done SETUP and got responses. Now client 1 is doing
191 play and client 2 is just closing the connection.
192 Then without patch there are a problem when
193 client1 is calling gst_rtsp_media_unsuspend in handle_play_request.
194 And client2 is doing closing connection we can end up in a call
195 to gst_rtsp_media_set_state when
196 priv->status == GST_RTSP_MEDIA_STATUS_PREPARING and all the logic for
197 shut down media is jumped over .
198 With this patch and this scenario we wait until
199 priv->status == GST_RTSP_MEDIA_STATUS_PREPARED and then continue to
200 execute after that and now we will execute the logic for
203 2019-03-04 09:13:30 +0000 Tim-Philipp Müller <tim@centricular.com>
211 === release 1.15.2 ===
213 2019-02-26 11:58:53 +0000 Tim-Philipp Müller <tim@centricular.com>
219 * gst-rtsp-server.doap:
223 2019-02-19 09:45:08 +0100 Göran Jönsson <goranjn@axis.com>
225 * gst/rtsp-server/rtsp-media.c:
226 * tests/check/gst/client.c:
227 rtsp-media: Fix multicast use case with common media
236 2019-01-16 12:59:11 +0100 Göran Jönsson <goranjn@axis.com>
238 * gst/rtsp-server/rtsp-client.c:
239 * gst/rtsp-server/rtsp-stream.c:
240 * gst/rtsp-server/rtsp-stream.h:
241 rtsp-server: remove recursive behavior
242 Introduce a threadpool to send rtp and rtcp to avoid recursive behavior.
244 2019-01-25 14:22:42 +0200 Sebastian Dröge <sebastian@centricular.com>
246 * gst/rtsp-server/rtsp-client.c:
247 rtsp-client: Only allow to set either a send_func or send_messages_func but not both
248 And route all messages through the send_func if no send_messages_func
250 We otherwise break backwards compatibility.
252 2018-09-17 22:18:46 +0300 Sebastian Dröge <sebastian@centricular.com>
254 * docs/libs/gst-rtsp-server-sections.txt:
255 * gst/rtsp-server/rtsp-client.c:
256 * gst/rtsp-server/rtsp-client.h:
257 * gst/rtsp-server/rtsp-stream.c:
258 rtsp-client: Add support for sending buffer lists directly
259 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
261 2018-06-27 12:17:07 +0200 Sebastian Dröge <sebastian@centricular.com>
263 * docs/libs/gst-rtsp-server-sections.txt:
264 * gst/rtsp-server/rtsp-client.c:
265 * gst/rtsp-server/rtsp-media.c:
266 * gst/rtsp-server/rtsp-stream-transport.c:
267 * gst/rtsp-server/rtsp-stream-transport.h:
268 * gst/rtsp-server/rtsp-stream.c:
269 * gst/rtsp-sink/gstrtspclientsink.c:
270 rtsp-server: Add support for buffer lists
271 This adds new functions for passing buffer lists through the different
272 layers without breaking API/ABI, and enables the appsink to actually
273 provide buffer lists.
274 This should already reduce CPU usage and potentially context switches a
275 bit by passing a whole buffer list from the appsink instead of
276 individual buffers. As a next step it would be necessary to
277 a) Add support for a vector of data for the GstRTSPMessage body
278 b) Add support for sending multiple messages at once to the
279 GstRTSPWatch and let it be handled internally
280 c) Adding API to GOutputStream that works like writev()
281 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
283 2018-12-04 14:12:04 +0100 Benjamin Berg <bberg@redhat.com>
285 * gst/rtsp-server/rtsp-client.c:
286 client: Fix crash in close handler
287 The close handler could trigger a crash because it invalidated the
288 watch_context while still leaving a source attached to it which would be
289 cleaned up at a later point.
291 2019-01-29 14:42:35 +0100 Edward Hervey <edward@centricular.com>
293 * gst/rtsp-server/rtsp-stream.c:
294 rtsp-stream: Use cached address when allocating sockets
295 If an address/port was previously decided upon (ex: multicast in the
296 SDP), then use that instead of re-creating another one
297 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/57
299 2018-12-27 11:28:17 +0100 Lars Wiréen <larswi@axis.com>
301 * gst/rtsp-server/rtsp-media.c:
302 rtsp-media: Fix race codition in finish_unprepare
303 The previous fix for race condition around finish_unprepare where the
304 function could be called twice assumed that the status wouldn't change
305 during execution of the function. This assumption is incorrect as the
306 state may change, for example if an error message arrives from the
308 Instead a flag keeping track on whether the finish_unprepare function
309 is currently executing is introduced and checked.
310 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/59
312 === release 1.15.1 ===
314 2019-01-17 02:26:48 +0000 Tim-Philipp Müller <tim@centricular.com>
320 * gst-rtsp-server.doap:
324 2018-12-05 15:07:25 +0100 Patricia Muscalu <patricia@axis.com>
326 * gst/rtsp-server/rtsp-stream.c:
327 Add source elements to the pipeline before activation
328 In plug_src we changed the element state before adding it to
329 the owner container. This prevented the pipeline from intercepting
330 a GST_STREAM_STATUS_TYPE_CREATE message from the pad in order
331 to assign a custom task pool.
332 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/53
334 2018-12-05 17:24:59 -0300 Thibault Saunier <tsaunier@igalia.com>
337 Automatic update of common submodule
338 From ed78bee to 59cb678
340 2018-11-20 19:12:09 +0100 Ingo Randolf <ingo.randolf@servus.at>
342 * examples/test-appsrc.c:
343 examples: test-appsrc: fix coding style error
345 2018-11-20 11:07:48 +0100 Ingo Randolf <ingo.randolf@servus.at>
347 * examples/test-appsrc.c:
348 examples: test-appsrc: fix buffer leak
350 2018-11-17 19:19:54 +0100 Patricia Muscalu <patricia@axis.com>
352 * gst/rtsp-server/rtsp-media.c:
353 rtsp-media: Update priv->blocked when linked streams are unblocked.
354 Media is considered to be blocked when all streams that belong to
355 that media are blocked.
356 This patch solves the problem of inconsistent updates of
357 priv->blocked that are not synchronized with the media state.
359 2018-11-17 18:18:27 +0100 Patricia Muscalu <patricia@axis.com>
361 * gst/rtsp-server/rtsp-media.c:
362 rtsp-media: Don't block streams before seeking
363 Before the seek operation is performed on media, it's required that
364 its pipeline is prepared <=> the pipeline is in the PAUSED state.
365 At this stage, all transport parts (transport sinks) have been successfully
366 added to the pipeline and there is no need for blocking the streams.
368 2018-11-17 16:11:53 +0100 Patricia Muscalu <patricia@axis.com>
370 * tests/check/gst/rtspserver.c:
371 tests: rtspserver: Add shared media test case for TCP
373 2018-11-06 18:21:54 +0100 Linus Svensson <linussn@axis.com>
375 * gst/rtsp-server/rtsp-stream.c:
376 rtsp-stream: Use seqnum-offset for rtpinfo
377 The sequence number in the rtpinfo is supposed to be the first RTP
378 sequence number. The "seqnum" property on a payloader is supposed to be
379 the number from the last processed RTP packet. The sequence number for
380 payloaders that inherit gstrtpbasepayload will not be correct in case of
381 buffer lists. In order to fix the seqnum property on the payloaders
382 gst-rtsp-server must get the sequence number for rtpinfo elsewhere and
383 "seqnum-offset" from the "stats" property contains the value of the
384 very first RTP packet in a stream. The server will, however, try to look
385 at the last simple in the sink element and only use properties on the
386 payloader in case there no sink elements yet, and by looking at the last
387 sample of the sink gives the server full control of which RTP packet it
388 looks at. If the payloader does not have the "stats" property, "seqnum"
389 is still used since "seqnum-offset" is only present in as part of
390 "stats" and this is still an issue not solved with this patch.
391 Needed for gst-plugins-base!17
393 2018-11-06 18:10:56 +0100 Linus Svensson <linussn@axis.com>
395 * gst/rtsp-server/rtsp-stream.c:
396 rtsp-stream: Plug memory leak
397 Attaching a GSource to a context will increase the refcount. The idle
398 source will never be free'd since the initial reference is never
401 2018-11-12 16:06:39 +0200 Jordan Petridis <jordan@centricular.com>
404 Add Gitlab CI configuration
405 This commit adds a .gitlab-ci.yml file, which uses a feature
406 to fetch the config from a centralized repository. The intent is
407 to have all the gstreamer modules use the same configuration.
408 The configuration is currently hosted at the gst-ci repository
409 under the gitlab/ci_template.yml path.
410 Part of https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/29
412 2018-11-05 05:56:35 +0000 Matthew Waters <matthew@centricular.com>
415 * gst-rtsp-server.doap:
416 Update git locations to gitlab
418 2018-11-01 14:20:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
420 * gst/rtsp-server/meson.build:
421 meson: add new onvif types
423 2018-11-01 12:49:51 +0200 Sebastian Dröge <sebastian@centricular.com>
425 * gst/rtsp-server/meson.build:
426 Add ONVIF subclass headers to the installed headers in meson.build too
428 2018-11-01 11:29:01 +0200 Sebastian Dröge <sebastian@centricular.com>
430 * gst/rtsp-server/rtsp-server-object.h:
431 * gst/rtsp-server/rtsp-server.h:
432 rtsp-server: Declare GstRTSPServer struct before anything else
433 It's needed by all kinds of other headers, including the ones that are
434 required for defining the GstRTSPServer struct itself and its API.
436 2018-11-01 10:23:02 +0200 Sebastian Dröge <sebastian@centricular.com>
438 * gst/rtsp-server/rtsp-onvif-client.h:
439 * gst/rtsp-server/rtsp-onvif-media-factory.h:
440 * gst/rtsp-server/rtsp-onvif-media.h:
441 * gst/rtsp-server/rtsp-onvif-server.h:
442 Mark all ONVIF-specific subclasses as Since 1.14
444 2018-11-01 10:18:22 +0200 Sebastian Dröge <sebastian@centricular.com>
446 * gst/rtsp-server/Makefile.am:
447 * gst/rtsp-server/meson.build:
448 * gst/rtsp-server/rtsp-context.h:
449 * gst/rtsp-server/rtsp-onvif-server.c:
450 * gst/rtsp-server/rtsp-onvif-server.h:
451 * gst/rtsp-server/rtsp-server-object.h:
452 * gst/rtsp-server/rtsp-server-prelude.h:
453 * gst/rtsp-server/rtsp-server.c:
454 * gst/rtsp-server/rtsp-server.h:
455 * gst/rtsp-server/rtsp-session.h:
456 Include ONVIF types from single-include rtsp-server.h
457 ... by actually making it a single-include header and moving everything
458 related to the GstRTSPServer type to rtsp-server-object.h instead.
459 Otherwise there are too many circular includes.
460 https://bugzilla.gnome.org/show_bug.cgi?id=797361
462 2018-10-18 07:25:05 +0200 Göran Jönsson <goranjn@axis.com>
464 * gst/rtsp-server/rtsp-client.c:
465 * gst/rtsp-server/rtsp-latency-bin.c:
466 * gst/rtsp-server/rtsp-stream.c:
467 * gst/rtsp-server/rtsp-stream.h:
468 rtsp-stream: use idle source in on_message_sent
469 When the underlying layers are running on_message_sent, this sometimes
470 causes the underlying layer to send more data, which will cause the
471 underlying layer to run callback on_message_sent again. This can go on
473 To break this chain, we introduce an idle source that takes care of
474 sending data if there are more to send when running callback
475 https://bugzilla.gnome.org/show_bug.cgi?id=797289
477 2018-10-20 16:14:53 +0200 Edward Hervey <edward@centricular.com>
479 * gst/rtsp-server/rtsp-client.c:
480 rtsp-client: Remove timeout GSource on cleanup
481 Avoids ending up with races where a timeout would still be around
482 *after* a client was gone. This could happen rather easily in
483 RTSP-over-HTTP mode on a local connection, where each RTSP message
484 would be sent as a different HTTP connection with the same tunnelid.
485 If not properly removed, that timeout would then try to free again
486 a client (and its contents).
488 2018-10-04 14:31:59 +0100 Tim-Philipp Müller <tim@centricular.com>
490 * gst/rtsp-server/Makefile.am:
491 autotools: fix distcheck
493 2018-09-12 11:55:15 +0200 Ognyan Tonchev <ognyan@axis.com>
495 * gst/rtsp-server/Makefile.am:
496 * gst/rtsp-server/meson.build:
497 * gst/rtsp-server/rtsp-latency-bin.c:
498 * gst/rtsp-server/rtsp-latency-bin.h:
499 * gst/rtsp-server/rtsp-onvif-media.c:
500 onvif: encapsulate onvif part into a bin
501 ...and thus do not let onvif affect pipelines latency
502 https://bugzilla.gnome.org/show_bug.cgi?id=797174
504 2018-09-27 19:57:13 +0200 Patricia Muscalu <patricia@dovakhiin.com>
506 * tests/check/gst/client.c:
507 tests: client: Avoid bind() failures in tests
508 https://bugzilla.gnome.org/show_bug.cgi?id=797059
510 2018-09-06 16:17:33 +0200 Patricia Muscalu <patricia@axis.com>
512 * gst/rtsp-server/rtsp-media-factory.c:
513 * gst/rtsp-server/rtsp-media-factory.h:
514 * gst/rtsp-server/rtsp-media.c:
515 * gst/rtsp-server/rtsp-media.h:
516 * gst/rtsp-server/rtsp-stream.c:
517 * gst/rtsp-server/rtsp-stream.h:
518 * tests/check/gst/client.c:
519 * tests/check/gst/mediafactory.c:
520 New property for socket binding to mcast addresses
521 By default the multicast sockets are bound to INADDR_ANY,
522 as it's not allowed to bind sockets to multicast addresses
523 in Windows. This default behaviour can be changed by setting
524 bind-mcast-address property on the media-factory object.
525 https://bugzilla.gnome.org/show_bug.cgi?id=797059
527 2018-09-24 09:36:21 +0100 Tim-Philipp Müller <tim@centricular.com>
530 * gst/rtsp-server/Makefile.am:
531 * gst/rtsp-server/meson.build:
532 * gst/rtsp-server/rtsp-address-pool.c:
533 * gst/rtsp-server/rtsp-auth.c:
534 * gst/rtsp-server/rtsp-client.c:
535 * gst/rtsp-server/rtsp-context.c:
536 * gst/rtsp-server/rtsp-media-factory-uri.c:
537 * gst/rtsp-server/rtsp-media-factory.c:
538 * gst/rtsp-server/rtsp-media.c:
539 * gst/rtsp-server/rtsp-mount-points.c:
540 * gst/rtsp-server/rtsp-params.c:
541 * gst/rtsp-server/rtsp-permissions.c:
542 * gst/rtsp-server/rtsp-sdp.c:
543 * gst/rtsp-server/rtsp-server-prelude.h:
544 * gst/rtsp-server/rtsp-server.c:
545 * gst/rtsp-server/rtsp-session-media.c:
546 * gst/rtsp-server/rtsp-session-pool.c:
547 * gst/rtsp-server/rtsp-session.c:
548 * gst/rtsp-server/rtsp-stream-transport.c:
549 * gst/rtsp-server/rtsp-stream.c:
550 * gst/rtsp-server/rtsp-thread-pool.c:
551 * gst/rtsp-server/rtsp-token.c:
553 libs: fix API export/import and 'inconsistent linkage' on MSVC
554 Export rtsp-server library API in headers when we're building the
555 library itself, otherwise import the API from the headers.
556 This fixes linker warnings on Windows when building with MSVC.
557 Fix up some missing config.h includes when building the lib which
558 is needed to get the export api define from config.h
559 https://bugzilla.gnome.org/show_bug.cgi?id=797185
561 2018-09-19 14:31:56 +0200 Edward Hervey <edward@centricular.com>
563 * gst/rtsp-server/rtsp-media-factory.c:
564 rtsp-media-factory: Add missing break statements
565 This resulted in warnings/assertions whenever one accessed the
566 max-mcast-ttl property.
570 2018-09-19 12:21:30 +0100 Tim-Philipp Müller <tim@centricular.com>
574 meson: add gobject-cast-checks, glib-asserts, glib-checks options
576 2018-09-19 12:17:57 +0100 Tim-Philipp Müller <tim@centricular.com>
580 * tests/check/meson.build:
581 meson: add option to disable build of rtspclientsink plugin
583 2018-09-19 12:10:14 +0100 Tim-Philipp Müller <tim@centricular.com>
586 meson: re-arrange options
588 2018-09-01 11:21:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
592 * tests/check/meson.build:
594 meson: Use feature option for tests option
595 This was somehow missed the last time around.
597 2018-08-31 14:42:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
599 * gst/rtsp-server/meson.build:
601 meson: Maintain macOS ABI through dylib versioning
602 Requires Meson 0.48, but the feature will be ignored on older versions
603 so it's safe to add it without bumping the requirement.
605 https://github.com/mesonbuild/meson/blob/master/docs/markdown/Reference-manual.md#shared_library
607 2018-08-31 17:20:47 +1000 Matthew Waters <matthew@centricular.com>
609 * gst/rtsp-sink/meson.build:
611 meson: add pkg-config file for the rtspclientsink plugin
613 2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
615 * gst/rtsp-server/rtsp-client.c:
616 * tests/check/gst/client.c:
617 rtsp-client: Avoid reuse of channel numbers for interleaved
618 If a (strange) client would reuse interleaved channel numbers in
619 multiple SETUP requests, we should not accept them. The channel
620 numbers are used for looking up stream transports in the
621 priv->transports hash table, and transports disappear from the table
622 if channel numbers are reused.
623 RFC 7826 (RTSP 2.0), Section 18.54, clarifies that it is OK for the
624 server to change the channel numbers suggested by the client.
625 https://bugzilla.gnome.org/show_bug.cgi?id=796988
627 2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
629 * tests/check/gst/client.c:
630 rtsp-client: Add unit test of SETUP for RTSP/RTP/TCP
631 Allow regex for matching transport header against expected pattern.
632 https://bugzilla.gnome.org/show_bug.cgi?id=796988
634 2018-08-15 18:57:27 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
636 * tests/check/meson.build:
637 meson: There is no gstreamer-plugins-good-1.0.pc
638 There is no installed version of that, only an uninstalled version.
640 2018-08-14 14:31:55 +0300 Sebastian Dröge <sebastian@centricular.com>
642 * gst/rtsp-server/rtsp-client.c:
643 * tests/check/gst/stream.c:
644 Fix indentation again
646 2018-07-26 12:01:16 +0200 Patricia Muscalu <patricia@axis.com>
648 * gst/rtsp-server/rtsp-client.c:
649 * gst/rtsp-server/rtsp-stream.c:
650 * gst/rtsp-server/rtsp-stream.h:
651 * tests/check/gst/client.c:
652 * tests/check/gst/stream.c:
653 stream: Added a list of multicast client addresses
654 When media is shared, the same media stream can be sent
655 to multiple multicast groups. Currently, there is no API
656 to retrieve multicast addresses from the stream.
657 When calling gst_rtsp_stream_get_multicast_address() function,
658 only the first multicast address is returned.
659 With this patch, each multicast destination requested in SETUP
660 will be stored in an internal list (call to
661 gst_rtsp_stream_add_multicast_client_address()).
662 The list of multicast groups requested by the clients can be
663 retrieved by calling gst_rtsp_stream_get_multicast_client_addresses().
664 There still exist some problems with the current implementation
665 in the multicast case:
666 1) The receiving part is currently only configured with
667 regard to the first multicast client (see
668 https://bugzilla.gnome.org/show_bug.cgi?id=796917).
669 2) Secondly, of security reasons, some constraints should be
670 put on the requested multicast destinations (see
671 https://bugzilla.gnome.org/show_bug.cgi?id=796916).
672 Change-Id: I6b060746e472a0734cc2fd828ffe4ea2956733ea
673 https://bugzilla.gnome.org/show_bug.cgi?id=793441
675 2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
677 * gst/rtsp-server/rtsp-client.c:
678 * gst/rtsp-server/rtsp-stream.c:
679 * gst/rtsp-server/rtsp-stream.h:
680 * tests/check/gst/client.c:
681 stream: Choose the maximum ttl value provided by multicast clients
682 The maximum ttl value provided so far by the multicast clients
683 will be chosen and reported in the response to the current
685 Change-Id: I5408646e3b5a0a224d907ae215bdea60c4f1905f
686 https://bugzilla.gnome.org/show_bug.cgi?id=793441
688 2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
690 * gst/rtsp-server/rtsp-stream.c:
691 * tests/check/gst/client.c:
692 rtsp-stream: Don't require address pool in the transport specific case
693 If "transport.client-settings" parameter is set to true, the client is
694 allowed to specify destination, ports and ttl.
695 There is no need for pre-configured address pool.
696 Change-Id: I6ae578fb5164d78e8ec1e2ee82dc4eaacd0912d1
697 https://bugzilla.gnome.org/show_bug.cgi?id=793441
699 2018-07-24 14:02:40 +0200 Patricia Muscalu <patricia@axis.com>
701 * gst/rtsp-server/rtsp-client.c:
702 * tests/check/gst/client.c:
703 client: Don't reserve multicast address in the client setting case
704 When two multicast clients request specific transport
705 configurations, and "transport.client-settings" parameter is
706 set to true, it's wrong to actually require that these two
707 clients request the same multicast group.
708 Removed test_client_multicast_invalid_transport_specific test
709 cases as they wrongly require that the requested destination
710 address is supposed to be present in the address pool, also in
711 the case when "transport.client-settings" parameter is set to true.
712 Change-Id: I4580182ef35996caf644686d6139f72ec599c9fa
713 https://bugzilla.gnome.org/show_bug.cgi?id=793441
715 2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
717 * gst/rtsp-server/rtsp-media-factory.c:
718 * gst/rtsp-server/rtsp-media-factory.h:
719 * gst/rtsp-server/rtsp-media.c:
720 * gst/rtsp-server/rtsp-media.h:
721 * gst/rtsp-server/rtsp-stream.c:
722 * gst/rtsp-server/rtsp-stream.h:
723 * tests/check/gst/mediafactory.c:
724 Add new API for setting/getting maximum multicast ttl value
725 Change-Id: I5ef4758188c14785e17fb8fbf42a3dc0cb054233
726 https://bugzilla.gnome.org/show_bug.cgi?id=793441
728 2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
730 * gst/rtsp-server/rtsp-stream.c:
731 rtsp-stream: avoid duplicating the first multicast client
732 In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
733 clients were dynamically added and removed to the multicast
734 udp sinks, as such we should no longer add a first client in
735 set_multicast_socket_for_udpsink
736 https://bugzilla.gnome.org/show_bug.cgi?id=793441
738 2018-08-14 14:25:53 +0300 Sebastian Dröge <sebastian@centricular.com>
740 * gst/rtsp-server/rtsp-stream.c:
741 Revert "rtsp-stream: avoid duplicating the first multicast client"
742 This reverts commit 33570944401747f44d8ebfec535350651413fb92.
743 Commits where accidentially squashed together
745 2018-08-14 14:25:42 +0300 Sebastian Dröge <sebastian@centricular.com>
747 * gst/rtsp-server/rtsp-client.c:
748 * gst/rtsp-server/rtsp-media-factory.c:
749 * gst/rtsp-server/rtsp-media-factory.h:
750 * gst/rtsp-server/rtsp-media.c:
751 * gst/rtsp-server/rtsp-media.h:
752 * gst/rtsp-server/rtsp-stream.c:
753 * gst/rtsp-server/rtsp-stream.h:
754 * tests/check/gst/client.c:
755 * tests/check/gst/mediafactory.c:
756 Revert "Add new API for setting/getting maximum multicast ttl value"
757 This reverts commit 7f0ae77e400fb8a0462a76a5dd2e63e12c4a2e52.
758 Commits where accidentially squashed together
760 2018-08-14 14:25:37 +0300 Sebastian Dröge <sebastian@centricular.com>
762 * gst/rtsp-server/rtsp-stream.c:
763 * tests/check/gst/client.c:
764 Revert "rtsp-stream: Don't require address pool in the transport specific case"
765 This reverts commit a9db3e7f092cfeb5475e9aa24b1e91906c141d52.
766 Commits where accidentially squashed together
768 2018-08-14 14:25:14 +0300 Sebastian Dröge <sebastian@centricular.com>
770 * gst/rtsp-server/rtsp-client.c:
771 * gst/rtsp-server/rtsp-stream.c:
772 * gst/rtsp-server/rtsp-stream.h:
773 * tests/check/gst/client.c:
774 * tests/check/gst/stream.c:
775 Revert "stream: Choose the maximum ttl value provided by multicast clients"
776 This reverts commit 499e437e501215849d24cdaa157e0edf4de097d0.
777 Commits where accidentially squashed together
779 2018-08-14 14:10:56 +0300 Sebastian Dröge <sebastian@centricular.com>
781 * examples/test-auth-digest.c:
782 examples: Fix indentation
784 2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
786 * gst/rtsp-server/rtsp-client.c:
787 * gst/rtsp-server/rtsp-stream.c:
788 * gst/rtsp-server/rtsp-stream.h:
789 * tests/check/gst/client.c:
790 * tests/check/gst/stream.c:
791 stream: Choose the maximum ttl value provided by multicast clients
792 The maximum ttl value provided so far by the multicast clients
793 will be chosen and reported in the response to the current
795 https://bugzilla.gnome.org/show_bug.cgi?id=793441
797 2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
799 * gst/rtsp-server/rtsp-stream.c:
800 * tests/check/gst/client.c:
801 rtsp-stream: Don't require address pool in the transport specific case
802 If "transport.client-settings" parameter is set to true, the client is
803 allowed to specify destination, ports and ttl.
804 There is no need for pre-configured address pool.
805 https://bugzilla.gnome.org/show_bug.cgi?id=793441
807 2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
809 * gst/rtsp-server/rtsp-client.c:
810 * gst/rtsp-server/rtsp-media-factory.c:
811 * gst/rtsp-server/rtsp-media-factory.h:
812 * gst/rtsp-server/rtsp-media.c:
813 * gst/rtsp-server/rtsp-media.h:
814 * gst/rtsp-server/rtsp-stream.c:
815 * gst/rtsp-server/rtsp-stream.h:
816 * tests/check/gst/client.c:
817 * tests/check/gst/mediafactory.c:
818 Add new API for setting/getting maximum multicast ttl value
819 https://bugzilla.gnome.org/show_bug.cgi?id=793441
821 2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
823 * gst/rtsp-server/rtsp-stream.c:
824 rtsp-stream: avoid duplicating the first multicast client
825 In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
826 clients were dynamically added and removed to the multicast
827 udp sinks, as such we should no longer add a first client in
828 set_multicast_socket_for_udpsink
829 https://bugzilla.gnome.org/show_bug.cgi?id=793441
831 2018-08-06 15:33:04 -0400 Thibault Saunier <tsaunier@igalia.com>
833 * gst/rtsp-server/Makefile.am:
834 rtsp-server: Add gstreamer-base gir dir in autotools
836 2018-07-25 19:54:55 +0200 Mathieu Duponchelle <mathieu@centricular.com>
838 * gst/rtsp-server/rtsp-client.c:
839 * gst/rtsp-server/rtsp-stream.c:
840 rtsp-client: always allocate both IPV4 and IPV6 sockets
841 multiudpsink does not support setting the socket* properties
842 after it has started, which meant that rtsp-server could no
843 longer serve on both IPV4 and IPV6 sockets since the patches
844 from https://bugzilla.gnome.org/show_bug.cgi?id=757488 were
846 When first connecting an IPV6 client then an IPV4 client,
847 multiudpsink fell back to using the IPV6 socket.
848 When first connecting an IPV4 client, then an IPV6 client,
849 multiudpsink errored out, released the IPV4 socket, then
850 crashed when trying to send a message on NULL nevertheless,
851 that is however a separate issue.
852 This could probably be fixed by handling the setting of
853 sockets in multiudpsink after it has started, that will
854 however be a much more significant effort.
855 For now, this commit simply partially reverts the behaviour
856 of rtsp-stream: it will continue to only create the udpsinks
857 when needed, as was the case since the patches were merged,
858 it will however when creating them, always allocate both
859 sockets and set them on the sink before it starts, as was
860 the case prior to the patches.
861 Transport configuration will only error out if the allocation
862 of UDP sockets fails for the actual client's family, this
863 also downgrades the GST_ERRORs in alloc_ports_one_family
864 to GST_WARNINGs, as failing to allocate is no longer
866 https://bugzilla.gnome.org/show_bug.cgi?id=796875
868 2018-07-25 17:22:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
872 meson: Convert common options to feature options
873 These are necessary for gst-build to set options correctly. The
874 remaining automagic option is cgroup support in examples.
875 https://bugzilla.gnome.org/show_bug.cgi?id=795107
877 2018-07-23 18:03:51 +0300 Sebastian Dröge <sebastian@centricular.com>
879 * gst/rtsp-server/rtsp-stream.c:
880 rtsp-stream: Slightly simplify locking
882 2018-06-28 11:22:21 +0200 David Svensson Fors <davidsf@axis.com>
884 * gst/rtsp-server/rtsp-client.c:
885 * gst/rtsp-server/rtsp-stream-transport.c:
886 * gst/rtsp-server/rtsp-stream-transport.h:
887 * gst/rtsp-server/rtsp-stream.c:
888 Limit queued TCP data messages to one per stream
889 Before, the watch backlog size in GstRTSPClient was changed
890 dynamically between unlimited and a fixed size, trying to avoid both
891 unlimited memory usage and deadlocks while waiting for place in the
892 queue. (Some of the deadlocks were described in a long comment in
894 In the previous commit, we changed to a fixed backlog size of 100.
895 This is possible, because we now handle RTP/RTCP data messages differently
896 from RTSP request/response messages.
897 The data messages are messages tunneled over TCP. We allow at most one
898 queued data message per stream in GstRTSPClient at a time, and
899 successfully sent data messages are acked by sending a "message-sent"
900 callback from the GstStreamTransport. Until that ack comes, the
901 GstRTSPStream does not call pull_sample() on its appsink, and
902 therefore the streaming thread in the pipeline will not be blocked
903 inside GstRTSPClient, waiting for a place in the queue.
904 pull_sample() is called when we have both an ack and a "new-sample"
905 signal from the appsink. Then, we know there is a buffer to write.
906 RTSP request/response messages are not acked in the same way as data
907 messages. The rest of the 100 places in the queue are used for
908 them. If the queue becomes full of request/response messages, we
909 return an error and close the connection to the client.
910 Change-Id: I275310bc90a219ceb2473c098261acc78be84c97
912 2018-06-28 11:22:13 +0200 David Svensson Fors <davidsf@axis.com>
914 * gst/rtsp-server/rtsp-client.c:
915 rtsp-client: Use fixed backlog size
916 Change to using a fixed backlog size WATCH_BACKLOG_SIZE.
917 Preparation for the next commit, which changes to a different way of
918 avoiding both deadlocks and unlimited memory usage with the watch
921 2018-07-16 21:57:08 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
923 * gst/rtsp-server/rtsp-media.c:
924 rtsp-media: unref clock (if set) when finalizing
925 https://bugzilla.gnome.org/show_bug.cgi?id=796814
927 2018-07-16 21:56:44 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
929 * docs/libs/gst-rtsp-server-sections.txt:
930 rtsp-media: add gst_rtsp_media_*_set_clock to docs
931 https://bugzilla.gnome.org/show_bug.cgi?id=796814
933 2018-07-12 19:01:54 +0100 Tim-Philipp Müller <tim@centricular.com>
935 * gst/rtsp-server/rtsp-media-factory.c:
936 media-factory: unref old clock when setting new clock
937 https://bugzilla.gnome.org/show_bug.cgi?id=796724
939 2018-06-29 15:20:57 -0700 Brendan Shanks <brendan.shanks@teradek.com>
941 * gst/rtsp-server/rtsp-media-factory.c:
942 media-factory: unref clock in finalize
943 https://bugzilla.gnome.org/show_bug.cgi?id=796724
945 2018-07-12 18:57:21 +0100 Tim-Philipp Müller <tim@centricular.com>
947 * gst/rtsp-server/rtsp-onvif-media.c:
948 rtsp-onvif-media: fix g-ir-scanner warnings
950 2018-07-10 23:56:23 +0100 Tim-Philipp Müller <tim@centricular.com>
953 .gitignore: add another example binary
955 2018-07-10 23:55:20 +0100 Tim-Philipp Müller <tim@centricular.com>
957 * examples/meson.build:
958 meson: add new test-appsrc2 example to meson build
960 2018-07-10 23:53:41 +0100 Tim-Philipp Müller <tim@centricular.com>
962 * examples/Makefile.am:
963 examples: fix build of new test-appsrc2 example
964 Need to link against libgstapp-1.0.
966 2018-07-11 01:25:51 +1000 Jan Schmidt <jan@centricular.com>
968 * examples/.gitignore:
969 * examples/Makefile.am:
970 * examples/test-appsrc2.c:
971 examples: Add test-appsrc2
972 Add an example of feeding both audio and video into an RTSP
975 2016-01-08 18:12:14 -0500 Louis-Francis Ratté-Boulianne <lfrb@collabora.com>
977 * gst/rtsp-server/rtsp-client.c:
978 client: Strip transport parts as whitespaces could be around commas
979 https://bugzilla.gnome.org/show_bug.cgi?id=758428
981 2018-06-27 08:30:42 +0200 Göran Jönsson <goranjn@axis.com>
983 * gst/rtsp-server/rtsp-stream.c:
984 rtsp-stream: avoid pushing data on unlinked udpsrc pad during setup
985 Fix race when setting up source elements.
986 Since we set the source element(s) to PLAYING state before hooking
987 them up to the downstream funnel, it's possible for the source element
988 to receive packets before we actually get to linking it to the funnel,
989 in which case buffers would be pushed out on an unlinked pad, causing
990 it to error out and stop receiving more data.
991 We fix this by blocking the source's srcpad until we have linked it.
992 https://bugzilla.gnome.org/show_bug.cgi?id=796160
994 2018-03-21 10:56:51 +0100 Ognyan Tonchev <ognyan@axis.com>
996 * gst/rtsp-server/rtsp-stream.c:
997 rtsp-stream: Fix mismatch between allowed and configured protocols
998 https://bugzilla.gnome.org/show_bug.cgi?id=796679
1000 2017-02-01 09:44:50 +0100 Ulf Olsson <ulfo@axis.com>
1002 * gst/rtsp-server/rtsp-stream.c:
1003 rtsp-stream: Emit a signal when the SRTP decoder is created
1004 https://bugzilla.gnome.org/show_bug.cgi?id=778080
1006 2018-03-13 11:10:35 +0100 Patricia Muscalu <patricia@axis.com>
1008 * gst/rtsp-server/rtsp-stream.c:
1009 rtsp-stream: Don't require presence of sinks in _get_*_socket()
1010 Transport specific sink elements are added to the pipeline
1011 in PLAY request and sockets are already created in SETUP so
1012 it's actually wrong to require the presence of sinks in
1013 _get_*_socket() functions.
1014 https://bugzilla.gnome.org/show_bug.cgi?id=793441
1016 2018-02-14 10:41:02 +0100 Patricia Muscalu <patricia@axis.com>
1018 * gst/rtsp-server/rtsp-stream.c:
1019 rtsp-stream: Update transport for multicast clients as well
1020 If a multicast client requests different transport settings
1021 than the existing one make sure that this new transport
1022 configuruation is propagated to the multicast udp sink.
1023 https://bugzilla.gnome.org/show_bug.cgi?id=793441
1025 2018-02-13 11:04:36 +0100 Patricia Muscalu <patricia@axis.com>
1027 * gst/rtsp-server/rtsp-stream.c:
1028 rtsp-stream: Set the multicast TTL parameter on multicast udp sinks
1029 And not on unicast udp sinks
1030 https://bugzilla.gnome.org/show_bug.cgi?id=793441
1032 2018-06-24 12:44:26 +0200 Tim-Philipp Müller <tim@centricular.com>
1034 * gst/rtsp-server/rtsp-address-pool.c:
1035 * gst/rtsp-server/rtsp-auth.c:
1036 * gst/rtsp-server/rtsp-client.c:
1037 * gst/rtsp-server/rtsp-media-factory-uri.c:
1038 * gst/rtsp-server/rtsp-media-factory.c:
1039 * gst/rtsp-server/rtsp-media.c:
1040 * gst/rtsp-server/rtsp-mount-points.c:
1041 * gst/rtsp-server/rtsp-server.c:
1042 * gst/rtsp-server/rtsp-session-media.c:
1043 * gst/rtsp-server/rtsp-session-pool.c:
1044 * gst/rtsp-server/rtsp-session.c:
1045 * gst/rtsp-server/rtsp-stream-transport.c:
1046 * gst/rtsp-server/rtsp-stream.c:
1047 * gst/rtsp-server/rtsp-thread-pool.c:
1048 Update for g_type_class_add_private() deprecation in recent GLib
1050 2018-06-24 12:45:49 +0200 Tim-Philipp Müller <tim@centricular.com>
1052 * gst/rtsp-server/rtsp-auth.c:
1053 * gst/rtsp-server/rtsp-media.c:
1054 * gst/rtsp-server/rtsp-sdp.c:
1055 * gst/rtsp-server/rtsp-stream.c:
1058 2018-06-22 23:17:08 +1000 Jan Schmidt <jan@centricular.com>
1060 * examples/Makefile.am:
1061 * examples/test-video-disconnect.c:
1062 examples: Add test-video-disconnect example
1063 Simple example which cuts off all clients 10 seconds
1064 after the first one connects.
1066 2018-06-20 04:37:11 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1068 * docs/libs/gst-rtsp-server-sections.txt:
1069 * examples/test-auth-digest.c:
1070 * gst/rtsp-server/rtsp-auth.c:
1071 * gst/rtsp-server/rtsp-auth.h:
1072 rtsp-auth: Add support for parsing .htdigest files
1073 Passwords are usually not stored in clear text, but instead
1074 stored already hashed in a .htdigest file.
1075 Add support for parsing such files, add API to allow setting
1076 a custom realm in RTSPAuth, and update the digest example.
1077 https://bugzilla.gnome.org/show_bug.cgi?id=796637
1079 2018-06-19 14:53:02 +1000 Matthew Waters <matthew@centricular.com>
1081 * gst/rtsp-sink/gstrtspclientsink.c:
1082 * gst/rtsp-sink/gstrtspclientsink.h:
1083 rtspclientsink: fix waiting for multiple streams
1084 We were previously only ever waiting for a single stream to notify it's
1085 blocked status through GstRTSPStreamBlocking. Actually count streams to
1087 Fixes rtspclientsink sending SDP's without out some of the input
1089 https://bugzilla.gnome.org/show_bug.cgi?id=796624
1091 2018-06-20 04:30:04 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1093 * docs/libs/gst-rtsp-server-sections.txt:
1094 docs: add missing auth methods
1096 2018-06-20 00:10:18 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1098 * gst/rtsp-server/rtsp-stream.c:
1099 rtsp-stream: only create funnel if it didn't exist already.
1100 This precented using multiple protocols for the same stream.
1101 https://bugzilla.gnome.org/show_bug.cgi?id=796634
1103 2018-06-20 01:35:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1105 * examples/meson.build:
1106 meson: build auth-digest example
1108 2018-06-05 08:44:44 +0200 Patricia Muscalu <patricia@axis.com>
1110 * gst/rtsp-server/rtsp-client.c:
1111 * gst/rtsp-server/rtsp-media.c:
1112 * gst/rtsp-server/rtsp-sdp.c:
1113 * gst/rtsp-server/rtsp-session-media.c:
1114 * gst/rtsp-server/rtsp-stream-transport.c:
1115 Get payloader stats only for the sending streams
1116 Get/set payloader properties only for streams that actually
1117 contain a payloader element.
1118 https://bugzilla.gnome.org/show_bug.cgi?id=796523
1120 2018-05-18 14:53:49 +0200 Edward Hervey <edward@centricular.com>
1122 * gst/rtsp-server/Makefile.am:
1123 Makefile: Don't hardcode libtool for g-i build
1124 Similar to the other commits in core/base/bad
1126 2018-05-08 14:13:31 +0200 Johan Bjäreholt <johanbj@axis.com>
1128 * gst/rtsp-server/rtsp-onvif-media-factory.h:
1129 rtsp-onvif-media-factory: export gst_rtsp_onvif_media_factory_requires_backchannel
1130 https://bugzilla.gnome.org/show_bug.cgi?id=796229
1132 2018-05-09 04:09:02 +1000 Jan Schmidt <jan@centricular.com>
1134 * gst/rtsp-sink/gstrtspclientsink.c:
1135 rtspclientsink: Don't deadlock in preroll on early close
1136 If the connection is closed very early, the flushing
1137 marker might not get set and rtspclientsink can get
1138 deadlocked waiting for preroll forever.
1139 https://bugzilla.gnome.org/show_bug.cgi?id=786961
1141 2018-05-05 19:51:52 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
1144 * meson_options.txt:
1145 meson: Update option names to omit disable_ and with- prefixes
1146 Also yield common options to the outer project (gst-build in our case)
1147 so that they don't have to be set manually.
1149 2018-04-25 11:00:32 +0100 Tim-Philipp Müller <tim@centricular.com>
1152 meson: use -Wl,-Bsymbolic-functions where supported
1153 Just like the autotools build.
1155 2018-04-22 20:09:01 +0100 Tim-Philipp Müller <tim@centricular.com>
1158 * tests/check/Makefile.am:
1159 configure: check for -good and -bad plugins only in uninstalled setup
1160 Avoids confusing configure messages looking or a -good .pc file
1162 Also use plugindir variables that common macros set while at it.
1163 https://bugzilla.gnome.org/show_bug.cgi?id=795466
1165 2018-04-17 11:03:11 +0200 Joakim Johansson <joakimj@axis.com>
1167 * gst/rtsp-server/rtsp-client.c:
1168 rtsp-client: Fix session timeout
1169 When streaming data over TCP then is not the keep-alive
1170 functionality working.
1171 The reason is that the function do_send_data have changed
1172 to boolean but the code is still checking the received result
1173 from send_func with GST_RTSP_OK.
1174 The result is that a successful send_func will always lead to
1175 that do_send_data is returning false and the keep-alive will
1177 https://bugzilla.gnome.org/show_bug.cgi?id=795321
1179 2018-04-02 22:49:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1181 * docs/libs/gst-rtsp-server-sections.txt:
1182 * gst/rtsp-server/rtsp-media.c:
1183 * gst/rtsp-server/rtsp-sdp.c:
1184 * gst/rtsp-server/rtsp-stream.c:
1185 * gst/rtsp-server/rtsp-stream.h:
1186 * gst/rtsp-sink/gstrtspclientsink.c:
1187 * gst/rtsp-sink/gstrtspclientsink.h:
1188 Implement support for ULP Forward Error Correction
1189 In this initial commit, interface is only exposed for RECORD,
1190 further work will be needed in rtspsrc to support this for
1192 https://bugzilla.gnome.org/show_bug.cgi?id=794911
1194 2018-04-17 17:47:30 +0300 Sebastian Dröge <sebastian@centricular.com>
1196 * gst/rtsp-server/rtsp-onvif-media.c:
1197 Revert "rtsp-server: Switch around sendonly/recvonly attributes"
1198 This reverts commit 3d275b1345b76151418e3f56ed014d9089ac1a57.
1199 While RFC 3264 (SDP) says that sendonly/recvonly are from the point of view of
1200 the requester, the actual RTSP RFCs (RFC 2326 / 7826) disagree and say
1201 the opposite, just like the ONVIF standard.
1202 Let's follow those RFCs as we're doing RTSP here, and add a property at
1203 a later time if needed to switch to the SDP RFC behaviour.
1204 https://bugzilla.gnome.org/show_bug.cgi?id=793964
1206 2018-04-16 10:53:52 +0100 Tim-Philipp Müller <tim@centricular.com>
1209 Automatic update of common submodule
1210 From 3fa2c9e to ed78bee
1212 2018-04-04 10:06:06 +0300 Sebastian Dröge <sebastian@centricular.com>
1214 * gst/rtsp-server/rtsp-client.c:
1215 * gst/rtsp-server/rtsp-media-factory.c:
1216 * gst/rtsp-server/rtsp-media.c:
1217 * gst/rtsp-server/rtsp-stream.c:
1218 * tests/check/gst/rtspclientsink.c:
1219 gst: Run everything through gst-indent again
1221 2018-04-03 08:57:47 +0200 Branko Subasic <branko@axis.com>
1223 * gst/rtsp-server/rtsp-media.c:
1224 * tests/check/gst/media.c:
1225 rtsp-media: query the position on active streams if media is complete
1226 If the media is complete, i.e. one or more streams have been configured
1227 with sinks, then we want to query the position on those streams only.
1228 A query on an incomplete stream may return a position that originates from
1230 https://bugzilla.gnome.org/show_bug.cgi?id=794964
1232 2018-04-02 12:35:04 +0100 Tim-Philipp Müller <tim@centricular.com>
1234 * gst/rtsp-sink/gstrtspclientsink.c:
1235 rtspclientsink: make sure not to use freed string
1236 Set transport string to NULL after freeing it, so that
1237 at worst we get a NULL pointer if constructing a new
1238 transport string fails (which shouldn't really fail here).
1239 Also check return value of that, just in case.
1242 2018-03-30 23:34:01 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1244 * gst/rtsp-server/rtsp-client.c:
1245 rtsp-client: do not free string passed to take_header
1247 2018-03-30 23:10:10 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1249 * gst/rtsp-server/rtsp-stream.c:
1250 rtsp-stream: do not take lock in request_aux_receiver
1251 Added it right before pushing the previous commit, it is
1252 incorrect and deadlocks because this function gets called
1253 from the join_bin thread, which already holds the lock,
1254 that's the reason why request_aux_sender didn't take the
1257 2018-03-29 22:49:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1259 * docs/libs/gst-rtsp-server-sections.txt:
1260 * gst/rtsp-server/rtsp-media-factory.c:
1261 * gst/rtsp-server/rtsp-media-factory.h:
1262 * gst/rtsp-server/rtsp-media.c:
1263 * gst/rtsp-server/rtsp-media.h:
1264 * gst/rtsp-server/rtsp-stream.c:
1265 * gst/rtsp-server/rtsp-stream.h:
1266 rtsp-server: add API to enable retransmission requests
1267 "do-retransmission" was previously set when rtx-time != 0,
1268 which made no sense as do-retransmission is used to enable
1269 the sending of retransmission requests, where as rtx-time
1270 is used by the peer to enable storing of buffers in order
1271 to respond to retransmission requests.
1272 rtsp-media now also provides a callback for the
1273 request-aux-receiver signal.
1274 https://bugzilla.gnome.org/show_bug.cgi?id=794822
1276 2018-03-29 16:18:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1278 * gst/rtsp-sink/gstrtspclientsink.c:
1279 rtspclientsink: add rtx ssrc to mikey's crypto sessions
1280 https://bugzilla.gnome.org/show_bug.cgi?id=794813
1282 2018-03-29 16:15:45 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1284 * gst/rtsp-sink/gstrtspclientsink.c:
1285 rtspclientsink: Handle the KeyMgmt header in ANNOUNCE response
1286 This in order to be able to decrypt the RTCP backchannel
1287 https://bugzilla.gnome.org/show_bug.cgi?id=794813
1289 2018-03-29 16:12:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1291 * gst/rtsp-server/rtsp-client.c:
1292 rtsp-client: Send KeyMgmt header in ANNOUNCE response
1293 When sending back an encrypted RTCP back channel, it is useful
1294 for the client to know the encryption key.
1295 https://bugzilla.gnome.org/show_bug.cgi?id=794813
1297 2018-03-29 16:06:31 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1299 * gst/rtsp-server/rtsp-client.c:
1300 * gst/rtsp-server/rtsp-stream.c:
1301 * gst/rtsp-server/rtsp-stream.h:
1302 rtsp-stream: extract handle_keymgmt from rtsp-client
1303 rtspclientsink will also need to parse KeyMgmt headers
1304 sent by the server to decrypt the RTCP backchannel stream
1305 https://bugzilla.gnome.org/show_bug.cgi?id=794813
1307 2018-03-29 02:51:02 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1309 * gst/rtsp-sink/gstrtspclientsink.c:
1310 * tests/check/gst/rtspclientsink.c:
1311 rtspclientsink: Fix client ports for the RTCP backchannel
1312 This was broken since the work for delayed transport creation
1313 was merged: the creation of the transports string depends on
1314 calling stream_get_server_port, which only starts returning
1315 something meaningful after a call to stream_allocate_udp_sockets
1316 has been made, this function expects a transport that we parse
1317 from the transport string ...
1318 Significant refactoring is in order, but does not look entirely
1319 trivial, for now we put a band aid on and create a second transport
1320 string after the stream has been completed, to pass it in
1321 the request headers instead of the previous, incomplete one.
1322 https://bugzilla.gnome.org/show_bug.cgi?id=794789
1324 2018-02-15 13:26:16 +0100 Göran Jönsson <goranjn@axis.com>
1326 * gst/rtsp-server/rtsp-client.c:
1327 rtsp-client:Error handling when equal http session cookie
1328 There are some clients that are sending same session cookie on random
1330 https://bugzilla.gnome.org/show_bug.cgi?id=753616
1332 2018-03-20 16:21:37 +0200 Sebastian Dröge <sebastian@centricular.com>
1334 * gst/rtsp-server/rtsp-media-factory-uri.c:
1335 rtsp-media-factory-uri: Fix compilation with latest GLib
1336 rtsp-media-factory-uri.c: In function ‘rtsp_media_factory_uri_create_element’:
1337 rtsp-media-factory-uri.c:621:17: error: assignment from incompatible pointer type [-Werror=incompatible-pointer-types]
1338 data->factory = g_object_ref (factory);
1341 2018-03-20 10:21:36 +0000 Tim-Philipp Müller <tim@centricular.com>
1349 === release 1.14.0 ===
1351 2018-03-19 20:27:04 +0000 Tim-Philipp Müller <tim@centricular.com>
1357 * gst-rtsp-server.doap:
1361 === release 1.13.91 ===
1363 2018-03-13 19:28:33 +0000 Tim-Philipp Müller <tim@centricular.com>
1369 * gst-rtsp-server.doap:
1373 2018-03-13 13:30:41 +0000 Tim-Philipp Müller <tim@centricular.com>
1375 * gst/rtsp-server/Makefile.am:
1376 * gst/rtsp-server/meson.build:
1377 * gst/rtsp-server/rtsp-address-pool.h:
1378 * gst/rtsp-server/rtsp-auth.h:
1379 * gst/rtsp-server/rtsp-client.h:
1380 * gst/rtsp-server/rtsp-context.h:
1381 * gst/rtsp-server/rtsp-media-factory-uri.h:
1382 * gst/rtsp-server/rtsp-media-factory.h:
1383 * gst/rtsp-server/rtsp-media.h:
1384 * gst/rtsp-server/rtsp-mount-points.h:
1385 * gst/rtsp-server/rtsp-onvif-client.h:
1386 * gst/rtsp-server/rtsp-onvif-media-factory.h:
1387 * gst/rtsp-server/rtsp-onvif-media.h:
1388 * gst/rtsp-server/rtsp-onvif-server.h:
1389 * gst/rtsp-server/rtsp-params.h:
1390 * gst/rtsp-server/rtsp-permissions.h:
1391 * gst/rtsp-server/rtsp-sdp.h:
1392 * gst/rtsp-server/rtsp-server-prelude.h:
1393 * gst/rtsp-server/rtsp-server.h:
1394 * gst/rtsp-server/rtsp-session-media.h:
1395 * gst/rtsp-server/rtsp-session-pool.h:
1396 * gst/rtsp-server/rtsp-session.h:
1397 * gst/rtsp-server/rtsp-stream-transport.h:
1398 * gst/rtsp-server/rtsp-stream.h:
1399 * gst/rtsp-server/rtsp-thread-pool.h:
1400 * gst/rtsp-server/rtsp-token.h:
1401 rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API
1402 We need different export decorators for the different libs.
1403 For now no actual change though, just rename before the release,
1404 and add prelude headers to define the new decorator to GST_EXPORT.
1406 2018-03-07 12:20:05 +0200 Sebastian Dröge <sebastian@centricular.com>
1408 * gst/rtsp-server/rtsp-onvif-media-factory.c:
1409 rtsp-onvif-media-factory: Document that backchannel pipelines must end with async=false sinks
1410 https://bugzilla.gnome.org/show_bug.cgi?id=794143
1412 === release 1.13.90 ===
1414 2018-03-03 22:49:34 +0000 Tim-Philipp Müller <tim@centricular.com>
1420 * gst-rtsp-server.doap:
1424 2018-03-02 16:24:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1426 * gst/rtsp-server/rtsp-media-factory.c:
1427 * gst/rtsp-server/rtsp-permissions.c:
1428 permissions: add Since tags and example for new API
1430 2018-03-02 01:36:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1432 * docs/libs/gst-rtsp-server-sections.txt:
1433 * gst/rtsp-server/rtsp-media-factory.c:
1434 * gst/rtsp-server/rtsp-media-factory.h:
1435 * gst/rtsp-server/rtsp-permissions.c:
1436 * gst/rtsp-server/rtsp-permissions.h:
1437 * tests/check/gst/permissions.c:
1438 permissions: more bindings-friendly API
1439 https://bugzilla.gnome.org/show_bug.cgi?id=793975
1441 2018-03-01 19:28:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1444 meson: enable more warnings
1446 2018-02-28 21:12:43 +0200 Sebastian Dröge <sebastian@centricular.com>
1448 * gst/rtsp-server/rtsp-client.c:
1449 rtsp-client: Place netaddress meta on packets received via TCP
1450 This allows us to later map signals from rtpbin/rtpsource back to the
1451 corresponding stream transport, and allows to do keep-alive based on
1452 RTCP packets in case of TCP media transport.
1453 https://bugzilla.gnome.org/show_bug.cgi?id=789646
1455 2018-02-27 20:34:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1457 * gst/rtsp-sink/gstrtspclientsink.c:
1458 rtspclientsink: if OPEN failed, unqueue next command
1459 As READY_TO_PAUSED can no longer return async, the RECORD
1460 command will be queued before the OPEN command fails
1461 (for example in case the server could not be connected),
1462 and record then waits for ever.
1463 https://bugzilla.gnome.org/show_bug.cgi?id=793896
1465 2018-02-26 22:59:17 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1467 * gst/rtsp-sink/gstrtspclientsink.c:
1468 rtspclientsink: fix retrieval of custom payloader caps
1469 If a bin is passed as the custom payloader, the caps of
1470 its factory will be empty, the correct way to obtain the caps
1471 is to query its sinkpad.
1473 2018-02-26 22:59:00 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1475 * gst/rtsp-sink/gstrtspclientsink.c:
1476 rtspclientsink: fix extra unref of custom payloader
1478 2018-02-26 22:57:39 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1480 * gst/rtsp-sink/gstrtspclientsink.c:
1481 rspclientsink: fix recent code indentation
1483 2018-02-26 20:27:57 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1485 * gst/rtsp-sink/gstrtspclientsink.c:
1486 rtspclientsink: add missing get_type prototype
1488 2018-02-24 03:52:15 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1490 * gst/rtsp-sink/gstrtspclientsink.c:
1491 rtspclientsink: allow setting payloader as pad property
1492 This was a FIXME item, and can be quite useful, also
1493 allowing to specify payloader properties from the command
1494 line, which is always nice.
1495 https://bugzilla.gnome.org/show_bug.cgi?id=793776
1497 2018-02-26 14:16:54 +0100 Carlos Rafael Giani <dv@pseudoterminal.org>
1499 * gst/rtsp-server/rtsp-media.c:
1500 rtsp-media: Replace g_print() log line
1501 https://bugzilla.gnome.org/show_bug.cgi?id=793838
1503 2018-02-22 20:17:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1505 * gst/rtsp-server/rtsp-media.c:
1506 * tests/check/gst/rtspclientsink.c:
1507 rtsp-media: fix RECORD getting stuck
1508 The test_record case was working because async=false had
1509 been added in https://bugzilla.gnome.org/show_bug.cgi?id=757488
1510 but that was incorrect, as it should not be needed.
1511 Removing async=false made the test fail as expected, this is
1512 fixed by not trying to preroll when preparing the media for
1513 RECORD, as start_prepare is called upon receiving ANNOUNCE,
1514 and our peer will not start sending media until it has received
1515 a response to that request, and sent and received a response
1516 to RECORD as well, thus obviously preventing preroll.
1517 https://bugzilla.gnome.org/show_bug.cgi?id=793738
1519 2018-02-23 03:26:21 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1521 * gst/rtsp-server/rtsp-auth.c:
1522 rtsp-auth: fix set_tls_authentication_mode annotation
1524 2018-02-19 11:57:29 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
1526 * gst/rtsp-server/rtsp-onvif-media.c:
1527 rtp-server: remove redefined variable
1528 res is a boolean variable which is defined in the function scope and
1529 redefined, with no reason, in the loop scope. This patch removes the
1531 https://bugzilla.gnome.org/show_bug.cgi?id=793592
1533 2018-02-05 11:49:07 +0100 Ognyan Tonchev <ognyan@axis.com>
1535 * gst/rtsp-server/rtsp-media.c:
1536 * gst/rtsp-server/rtsp-stream.c:
1537 * gst/rtsp-server/rtsp-stream.h:
1538 stream: Add functions for checking if stream is receiver or sender
1539 ...and replace all checks for RECORD in GstRTSPMedia which are really
1540 for "sender-only". This way the code becomes more generic and introducing
1541 support for onvif-backchannel later on will require no changes in
1544 2017-10-21 14:06:30 +0200 Ognyan Tonchev <ognyan@axis.com>
1546 * gst/rtsp-server/rtsp-onvif-media-factory.c:
1547 * gst/rtsp-server/rtsp-onvif-media-factory.h:
1548 onvif: Make requires_backchannel() public
1549 ...in order to let subclasses building the onvif part of the pipeline
1550 check whether backchannel shall be included or not.
1552 2018-01-22 12:46:34 +0200 Sebastian Dröge <sebastian@centricular.com>
1554 * gst/rtsp-server/rtsp-onvif-media.c:
1555 rtsp-server: Switch around sendonly/recvonly attributes
1556 They are wrong in the ONVIF streaming spec. The backchannel should be
1557 recvonly and the normal media should be sendonly: direction is always
1558 from the point of view of the SDP offerer (the server) according to
1561 2017-09-25 19:41:05 +0300 Sebastian Dröge <sebastian@centricular.com>
1563 * docs/libs/gst-rtsp-server-docs.sgml:
1564 * docs/libs/gst-rtsp-server-sections.txt:
1565 * examples/.gitignore:
1566 * examples/Makefile.am:
1567 * examples/test-onvif-backchannel.c:
1568 * gst/rtsp-server/Makefile.am:
1569 * gst/rtsp-server/rtsp-media.h:
1570 * gst/rtsp-server/rtsp-onvif-client.c:
1571 * gst/rtsp-server/rtsp-onvif-client.h:
1572 * gst/rtsp-server/rtsp-onvif-media-factory.c:
1573 * gst/rtsp-server/rtsp-onvif-media-factory.h:
1574 * gst/rtsp-server/rtsp-onvif-media.c:
1575 * gst/rtsp-server/rtsp-onvif-media.h:
1576 * gst/rtsp-server/rtsp-onvif-server.c:
1577 * gst/rtsp-server/rtsp-onvif-server.h:
1578 * gst/rtsp-server/rtsp-sdp.c:
1579 * gst/rtsp-server/rtsp-sdp.h:
1580 rtsp: Add support for ONVIF backchannel
1581 This adds a new RTSP server, client, media-factory and media subclass
1582 for handling the specifics of the backchannel. Ideally this later can be
1583 extended with other ONVIF specific features.
1585 2017-10-12 21:00:16 +0300 Sebastian Dröge <sebastian@centricular.com>
1587 * gst/rtsp-server/rtsp-media.c:
1588 rtsp-media: Add support for sending+receiving medias
1589 We need to add an appsrc/appsink in that case because otherwise the
1590 media bin will be a sink and a source for rtpbin, causing a pipeline
1592 https://bugzilla.gnome.org/show_bug.cgi?id=788950
1594 2018-02-15 19:44:28 +0000 Tim-Philipp Müller <tim@centricular.com>
1600 === release 1.13.1 ===
1602 2018-02-15 17:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
1606 * gst-rtsp-server.doap:
1610 2018-02-14 17:11:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1612 * gst/rtsp-server/rtsp-session-pool.c:
1613 session-pool: remove nullable return annotation
1614 create_watch can only return NULL from the API guards, no
1617 2018-02-13 18:59:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1619 * gst/rtsp-server/rtsp-media-factory.c:
1620 * gst/rtsp-server/rtsp-media.c:
1621 set_clock functions: Add nullable annotations
1623 2018-02-10 00:07:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1625 * gst/rtsp-server/rtsp-auth.c:
1626 * gst/rtsp-server/rtsp-client.c:
1627 * gst/rtsp-server/rtsp-media-factory.c:
1628 * gst/rtsp-server/rtsp-media.c:
1629 * gst/rtsp-server/rtsp-mount-points.c:
1630 * gst/rtsp-server/rtsp-server.c:
1631 * gst/rtsp-server/rtsp-session-media.c:
1632 * gst/rtsp-server/rtsp-session-pool.c:
1633 * gst/rtsp-server/rtsp-session.c:
1634 * gst/rtsp-server/rtsp-stream-transport.c:
1635 * gst/rtsp-server/rtsp-stream.c:
1636 * gst/rtsp-server/rtsp-thread-pool.c:
1637 All around: add annotations and API guards
1639 2018-02-12 19:12:35 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1641 * tests/test-cleanup.c:
1642 test-cleanup: bind any port
1643 The meson test suite runs tests in parallel, trying to bind
1644 a single port made the test fail.
1646 2018-02-08 19:15:10 +0000 Tim-Philipp Müller <tim@centricular.com>
1649 meson: make version numbers ints and fix int/string comparison
1650 WARNING: Trying to compare values of different types (str, int).
1651 The result of this is undefined and will become a hard error
1652 in a future Meson release.
1654 2018-02-06 18:00:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1656 * gst/rtsp-server/rtsp-context.c:
1657 gst_rtsp_context_get_current: add (skip) annotation
1658 The return value type is defined with G_DEFINE_POINTER_TYPE,
1659 and gi emits the following warning:
1660 Invalid non-constant return of bare structure or union; register as
1661 boxed type or (skip)
1663 2018-02-06 17:58:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1665 * gst/rtsp-server/rtsp-client.c:
1666 rtsp-client: add type annotations
1667 gi doesn't seem to be able to figure out the type of the
1668 signal parameters when defined with G_DEFINE_POINTER_TYPE
1670 2018-02-04 12:24:09 +0100 Tim-Philipp Müller <tim@centricular.com>
1673 autotools: use -fno-strict-aliasing where supported
1674 https://bugzilla.gnome.org/show_bug.cgi?id=769183
1676 2018-01-30 20:35:21 +0000 Tim-Philipp Müller <tim@centricular.com>
1679 meson: use -fno-strict-aliasing where supported
1680 https://bugzilla.gnome.org/show_bug.cgi?id=769183
1682 2018-01-25 12:09:03 +0000 Tim-Philipp Müller <tim@centricular.com>
1684 * gst/rtsp-server/rtsp-mount-points.c:
1685 mount-points: bail out of loop again when matching mount points
1686 Previous patch led to us iterating the entire sequence. Bail out
1687 of the loop again if we have a match but are moving away from it.
1688 https://bugzilla.gnome.org/show_bug.cgi?id=771555
1690 2018-01-25 12:06:57 +0000 Tim-Philipp Müller <tim@centricular.com>
1692 * tests/check/gst/mountpoints.c:
1693 tests: mountpoints: add more checks for mount point path matching
1694 https://bugzilla.gnome.org/show_bug.cgi?id=771555
1696 2016-09-16 20:41:19 +0000 Andrew Bott <andrew.bott@blackmoth.com>
1698 * gst/rtsp-server/rtsp-mount-points.c:
1699 mount-points: fix matching of paths where there's also an entry with a common prefix
1700 e.g. with the following mount points
1704 _match() would not match /raw/video and /raw/snapshot correctly.
1705 https://bugzilla.gnome.org/show_bug.cgi?id=771555
1707 2018-01-18 23:53:20 +0000 Tim-Philipp Müller <tim@centricular.com>
1709 * docs/libs/gst-rtsp-server-sections.txt:
1710 * gst/rtsp-server/rtsp-permissions.c:
1711 * gst/rtsp-server/rtsp-permissions.h:
1712 * tests/check/gst/permissions.c:
1713 permissions: add some new API to make this usable from bindings
1714 https://bugzilla.gnome.org/show_bug.cgi?id=787073
1716 2018-01-18 11:32:32 +0000 Tim-Philipp Müller <tim@centricular.com>
1718 * gst/rtsp-server/rtsp-token.c:
1719 rtsp-token: annotate constructors for bindings
1720 This maps _new_empty() to _new(), which also makes RTSPToken()
1721 work properly now. Since this API wasn't usable from bindings
1722 before, this should hopefully be fine.
1723 https://bugzilla.gnome.org/show_bug.cgi?id=787073
1725 2018-01-18 11:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
1727 * docs/libs/gst-rtsp-server-sections.txt:
1728 * gst/rtsp-server/rtsp-token.c:
1729 * gst/rtsp-server/rtsp-token.h:
1730 * tests/check/gst/token.c:
1731 rtsp-token: add some API to set fields from bindings
1732 The existing functions are all vararg-based and as such
1733 not usable from bindings.
1734 https://bugzilla.gnome.org/show_bug.cgi?id=787073
1736 2018-01-13 15:02:28 +0000 Tim-Philipp Müller <tim@centricular.com>
1738 * tests/check/gst/rtspclientsink.c:
1739 * tests/check/gst/rtspserver.c:
1740 * tests/check/gst/sessionpool.c:
1741 * tests/check/gst/stream.c:
1742 tests: fix indentation
1745 2018-01-13 14:58:55 +0000 Tim-Philipp Müller <tim@centricular.com>
1747 * tests/check/gst/rtspserver.c:
1748 tests: rtspserver: fix another ref leak
1749 Even if this didn't show up in valgrind.
1751 2018-01-13 14:58:00 +0000 Tim-Philipp Müller <tim@centricular.com>
1753 * tests/check/gst/rtspclientsink.c:
1754 tests: rtspclientsink: fix leak
1756 2018-01-02 14:19:31 +0100 Branko Subasic <branko@axis.com>
1758 * tests/check/gst/rtspserver.c:
1759 test: rtspserver: plug memory leak in test_no_session_timeout
1760 In test_no_session_timeout, unref the rtsp session object when the
1762 https://bugzilla.gnome.org/show_bug.cgi?id=792127
1764 2017-12-20 14:17:02 +0100 Edward Hervey <edward@centricular.com>
1766 * gst/rtsp-sink/gstrtspclientsink.c:
1767 rtpsclientsink: Initialize and clear newly added mutex and cond
1768 While it *did* work, glib would automatically create new mutex and cond
1769 ... which never got freed
1771 2017-12-19 11:34:37 +0200 Sebastian Dröge <sebastian@centricular.com>
1773 * gst/rtsp-server/rtsp-stream.c:
1774 rtsp-stream: Set multicast TTL on the multicast sockets
1775 And not if we do unicast UDP.
1776 https://bugzilla.gnome.org/show_bug.cgi?id=791743
1778 2017-12-19 11:14:48 +0200 Sebastian Dröge <sebastian@centricular.com>
1780 * gst/rtsp-server/rtsp-stream.c:
1781 rtsp-stream: Decide based on the sockets, not the addresses if we already allocated a socket
1782 In the multicast case (as in test-multicast, not test-multicast2), the
1783 address could be allocated/reserved (and thus set) already without
1784 allocating the actual socket. We need to allocate the socket here still
1785 instead of just claiming that it was already allocated.
1786 See https://bugzilla.gnome.org/show_bug.cgi?id=791743#c2
1788 2017-12-16 21:46:53 +0100 Patricia Muscalu <patricia@dovakhiin.com>
1790 * gst/rtsp-sink/gstrtspclientsink.c:
1791 * gst/rtsp-sink/gstrtspclientsink.h:
1792 rtspclientsink: Use the new rtsp-stream API
1793 https://bugzilla.gnome.org/show_bug.cgi?id=790412
1795 2017-12-16 21:01:43 +0100 Patricia Muscalu <patricia@dovakhiin.com>
1797 * gst/rtsp-sink/gstrtspclientsink.c:
1798 * gst/rtsp-sink/gstrtspclientsink.h:
1799 rtspclientsink: Wait until OPEN has been scheduled
1800 Make sure that the sink thread has started opening connection
1801 to the server before continuing.
1802 https://bugzilla.gnome.org/show_bug.cgi?id=790412
1804 2017-12-14 14:53:35 +1100 Matthew Waters <matthew@centricular.com>
1807 Automatic update of common submodule
1808 From e8c7a71 to 3fa2c9e
1810 2017-12-07 16:08:29 +0100 Edward Hervey <edward@centricular.com>
1812 * gst/rtsp-server/rtsp-media.c:
1813 * gst/rtsp-server/rtsp-session-media.c:
1814 * gst/rtsp-server/rtsp-stream.c:
1815 rtsp-server: Minor doc fixes
1818 2017-12-06 20:47:22 +0000 Tim-Philipp Müller <tim@centricular.com>
1821 * tests/Makefile.am:
1822 tests: disable all tests when --disable-tests is used
1823 Move conditional subdir include into top level.
1824 Based on patch by: Joel Holdsworth
1825 https://bugzilla.gnome.org/show_bug.cgi?id=757703
1827 2017-12-06 20:42:39 +0000 Tim-Philipp Müller <tim@centricular.com>
1830 * meson_options.txt:
1831 * tests/meson.build:
1832 meson: build more tests and add options to disable tests and examples
1834 2017-11-26 13:26:39 -0300 Thibault Saunier <tsaunier@gnome.org>
1836 * gst/rtsp-server/rtsp-session.c:
1837 Fix build when -Werror=deprecated-declarations is on
1838 As gst_rtsp_session_next_timeout is deprecated.
1840 ../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]
1841 res = (gst_rtsp_session_next_timeout (session, now) == 0);
1843 ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-session.c:685:1: note: declared here
1844 gst_rtsp_session_next_timeout (GstRTSPSession * session, GTimeVal * now)
1845 ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~
1848 2017-11-27 20:18:24 +1100 Matthew Waters <matthew@centricular.com>
1851 Automatic update of common submodule
1852 From 3f4aa96 to e8c7a71
1854 2017-11-25 20:34:16 +0100 Patricia Muscalu <patricia@dovakhiin.com>
1856 * tests/check/gst/media.c:
1857 check/media: Add seekability test case: not all streams are active
1858 Media contains two streams but only one is complete and prepared
1860 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1862 2017-11-25 20:32:02 +0100 Patricia Muscalu <patricia@dovakhiin.com>
1864 * gst/rtsp-server/rtsp-stream.c:
1865 rtsp-stream: Do not reset 'blocking' if stream is already blocked
1866 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1868 2017-11-25 20:45:44 +0100 Patricia Muscalu <patricia@dovakhiin.com>
1870 * gst/rtsp-server/rtsp-media.c:
1871 rtsp-media: Fix missing lock in gst_rtsp_media_seekable()
1872 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1874 2017-11-26 16:29:49 +0000 Tim-Philipp Müller <tim@centricular.com>
1877 meson: remove vs_module_defs_dir variable which is no longer needed
1879 2017-11-26 14:46:05 +0000 Tim-Philipp Müller <tim@centricular.com>
1881 * gst/rtsp-server/rtsp-session.h:
1884 2017-11-26 12:53:42 +0000 Tim-Philipp Müller <tim@centricular.com>
1887 * gst/rtsp-server/meson.build:
1889 * win32/common/libgstrtspserver.def:
1890 win32: remove .def file with exports
1891 They're no longer needed, symbol exporting is now explicit
1892 via GST_EXPORT in all cases (autotools, meson, incl. MSVC).
1894 2017-11-26 12:28:40 +0000 Tim-Philipp Müller <tim@centricular.com>
1897 autotools: stop controlling symbol visibility with -export-symbols-regex
1898 Instead, use -fvisibility=hidden and explicit exports via GST_EXPORT.
1899 This should result in consistent behaviour for the autotools and
1902 2017-11-26 12:47:08 +0000 Tim-Philipp Müller <tim@centricular.com>
1904 * gst/rtsp-server/rtsp-media.h:
1905 * gst/rtsp-server/rtsp-server.h:
1906 * gst/rtsp-server/rtsp-session.c:
1907 * gst/rtsp-server/rtsp-session.h:
1908 rtsp-server: add missing GST_EXPORT and export deprecated funcs
1910 2017-11-25 07:53:30 +0100 Edward Hervey <edward@centricular.com>
1912 * tests/check/gst/media.c:
1913 check: Add seekability testing on medias
1914 Make sure that once GstRTSPMedia are prepared they returned
1915 the expected seekability results
1916 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1918 2017-11-24 17:34:31 +0100 Edward Hervey <edward@centricular.com>
1920 * docs/libs/gst-rtsp-server-sections.txt:
1921 * gst/rtsp-server/rtsp-media.c:
1922 * gst/rtsp-server/rtsp-stream.c:
1923 * gst/rtsp-server/rtsp-stream.h:
1924 * win32/common/libgstrtspserver.def:
1925 rtsp-media: Enable seeking query before pipeline is complete
1926 SDP are now provided *before* the pipeline is fully complete. In order
1927 to know whether a media is seekable or not therefore requires asking
1928 the invididual streams.
1929 API: gst_rtsp_stream_seekable
1930 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1932 2017-11-23 20:34:03 +0100 Patricia Muscalu <patricia@axis.com>
1934 * gst/rtsp-server/rtsp-media.c:
1935 rtsp-media: Fix handling in default_unsuspend()
1936 Handle the case when streams are not blocked and media
1937 is suspended from PAUSED.
1938 Change-Id: I2f3d222ea7b9b20a0732ea5dc81a32d17ab75040
1939 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1941 2017-11-23 18:51:21 +0100 Patricia Muscalu <patricia@axis.com>
1943 * tests/check/gst/media.c:
1944 check/media: Fix thread pool leak.
1945 Change-Id: I0f92b1caca0ee518ae64a7dacfbd28a214c3eea1
1946 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1948 2017-11-23 18:39:44 +0100 Patricia Muscalu <patricia@axis.com>
1950 * gst/rtsp-server/rtsp-media.c:
1951 rtsp-media: Removed fakesink elements
1952 There is not need of adding fakesink elements to the media
1953 pipeline in the dynamic-payloader case.
1954 The media pipeline itself is dynamically updated with
1955 the receiver and sender parts that are based on the client
1956 transport information known after SETUP has been received.
1957 Change-Id: I4e88c9b500c04030669822f0d03b1842913f6cb9
1958 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1960 2017-11-23 09:10:54 +0100 Patricia Muscalu <patricia@axis.com>
1962 * gst/rtsp-server/rtsp-media.c:
1963 rtsp-media: Corrected ASYNC_DONE handling
1964 Media is complete when all the transport based parts are
1965 added to the media pipeline. At this point ASYNC_DONE is
1966 posted by the media pipeline and media is ready to enter
1968 Change-Id: I50fb8dfed88ebaf057d9a35fca2d7f0a70e9d1fa
1969 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1971 2017-11-22 12:24:38 +0100 Edward Hervey <bilboed@bilboed.com>
1973 * tests/check/gst/media.c:
1974 check/media: Check that prepared media can provide a SDP
1975 Whenever a RTSPMedia is prepared, it should be able to provide a SDP
1977 2017-11-21 09:53:19 +0100 Edward Hervey <edward@centricular.com>
1979 * gst/rtsp-server/rtsp-client.c:
1980 rtsp-client: Don't leak addr
1983 2017-11-21 09:53:08 +0100 Edward Hervey <bilboed@bilboed.com>
1985 * gst/rtsp-server/rtsp-client.c:
1986 * gst/rtsp-server/rtsp-session-media.c:
1987 * gst/rtsp-server/rtsp-stream.c:
1990 2017-11-20 18:30:19 +0100 Edward Hervey <bilboed@bilboed.com>
1992 * gst/rtsp-server/rtsp-media.c:
1993 rtsp-media: Don't unblock with remaining dynamic payloaders
1994 If we still have some dynamic paylaoders which haven't posted
1995 no-more-pads yet, don't go to PREPARED if one of the streams
1997 The risk was that we would end up not exposing/using all specified
1999 The downside is that if you have _multiple_ _live_ _dynamic_ payloaders
2000 then it will take a bit more time to start. But only if those 3
2001 conditions are present.
2002 https://bugzilla.gnome.org/show_bug.cgi?id=769521
2004 2017-11-20 16:49:29 +0100 Edward Hervey <edward@centricular.com>
2006 * gst/rtsp-server/rtsp-media.c:
2009 2017-11-20 16:48:55 +0100 Edward Hervey <edward@centricular.com>
2011 * gst/rtsp-server/rtsp-media.c:
2012 rtsp-media: Don't set float on a gint64 variable
2013 Just use 0. Fixes 'undefined' behaviour from clang
2015 2017-11-20 18:29:02 +0100 Edward Hervey <edward@centricular.com>
2017 * gst/rtsp-server/rtsp-media.c:
2018 rtsp-media: Fix previous commit
2019 We only want to count dynamic payloaders
2021 2017-11-20 09:32:07 +0100 Edward Hervey <edward@centricular.com>
2023 * gst/rtsp-server/rtsp-media.c:
2024 * tests/check/gst/media.c:
2025 rtsp-media: Handle multiple dynamic elements
2026 If we have more than one dynamic payloader in the pipeline, we need
2027 to wait until the *last* one emits 'no-more-pads' before switching
2029 Failure to do so would result in a race where some of the streams
2030 wouldn't properly be prepared
2031 https://bugzilla.gnome.org/show_bug.cgi?id=769521
2033 2017-11-16 12:18:20 +0200 Sebastian Dröge <sebastian@centricular.com>
2035 * win32/common/libgstrtspserver.def:
2036 win32: Fix exported symbols list
2038 2017-11-15 19:52:29 +0200 Sebastian Dröge <sebastian@centricular.com>
2040 * gst/rtsp-server/rtsp-stream.c:
2041 rtsp-stream: Only update the RTP udpsink if it actually exists
2042 For send-only streams it does not exist, but the RTCP udpsink might.
2044 2017-11-15 18:15:53 +0200 Sebastian Dröge <sebastian@centricular.com>
2046 * win32/common/libgstrtspserver.def:
2047 win32: Update exports
2049 2017-10-23 09:49:09 +0200 Patricia Muscalu <patricia@axis.com>
2051 * gst/rtsp-server/rtsp-media.c:
2052 * gst/rtsp-server/rtsp-stream.c:
2053 * gst/rtsp-server/rtsp-stream.h:
2054 rtsp-media: seek on media pipelines that are complete
2055 Make sure that a seek is performed on pipelines that
2056 contain at least one sink element.
2057 Change-Id: Icf398e10add3191d104b1289de612412da326819
2058 https://bugzilla.gnome.org/show_bug.cgi?id=788340
2060 2017-10-17 10:44:33 +0200 Patricia Muscalu <patricia@axis.com>
2062 * gst/rtsp-server/rtsp-client.c:
2063 * gst/rtsp-server/rtsp-media.c:
2064 * gst/rtsp-server/rtsp-media.h:
2065 * gst/rtsp-server/rtsp-stream.c:
2066 * gst/rtsp-server/rtsp-stream.h:
2067 * tests/check/gst/client.c:
2068 * tests/check/gst/media.c:
2069 * tests/check/gst/rtspserver.c:
2070 * tests/check/gst/stream.c:
2071 Dynamically reconfigure pipeline in PLAY based on transports
2072 The initial pipeline does not contain specific transport
2073 elements. The receiver and the sender parts are added
2075 If the media is shared, the streams are dynamically
2076 reconfigured after each PLAY.
2077 https://bugzilla.gnome.org/show_bug.cgi?id=788340
2079 2017-10-16 12:40:57 +0200 Patricia Muscalu <patricia@axis.com>
2081 * gst/rtsp-server/rtsp-stream.c:
2082 rtsp-stream: obtain stream position from pad
2083 If no sinks have been added yet, obtain the current and
2084 the stop position of the stream from the send_src pad.
2085 Change-Id: Iacd4ab4bdc69f6b49370d06012880ce48a7d595a
2086 https://bugzilla.gnome.org/show_bug.cgi?id=788340
2088 2017-10-16 11:35:10 +0200 Patricia Muscalu <patricia@axis.com>
2090 * gst/rtsp-server/rtsp-session-media.c:
2091 * gst/rtsp-server/rtsp-session-media.h:
2092 rtsp-session-media: add function to get a list of transports
2093 Change-Id: I817e10624da0f3200f24d1b232cff481099278e3
2094 https://bugzilla.gnome.org/show_bug.cgi?id=788340
2096 2017-10-16 11:15:55 +0200 Patricia Muscalu <patricia@axis.com>
2098 * gst/rtsp-server/rtsp-stream.c:
2099 * gst/rtsp-server/rtsp-stream.h:
2100 rtsp-stream: add functions to get rtp and rtcp multicast sockets
2101 Change-Id: Iddfe6e0bd250cb0159096d5eba9e4202d22b56db
2102 https://bugzilla.gnome.org/show_bug.cgi?id=788340
2104 2017-10-20 12:21:48 +0200 Patricia Muscalu <patricia@axis.com>
2106 * gst/rtsp-server/rtsp-stream.c:
2107 stream: set async=sync=false only for RTCP appsink
2108 Change-Id: I929a218a9adf4759f61322b6f2063aacc5595f90
2109 https://bugzilla.gnome.org/show_bug.cgi?id=788340
2111 2017-10-16 10:10:17 +0200 Patricia Muscalu <patricia@axis.com>
2113 * gst/rtsp-server/rtsp-media.c:
2114 rtsp-media: return minimum value in query position case
2115 The minimum position should be returned as we are interested
2116 in the whole interval.
2117 Change-Id: I30e297fc040c995ae40c25dee8ff56321612fe2b
2118 https://bugzilla.gnome.org/show_bug.cgi?id=788340
2120 2017-08-09 11:52:38 +0200 Jonathan Karlsson <jonakn@axis.com>
2122 * gst/rtsp-server/rtsp-session.c:
2123 * tests/check/gst/rtspserver.c:
2124 rtsp-session: Handle the case when timeout=0
2125 According to the documentation, a timeout of value 0 means
2126 that the session never timeouts. This adds handling of that.
2127 If timeout=0 we just return with a -1 from
2128 gst_rtsp_session_next_timeout_usec ().
2129 https://bugzilla.gnome.org/show_bug.cgi?id=785058
2131 2017-07-17 17:15:22 +0300 Sebastian Dröge <sebastian@centricular.com>
2133 * gst/rtsp-sink/gstrtspclientsink.c:
2134 rtspclientsink: Add "accept-certificate" signal for manually checking a TLS certificate for validity
2135 https://bugzilla.gnome.org/show_bug.cgi?id=785024
2137 2017-10-26 14:43:19 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2139 * docs/libs/gst-rtsp-server-sections.txt:
2140 * gst/rtsp-server/rtsp-media-factory.c:
2141 docs: add media factory transport mode accessors
2142 and fix the documentation for the return value of the getter
2144 2017-10-09 12:43:01 +0200 Branko Subasic <branko@axis.com>
2146 * gst/rtsp-server/rtsp-client.c:
2147 rtsp-client: unref 'pipelined_requests' in finalize
2148 The hash table priv->pipelined_requests is not unref:ed in the
2149 finalize funktion. Make sure it is.
2150 https://bugzilla.gnome.org/show_bug.cgi?id=788704
2152 2017-10-09 14:44:40 +0200 Thibault Saunier <thibault.saunier@osg.samsung.com>
2154 * gst/rtsp-server/rtsp-media.c:
2155 rtsp-media: Initialize scalar variable
2158 2017-10-06 10:27:34 +0200 Edward Hervey <edward@centricular.com>
2160 * win32/common/libgstrtspserver.def:
2161 win32: Update export file
2163 2017-04-22 09:26:07 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
2165 * gst/rtsp-server/rtsp-client.c:
2166 * gst/rtsp-server/rtsp-media.c:
2167 * gst/rtsp-server/rtsp-media.h:
2168 Start support for RTSP 2.0
2169 This adds basic support for new 2.0 features, though the protocol is
2170 subposdely backward incompatible, most semantics are the sames.
2173 * version negotiation
2174 * pipelined requests support
2175 * Media-Properties support
2176 * Accept-Ranges support
2178 * gst_rtsp_media_seekable
2179 The RTSP methods that have been removed when using 2.0 now return
2181 https://bugzilla.gnome.org/show_bug.cgi?id=781446
2183 2017-06-02 15:37:54 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
2185 * gst/rtsp-server/rtsp-stream.c:
2186 stream: Use stream duration as stream-stop if segment was not configured with a stop
2187 Allowing client to know stream duration when no seeking happened.
2188 https://bugzilla.gnome.org/show_bug.cgi?id=783435
2190 2017-09-25 19:40:17 +0300 Sebastian Dröge <sebastian@centricular.com>
2192 * gst/rtsp-server/rtsp-media-factory.c:
2193 rtsp-media-factory: Don't cache any media if NULL was returned as key
2194 The docs already mentioned this, but we actually stored it in the hash
2195 table with key==NULL and leaked its reference forever.
2197 2017-09-18 19:31:31 +0200 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
2199 * gst/rtsp-sink/gstrtspclientsink.c:
2200 * gst/rtsp-sink/gstrtspclientsink.h:
2201 rtspclientsink: Use a mutex for protecting against concurrent send/receives
2202 This is a simple port of:
2203 * a722f6e8329032c6eda4865d6a07f4ba5981d7ea
2204 * c438545dc9e2f14f657bc0ef261fff726449867b
2205 * cd17c71dcea5c9310d21f1347c7520983e5869ac
2206 in gst-plugins-good.
2208 2017-08-31 13:24:15 +0530 Satya Prakash Gupta <sp.gupta@samsung.com>
2210 * gst/rtsp-server/rtsp-sdp.c:
2211 sdp: fix Memory leak in error case
2212 https://bugzilla.gnome.org/show_bug.cgi?id=787059
2214 2017-08-18 17:37:01 +0100 Tim-Philipp Müller <tim@centricular.com>
2216 * pkgconfig/meson.build:
2217 meson: don't install -uninstalled.pc file
2218 https://bugzilla.gnome.org/show_bug.cgi?id=786457
2220 2017-08-17 12:26:17 +0100 Tim-Philipp Müller <tim@centricular.com>
2223 Automatic update of common submodule
2224 From 48a5d85 to 3f4aa96
2226 2017-08-14 21:04:23 +0300 Sebastian Dröge <sebastian@centricular.com>
2228 * gst/rtsp-server/rtsp-client.c:
2229 rtsp-client: Fix typo in debug message
2231 2017-08-11 14:14:32 +0100 Tim-Philipp Müller <tim@centricular.com>
2234 meson: hide symbols by default unless explicitly exported
2236 2017-08-10 14:20:12 +0100 Tim-Philipp Müller <tim@centricular.com>
2238 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
2239 pkgconfig: remove -I@srcdir@/.. which duplicates abs_top_srcdir
2240 Fixes meson warning about undefined @srcdir@.
2242 2017-07-21 13:36:00 +0100 Tim-Philipp Müller <tim@centricular.com>
2244 * tests/meson.build:
2245 meson: skip tests on windows for now
2246 As we do in the other modules. As libgstcheck is currently not
2247 built on windows. Fixes "Fallback variable 'gst_check_dep' in
2248 the subproject 'gstreamer' does not exist"" Meson error.
2250 2017-06-22 07:25:07 -0700 Julien Isorce <julien.isorce@gmail.com>
2252 * gst/rtsp-server/rtsp-stream.c:
2253 rtsp-stream: fix connection delay due to wrong assumption on last-sample
2254 Commit 852cc09f542af5cadd79ffd7fe79d6475cf57e14 assumed that
2255 multiudpsink's last-sample always comes from the payloader. Which
2256 is wrong if auxiliary streams are multiplexed in the same stream.
2257 So check the buffer's ssrc against the caps'ssrc before to use its
2258 seqnum. If not the same ssrc just use the payloader as done prior
2259 the commit above or when there is no last-sample yet.
2260 https://bugzilla.gnome.org/show_bug.cgi?id=784094
2262 2017-06-23 16:19:04 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
2265 meson: Allow using glib as a subproject
2267 2017-06-26 09:55:49 +0100 Tim-Philipp Müller <tim@centricular.com>
2270 meson: fix with-package-name option
2271 https://bugzilla.gnome.org/show_bug.cgi?id=784082
2273 2017-06-09 20:16:28 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
2276 Distribute meson_options.txt
2278 2017-06-09 20:11:47 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
2281 And config.h.meson is no longer dist either
2283 2017-06-09 21:27:09 +0100 Tim-Philipp Müller <tim@centricular.com>
2287 meson: config.h.meson is no longer needed
2289 2017-06-07 13:04:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
2291 * tests/check/meson.build:
2292 * tests/meson.build:
2293 meson: Fix building tests and activate them again
2295 2017-06-07 12:55:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
2297 * tests/check/meson.build:
2298 meson: Do not use path separator in test names
2299 Avoiding warnings like:
2300 WARNING: Target "elements/audioamplify" has a path separator in its name.
2302 2017-05-20 15:07:31 +0100 Tim-Philipp Müller <tim@centricular.com>
2305 * meson_options.txt:
2306 meson: add options to set package name and origin
2307 https://bugzilla.gnome.org/show_bug.cgi?id=782172
2309 2017-05-18 10:35:18 +0100 Tim-Philipp Müller <tim@centricular.com>
2311 * gst/rtsp-server/rtsp-address-pool.h:
2312 * gst/rtsp-server/rtsp-auth.h:
2313 * gst/rtsp-server/rtsp-client.h:
2314 * gst/rtsp-server/rtsp-context.h:
2315 * gst/rtsp-server/rtsp-media-factory-uri.h:
2316 * gst/rtsp-server/rtsp-media-factory.h:
2317 * gst/rtsp-server/rtsp-media.h:
2318 * gst/rtsp-server/rtsp-mount-points.h:
2319 * gst/rtsp-server/rtsp-params.h:
2320 * gst/rtsp-server/rtsp-permissions.h:
2321 * gst/rtsp-server/rtsp-sdp.h:
2322 * gst/rtsp-server/rtsp-server.h:
2323 * gst/rtsp-server/rtsp-session-media.h:
2324 * gst/rtsp-server/rtsp-session-pool.h:
2325 * gst/rtsp-server/rtsp-session.h:
2326 * gst/rtsp-server/rtsp-stream-transport.h:
2327 * gst/rtsp-server/rtsp-stream.h:
2328 * gst/rtsp-server/rtsp-thread-pool.h:
2329 * gst/rtsp-server/rtsp-token.h:
2330 Mark symbols explicitly for export with GST_EXPORT
2332 2017-05-16 14:44:43 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
2335 * gst/rtsp-sink/Makefile.am:
2336 Remove plugin specific static build option
2337 Static and dynamic plugins now have the same interface. The standard
2338 --enable-static/--enable-shared toggle are sufficient.
2340 2017-05-04 18:59:14 +0300 Sebastian Dröge <sebastian@centricular.com>
2346 === release 1.12.0 ===
2348 2017-05-04 15:40:46 +0300 Sebastian Dröge <sebastian@centricular.com>
2354 * gst-rtsp-server.doap:
2358 === release 1.11.91 ===
2360 2017-04-27 17:42:02 +0300 Sebastian Dröge <sebastian@centricular.com>
2366 * gst-rtsp-server.doap:
2370 2017-04-24 20:30:37 +0100 Tim-Philipp Müller <tim@centricular.com>
2373 Automatic update of common submodule
2374 From 60aeef6 to 48a5d85
2376 2017-04-13 14:20:10 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
2378 * gst/rtsp-server/rtsp-media-factory.c:
2379 * gst/rtsp-server/rtsp-media.c:
2380 * gst/rtsp-server/rtsp-session.c:
2381 * gst/rtsp-server/rtsp-stream.c:
2382 gi: Fix some annotations and docstrings
2384 2017-04-13 13:52:26 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
2386 * gst/rtsp-server/meson.build:
2388 * meson_options.txt:
2391 2017-04-10 23:51:12 +0100 Tim-Philipp Müller <tim@centricular.com>
2395 Automatic update of common submodule
2396 From 39ac2f5 to 60aeef6
2398 === release 1.11.90 ===
2400 2017-04-07 16:35:03 +0300 Sebastian Dröge <sebastian@centricular.com>
2406 * gst-rtsp-server.doap:
2410 2017-03-27 18:19:33 +0100 Tim-Philipp Müller <tim@centricular.com>
2412 * examples/test-launch.c:
2413 examples: make test-launch pipeline shared by default as well
2415 2017-02-27 19:10:44 +0200 Sebastian Dröge <sebastian@centricular.com>
2417 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
2418 gstreamer-rtsp-server: Add both srcdir and builddir to the include path
2419 Just the build dir is not going to work for srcdir!=builddir.
2421 2017-02-24 15:59:54 +0200 Sebastian Dröge <sebastian@centricular.com>
2424 meson: Update version
2426 2017-02-24 15:37:49 +0200 Sebastian Dröge <sebastian@centricular.com>
2431 === release 1.11.2 ===
2433 2017-02-24 15:10:07 +0200 Sebastian Dröge <sebastian@centricular.com>
2439 * gst-rtsp-server.doap:
2442 2017-02-14 20:40:26 +0000 Tim-Philipp Müller <tim@centricular.com>
2445 meson: dist meson build files
2446 Ship meson build files in tarballs, so people who use tarballs
2447 in their builds can start playing with meson already.
2449 2017-02-07 23:39:37 +1100 Jan Schmidt <jan@centricular.com>
2451 * examples/test-record.c:
2452 examples/test-record: Add extra line to initial printout
2453 Add an example line of how to deliver a stream to the
2456 2017-01-19 14:57:19 +0200 Sebastian Dröge <sebastian@centricular.com>
2458 * gst/rtsp-server/rtsp-client.c:
2459 rtsp-client: Also handle the (S|G)ET_PARAMETER case of size==0 || !data as keep-alive
2460 If there is no Content-Length header, no body would be allocated and the
2461 '\0' would also not be appended to the body.
2463 2017-01-19 14:24:07 +0200 Sebastian Dröge <sebastian@centricular.com>
2465 * gst/rtsp-server/rtsp-client.c:
2466 rtsp-client: Fix handling of keep-alive GET_PARAMETER/SET_PARAMETER
2467 While they logically have 0 bytes length, GstRTSPConnection is appending
2468 a '\0' to everything making the size be 1 instead.
2470 2017-01-13 12:39:36 +0000 Tim-Philipp Müller <tim@centricular.com>
2475 2017-01-12 19:04:23 +0200 Sebastian Dröge <sebastian@centricular.com>
2477 * gst/rtsp-server/rtsp-session.c:
2478 rtsp-session: Only remove deprecated API if requested to do so, not just when disabling
2479 gst_rtsp_session_is_expired() and gst_rtsp_session_next_timeout() were
2482 2017-01-12 16:32:59 +0200 Sebastian Dröge <sebastian@centricular.com>
2487 === release 1.11.1 ===
2489 2017-01-12 16:14:46 +0200 Sebastian Dröge <sebastian@centricular.com>
2495 * gst-rtsp-server.doap:
2496 * win32/common/libgstrtspserver.def:
2499 2017-01-10 08:34:50 +0100 Patricia Muscalu <patricia@axis.com>
2501 * gst/rtsp-server/rtsp-stream.c:
2502 rtsp-stream: corrected if-statement in _get_server_port()
2503 This bug was accidentally introduced while fixing a segfault
2504 in _get_server_port() function.
2505 https://bugzilla.gnome.org/show_bug.cgi?id=776345
2507 2017-01-09 14:12:05 +0100 Patricia Muscalu <patricia@axis.com>
2509 * gst/rtsp-server/rtsp-stream.c:
2510 * tests/check/gst/stream.c:
2511 rtsp-stream: fixed segmenation fault in _get_server_port()
2512 Calling function gst_rtsp_stream_get_server_port() results in
2513 segmenation fault in the RTP/RTSP/TCP case.
2514 Port that the server will use to receive RTCP makes only
2515 sense in the UDP case, however the function should handle
2516 the TCP case in a nicer way.
2517 https://bugzilla.gnome.org/show_bug.cgi?id=776345
2519 2017-01-09 12:22:40 +0300 Aleksandr Slobodeniuk <alenuke@yandex.ru>
2521 * gst/rtsp-server/rtsp-media-factory.c:
2522 dosc: Fix a little typo
2523 https://bugzilla.gnome.org/show_bug.cgi?id=777037
2525 2017-01-04 16:20:54 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
2527 * pkgconfig/Makefile.am:
2528 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
2529 * pkgconfig/meson.build:
2530 meson: generate pkg-config -uninstalled pc files
2531 Generating those files is useful for users building the GStreamer stack
2532 using meson and having to link it to another project which is still
2533 using the autotools.
2534 https://bugzilla.gnome.org/show_bug.cgi?id=776810
2536 2017-01-04 16:11:08 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
2538 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
2539 pkgconfig: fix -uninstalled pc file
2540 pcfiledir was never defined so the paths were wrong.
2541 https://bugzilla.gnome.org/show_bug.cgi?id=776867
2543 2016-12-21 13:41:50 +0100 Patricia Muscalu <patricia@axis.com>
2545 * gst/rtsp-server/rtsp-stream.c:
2546 * tests/check/gst/rtspserver.c:
2547 rtsp-stream: Fixed TCP transport case
2548 Make sure that the appsink element is actually added to
2549 the bin before trying to link it with the elements in it.
2550 https://bugzilla.gnome.org/show_bug.cgi?id=776343
2552 2016-12-16 17:26:04 +0000 Tim-Philipp Müller <tim@centricular.com>
2558 Remove generated .spec file
2559 Likely extremely bitrotten, and we should not ship this anyway.
2561 2016-12-03 08:21:02 +0100 Edward Hervey <bilboed@bilboed.com>
2564 Automatic update of common submodule
2565 From f980fd9 to 39ac2f5
2567 2016-12-02 15:40:09 +0100 Edward Hervey <edward@centricular.com>
2569 * gst/rtsp-server/rtsp-media.c:
2570 media: Fix pt map caps
2571 Since decryption is handled within rtpbin, all outcoming stream
2572 caps will be application/x-rtp (i.e. regular rtp)
2573 Fixes RECORD with SRTP streams
2575 2016-12-02 15:38:04 +0100 Edward Hervey <edward@centricular.com>
2577 * gst/rtsp-server/rtsp-media-factory.c:
2578 media-factory: Create media objects with the proper transport mode
2579 The function called immediately afterwards (collect_streams()) will
2580 need it to work properly
2582 2016-12-02 14:36:50 +0200 Sebastian Dröge <sebastian@centricular.com>
2584 * gst/rtsp-server/rtsp-auth.c:
2585 rtsp-auth: Don't remove digest-auth nonces that already/still have a client connected
2587 2016-12-01 18:04:34 +0200 Sebastian Dröge <sebastian@centricular.com>
2589 * gst/rtsp-server/rtsp-media-factory.c:
2590 rtsp-media-factory: Don't create a pipeline for the media pipeline string
2591 We're going to put a pipeline into a pipeline otherwise, which is not
2594 2016-10-25 15:41:28 +0300 Kseniia Vasilchuk <vasilchukkseniia@gmail.com>
2596 * gst/rtsp-server/rtsp-media.c:
2597 media: Fix race condition around finish_unprepare() if called multiple time
2598 https://bugzilla.gnome.org/show_bug.cgi?id=755329
2600 2016-11-30 14:06:36 +1100 Jan Schmidt <jan@centricular.com>
2602 * gst/rtsp-sink/gstrtspclientsink.c:
2603 rtspclientsink: Don't leave stale pointer after unref
2604 Fix a warning on shutdown - don't keep a pointer to an
2605 alread-unreffed object.
2607 2016-11-26 11:24:50 +0000 Tim-Philipp Müller <tim@centricular.com>
2610 common: use https protocol for common submodule
2611 https://bugzilla.gnome.org/show_bug.cgi?id=775110
2613 2016-11-21 23:29:56 +1100 Matthew Waters <matthew@centricular.com>
2615 * gst/rtsp-server/rtsp-stream.c:
2616 stream: block the output of rtpbin instead of the source pipeline
2617 85c52e194bcb81928b96614be0ae47d59eccb1ce introduced a more correct
2618 detection of the srtp rollover counter to add to the SDP.
2619 Unfortunately, it was incomplete for live pipelines where the logic
2620 blocks the source bin before creating the SDP and thus would never have
2621 the necessary informaiton to create a correct SDP with srtp encryption.
2622 Move the pad blocks to rtpbin's output pads instead so that the
2623 necessary information can be created before we need the information for
2625 https://bugzilla.gnome.org/show_bug.cgi?id=770239
2627 2016-11-21 16:02:39 +0100 Dag Gullberg <dagg@axis.com>
2629 * gst/rtsp-server/rtsp-client.c:
2630 rtsp-client: add IDLE timeout, before session exists
2631 The RTSP server will not timeout an idle RTSP connection
2632 (note this is different from doing timeout on a RTSP
2634 At least for Apache this is a problem when running RTSP over
2635 HTTPS since it uses one of the threads (there is a rather
2636 limited number) that are available for handling requests.
2637 https://bugzilla.gnome.org/show_bug.cgi?id=771830
2639 2016-11-23 09:45:08 +0000 Tim-Philipp Müller <tim@centricular.com>
2644 2016-11-21 13:05:50 +0100 Göran Jönsson <goranjn@axis.com>
2646 * gst/rtsp-server/rtsp-stream.c:
2647 rtsp-stream: Set close-socket FALSE on UDP src:es
2648 With this RTSP server can use the sockets independent on the udpsrc
2650 When the udp src is finalized it will unref socket and when g_socket
2651 is finalized the socket will be closed.
2652 https://bugzilla.gnome.org/show_bug.cgi?id=765673
2654 2016-11-18 17:47:13 +0200 Sebastian Dröge <sebastian@centricular.com>
2656 * gst/rtsp-sink/gstrtspclientsink.c:
2657 rtspclientsink: Move to new helper function to parse authentication responses
2658 https://bugzilla.gnome.org/show_bug.cgi?id=774416
2660 2016-11-16 08:42:24 +0200 Sebastian Dröge <sebastian@centricular.com>
2662 * examples/Makefile.am:
2663 * examples/test-auth-digest.c:
2664 * gst/rtsp-server/rtsp-auth.c:
2665 * gst/rtsp-server/rtsp-auth.h:
2666 * win32/common/libgstrtspserver.def:
2667 rtsp-auth: Add support for Digest authentication
2668 https://bugzilla.gnome.org/show_bug.cgi?id=774416
2670 2016-11-17 09:41:53 -0800 Scott D Phillips <scott.d.phillips@intel.com>
2673 * gst/rtsp-server/meson.build:
2675 * tests/check/meson.build:
2677 * win32/common/libgstrtspserver.def:
2678 Enable building with MSVC
2679 https://bugzilla.gnome.org/show_bug.cgi?id=774640
2681 2016-11-18 20:23:14 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
2684 meson: gstreamer gst_check_dep does not exist on windows
2686 2016-11-17 09:43:37 -0800 Scott D Phillips <scott.d.phillips@intel.com>
2688 * gst/rtsp-server/rtsp-client.c:
2689 client: update do_send_message to match type GstRTSPClientSendFunc
2690 This type mismatch fails building with MSVC
2691 https://bugzilla.gnome.org/show_bug.cgi?id=774640
2693 2016-11-11 14:42:08 +0200 Sebastian Dröge <sebastian@centricular.com>
2695 * gst/rtsp-server/rtsp-sdp.c:
2696 rtsp-sdp: Fix indentation
2698 2016-11-10 05:16:00 +0000 Neha Arora <arora.neha@samsung.com>
2700 * gst/rtsp-server/rtsp-media.c:
2701 rtsp-media: Only signal "new-state" if the state has actually changed
2702 https://bugzilla.gnome.org/show_bug.cgi?id=774173
2704 2016-08-24 11:39:13 +0200 Branko Subasic <branko@axis.com>
2706 * gst/rtsp-server/rtsp-client.c:
2707 * gst/rtsp-server/rtsp-client.h:
2708 client: emit signal in the beginning of each rtsp request
2709 These signals let the application validate the requests, configure the
2710 media/stream in a certain way and also generate error status code in
2711 case of error or bad request.
2712 https://bugzilla.gnome.org/show_bug.cgi?id=758062
2714 2016-11-01 18:10:35 +0000 Tim-Philipp Müller <tim@centricular.com>
2717 meson: update version
2719 === release 1.11.0 ===
2721 2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com>
2726 === release 1.10.0 ===
2728 2016-11-01 18:06:46 +0200 Sebastian Dröge <sebastian@centricular.com>
2734 * gst-rtsp-server.doap:
2737 2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com>
2739 * tests/check/gst/rtspserver.c:
2740 * tests/check/gst/stream.c:
2741 tests: try to avoid using the same ports in different tests
2742 Causes problems with client multicast tests otherwise if
2743 tests are run in parallel.
2744 https://bugzilla.gnome.org/show_bug.cgi?id=773640
2746 2016-10-28 17:50:59 +0100 Tim-Philipp Müller <tim@centricular.com>
2748 * tests/check/gst/client.c:
2749 tests: client: use fail_unless_equals_foo() for better failure reporting
2751 2016-09-26 11:16:04 +0200 Göran Jönsson <goranjn@axis.com>
2753 * gst/rtsp-server/rtsp-client.c:
2754 rtsp-client: Session filter in unwatch session
2755 Call session filter with filter_session_media as paramer in
2756 client_unwatch_session if using drop_backlog = FALSE.
2757 In client_unwatch_session its allowed to grow the watchs backlog.
2758 If using drop_backlog = FALSE and the backlog is full it will cause
2759 a deadlock when setting session media state to NULL
2760 if the backlog is not allowed to grow.
2761 https://bugzilla.gnome.org/show_bug.cgi?id=771983
2763 2016-10-20 21:40:18 +0100 Tim-Philipp Müller <tim@centricular.com>
2766 meson: add fallbacks for gst modules
2769 2016-09-14 17:48:39 +0300 Nikita Bobkov <NikitaDBobkov@gmail.com>
2771 * gst/rtsp-server/rtsp-client.c:
2772 rtsp-client: Fix factory leaking in find_media() in error cases
2773 https://bugzilla.gnome.org/show_bug.cgi?id=771488
2775 2016-10-06 11:47:50 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2777 * gst/rtsp-server/rtsp-stream.c:
2778 stream: Fix randomly missing streams from SDP with dynamic elements
2779 When using dynamic elements, gst_rtsp_stream_join_bin() is called from
2780 "pad-added" signal. In that case priv->srcpad could already have its caps,
2781 and they'll be sent to priv->send_src[0] pad. That means that when it
2782 connects "notify::caps" signal, that pad could already have received its
2783 caps and the signal won't be emitted anymore.
2784 In that case priv->caps stay to NULL and when building the SDP that stream
2785 gets ignored. Leading to missing video or audio when playing in client side.
2786 https://bugzilla.gnome.org/show_bug.cgi?id=772478
2788 2016-09-30 11:42:08 +0100 Tim-Philipp Müller <tim@centricular.com>
2791 meson: update version
2793 === release 1.9.90 ===
2795 2016-09-30 13:04:12 +0300 Sebastian Dröge <sebastian@centricular.com>
2801 * gst-rtsp-server.doap:
2804 2016-09-17 13:17:19 +0100 Ian Jamison <ian.dev@arkver.com>
2806 * gst/rtsp-server/rtsp-media-factory.c:
2807 * gst/rtsp-server/rtsp-media.c:
2808 * gst/rtsp-server/rtsp-stream.c:
2809 rtsp-server: Hint that set_multicast_iface expects the name of the interface
2810 To prevent any possibly confusion with IPs or anything else.
2811 https://bugzilla.gnome.org/show_bug.cgi?id=771530
2813 2016-09-18 09:58:55 -0400 Sebastian Dröge <sebastian@centricular.com>
2815 * gst/rtsp-server/rtsp-media-factory.c:
2816 * gst/rtsp-server/rtsp-media.c:
2817 rtsp-media: Call g_free() instead of g_object_unref() on multicast-iface strings
2818 https://bugzilla.gnome.org/show_bug.cgi?id=763000#c5
2820 2016-09-14 11:31:15 +0200 Sebastian Dröge <sebastian@centricular.com>
2823 configure: Depend on gstreamer 1.9.2.1
2825 2016-09-10 20:52:31 +1000 Jan Schmidt <jan@centricular.com>
2829 Automatic update of common submodule
2830 From b18d820 to f980fd9
2832 2016-09-10 09:58:31 +1000 Jan Schmidt <jan@centricular.com>
2836 Automatic update of common submodule
2837 From 6f2d209 to b18d820
2839 2016-09-07 18:44:34 +0300 Sebastian Dröge <sebastian@centricular.com>
2841 * gst/rtsp-server/rtsp-stream.c:
2842 rtsp-stream: Remove unused _locked() variant of a function
2843 It was added during refactoring.
2845 2016-09-07 10:21:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2847 * gst/rtsp-server/rtsp-stream.c:
2848 stream: cosmetic cleanup
2849 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2851 2016-09-07 10:16:19 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2853 * gst/rtsp-server/rtsp-stream.c:
2854 stream: Compare IP addresses case insensitive in more places
2855 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2857 2016-09-07 10:12:18 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2860 * gst/rtsp-server/rtsp-stream.c:
2861 stream: Fix leaked joined_bin
2862 There is no need to keep a strong ref on it, and _leave_bin() was
2863 setting it to NULL before calling g_clear_object() so it was leaked.
2864 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2866 2016-09-06 19:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
2868 * gst/rtsp-server/rtsp-stream.c:
2869 rtsp-stream: Compare IP address strings case insensitive
2870 Otherwise IPv6 addresses might fail this comparision.
2872 2016-09-06 19:10:21 +0300 Sebastian Dröge <sebastian@centricular.com>
2874 * gst/rtsp-server/rtsp-stream.c:
2875 rtsp-stream: Bind multicast sockets to ANY as before
2876 https://bugzilla.gnome.org/show_bug.cgi?id=766612#c48
2878 2016-09-05 18:31:36 +0300 Kseniia <vasilchukkseniia@gmail.com>
2880 * gst/rtsp-server/rtsp-session.c:
2881 rtsp-session: Fix segfault when doing keep-alive after removing the session
2882 If keep-alive happens after removing the session but before finalizing the
2883 stream transport, we would segfault.
2884 https://bugzilla.gnome.org/show_bug.cgi?id=750544
2886 2016-09-05 18:04:50 +0300 Sebastian Dröge <sebastian@centricular.com>
2888 * gst/rtsp-server/rtsp-stream.c:
2889 rtsp-stream: Always create multicast UDP elements if the protocol flag is set
2890 Adding them later will cause deadlocks due to
2891 1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
2892 2) adding the multicast sink
2893 3) waiting for it to get data to preroll again
2894 3) never happens because the queues after the tee are full.
2896 2016-09-05 16:32:57 +0300 Sebastian Dröge <sebastian@centricular.com>
2898 * gst/rtsp-server/rtsp-stream.c:
2899 rtsp-stream: Fix up various multicast related issues
2901 2016-09-05 13:40:59 +0300 Sebastian Dröge <sebastian@centricular.com>
2903 * tests/check/gst/stream.c:
2904 tests: Fix compilation
2906 2016-07-28 15:33:05 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2908 * gst/rtsp-server/rtsp-client.c:
2909 * gst/rtsp-server/rtsp-stream.c:
2910 * tests/check/gst/stream.c:
2911 stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast
2912 This is basically reverting changes introduced in commit f62a9a7,
2913 because it was introducing various regressions:
2914 - It introduces a leak of udpsrc elements that got wrongly fixed by adding
2915 an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
2916 ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
2917 - If a mcast client connects, it creates a new socket in SETUP to try to respect
2918 the destination/port given by the client in the transport, and overrides the
2919 socket already set on the udpsink element. That means that if we already had a
2920 client connected, the source address on the udp packets it receives suddenly
2922 - If a 2nd mcast client connects, the destination/port in its transport is
2923 ignored but its transport wasn't updated.
2924 What this patch does:
2925 - Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
2926 - Always have a tee+queue when udp is enabled. This could be optimized
2927 again in a later patch, but is more complicated. If no unicast clients
2928 connects then those elements are useless, this could be also optimized
2930 - When mcast transport is added, it creates a new set of udpsrc/udpsink,
2931 seperated from those for unicast clients. Since we already support only
2932 one mcast address, we also create only one set of elements.
2933 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2935 2016-07-28 15:20:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2937 * gst/rtsp-server/rtsp-stream.c:
2938 stream: factor our plug_src function
2939 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2941 2016-07-21 21:46:16 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2943 * gst/rtsp-server/rtsp-stream.c:
2944 stream: factor out plug_sink function
2945 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2947 2016-07-20 23:05:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2949 * gst/rtsp-server/rtsp-stream.c:
2950 stream: small documentation clarification
2951 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2953 2016-07-20 15:35:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2955 * gst/rtsp-server/rtsp-stream.c:
2956 stream: rename addr_v4/6 to mcast_addr_v4/6 for clarity
2957 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2959 2016-07-14 11:10:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2961 * gst/rtsp-server/rtsp-stream.c:
2962 stream: Keep a ref on joined bin
2963 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2965 2016-07-20 15:11:32 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2967 * gst/rtsp-server/rtsp-stream.c:
2968 stream: code cleanup
2969 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2971 2016-07-20 23:18:23 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2973 * gst/rtsp-server/rtsp-stream.c:
2974 stream: small fix in error code path
2975 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2977 2016-07-20 20:09:57 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2979 * gst/rtsp-server/rtsp-stream.c:
2980 Revert "rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc"
2981 This partly reverts commit cba045e1b19fad6e689e10206f57903e15f1229a,
2982 but keeps unit tests.
2983 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2985 2016-09-01 12:33:00 +0300 Sebastian Dröge <sebastian@centricular.com>
2990 === release 1.9.2 ===
2992 2016-09-01 12:32:51 +0300 Sebastian Dröge <sebastian@centricular.com>
2998 * gst-rtsp-server.doap:
3001 2016-01-27 01:03:52 +0000 Tim-Philipp Müller <tim@centricular.com>
3004 * examples/meson.build:
3006 * gst/rtsp-server/meson.build:
3007 * gst/rtsp-sink/meson.build:
3009 * pkgconfig/meson.build:
3010 * tests/check/meson.build:
3011 * tests/meson.build:
3012 Add support for Meson as alternative/parallel build system
3013 https://github.com/mesonbuild/meson
3015 2016-08-26 21:56:13 +0200 Josep Torra <n770galaxy@gmail.com>
3018 * tests/check/Makefile.am:
3019 build: silence error about pthread for 'make check' in osx
3020 Fixes "clang: error: argument unused during compilation: '-pthread'"
3022 2015-09-25 15:04:00 +0000 Nikita Bobkov <NikitaDBobkov@gmail.com>
3024 * gst/rtsp-server/rtsp-client.c:
3025 rtsp-client: Fix leaking of media in error cases
3026 With additional fixes by Kseniya Vasilchuk <vasilchukkseniia@gmail.com>
3027 and myself to make the media refcounting a bit easier to follow.
3028 https://bugzilla.gnome.org/show_bug.cgi?id=755632
3030 2016-08-02 15:08:22 +0300 Sebastian Dröge <sebastian@centricular.com>
3032 * gst/rtsp-server/rtsp-client.c:
3033 rtsp-client: Fix leaking of session in error cases
3034 https://bugzilla.gnome.org/show_bug.cgi?id=755632
3036 2016-07-11 21:16:04 +0200 Stefan Sauer <ensonic@users.sf.net>
3039 Automatic update of common submodule
3040 From f363b32 to f49c55e
3042 2016-07-06 13:51:15 +0300 Sebastian Dröge <sebastian@centricular.com>
3047 === release 1.9.1 ===
3049 2016-07-06 13:28:12 +0300 Sebastian Dröge <sebastian@centricular.com>
3055 * gst-rtsp-server.doap:
3058 2016-06-24 02:02:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
3061 configure: Need to add -DGST_STATIC_COMPILATION when building only statically
3062 https://bugzilla.gnome.org/show_bug.cgi?id=767463
3064 2016-06-21 11:49:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
3067 Automatic update of common submodule
3068 From ac2f647 to f363b32
3070 2016-04-14 22:56:11 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
3072 * gst/rtsp-server/rtsp-sdp.c:
3073 * gst/rtsp-server/rtsp-sdp.h:
3074 * gst/rtsp-server/rtsp-stream.c:
3075 * gst/rtsp-server/rtsp-stream.h:
3076 sdp: add rollover counters for all sender SSRC
3077 We add different crypto sessions in MIKEY, one for each sender
3078 SSRC. Currently, all of them will have the same security policy, 0.
3079 The rollover counters are obtained from the srtpenc element using the
3081 https://bugzilla.gnome.org/show_bug.cgi?id=730539
3083 2016-06-07 20:44:42 +0100 Tim-Philipp Müller <tim@centricular.com>
3085 * gst/rtsp-server/rtsp-media-factory.h:
3086 * gst/rtsp-server/rtsp-server.h:
3087 docs: fix some typos
3089 2016-05-25 10:28:43 +0100 Tim-Philipp Müller <tim@centricular.com>
3091 * gst/rtsp-server/Makefile.am:
3092 g-i: pass compiler env to g-ir-scanner
3093 It's what introspection.mak does as well. Should
3094 fix spurious build failures on gnome-continuous
3095 (caused by g-ir-scanner getting compiler details
3096 via python which is broken in some environments
3097 so passing the compiler details bypasses that).
3099 2016-05-18 16:48:44 +0100 Ian <ian.arkver.dev@gmail.com>
3101 * gst/rtsp-server/rtsp-session.c:
3102 rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header
3103 This works with rtspsrc and live555, but fails with e.g. ffmpeg.
3104 https://bugzilla.gnome.org/show_bug.cgi?id=766619
3106 2016-03-07 14:48:38 +0100 Edward Hervey <bilboed@bilboed.com>
3108 * gst/rtsp-sink/gstrtspclientsink.c:
3109 rtspclientsink: Check return value of sscanf
3110 And just make sure we always have 0/0 if we have an error
3113 2016-04-25 08:55:25 -0400 Jake Foytik <jake.foytik@ipconfigure.com>
3115 * gst/rtsp-server/rtsp-stream.c:
3116 * tests/check/gst/rtspserver.c:
3117 * tests/check/gst/stream.c:
3118 rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
3119 - Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
3120 - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
3121 - Create unit test for shared media.
3122 https://bugzilla.gnome.org/show_bug.cgi?id=764744
3124 2016-04-11 10:55:23 +0300 Sebastian Dröge <sebastian@centricular.com>
3126 * gst/rtsp-server/rtsp-stream.c:
3127 rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
3128 For IPv6 addresses, binding to a multicast group does not work on Linux
3129 either. Always bind to ANY and then later join the multicast group.
3130 https://bugzilla.gnome.org/show_bug.cgi?id=764679
3132 2016-04-14 10:05:02 +0100 Julien Isorce <j.isorce@samsung.com>
3135 Automatic update of common submodule
3136 From 6f2d209 to ac2f647
3138 2016-04-06 10:09:46 +0200 Patricia Muscalu <patricia@axis.com>
3140 * gst/rtsp-server/rtsp-thread-pool.c:
3141 rtsp-thread-pool: explained why GSource is a part of ThreadImpl
3142 Clarified why it is necessary to add source information to
3143 GstRTSPThreadImpl. See the reported bug in GLib:
3144 https://bugzilla.gnome.org/show_bug.cgi?id=720186
3145 for more information.
3146 https://bugzilla.gnome.org/show_bug.cgi?id=761702
3148 2016-04-04 12:58:38 +0300 Sebastian Dröge <sebastian@centricular.com>
3150 * examples/Makefile.am:
3151 examples: Clean up CFLAGS/LDADD even more
3152 The internal .la should come first and is part of LDADD, as is
3155 2016-04-04 12:39:39 +0300 Sebastian Dröge <sebastian@centricular.com>
3157 * examples/Makefile.am:
3158 examples: Clean up CFLAGS/LDADD to link with the correct versions of all libraries
3160 2016-04-03 12:06:29 +0300 Sebastian Dröge <sebastian@centricular.com>
3162 * gst/rtsp-server/Makefile.am:
3163 rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS)
3165 2015-12-30 18:39:05 +0200 Sebastian Dröge <sebastian@centricular.com>
3167 * gst/rtsp-server/rtsp-client.c:
3168 * gst/rtsp-server/rtsp-media-factory.c:
3169 * gst/rtsp-server/rtsp-media-factory.h:
3170 * gst/rtsp-server/rtsp-media.c:
3171 * gst/rtsp-server/rtsp-media.h:
3172 * gst/rtsp-server/rtsp-sdp.c:
3173 * gst/rtsp-server/rtsp-stream.c:
3174 * gst/rtsp-server/rtsp-stream.h:
3175 rtsp-server: Implement clock signalling according to RFC7273
3176 For NTP and PTP clocks we signal the actual clock that is used and signal
3177 the direct media clock offset.
3178 For all other clocks we at least signal that it's the local sender clock.
3179 This allows receivers to know which clock was used to generate the media and
3180 its RTP timestamps. Receivers can then implement network synchronization,
3181 either absolute or at least relative by getting the sender clock rate directly
3182 via NTP/PTP instead of estimating it from RTP timestamps and packet receive
3184 https://bugzilla.gnome.org/show_bug.cgi?id=760005
3186 2016-03-02 19:42:58 +0200 Sebastian Dröge <sebastian@centricular.com>
3188 * gst/rtsp-sink/gstrtspclientsink.c:
3189 rtspclientsink: Add support for setting the multicast interface
3190 https://bugzilla.gnome.org/show_bug.cgi?id=763000
3192 2016-03-02 19:42:13 +0200 Sebastian Dröge <sebastian@centricular.com>
3194 * gst/rtsp-server/rtsp-media-factory.c:
3195 * gst/rtsp-server/rtsp-media-factory.h:
3196 * gst/rtsp-server/rtsp-media.c:
3197 * gst/rtsp-server/rtsp-media.h:
3198 * gst/rtsp-server/rtsp-stream.c:
3199 * gst/rtsp-server/rtsp-stream.h:
3200 rtsp-media: Add support for setting the multicast interface
3201 https://bugzilla.gnome.org/show_bug.cgi?id=763000
3203 2016-03-07 08:50:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
3205 * gst/rtsp-sink/gstrtspclientsink.c:
3206 rtspclientsink: use new gst_element_class_add_static_pad_template()
3207 https://bugzilla.gnome.org/show_bug.cgi?id=763196
3209 2016-03-24 13:33:43 +0200 Sebastian Dröge <sebastian@centricular.com>
3214 === release 1.8.0 ===
3216 2016-03-24 13:00:35 +0200 Sebastian Dröge <sebastian@centricular.com>
3222 * gst-rtsp-server.doap:
3225 2016-03-16 23:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
3227 * gst/rtsp-server/rtsp-stream.c:
3228 rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
3229 This would get us NO_PREROLL in the bin again and break seeking.
3230 Thanks to Carlos Rafael Giani for helping to debug this!
3231 https://bugzilla.gnome.org/show_bug.cgi?id=740509
3233 === release 1.7.91 ===
3235 2016-03-15 12:26:13 +0200 Sebastian Dröge <sebastian@centricular.com>
3241 * gst-rtsp-server.doap:
3244 2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
3246 * gst/rtsp-server/rtsp-stream.c:
3247 rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
3248 Without this, RECORD pipelines are broken because
3249 a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
3250 added later. Previously it was there earlier and due to NO_PREROLL caused the
3251 pipeline to preroll immediately
3252 b) the udpsrc for the pipeline is added later and never set to PLAYING state,
3253 as the corresponding code previously was only for PLAY pipelines.
3254 https://bugzilla.gnome.org/show_bug.cgi?id=763281
3256 2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
3258 * gst/rtsp-server/rtsp-stream.c:
3259 rtsp-stream: Fix typo in the docstring
3260 gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
3262 2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
3264 * gst/rtsp-server/rtsp-stream.c:
3265 rtsp-stream: Disable multicast loopback for all our sockets
3266 On Windows this is a receiver-side setting, on Linux a sender-side setting. As
3267 we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
3268 loopback setting on the socket... while udpsink does which unfortunately has
3269 no effect here on Windows but on Linux.
3270 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3272 2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
3274 * tests/check/gst/stream.c:
3275 stream tests: added new tests
3276 Test a case when the address pool only contains multicast addresses
3277 and the client is requesting unicast udp.
3278 Added tests for multicast ports allocation.
3279 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3281 2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
3283 * gst/rtsp-server/rtsp-stream.c:
3284 rtsp-stream: Only bind multicast sockets to ANY on Windows
3285 On Linux it is still needed to bind to the multicast address
3286 to filter out random other packets, while on Windows binding
3287 to multicast addresses just fails.
3289 2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
3291 * gst/rtsp-server/rtsp-stream.c:
3292 rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
3293 Otherwise we fail to allocate UDP ports if the pool only contains multicast
3294 addresses, which is something that used to work before. For unicast addresses
3295 if the pool contains none, we just allocate them as if there is no pool at
3297 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3299 2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
3301 * gst/rtsp-server/rtsp-client.c:
3302 * gst/rtsp-server/rtsp-stream.c:
3303 rtsp-server: Fix indentation
3305 2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
3307 * gst/rtsp-server/rtsp-stream.c:
3308 rtsp-stream: Don't bind the sockets to multicast addresses
3309 This works on Linux but fails completely on Windows. You're supposed
3310 to bind to ANY and then join the multicast group.
3311 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3313 === release 1.7.90 ===
3315 2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
3321 * gst-rtsp-server.doap:
3324 2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
3327 Automatic update of common submodule
3328 From b64f03f to 6f2d209
3330 2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
3332 * gst/rtsp-sink/gstrtspclientsink.c:
3333 * tests/check/gst/rtspclientsink.c:
3334 rtspsink: Fix some leaks in rtspclientsink and the unit test.
3335 https://bugzilla.gnome.org/show_bug.cgi?id=762525
3337 2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
3339 * tests/check/gst/media.c:
3340 * tests/check/gst/rtspclientsink.c:
3341 * tests/check/gst/rtspserver.c:
3342 * tests/check/gst/stream.c:
3343 tests: unit test fixes
3344 Removed port allocation test from the media suite.
3345 The port allocation failure is now in the stream suite.
3347 Make sure that the media is suspended after the DESCRIBE request
3348 before reconfiguring the UDP sinks.
3350 In the RECORD case we have to set async property to false
3351 for the appsink element in the test in order to make sure
3352 that the media pipeline doesn't hang in start_preroll().
3353 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3355 2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
3357 * gst/rtsp-server/rtsp-client.c:
3358 * gst/rtsp-server/rtsp-stream.c:
3359 * gst/rtsp-server/rtsp-stream.h:
3360 rtsp-stream: postpone UDP socket allocation until SETUP
3361 Postpone the allocation of the UDP sockets until we know
3362 what transport has been chosen by the client.
3363 Both unicast and multicast UDP sources are created in one
3365 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3367 2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
3369 * gst/rtsp-server/rtsp-stream.c:
3370 rtsp-stream: postpone the creation of the UDP sources
3371 Code refactoring: allocate the UDP ports after the sender and
3372 the reciver parts have been created.
3373 We postpone the creation of the UDP sources until the UDP
3374 ports have been allocated.
3375 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3377 2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
3379 * gst/rtsp-server/rtsp-stream.c:
3380 rtsp-stream: added function for setting UDP sources to PLAYING state
3381 Code refactoring: Introduced a function for setting UDP sources
3383 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3385 2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
3387 * gst/rtsp-server/rtsp-stream.c:
3388 rtsp-stream: added function for creating and configuring UDP sources
3389 Code refactoring: create and configure UDP sources in a separate function.
3390 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3392 2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
3394 * gst/rtsp-server/rtsp-stream.c:
3395 rtsp-stream: added function for RTP/RTCP socket configuration
3396 Code refactoring: configure RTP and RTCP sockets for UDP sinks
3397 in a separate function.
3398 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3400 2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
3402 * gst/rtsp-server/rtsp-stream.c:
3403 rtsp-stream: added function for creating and configuring UDP sinks
3404 Code refactoring: create and configure UDP sinks in a separate function.
3405 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3407 2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
3409 * gst/rtsp-server/rtsp-stream.c:
3410 rtsp-stream: added helper function for creating the sender/receiver parts
3411 Code refactoring: introduced helper function for creating
3412 the receiver and the sender parts of the streaming pipeline.
3413 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3415 2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
3420 === release 1.7.2 ===
3422 2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
3428 * gst-rtsp-server.doap:
3431 2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
3433 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
3434 uninstalled.pc: add support for non libtool build systems
3435 Currently the .la path is provided which requires to use libtool as
3436 mentioned in the GStreamer manual section-helloworld-compilerun.html.
3437 It is fine as long as the application is built using libtool.
3438 So currently it is not possible to compile a GStreamer application
3439 within gst-uninstalled with CMake or other build system different
3441 This patch allows to do the following in gst-uninstalled env:
3442 gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
3443 gstreamer-rtsp-server-1.0)
3444 Previously it required to prepend libtool --mode=link
3445 https://bugzilla.gnome.org/show_bug.cgi?id=720778
3447 2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
3449 * gst/rtsp-sink/gstrtspclientsink.c:
3450 rtspclientsink: remove check for impossible condition
3451 Goto error label checks stream to see if it needs to be unreferenced before
3452 returning, but this goto jumps happens before the stream is ever set, so it
3453 will always be NULL in this error label.
3456 2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
3458 * gst/rtsp-sink/gstrtspclientsink.c:
3459 rtspclientsink: clean switch statements
3460 Coverity demands for fallthrough statements to be clearly commented,
3461 to distinguish from accidental fall throughs. And it also needs all
3462 cases to finish with a break, even if the break is never going to be
3463 executed like in the case of a continue jump.
3467 2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
3469 * tests/check/Makefile.am:
3470 tests: extend the AM_TESTS_ENVIRONMENT from check.mak
3471 To get the CK_DEFAULT_TIMEOUT defined for all tests
3472 Also removes a 120 seconds timeout that was set as default
3473 explicitly in this module
3474 https://bugzilla.gnome.org/show_bug.cgi?id=761472
3476 2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
3480 Automatic update of common submodule
3481 From 86e4663 to b64f03f
3483 2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
3485 * gst/rtsp-server/rtsp-media.c:
3486 rtsp-media: fix state_lock not locked again when preroll fails
3487 https://bugzilla.gnome.org/show_bug.cgi?id=761399
3489 2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
3492 configure: Move plugin specific flags below all the others
3493 They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
3494 -no-undefined. And -no-undefined is required on Windows to build DLLs.
3496 2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
3498 * gst/rtsp-sink/gstrtspclientsink.c:
3499 rtspclientsink: Simplify slightly using new -base API
3500 Use the new Mikey and SDP API in the base plugins libs
3501 to simplify some code.
3502 https://bugzilla.gnome.org/show_bug.cgi?id=758180
3504 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3509 * gst/rtsp-sink/Makefile.am:
3510 * gst/rtsp-sink/gstrtspclientsink.c:
3511 * gst/rtsp-sink/gstrtspclientsink.h:
3512 * gst/rtsp-sink/plugin.c:
3513 * tests/check/Makefile.am:
3514 * tests/check/gst/rtspclientsink.c:
3515 rtspsink: Add rtspclientsink element
3516 Add an rtspclientsink element that accepts streams for which
3517 there is a registered payloader and sends them to
3518 an RTSP server using RECORD.
3519 Sending is synchronised to the pipeline clock. Payload-types
3520 are automatically selected. The 'new-payloader' signal is fired
3521 for custom configuration of payloaders when they are created.
3522 Can now stream a movie like this:
3524 ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
3525 decodebin name=depay1 ! audioconvert ! autoaudiosink )"
3527 gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
3528 queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
3529 https://bugzilla.gnome.org/show_bug.cgi?id=758180
3531 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3533 * gst/rtsp-server/rtsp-stream.c:
3534 * gst/rtsp-server/rtsp-stream.h:
3535 rtsp-stream: Add functions for using rtsp-stream from the client
3536 Add a boolean to indicate that the rtsp-stream is running on the
3537 'client' side of an RTSP connection, for sending streams via
3538 RECORD. In that case, the roles of the client/server ports
3539 in transport setup are swapped.
3540 https://bugzilla.gnome.org/show_bug.cgi?id=758180
3542 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3544 * gst/rtsp-server/rtsp-sdp.c:
3545 * gst/rtsp-server/rtsp-sdp.h:
3546 rtsp-sdp: Add gst_rtsp_sdp_from_stream()
3547 A new function that adds info from a GstRTSPStream into an SDP message.
3548 https://bugzilla.gnome.org/show_bug.cgi?id=758180
3550 2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
3552 * gst/rtsp-server/rtsp-media.c:
3553 rtsp-media: Fix mutex beeing unlocked while they should be locked
3554 https://bugzilla.gnome.org/show_bug.cgi?id=761226
3556 2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
3558 * gst/rtsp-server/rtsp-media-factory.c:
3559 rtsp-media-factory: add missing break in "clock" property setter
3562 2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
3564 * gst/rtsp-server/rtsp-stream.c:
3565 rtsp-stream: fixed assert during update transport
3566 When RTSP server trying update transport during multicast, it throws an
3567 assert. The assert is thrown because it is trying to get the parent of
3568 an non-existing funnel element.
3569 https://bugzilla.gnome.org/show_bug.cgi?id=760150
3571 2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
3573 * gst/rtsp-server/rtsp-permissions.h:
3574 * gst/rtsp-server/rtsp-thread-pool.h:
3575 * gst/rtsp-server/rtsp-token.h:
3576 docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
3577 gtk-doc can handle static inline functions just fine these days,
3578 there's no need for this stuff any more.
3580 2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
3582 * gst/rtsp-server/rtsp-media.c:
3583 * gst/rtsp-server/rtsp-sdp.c:
3584 sdp: replace duplicated codes to call new base sdp apis
3585 https://bugzilla.gnome.org/show_bug.cgi?id=745880
3587 2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
3589 * examples/test-netclock.c:
3590 test-netclock: Use the new API to configure a clock directly
3592 2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
3594 * gst/rtsp-server/rtsp-media-factory.c:
3595 * gst/rtsp-server/rtsp-media-factory.h:
3596 * gst/rtsp-server/rtsp-media.c:
3597 * gst/rtsp-server/rtsp-media.h:
3598 rtsp-media: Add API to directly configure a clock on the media pipelines
3600 2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
3602 * gst/rtsp-server/rtsp-media.c:
3603 rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
3605 2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
3607 * gst/rtsp-server/rtsp-media-factory.c:
3608 rtsp-media-factory: Add FIXME for 2.0
3610 2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
3612 * gst/rtsp-server/rtsp-stream.c:
3613 rtsp-stream: Fix indentation
3615 2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3617 * gst/rtsp-server/rtsp-media.c:
3618 rtsp-media: Do not prepare media after media times out
3619 Deferred calls to start_prepare() can be deferred past the point until
3620 which wait_preroll() and by proxy gst_rtsp_media_get_status() is
3621 prepared to wait. Previously there was no lock and no check for this
3622 situation. This meant that a media could be prepared and unprepared
3623 simultaneously by two different threads. Now a lock is in place and a
3624 suitable check is done.
3625 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
3627 2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
3629 * gst/rtsp-server/rtsp-client.c:
3630 * gst/rtsp-server/rtsp-media-factory.c:
3631 * gst/rtsp-server/rtsp-media-factory.h:
3632 * gst/rtsp-server/rtsp-media.c:
3633 * gst/rtsp-server/rtsp-media.h:
3634 rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
3635 Without TEARDOWN it might be desireable to keep the media running and continue
3636 sending data to the client, even if the RTSP connection itself is
3638 Only do this for session medias that have only UDP transports. If there's at
3639 least on TCP transport, it will stop working and cause problems when the
3640 connection is disconnected.
3641 https://bugzilla.gnome.org/show_bug.cgi?id=758999
3643 2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
3648 === release 1.7.1 ===
3650 2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
3656 * gst-rtsp-server.doap:
3659 2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
3662 configure: Make -Bsymbolic check work with clang.
3663 Update the -Bsymbolic check with the version glib has. This version
3665 https://bugzilla.gnome.org/show_bug.cgi?id=759713
3667 2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
3669 * gst/rtsp-server/rtsp-session-pool.c:
3670 rtsp-session-pool: Avoid dollar sign ($) in session ids
3671 Live555 in VLC strips off dollar signs and then gets very confused,
3672 we don't loose too much entropy by just skipping it.
3674 2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
3676 * gst/rtsp-server/rtsp-address-pool.h:
3677 * gst/rtsp-server/rtsp-auth.h:
3678 * gst/rtsp-server/rtsp-client.h:
3679 * gst/rtsp-server/rtsp-media-factory-uri.h:
3680 * gst/rtsp-server/rtsp-media-factory.h:
3681 * gst/rtsp-server/rtsp-media.h:
3682 * gst/rtsp-server/rtsp-mount-points.h:
3683 * gst/rtsp-server/rtsp-permissions.h:
3684 * gst/rtsp-server/rtsp-server.h:
3685 * gst/rtsp-server/rtsp-session-media.h:
3686 * gst/rtsp-server/rtsp-session-pool.h:
3687 * gst/rtsp-server/rtsp-session.h:
3688 * gst/rtsp-server/rtsp-stream-transport.h:
3689 * gst/rtsp-server/rtsp-stream.h:
3690 * gst/rtsp-server/rtsp-thread-pool.h:
3691 * gst/rtsp-server/rtsp-token.h:
3692 rtsp-server: Add g_autoptr() support to all types
3693 https://bugzilla.gnome.org/show_bug.cgi?id=754464
3695 2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
3697 * gst/rtsp-server/rtsp-stream.c:
3698 rtsp-stream: fixed valgrind error
3699 Fixed the valgrind error in unit test. The UDP source created during
3700 gst_rtsp_stream_join_bin() was not released while destroying the rtp
3702 https://bugzilla.gnome.org/show_bug.cgi?id=759010
3704 2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
3708 Automatic update of common submodule
3709 From b319909 to 86e4663
3711 2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
3713 * gst/rtsp-server/rtsp-client.c:
3714 rtsp-client: suspend media during setup request
3715 SETUP request from clients needs to suspend the media to clear the
3716 prerolled buffers. Otherwise it will not affect the prerolled buffer
3717 and the prerolled buffers will be incorrect (for example block-size
3718 from setup request will not affect the prerolled buffer unless the
3719 media is suspended).
3720 https://bugzilla.gnome.org/show_bug.cgi?id=758268
3722 2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
3724 * gst/rtsp-server/rtsp-stream.c:
3725 rtsp-stream: create stream pipeline based on transport
3726 Based on the protocol, create the rtsp stream pipeline. If only TCP or
3727 only UDP is set as the transport protocol, it will not add the extra tee
3728 or queue element to the pipeline. Both these elements will be added, if
3729 it supports both TCP and UDP protocols. This improves the pipeline
3730 performance when one protocol is present.
3731 https://bugzilla.gnome.org/show_bug.cgi?id=758179
3733 2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
3735 * gst/rtsp-server/rtsp-stream.c:
3736 rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
3737 Adding them when not needed will start some logic inside rtpbin that might be
3738 problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
3739 would start up a rtpjitterbuffer and behave in weird ways.
3740 We still set up the UDP sources for RTP receiving for a sender media to be
3741 able to receive any packets sent by the client for NAT traversal. They will
3742 all go to a fakesink though.
3743 Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
3744 NO_PREROLL, which will cause deadlocks when seeking the media as it will never
3745 receive ASYNC_DONE after a seek.
3746 https://bugzilla.gnome.org/show_bug.cgi?id=758319
3748 2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
3750 * gst/rtsp-server/rtsp-stream.c:
3751 rtsp-stream: Disable multicast loopback for the multicast udp sources too
3752 On POSIX this setting is for sender sockets, on Windows for receiver sockets.
3753 Previously we were only setting this for sender sockets, which caused looped
3754 back packets to be received on Windows if a multicast transport was used.
3756 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3758 * examples/test-record-auth.c:
3759 * examples/test-record.c:
3760 examples: Actually use the provided port in the record examples
3762 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3764 * examples/test-record-auth.c:
3765 test-record-auth: Add the option to build in TLS support
3767 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3769 * examples/test-auth.c:
3770 test-auth: Use an 'anonymous' user for unauthenticated default
3771 There's a comment on one of the resources that 'user' and 'admin'
3772 shouldn't even be able to see it, but they can if the default
3773 token is 'admin2', since that gives them access anyway.
3775 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3777 * examples/.gitignore:
3778 * examples/Makefile.am:
3779 * examples/test-record-auth.c:
3780 Add test-record-auth example
3782 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3784 * gst/rtsp-server/rtsp-client.c:
3785 * tests/check/gst/client.c:
3786 rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
3788 2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
3790 * gst/rtsp-server/rtsp-server.c:
3791 rtsp-server: Change the logic so we don't pop a NULL context
3792 When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
3793 will sometimes fail. This call is made before any context is pushed
3794 resulting in an attempt to pop a NULL context.
3795 https://bugzilla.gnome.org/show_bug.cgi?id=757949
3797 2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
3799 * tests/check/gst/rtspserver.c:
3800 rtspserver: Add udp-mcast transport SETUP test
3801 Refactor utility functions in the test file so they can handle
3802 more than UDP and TCP as lower transport.
3803 https://bugzilla.gnome.org/show_bug.cgi?id=756969
3805 2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
3807 * gst/rtsp-server/rtsp-stream.c:
3808 rtsp-stream: Always unref return value of gst_object_get_parent()
3809 Fixes a leak of a GstBin in the udp-mcast case.
3810 https://bugzilla.gnome.org/show_bug.cgi?id=756968
3812 2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
3815 Automatic update of common submodule
3816 From b99800a to b319909
3818 2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
3821 Use new GST_ENABLE_EXTRA_CHECKS #define
3822 https://bugzilla.gnome.org/show_bug.cgi?id=756870
3824 2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
3827 Automatic update of common submodule
3828 From 6babecd to b99800a
3830 2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
3833 Update GLib dependency to 2.40.0
3835 2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
3837 * examples/test-mp4.c:
3838 * gst/rtsp-server/rtsp-stream.c:
3839 stream: listen to sender ssrc signals
3840 https://bugzilla.gnome.org/show_bug.cgi?id=746747
3842 2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
3845 common: update for new suppression
3846 Makes check-valgrind pass with glib 2.46
3848 2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3850 * gst/rtsp-server/rtsp-media.c:
3851 rtsp-media: Take reference to media that will be prepared
3852 default_prepare() takes a transfer-none reference GstRTSPMedia object.
3853 Later on a g_idle_source_new() is created and a pointer to the media
3854 object is passed as user data. If the media is freed before the idle
3855 source is dispatched the media object pointer is invalid, but the idle
3856 source callback expects it to still be valid. To fix this a reference to
3857 the media object is taken when registering the source callback function
3858 and a corresponding release of the reference is done when the souce is
3860 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
3862 2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
3864 * examples/test-launch.c:
3865 * examples/test-mp4.c:
3866 * examples/test-ogg.c:
3867 * examples/test-record.c:
3868 * examples/test-uri.c:
3869 rtsp-server: Fix memory leaks when context parse fails
3870 When g_option_context_parse fails, context and error variables are not getting free'd
3871 which results in memory leaks. Free'ing the same.
3872 And replacing g_error_free with g_clear_error, which checks if the error being passed
3873 is not NULL and sets the variable to NULL on free'ing.
3874 https://bugzilla.gnome.org/show_bug.cgi?id=753863
3876 2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
3881 === release 1.6.0 ===
3883 2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
3889 * gst-rtsp-server.doap:
3892 === release 1.5.91 ===
3894 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
3900 * gst-rtsp-server.doap:
3903 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
3905 * docs/libs/gst-rtsp-server-sections.txt:
3906 * gst/rtsp-server/rtsp-stream.c:
3907 stream: fix docs for recently-added get/set_buffer_size API
3908 https://bugzilla.gnome.org/show_bug.cgi?id=749095
3910 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
3912 * gst/rtsp-server/rtsp-media.c:
3913 rtsp-media: Don't crash on encrypted RTX SDP
3914 In parse_keymgmt(), don't mutate the input string that's been passed
3915 as const, especially since we might need the original value again if
3916 the same key info applies to multiple streams (RTX, for example).
3917 https://bugzilla.gnome.org/show_bug.cgi?id=754753
3919 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
3921 * examples/test-mp4.c:
3922 test-mp4: Support filenames with spaces in them. Error out on too few arguments
3924 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
3926 * examples/test-record.c:
3927 test-record: Check parameter count and print out help
3928 If no launch pipeline was supplied, print out some help
3930 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
3932 * gst/rtsp-server/rtsp-media.c:
3933 * gst/rtsp-server/rtsp-stream.c:
3934 * gst/rtsp-server/rtsp-stream.h:
3935 rtsp-stream: Implement UDP buffer size setting.
3936 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
3938 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
3939 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
3941 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
3943 * gst/rtsp-server/rtsp-media.h:
3944 rtsp-media: Fix small typo causing gtk-doc to complain
3946 === release 1.5.90 ===
3948 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
3954 * gst-rtsp-server.doap:
3957 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
3959 * gst/rtsp-server/rtsp-media-factory.c:
3960 media-factory: get port number through gst_rtsp_url_get_port
3961 https://bugzilla.gnome.org/show_bug.cgi?id=753473
3963 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
3965 * tests/check/gst/media.c:
3966 media-test: Removing unnecessary assertion
3967 https://bugzilla.gnome.org/show_bug.cgi?id=753385
3969 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
3971 * gst/rtsp-server/rtsp-server.c:
3972 Document that source keeps a ref on server until it's destroyed
3973 https://bugzilla.gnome.org/show_bug.cgi?id=749227
3975 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
3977 * tests/check/gst/media.c:
3978 media-test: Test for multiple dynamic payload
3979 https://bugzilla.gnome.org/show_bug.cgi?id=753385
3981 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
3983 * gst/rtsp-server/rtsp-media.c:
3984 media: Only add fakesink once per pipeline
3985 The intention is to prevent going PLAYING state before pads are created.
3986 If there was mutilple dynamic payload, it would leak few fakesink and
3987 actually prevent from ever reaching playing state.
3988 https://bugzilla.gnome.org/show_bug.cgi?id=753385
3990 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
3992 * gst/rtsp-server/rtsp-media.c:
3993 Revert "rtsp-media: Only add 1 fakesink per pipeline"
3994 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
3996 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
3998 * gst/rtsp-server/rtsp-media.c:
3999 rtsp-media: Only add 1 fakesink per pipeline
4000 There should be only one fakesink per pipeline, not per dynpay. This
4001 would lead to element naming clash.
4003 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
4005 * gst/rtsp-server/rtsp-media.c:
4006 rtsp-media: assertion error due to wrong condition check
4007 In media to caps function, reserved_keys array is being used for variable i,
4008 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
4009 changed it to variable j
4010 https://bugzilla.gnome.org/show_bug.cgi?id=753009
4012 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
4014 * gst/rtsp-server/rtsp-media.c:
4015 rtsp-media: Strip keys from the fmtp that we use internally in our caps
4016 Skip keys from the fmtp, which we already use ourselves for the
4017 caps. Some software is adding random things like clock-rate into
4018 the fmtp, and we would otherwise here set a string-typed clock-rate
4019 in the caps... and thus fail to create valid RTP caps
4020 https://bugzilla.gnome.org/show_bug.cgi?id=753009
4022 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4024 * gst/rtsp-server/rtsp-thread-pool.c:
4025 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
4026 https://bugzilla.gnome.org/show_bug.cgi?id=752640
4028 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
4031 Automatic update of common submodule
4032 From f74b2df to 9aed1d7
4034 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
4039 === release 1.5.2 ===
4041 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
4047 * gst-rtsp-server.doap:
4050 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
4052 * gst/rtsp-server/rtsp-client.c:
4053 * gst/rtsp-server/rtsp-client.h:
4054 * tests/check/gst/client.c:
4055 rtsp-client: allow application to decide what requirements are supported
4056 Add "check-requirements" signal and vfunc to allow application
4057 (and subclasses) to check the requirements.
4058 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
4059 https://bugzilla.gnome.org/show_bug.cgi?id=749417
4061 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
4064 Automatic update of common submodule
4065 From 6015d26 to f74b2df
4067 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
4069 * gst/rtsp-server/rtsp-media.c:
4070 rtsp-media: Always use real payloader when creating streams
4071 A bin that contains the real payloader might be used as payloader. In this
4072 case we have to get the real payloader for the various properties it provides.
4073 Example use cases for this are bins that payload some media and then have
4074 additional elements that add metadata or RTP extension headers to the stream.
4075 https://bugzilla.gnome.org/show_bug.cgi?id=750800
4077 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
4079 * examples/test-netclock-client.c:
4080 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
4082 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
4084 * examples/test-netclock-client.c:
4085 * examples/test-netclock.c:
4086 test-netclock: Use new ntp-time-source property on rtpbin
4087 Select the clock time to be used as NTP time source. This allows proper
4088 synchronization between receivers, independent of sharing base times, and just
4089 requires them to use the same clock.
4091 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
4093 * examples/test-netclock-client.c:
4094 * examples/test-netclock.c:
4095 test-netclock: Setting the same base time on sender and receiver is not necessary
4096 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
4098 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
4100 * gst/rtsp-server/rtsp-stream.c:
4101 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
4102 https://bugzilla.gnome.org/show_bug.cgi?id=750764
4104 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
4106 * docs/libs/gst-rtsp-server.types:
4107 docs: add missing types
4108 https://bugzilla.gnome.org/show_bug.cgi?id=750764
4110 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
4112 * docs/libs/gst-rtsp-server-sections.txt:
4113 docs: add missing apis
4114 https://bugzilla.gnome.org/show_bug.cgi?id=750764
4116 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
4118 * examples/test-netclock-client.c:
4119 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
4121 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4123 * docs/libs/gst-rtsp-server-sections.txt:
4124 * gst/rtsp-server/rtsp-auth.c:
4125 * gst/rtsp-server/rtsp-auth.h:
4126 GstRTSPAuth: Add client certificate authentication support
4127 https://bugzilla.gnome.org/show_bug.cgi?id=750471
4129 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
4131 * examples/test-netclock-client.c:
4132 test-netclock-client: Use new GstClock API to wait for clock synchronization
4134 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
4136 * examples/test-netclock-client.c:
4137 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
4138 A mainloop is needed to get glimagesink to display something on OSX, and
4139 the source-setup signal just makes things a little bit easier.
4141 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
4144 Automatic update of common submodule
4145 From d9a3353 to 6015d26
4147 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
4150 Automatic update of common submodule
4151 From d37af32 to d9a3353
4153 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
4156 Automatic update of common submodule
4157 From 21ba2e5 to d37af32
4159 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
4162 Automatic update of common submodule
4163 From c408583 to 21ba2e5
4165 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
4167 * docs/libs/Makefile.am:
4168 docs: remove variables that we define in the snippet from common
4169 This is syncing our Makefile.am with upstream gtkdoc.
4171 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
4174 Automatic update of common submodule
4175 From 44a3517 to c408583
4177 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
4182 === release 1.5.1 ===
4184 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
4190 * gst-rtsp-server.doap:
4193 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
4195 * gst/rtsp-server/rtsp-client.c:
4196 rtsp-client: No flush during Teardown.
4197 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
4198 backlog is empty it can happen that just a part of a message will be
4199 sent and rest is in backlog queue. If then flush during teardown
4200 just a part of message will be sent.This can lead to client miss
4201 teardown response since it expect to get the last part of message.
4202 The flushing during teardown was introduced to fix a deadlock that now
4203 is fixed more generally in handle_request by temporary setting backlog
4205 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
4207 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
4209 * tests/check/Makefile.am:
4210 tests: Use AM_TESTS_ENVIRONMENT
4211 Needed by the new automake test runner and the
4212 current version of the common submodule.
4214 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
4216 * gst/rtsp-server/rtsp-media.h:
4217 * gst/rtsp-server/rtsp-stream.h:
4218 rtsp-server: Use single-include rtsp header to make sure we get all definitions
4220 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
4222 * gst/rtsp-server/rtsp-media.c:
4223 rtsp-media: Mark some more functions static
4225 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
4227 * gst/rtsp-server/rtsp-media.c:
4228 rtsp-media: Only unblock the media in suspend() when actually changing the state
4229 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
4231 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
4233 * examples/test-video-rtx.c:
4234 examples: Use AVPF profile for the RTX example
4236 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
4238 * gst/rtsp-server/rtsp-sdp.c:
4239 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
4241 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
4243 * gst/rtsp-server/rtsp-stream.c:
4244 rtsp-stream: get valid clock-rate from last-sample
4245 clock-rate in last-sample's caps is integer, not unsigned.
4246 To get this value properly, variable needs to be type-casted to int.
4247 https://bugzilla.gnome.org/show_bug.cgi?id=747614
4249 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
4253 autogen.sh: only run autopoint if gettext requested in configure.ac
4254 Not just because there happens to be a po directory.
4255 https://bugzilla.gnome.org/show_bug.cgi?id=748058
4257 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
4260 Revert "configure.ac: uncomment gettext version setup"
4261 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
4262 We don't need a gettext setup here and there's no po
4263 directory either, so no reason why autopoint would be
4264 run in the first place.
4265 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
4267 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
4269 * examples/test-multicast.c:
4270 * examples/test-multicast2.c:
4271 * examples/test-sdp.c:
4272 * examples/test-video-rtx.c:
4273 * examples/test-video.c:
4274 * tests/test-cleanup.c:
4275 * tests/test-reuse.c:
4276 Fix timeout function signatures across tests and examples
4278 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
4280 * tests/check/Makefile.am:
4281 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
4282 Make sure the test environment is set up.
4283 https://bugzilla.gnome.org//show_bug.cgi?id=747624
4285 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
4288 configure: bump automake requirement to 1.14 and autoconf to 2.69
4289 This is only required for builds from git, people can still
4290 build tarballs if they only have older autotools.
4291 https://bugzilla.gnome.org//show_bug.cgi?id=747624
4293 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
4296 configure.ac: uncomment gettext version setup
4297 Fixes autogen.sh. It would run autopoint, which would complain
4298 that it could not find the gettext version in configure.ac.
4299 https://bugzilla.gnome.org/show_bug.cgi?id=748058
4301 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
4303 * examples/test-video-rtx.c:
4304 test-video-rtx: set exact payload type to PCMA payloader
4305 Setting wrong payload type causes failure to do retransmission through audio stream
4306 https://bugzilla.gnome.org/show_bug.cgi?id=747839
4308 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
4310 * gst/rtsp-server/rtsp-media.c:
4311 * gst/rtsp-server/rtsp-stream.c:
4312 * gst/rtsp-server/rtsp-stream.h:
4313 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
4314 Because of duplicated g_signal_connect for request-aux-sender signal,
4315 wrong stream pointer is passed to the signal handler.
4316 Instead of passing each stream, pass stream array and get the relevant stream.
4317 https://bugzilla.gnome.org/show_bug.cgi?id=747839
4319 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
4323 Update autogen.sh to latest version from common
4324 Fixes build after aclocal_check etc. helpers have been removed.
4326 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
4329 Automatic update of common submodule
4330 From bc76a8b to c8fb372
4332 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
4334 * gst/rtsp-server/rtsp-stream.c:
4335 rtsp-stream: Limit the queues to 1 buffer
4336 We only need them to be able to pre-roll, queueing up more data here
4337 is only going to harm latency and memory usage.
4339 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
4341 * gst/rtsp-server/rtsp-stream.c:
4342 rtsp-stream: Update comment and ASCII art to the latest code
4343 We have a queue in front of the udpsink too to prevent the pipeline from
4346 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4348 * gst/rtsp-server/rtsp-stream.c:
4349 rtsp-media: Properly return first rtptime
4350 Instead we where returning first GstBuffer timestamp. This would result
4351 in clock skew and unwanted behaviour in RTSP playback.
4352 https://bugzilla.gnome.org/show_bug.cgi?id=746479
4354 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4356 * gst/rtsp-server/rtsp-stream.c:
4357 rtsp-stream: Don't leave buffer mapped
4358 If the seq is NULL, the RTP buffer was left mapped. We should always
4361 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
4366 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
4368 * gst/rtsp-server/rtsp-media-factory.c:
4369 * tests/check/gst/client.c:
4370 Fix double semicolons
4372 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
4374 * gst/rtsp-server/rtsp-stream.c:
4375 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
4376 This gives more accurate values than asking the payloader. There might be
4377 queueing happening between the payloader and the sink.
4378 https://bugzilla.gnome.org/show_bug.cgi?id=745704
4380 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
4382 * gst/rtsp-server/rtsp-media.c:
4383 rtsp-media: Don't seek for PLAY if the position will not change
4384 https://bugzilla.gnome.org/show_bug.cgi?id=745704
4386 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
4388 * gst/rtsp-server/rtsp-media.c:
4389 rtsp-media: Don't include payload type in the caps for framesize
4390 When the sdp media attribute framesize are converted to caps
4391 the <payload> should not be included.
4392 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
4393 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
4395 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
4397 * gst/rtsp-server/rtsp-sdp.c:
4398 rtsp-sdp: add payload type to the sdp framesize attribute
4399 The sdp framesize attribute is desribed in RFC6064. It is specified
4400 for payloading of H263 and has the following form
4401 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
4402 should be added to the caps in a payloader and the <payload type> should
4403 be added by the rtsp-server.
4404 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
4406 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
4408 * examples/test-uri.c:
4409 examples: test-uri: fix tainted variable
4410 Insignificant but this keeps Coverity happy.
4413 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
4415 * examples/.gitignore:
4416 * examples/Makefile.am:
4417 * examples/test-netclock-client.c:
4418 * examples/test-netclock.c:
4419 examples: Add a simple example of network synch for live streams.
4420 An example server and client that works for synchronising live streams
4421 only - as it can't support pause/play.
4423 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
4425 * gst/rtsp-server/rtsp-media-factory.c:
4426 * gst/rtsp-server/rtsp-media-factory.h:
4427 rtsp-media-factory: Add functions to set/get the media gtype
4428 Allow specifying the GType of a GstRtspMedia subclass to create
4429 as a simpler way to get the factory to create a custom
4430 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
4432 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
4434 * gst/rtsp-server/rtsp-media.c:
4435 rtsp-media: fix double unlock in _get_buffer_size()
4436 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
4437 because of double g_mutex_unlock () usage.
4438 https://bugzilla.gnome.org/show_bug.cgi?id=745434
4440 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
4442 * gst/rtsp-server/rtsp-session-pool.c:
4443 * gst/rtsp-server/rtsp-session.c:
4444 * gst/rtsp-server/rtsp-session.h:
4445 rtsp-session: Use monotonic time for RTSP session timeout
4446 Changed RTSP session timeout handling to monotonic time
4447 and deprecating the API for current system time.
4448 This fixes timeouts when the system time changes.
4449 https://bugzilla.gnome.org/show_bug.cgi?id=743346
4451 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
4453 * gst/rtsp-server/rtsp-client.c:
4454 * gst/rtsp-server/rtsp-media.c:
4455 rtsp-client: Only error out in PLAY if seeking actually failed
4456 If the media was just not seekable, we continue from whatever position we are
4457 and let the client decide if that is what is wanted or not.
4458 Only if the actual seek failed, we can't really recover and should error out.
4460 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
4462 * gst/rtsp-server/rtsp-stream.c:
4463 rtsp-stream: Add necessary queues between tee and multiudpsink
4464 https://bugzilla.gnome.org/show_bug.cgi?id=744379
4466 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
4468 * gst/rtsp-server/rtsp-client.c:
4469 * gst/rtsp-server/rtsp-media.c:
4470 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
4471 Instead error out properly the same way as if the SEEKING query already
4474 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
4476 * gst/rtsp-server/rtsp-stream.h:
4477 rtsp-stream: minor code formatting fix
4479 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
4481 * gst/rtsp-server/rtsp-media.c:
4482 rtsp-media: fix logic for collect_streams
4483 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
4484 all streams it knows if it got any, and can check if the transport mode is OK.
4487 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
4489 * gst/rtsp-server/rtsp-media.c:
4490 rtsp-media: Don't set the transport mode based on what elements we find
4491 Just print a warning if the one that was set before disagrees with what
4492 elements we found. It must already be set to something before as this
4493 function is called after we received the SDP from ANNOUNCE in RECORD mode,
4494 and we would reject ANNOUNCE if the RECORD flag was not set.
4496 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
4498 * tests/check/gst/rtspserver.c:
4499 tests: rtspserver: rename shadowed variable
4500 We have two different 'sink' variables here,
4501 rename one of them for clarity.
4503 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
4505 * gst/rtsp-server/rtsp-client.c:
4506 rtsp-client: fix awkward if clause
4508 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
4510 * examples/test-uri.c:
4511 examples: test-uri: improve uri argument handling and accept file names
4512 Print an error if the argument passed is not a URI and can't
4513 be converted into one, or no arguments have been provided.
4515 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
4517 * examples/test-uri.c:
4518 examples: test-uri: don't remove mount point after 10 seconds
4519 It's very irritating when trying to test stuff repeatedly
4520 and serves no real purpose other than showing that it can
4523 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
4525 * examples/.gitignore:
4526 examples: add new test-record to .gitignore
4528 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
4530 * examples/test-record.c:
4531 * gst/rtsp-server/rtsp-client.c:
4532 * gst/rtsp-server/rtsp-media-factory.c:
4533 * gst/rtsp-server/rtsp-media-factory.h:
4534 * gst/rtsp-server/rtsp-media.c:
4535 * gst/rtsp-server/rtsp-media.h:
4536 * tests/check/gst/rtspserver.c:
4537 rtsp-media: Use flags to distinguish between PLAY and RECORD media
4539 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
4541 * examples/test-record.c:
4542 test-record: Set latency for playback-style example to 2s instead of 200ms
4544 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
4546 * tests/check/gst/rtspserver.c:
4547 tests: add some unit tests for ANNOUNCE and RECORD
4548 https://bugzilla.gnome.org/show_bug.cgi?id=743175
4550 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
4552 * gst/rtsp-server/rtsp-client.c:
4553 rtsp-client: fix a couple of leaks in handle_announce
4555 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
4557 * gst/rtsp-server/rtsp-media-factory.c:
4558 * gst/rtsp-server/rtsp-media-factory.h:
4559 * gst/rtsp-server/rtsp-media.c:
4560 * gst/rtsp-server/rtsp-media.h:
4561 rtsp-media: Expose latency setting for setting the rtpbin latency
4563 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
4565 * examples/test-record.c:
4566 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
4568 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
4570 * gst/rtsp-server/rtsp-stream.c:
4571 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
4573 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
4575 * examples/Makefile.am:
4576 * examples/test-record.c:
4577 * gst/rtsp-server/rtsp-client.c:
4578 * gst/rtsp-server/rtsp-client.h:
4579 * gst/rtsp-server/rtsp-media-factory.c:
4580 * gst/rtsp-server/rtsp-media-factory.h:
4581 * gst/rtsp-server/rtsp-media.c:
4582 * gst/rtsp-server/rtsp-media.h:
4583 * gst/rtsp-server/rtsp-session-media.c:
4584 * gst/rtsp-server/rtsp-stream.c:
4585 * gst/rtsp-server/rtsp-stream.h:
4586 Add initial support for RECORD
4587 We currently only support media that is RECORD or PLAY only, not both at once.
4588 https://bugzilla.gnome.org/show_bug.cgi?id=743175
4590 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
4592 * gst/rtsp-server/rtsp-stream.c:
4593 rtsp-stream: RTCP and RTP transport cache cookies seperated
4594 RTCP packets were not sent because the same tr_cache_cookie was used for
4595 both RTP and RTCP. So only one of the tr_cache lists were populated
4596 depending on which one was sent first. If the tr_cache list is not
4597 populated then no packets can be sent. Most often this happened to be
4598 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
4599 resulted in both the tr_cache_lists to be populated regardless of which
4601 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
4603 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
4605 * gst/rtsp-server/rtsp-stream.c:
4606 rtsp-stream: fix false compiler warning
4607 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
4609 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
4611 * gst/rtsp-server/rtsp-client.c:
4612 rtsp-client: log interleaved data received
4614 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
4616 * gst/rtsp-server/rtsp-client.c:
4617 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
4619 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
4621 * gst/rtsp-server/rtsp-client.c:
4622 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
4624 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
4626 * gst/rtsp-server/rtsp-client.c:
4627 rtsp-client: Use a random session ID in the SDP
4628 RFC4566 Section 5.2 says that it should make the username, session id,
4629 nettype, addrtype and unicast address tuple globally unique. Always using
4630 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
4631 Instead let's create a 64 bit random number, which at least brings us
4632 closer to the goal of global uniqueness.
4633 https://tools.ietf.org/html/rfc4566#section-5.2
4635 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
4637 * examples/test-launch.c:
4638 * examples/test-mp4.c:
4639 * examples/test-ogg.c:
4640 * examples/test-uri.c:
4641 examples: Don't call gst_init() and gst_get_option_group()
4642 The latter calls the former at the appropriate time.
4644 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
4646 * gst/rtsp-server/rtsp-client.c:
4647 rtsp-client: Drop trailing \0 of RTSP DATA messages
4648 We add a trailing \0 in GstRTSPConnection to make parsing of
4649 string message bodies easier (e.g. the SDP from DESCRIBE) but
4650 for actual data this means we have to drop it or otherwise
4651 create invalid data.
4653 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
4655 * gst/rtsp-server/rtsp-stream.c:
4656 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
4657 Fixes crash when two threads access handle_new_sample() at the same
4658 time, one for RTP, one for RTCP.
4659 Otherwise, when iterating over the transports cache, it might be modified by
4660 another thread at the same time if the transports cookie has changed.
4661 https://bugzilla.gnome.org/show_bug.cgi?id=742954
4663 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
4665 * gst/rtsp-server/rtsp-stream.c:
4666 rtsp-stream: Set format=TIME on our app sources for TCP
4668 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
4670 * gst/rtsp-server/rtsp-session-pool.c:
4671 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
4672 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
4673 RFC 2326 states that session IDs may consist of alphanumeric as well as
4674 the safe characters $-_.+ -- N.B. the percent character is not allowed.
4675 Previously the session ID was URI-escaped, this meant that any character
4676 which was not alphanumeric or any of the characters +-._~ would be
4677 percent encoded. While the RFC (surprisingly) mentions that linear white
4678 space in session IDs should be URI-escaped, it does not say anything
4679 about other characters. Moreover no white space is allowed in the
4680 session ID. Finally the percent character which is the result of
4681 URI-escaping is not allowed in a session ID.
4682 So there is no reason to do any URI-escaping, and now it is removed.
4683 https://bugzilla.gnome.org/show_bug.cgi?id=742869
4685 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
4688 Automatic update of common submodule
4689 From f2c6b95 to bc76a8b
4691 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
4694 Fix 'make check' from top-level directory
4696 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
4698 * examples/test-launch.c:
4699 * examples/test-mp4.c:
4700 * examples/test-ogg.c:
4701 * examples/test-uri.c:
4702 examples: Add command-line parsing and take a 'port' argument
4703 This allows users to run multiple servers on different ports for testing.
4704 Only done for examples that actually take arguments and hence are capable of
4705 outputting different streams for each instance on each port.
4706 https://bugzilla.gnome.org/show_bug.cgi?id=742115
4708 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
4710 * gst/rtsp-server/rtsp-client.c:
4711 * gst/rtsp-server/rtsp-client.h:
4712 rtsp-client: Add a send_message default signal handler
4713 This allows subclasses to easily hook into the response sending
4714 mechanism without doing everything from a signal, which seems
4715 awkward from subclasses.
4717 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
4720 Automatic update of common submodule
4721 From ef1ffdc to f2c6b95
4723 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4727 configure: add --disable-examples switch
4728 https://bugzilla.gnome.org/show_bug.cgi?id=741678
4730 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
4732 * examples/.gitignore:
4733 * examples/Makefile.am:
4734 * examples/test-video-rtx.c:
4735 examples: add a retransmisison example implementing RFC4588
4736 Currently only SSRC-multiplexed rtx streams are supported
4738 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
4740 * gst/rtsp-server/rtsp-stream.c:
4741 rtsp-stream: Fix some minor memory leaks
4743 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
4745 * gst/rtsp-server/rtsp-media.c:
4746 rtsp-media: Some minor cleanup
4748 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
4750 * gst/rtsp-server/rtsp-stream.c:
4751 rtsp-stream: Fix compiler warnings
4752 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
4753 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
4755 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
4756 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
4759 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
4761 * docs/libs/gst-rtsp-server-sections.txt:
4762 * gst/rtsp-server/rtsp-media-factory.c:
4763 * gst/rtsp-server/rtsp-media-factory.h:
4764 * gst/rtsp-server/rtsp-media.c:
4765 * gst/rtsp-server/rtsp-media.h:
4766 * gst/rtsp-server/rtsp-sdp.c:
4767 * gst/rtsp-server/rtsp-stream.c:
4768 * gst/rtsp-server/rtsp-stream.h:
4769 media: implement ssrc-multiplexed retransmission support
4770 based off RFC 4588 and the server-rtpaux example in -good
4772 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
4774 * gst/rtsp-server/rtsp-client.c:
4775 * gst/rtsp-server/rtsp-stream-transport.c:
4776 * gst/rtsp-server/rtsp-stream.c:
4777 rtsp: Ref transports in hash table.
4778 Also ref streams for transports.
4779 This solves a crash when reciving a rtcp after teardown but before
4781 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
4783 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
4786 Automatic update of common submodule
4787 From 7bb2bce to ef1ffdc
4789 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
4791 * gst/rtsp-server/rtsp-client.c:
4792 client: refactor cleanup of cached media
4794 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
4796 * tests/check/gst/client.c:
4798 The session leak is now fixed, lets remove those FIXME comments.
4800 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
4802 * tests/check/gst/rtspserver.c:
4803 tests: Test to setup two sessions on one connection
4804 https://bugzilla.gnome.org/show_bug.cgi?id=739112
4806 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
4808 * tests/check/gst/rtspserver.c:
4809 tests: Test setup with tcp transport
4810 https://bugzilla.gnome.org/show_bug.cgi?id=739112
4812 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
4814 * gst/rtsp-server/rtsp-client.c:
4815 client: Configure transport after creating session media
4816 The default implementation of configure_client_transport() in
4817 rtsp-client uses the session media when it chooses channels for
4818 interleaved traffic.
4819 https://bugzilla.gnome.org/show_bug.cgi?id=739112
4821 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
4823 * gst/rtsp-server/rtsp-client.c:
4824 * gst/rtsp-server/rtsp-session-media.c:
4825 client: Stop caching media in client when doing setup
4826 If the media has been managed by a session media, it should not be
4827 cached in the client any longer. The GstRTSPSessionMedia object is now
4828 responsible for unpreparing the GstRTSPMedia object using
4829 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
4831 https://bugzilla.gnome.org/show_bug.cgi?id=739112
4833 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4835 * gst/rtsp-server/rtsp-stream.c:
4836 rtsp-stream: unref srtp decoder when leaving bin
4837 https://bugzilla.gnome.org/show_bug.cgi?id=739481
4839 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4841 * gst/rtsp-server/rtsp-client.c:
4842 rtsp-client: mikey memory leaks
4843 https://bugzilla.gnome.org/show_bug.cgi?id=739383
4845 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
4848 Automatic update of common submodule
4849 From 84d06cd to 7bb2bce
4851 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
4854 Parallelise 'make check-valgrind'
4856 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
4859 Automatic update of common submodule
4860 From a8c8939 to 84d06cd
4862 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
4865 Automatic update of common submodule
4866 From 36388a1 to a8c8939
4868 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
4870 * gst/rtsp-server/rtsp-media.c:
4871 rtsp-media: deactivate media when shutting down from paused
4872 This was only done when going directly from playing.
4873 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
4875 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4877 * gst/rtsp-server/rtsp-client.c:
4878 * gst/rtsp-server/rtsp-context.h:
4879 rtsp-client: add stream transport to context
4880 We add the stream transport to the context so we can get the configured
4881 client stream transport in the setup request signal.
4882 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
4884 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4886 * gst/rtsp-server/rtsp-stream.c:
4887 stream: release lock even not all transports have been removed
4888 We don't want to keep the lock even we return FALSE because not all the
4889 transports have been removed. This could lead into a deadlock.
4890 https://bugzilla.gnome.org/show_bug.cgi?id=737797
4892 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
4894 * gst/rtsp-server/rtsp-sdp.c:
4895 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
4896 These were renamed in GstRTPBasePayload in 1.0
4898 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4900 * gst/rtsp-server/rtsp-client.c:
4901 client: set session media to NULL without the lock
4902 We need to set session medias to NULL without the client lock otherwise
4903 we can end up in a deadlock if another thread is waiting for the lock
4904 and media unprepare is also waiting for that thread to end.
4905 https://bugzilla.gnome.org/show_bug.cgi?id=737690
4907 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
4909 * gst/rtsp-server/rtsp-media.c:
4910 rtsp-media: Set state to UNPREPARING in all cases
4912 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
4914 * gst/rtsp-server/rtsp-media.c:
4915 media: set state to unpreparing when unprepare is initiated
4916 https://bugzilla.gnome.org/show_bug.cgi?id=737675
4918 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
4920 * gst/rtsp-server/rtsp-client.c:
4921 rtsp-client: Remove backlog limit while processings requests
4922 If the backlog limit is kept two cases of deadlocks may be
4923 encountered when streaming over TCP. Without the backlog
4924 limit this deadlocks can not happen, at the expence of
4926 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
4928 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
4930 * gst/rtsp-server/rtsp-client.c:
4931 rtsp-client: do not free main context before rtsp watch
4932 https://bugzilla.gnome.org/show_bug.cgi?id=737110
4934 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
4936 * tests/check/gst/rtspserver.c:
4937 tests: Extend unit test timeout to accomodate for valgrind
4938 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
4940 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
4942 * gst/rtsp-server/rtsp-client.c:
4943 * gst/rtsp-server/rtsp-session.c:
4944 * gst/rtsp-server/rtsp-stream-transport.c:
4945 rtsp-*: Treat sending packets to clients as keepalive
4946 As long as gst-rtsp-server can successfully send RTP/RTCP data to
4947 clients then the client must be reading. This change makes the server
4948 timeout the connection if the client stops reading.
4949 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
4951 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
4953 * gst/rtsp-server/rtsp-client.c:
4954 rtsp-client: Allow backlog to grow while expiring session
4955 Allow the send backlog in the RTSP watch to grow to unlimited size while
4956 attempting to bring the media pipeline to NULL due to a session
4957 expiring. Without this change the appsink element cannot change state
4958 because it is blocked while rendering data in the new_sample callback.
4959 This callback will block until it has successfully put the data into the
4960 send backlog. There is a chance that the send backlog is full at this
4961 point which means that the callback may block for a long time, possibly
4962 forever. Therefore the media pipeline may also be prevented from
4963 changing state for a long time.
4964 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
4966 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
4968 * gst/rtsp-server/rtsp-client.c:
4969 rtsp-client: Make old compilers happy
4970 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
4971 Just in case that guint8 doesn't fit in a pointer. Just in case ...
4973 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
4975 * gst/rtsp-server/rtsp-client.c:
4976 client: raise the backlog limits before pausing
4977 We need to raise the backlog limits before pausing the pipeline or else
4978 the appsink might be blocking in the render method in wait_backlog() and
4979 we would deadlock waiting for paused.
4980 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
4982 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
4984 * gst/rtsp-server/rtsp-client.c:
4985 client: make define for the WATCH_BACKLOG
4986 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
4988 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
4990 * gst/rtsp-server/rtsp-client.c:
4991 client: simplify session transport handling
4992 link/unlink of the transport in a session was done to keep track of all
4993 TCP transports and to send RTP/RTCP data to the streams. We can simplify
4994 that by putting all the TCP transports in a hashtable indexed with the
4996 We also don't need to link/unlink the transports when we pause/resume
4997 the streams. The same effect is already achieved when we pause/play the
4998 media. Indeed, when we pause the media, the transport is removed from
4999 the media and the callbacks will not be called anymore.
5000 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
5002 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
5004 * gst/rtsp-server/rtsp-stream-transport.c:
5005 * gst/rtsp-server/rtsp-stream-transport.h:
5006 stream-transport: make method to handle received data
5007 Make a method to handle the data received on a channel. It sends the
5008 data to the stream of the transport on the RTP or RTCP pads based on
5011 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
5013 * examples/test-mp4.c:
5014 test: add example of dumping RTCP reports
5016 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
5018 * gst/rtsp-server/rtsp-media.c:
5019 * gst/rtsp-server/rtsp-stream.c:
5020 * gst/rtsp-server/rtsp-stream.h:
5021 rtsp-media: Make sure that sequence numbers are monotonic after pause
5022 The sequence number is not monotonic for RTP packets after pause. The
5023 reason is basepayloader generates a randon sequence number when the
5024 pipeline goes from ready to pause. With this fix generation of sequence
5025 number will be monotonic when going from pause to play request.
5026 https://bugzilla.gnome.org/show_bug.cgi?id=736017
5028 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
5030 * gst/rtsp-server/rtsp-client.c:
5031 rtsp-client: Protect saved clients watch with a mutex
5032 Fixes a crash when close() is called while merging clients
5033 in handle_tunnel(). In that case close() would destroy the
5034 watch while it is still being used in handle_tunnel().
5035 https://bugzilla.gnome.org/show_bug.cgi?id=735570
5037 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
5039 * gst/rtsp-server/rtsp-stream.c:
5040 rtsp-stream: Remove the multicast group udp sources when removing from the bin
5042 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
5044 * gst/rtsp-server/rtsp-media.c:
5045 * gst/rtsp-server/rtsp-stream.c:
5046 * gst/rtsp-server/rtsp-stream.h:
5047 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
5048 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
5049 seeking and will always continue counting the time. This leads to
5050 the NPT after a backwards seek to be something completely different
5051 to the actual seek position.
5052 https://bugzilla.gnome.org/show_bug.cgi?id=732644
5054 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
5056 * examples/test-appsrc.c:
5057 examples: fix another reference leak
5058 gst_rtsp_media_get_element() returns a new ref.
5060 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5062 * examples/test-appsrc.c:
5063 examples: unref element after usage
5064 gst_bin_get_by_name_recurse_up() returns an element
5065 reference that must be unreffed after usage.
5066 https://bugzilla.gnome.org/show_bug.cgi?id=734546
5068 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
5070 * gst/rtsp-server/rtsp-media.c:
5071 signals: Fix copy-pasto in target-state signal offset
5073 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
5077 Makefile: Add usage of build-checks step
5078 Allows building checks without running them
5080 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
5082 * gst/rtsp-server/rtsp-stream.c:
5083 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
5084 When a UDP multicast transport is used it is expected that the server listens
5085 for RTP and RTCP packets on the multicast group with the corresponding port.
5086 Without this we will never get RTCP packets from clients in multicast mode.
5087 https://bugzilla.gnome.org/show_bug.cgi?id=732238
5089 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
5094 === release 1.4.0 ===
5096 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
5102 * gst-rtsp-server.doap:
5105 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
5107 * gst/rtsp-server/rtsp-media.h:
5108 media: correct misspelled words in description
5109 https://bugzilla.gnome.org/show_bug.cgi?id=733244
5111 === release 1.3.91 ===
5113 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
5119 * gst-rtsp-server.doap:
5122 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
5124 * docs/libs/gst-rtsp-server-sections.txt:
5127 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
5129 * gst/rtsp-server/rtsp-server.c:
5130 server: implement client REMOVE filter
5132 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
5134 * gst/rtsp-server/rtsp-client.c:
5135 * gst/rtsp-server/rtsp-client.h:
5136 client: expose _close() method
5137 Expose a previously internal close method to close the client
5140 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
5142 * gst/rtsp-server/rtsp-session-pool.c:
5143 session-pool: signal session-removed outside of the lock
5144 Release the lock before emiting the session-removed signal.
5146 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
5148 * gst/rtsp-server/rtsp-client.c:
5149 * gst/rtsp-server/rtsp-server.c:
5150 * gst/rtsp-server/rtsp-session-pool.c:
5151 * gst/rtsp-server/rtsp-session.c:
5152 * gst/rtsp-server/rtsp-stream.c:
5153 filter: Release lock in filter functions
5154 Release the object lock before calling the filter functions. We need to
5155 keep a cookie to detect when the list changed during the filter
5156 callback. We also keep a hashtable to make sure we only call the filter
5157 function once for each object in case of concurrent modification.
5158 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
5160 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
5162 * gst/rtsp-server/rtsp-client.c:
5163 client: check if watch is set in handle_teardown()
5164 The unit tests run without a watch
5166 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
5168 * tests/check/gst/client.c:
5169 client tests: send teardown to cleanup session
5171 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
5173 * tests/check/gst/rtspserver.c:
5174 server tests: send teardown to cleanup session
5176 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
5178 * gst/rtsp-server/rtsp-client.c:
5179 client: keep ref to client for the session removed handler
5180 This extra ref will be dropped when all client sessions have been
5181 removed. A session is removed when a client sends teardown, closes its
5182 endpoint of the TCP connection or the sessions expires.
5183 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
5185 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
5187 * gst/rtsp-server/rtsp-client.c:
5188 * gst/rtsp-server/rtsp-session.c:
5189 * tests/check/gst/client.c:
5190 client: manage media in session as a last step
5191 Once we manage a media in a session, we can't unmanage it anymore
5192 without destroying it. Therefore, first check everything before we
5193 manage the media, otherwise if something is wrong we have no way to
5195 If we created a new session and something went wrong, remove the session
5196 again. Fixes a leak in the unit test.
5198 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
5200 * examples/test-mp4.c:
5201 * examples/test-ogg.c:
5202 examples: print 'stream ready at url' for mp4 and ogg example
5204 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
5206 * gst/rtsp-server/rtsp-client.c:
5207 * gst/rtsp-server/rtsp-sdp.c:
5208 rtsp: fix for MIKEY api change
5210 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
5212 * gst/rtsp-server/rtsp-client.c:
5213 client: free watch context only once
5214 The watch context is freed when the source is destroyed. Avoids
5215 a CRITICAL when we try to unref the context twice.
5217 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
5219 * gst/rtsp-server/rtsp-client.c:
5222 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
5224 * gst/rtsp-server/rtsp-client.c:
5225 client: protect sessions with lock
5226 Protect the list of sessions with the lock.
5227 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
5229 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
5231 * gst/rtsp-server/rtsp-client.c:
5232 Client: keep a ref to the session
5233 Don't just keep a weak ref to the session objects but use a hard ref. We
5234 will be notified when a session is removed from the pool (expired) with
5235 the new session-removed signal.
5236 Don't automatically close the RTSP connection when all the sessions of
5237 a client are removed, a client can continue to operate and it can create
5238 a new session if it wants. If you want to remove the client from the
5239 server, you have to use gst_rtsp_server_client_filter() now.
5240 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
5241 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
5243 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
5245 * gst/rtsp-server/rtsp-session-pool.c:
5246 * gst/rtsp-server/rtsp-session-pool.h:
5247 session-pool: add session-removed signal
5248 Add a signal to be notified when a session is removed from the pool.
5250 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
5252 * gst/rtsp-server/Makefile.am:
5253 * gst/rtsp-server/rtsp-server.h:
5254 Make rtsp-server.h a single-include header, use it for G-I
5255 https://bugzilla.gnome.org/show_bug.cgi?id=732411
5257 === release 1.3.90 ===
5259 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
5265 * gst-rtsp-server.doap:
5268 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
5270 * gst/rtsp-server/rtsp-stream.c:
5271 stream: crypto can be NULL
5273 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
5275 * gst/rtsp-server/rtsp-client.c:
5276 * gst/rtsp-server/rtsp-media.c:
5277 * gst/rtsp-server/rtsp-mount-points.c:
5278 introspection: add missing allow-none annotations
5279 https://bugzilla.gnome.org/show_bug.cgi?id=730952
5281 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
5283 * gst/rtsp-server/rtsp-address-pool.c:
5284 * gst/rtsp-server/rtsp-media.c:
5285 * gst/rtsp-server/rtsp-session-media.c:
5286 * gst/rtsp-server/rtsp-session-pool.c:
5287 * gst/rtsp-server/rtsp-stream-transport.c:
5288 * gst/rtsp-server/rtsp-stream.c:
5289 * gst/rtsp-server/rtsp-token.c:
5290 introspection: add (nullable) annotations to return values
5291 https://bugzilla.gnome.org/show_bug.cgi?id=730952
5293 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
5295 * gst/rtsp-server/rtsp-client.c:
5296 * gst/rtsp-server/rtsp-stream.c:
5297 gi: improve annotations
5298 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
5300 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
5302 * gst/rtsp-server/rtsp-client.c:
5303 * gst/rtsp-server/rtsp-media-factory.c:
5304 * gst/rtsp-server/rtsp-media.c:
5305 * gst/rtsp-server/rtsp-server.c:
5306 signals: use generic marshal function
5307 Use the generic C marshal function.
5308 Use more explicit type instead of G_TYPE_POINTER
5310 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
5312 * gst/rtsp-server/rtsp-context.h:
5313 context: add type macro
5315 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
5317 * gst/rtsp-server/rtsp-client.c:
5318 * gst/rtsp-server/rtsp-sdp.c:
5319 * gst/rtsp-server/rtsp-sdp.h:
5320 sdp: hide key length defines
5321 They don't have a namespace.
5323 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
5328 === release 1.3.3 ===
5330 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
5336 * gst-rtsp-server.doap:
5339 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
5341 * gst/rtsp-server/rtsp-client.c:
5342 * gst/rtsp-server/rtsp-sdp.c:
5343 * gst/rtsp-server/rtsp-sdp.h:
5344 mikey: add different key length parameters
5345 Add encryption and authentication key length parameters to MIKEY. For
5346 the encoders, the key lengths are obtained from the cipher and auth
5347 algorithms set in the caps. For the decoders, they are obtained while
5348 parsing the key management from the client.
5349 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
5351 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
5353 * tests/check/gst/stream.c:
5354 stream tests: Make sure we get right multicast address from stream
5355 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
5357 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
5359 * gst/rtsp-server/rtsp-client.c:
5360 client: ref the context until rtsp watch is alive
5361 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
5363 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
5365 * gst/rtsp-server/rtsp-client.c:
5366 client: Destroy the rtsp watch after connection close
5368 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
5370 * gst/rtsp-server/rtsp-media.c:
5371 media: fix confusing comment
5373 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
5375 * gst/rtsp-server/rtsp-session.c:
5376 rtsp-session: Timeout in header.
5377 Adding the possbilty to always have timout in header.
5378 This is configurabe with setting "timeout-always-visible".
5379 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
5381 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
5386 === release 1.3.2 ===
5388 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
5395 * gst-rtsp-server.doap:
5398 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
5401 Automatic update of common submodule
5402 From 211fa5f to 1f5d3c3
5404 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
5406 * gst/rtsp-server/rtsp-client.c:
5407 client: store TCP ports in transport
5408 Store the TCP ports in the transport when we are doing RTSP over TCP.
5409 This way, we can easily get to the ports from the transport.
5410 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
5412 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
5414 * gst/rtsp-server/rtsp-stream.c:
5415 stream: add signals for new RTP/RTCP encoders
5416 New signals to allow the user to configure the dynamically created
5418 https://bugzilla.gnome.org/show_bug.cgi?id=730228
5420 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
5422 * gst/rtsp-server/rtsp-media.c:
5423 * gst/rtsp-server/rtsp-media.h:
5424 media: Make suspend()/unsuspend() virtual
5425 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
5427 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
5429 * gst/rtsp-server/rtsp-client.c:
5430 client: fix send-message signal marshaller
5431 Use generic marshalling for the send-message signal. It has
5432 two POINTER arguments, not just one.
5433 https://bugzilla.gnome.org/show_bug.cgi?id=729900
5435 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
5437 * tests/check/gst/media.c:
5438 tests: add and remove pads only once
5439 In this test we simulate a dynamic pad by watching the caps event.
5440 Because of renegotiation in the base payloader now, this caps is sent
5441 multiple times but we can only deal with 1 invocation, use a variable to
5442 only 'add and remove' the pad once.
5444 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
5446 * tests/check/gst/rtspserver.c:
5447 tests: add unit test for correct handling of Require headers
5448 https://bugzilla.gnome.org/show_bug.cgi?id=729426
5450 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
5452 * gst/rtsp-server/rtsp-client.c:
5453 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
5454 Servers must handle Require headers and must report a failure
5455 if they don't handle any of the Required options, see RFC 2326,
5456 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
5457 https://bugzilla.gnome.org/show_bug.cgi?id=729426
5459 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
5464 === release 1.3.1 ===
5466 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
5472 * gst-rtsp-server.doap:
5475 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
5478 Automatic update of common submodule
5479 From bcb1518 to 211fa5f
5481 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
5486 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
5488 * tests/check/gst/sessionmedia.c:
5489 tests: fix memory leak in sessionmedia unit test
5491 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
5493 * gst/rtsp-server/rtsp-client.c:
5494 client: emit a signal before sending a message
5495 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
5497 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
5499 * gst/rtsp-server/rtsp-client.c:
5500 client: pass context to send_message
5501 Pass the current context to send_message, we will need it later.
5503 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
5505 * gst/rtsp-server/rtsp-client.c:
5506 client: fix typo in comment
5508 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
5510 * gst/rtsp-server/rtsp-media.c:
5511 media: Do not stop thread twice if default_prepare() fails
5513 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
5515 * gst/rtsp-server/rtsp-client.c:
5516 client: set the watch to flushing before going to NULL
5517 First set the watch to flushing so that we unblock any current and
5518 future attempt to send data on the watch, Then set the pipeline to
5520 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
5522 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
5524 * gst/rtsp-server/rtsp-session-pool.c:
5525 * tests/check/gst/sessionpool.c:
5526 rtsp-session-pool: Fixes annotation
5527 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
5528 in the sessionpool test.
5529 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
5531 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
5533 * gst/rtsp-server/rtsp-media.c:
5534 * gst/rtsp-server/rtsp-media.h:
5535 media: make media_prepare virtual
5536 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
5538 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
5540 * gst/rtsp-server/rtsp-media.c:
5541 * tests/check/gst/media.c:
5542 media: stop the thread in more error cases
5544 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
5546 * gst/rtsp-server/rtsp-media.c:
5547 * tests/check/gst/media.c:
5548 media: allow NULL as the thread
5549 Use the default context whan passing a NULL thread.
5551 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
5553 * gst/rtsp-server/rtsp-client.c:
5554 rtsp-client: indent cleanup
5555 Coverity was moaning about unreachable code, and I think it was just
5556 confused by { being before the label. We'll see if it pops up again.
5559 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
5561 * gst/rtsp-server/rtsp-client.c:
5562 * gst/rtsp-server/rtsp-media.c:
5563 client: Add drop-backlog property
5564 When we have too many messages queued for a client (currently hardcoded
5565 to 100) we overflow and drop the messages. Add a drop-backlog property
5566 to control this behaviour. Setting this property to FALSE will retry
5567 to send the messages to the client by waiting for more room in the
5569 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
5571 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
5573 * gst/rtsp-server/rtsp-client.c:
5574 client: support for POST before GET when setting up a tunnel
5576 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
5578 * gst/rtsp-server/rtsp-client.c:
5579 client: remove watch of the second client after http tunnel setup
5580 The second client will be freed after the HTTP tunnel has been set up.
5581 Make sure it's RTSP watch is never dispatched again.
5582 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
5584 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
5586 * gst/rtsp-server/rtsp-media.c:
5587 * tests/check/gst/media.c:
5588 media: Make media_prepare() fail if port allocation fails
5589 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
5591 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
5593 * tests/check/gst/media.c:
5594 media test: cleanup the thread pool in tests
5596 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
5598 * gst/rtsp-server/rtsp-media.c:
5599 * tests/check/gst/media.c:
5600 rtsp-media: Unblock blocked streams in unprepare
5601 The streams will be blocked when a live media is prepared.
5602 The streams should be unblocked in gst_rtsp_media_unprepare.
5603 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
5605 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
5607 * gst/rtsp-server/rtsp-media.c:
5608 media: release the state lock when going to NULL
5609 Set our state to UNPREPARING and release the state-lock before
5610 setting the pipeline to the NULL state. This way, any pad-added
5611 callback will be able to take the state-lock and check that we are now
5612 unpreparing instead of deadlocking.
5613 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
5615 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
5617 * gst/rtsp-server/rtsp-media.c:
5618 media: protect status with lock
5619 Make sure we only update the status with the lock.
5621 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
5623 * gst/rtsp-server/rtsp-client.c:
5624 * gst/rtsp-server/rtsp-sdp.c:
5625 rtsp: update for MIKEY API changes
5627 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
5629 * gst/rtsp-server/rtsp-client.c:
5630 client: parse the mikey response from the client
5631 Parse the mikey response from the client and update the policy for
5634 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
5636 * gst/rtsp-server/rtsp-stream.c:
5637 * gst/rtsp-server/rtsp-stream.h:
5638 stream: add method to set crypto info
5639 Make a method to configure the crypto information of a stream.
5640 Set udpsrc in READY instead of PAUSED so that we can configure caps
5643 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
5645 * gst/rtsp-server/rtsp-client.c:
5646 client: cleanup error paths
5648 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
5650 * gst/rtsp-server/rtsp-media.c:
5653 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
5655 * examples/test-video.c:
5656 test: enable SRTP only on RTSPS
5657 We only want to enable SRTP when doing rtsp over TLS so that we can
5658 exchange the keys in a secure way.
5660 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
5662 * examples/test-video.c:
5663 test: print an error on failure
5665 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
5668 * examples/test-video.c:
5669 * gst/rtsp-server/rtsp-sdp.c:
5670 * gst/rtsp-server/rtsp-stream.c:
5671 * tests/check/Makefile.am:
5672 stream: add SRTP support
5673 Install srtp encoder and decoder elements in rtpbin
5676 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5678 * tests/check/Makefile.am:
5679 * tests/check/gst/sessionpool.c:
5680 tests: Add unit tests for sessionpool
5681 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
5683 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5685 * tests/check/gst/threadpool.c:
5686 tests: Improve code coverage of rtsp-threadpool tests
5687 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
5689 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5691 * tests/check/gst/sessionmedia.c:
5692 tests: Improve code coverage for rtsp-session-media
5693 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
5695 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5697 gobject-introspection: Add annotations to support language bindings
5698 In addition a few cosmetic changes:
5699 * Adjust the order of arguments
5700 * Fix typo: occured -> occurred
5701 * Fix indentation after Return:-clauses
5702 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
5704 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5706 * gst/rtsp-server/rtsp-stream.c:
5707 rtsp-stream: Don't mix IPv4 and IPv6 addresses
5708 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
5710 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
5712 * gst/rtsp-server/rtsp-stream.c:
5713 stream: take caps after the session manager
5714 Take the caps for the SDP after they leave the rtpbin so that we can
5715 also get the properties added by rtpbin elements.
5717 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
5719 * gst/rtsp-server/rtsp-stream.c:
5720 stream: release lock while pushing out packets
5721 Keep a cache of the transports and use this to iterate the transport
5722 while pushing packets. This allows us to release the lock early.
5723 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
5725 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
5727 * gst/rtsp-server/rtsp-client.c:
5728 * gst/rtsp-server/rtsp-client.h:
5729 rtsp-client: vmethod for modifying tunnel GET response
5730 Add a vmethod tunnel_http_response where the response to the HTTP GET
5731 for tunneled connections can be modified.
5732 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
5734 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
5736 * gst/rtsp-server/rtsp-sdp.c:
5737 sdp: make 1 media line per profile
5738 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
5739 line in the SDP for each profile. The client is then supposed to pick
5740 one of the profiles in the SETUP request. Because the m= lines have the
5741 same pt, the client also knows that only 1 option is possible.
5743 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
5745 * gst/rtsp-server/rtsp-media-factory.c:
5746 * gst/rtsp-server/rtsp-media-factory.h:
5747 * gst/rtsp-server/rtsp-media.c:
5748 factory: add profile property and pass to media and streams
5750 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
5752 * examples/test-multicast.c:
5753 * gst/rtsp-server/rtsp-sdp.c:
5754 sdp: pass multicast connection for multicast-only stream
5755 Pass the multicast address of the stream in the connection info in the
5756 SDP so that clients try a multicast connection first.
5757 Only allow multicast connections in the test-multicast example. Also
5758 increase the TTL a little.
5760 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5763 .gitignore: Ignore gcov intermediate files
5764 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
5766 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
5768 * gst/rtsp-server/rtsp-stream.c:
5769 stream: release some locks in error cases
5771 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5773 docs: Enable and fix gtk-doc warnings
5774 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
5775 * addresspool/mediafactory: Add missing annotation colon
5776 * stream: Annotate return value
5777 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
5779 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
5782 Automatic update of common submodule
5783 From fe1672e to bcb1518
5785 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
5788 Automatic update of common submodule
5789 From 1a07da9 to fe1672e
5791 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
5793 * examples/Makefile.am:
5794 examples: use LDADD for libs instead of LDFLAGS
5796 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
5799 configure: make sure releases are in .doap file
5801 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
5803 * examples/test-cgroups.c:
5804 examples: test-cgroups: don't put code with side effects into g_assert()
5805 The g_assert() might get compiled out with the right
5806 compiler/preprocessor flags.
5808 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
5810 * examples/.gitignore:
5811 examples: add cgroup test binary to .gitignore
5813 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
5815 * examples/test-cgroups.c:
5816 examples: fix cgroup test build
5817 Fixes build failure caused by compiler warning:
5818 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
5820 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
5823 .gitignore: ignore temp files created in the course of 'make check'
5825 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
5827 * gst/rtsp-server/rtsp-media.c:
5828 rtsp-media: don't loose frames handling new PLAY request
5829 If client supplied a range check if the range specifies the start point.
5830 If not, then do an accurate seek to the current position. If a start
5831 point was specified do do a key unit seek to make sure the streaming
5832 starts with decodeable frames.
5833 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
5835 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
5837 * gst/rtsp-server/rtsp-media.c:
5838 Revert "media: only flush when setting a new start position"
5839 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
5840 We need to do the flush in all cases, demuxer block currently for
5843 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
5845 * gst/rtsp-server/rtsp-media.c:
5846 media: only flush when setting a new start position
5847 Only flush the pipeline when we change the start position with
5849 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
5851 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
5853 * gst/rtsp-server/rtsp-stream.c:
5854 stream: set ttl-mc before adding the socket
5855 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
5856 never be set on socket.
5857 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
5859 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
5861 * gst/rtsp-server/rtsp-media.c:
5862 media: stop thread if media is already prepared
5863 in gst_rtsp_media_prepare() the thread is not used if media is already
5864 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
5866 https://bugzilla.gnome.org/show_bug.cgi?id=724182
5868 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
5871 build: Ship gst-rtsp-server.doap file
5873 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
5875 * tests/check/gst/rtspserver.c:
5876 tests: Fix another compiler warning with gcc
5878 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
5880 * gst/rtsp-server/rtsp-client.c:
5881 * gst/rtsp-server/rtsp-mount-points.c:
5882 * gst/rtsp-server/rtsp-stream.c:
5883 * tests/check/gst/client.c:
5884 rtsp-server: Fix lots of compiler warnings with clang
5886 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
5889 * gst-rtsp-server.doap:
5890 * tests/Makefile.am:
5891 configure: Synchronise with the configure scripts of the other modules
5893 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
5896 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
5898 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
5900 * gst/rtsp-server/rtsp-media.c:
5901 * gst/rtsp-server/rtsp-stream.c:
5902 Revert "rtsp-server: support build against last stable release"
5903 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
5904 Let us require 1.2.3 now, which is going to be released in a few
5907 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
5909 * gst/rtsp-server/rtsp-session-media.c:
5910 * gst/rtsp-server/rtsp-stream-transport.c:
5911 session: improve RTP-Info
5912 Ignore streams that can't generate RTP-Info instead of failing.
5913 Don't return the empty string when all streams are unconfigured but
5914 return NULL so that we don't generate and empty RTP-Info header.
5915 Improve docs a little.
5917 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
5919 * gst/rtsp-server/rtsp-session-media.c:
5920 Don't free rtpinfo GString when it is NULL
5921 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
5923 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
5925 * gst/rtsp-server/rtsp-media.c:
5926 media: only set keyframe flag when modifying start
5927 Only set the keyframe flag when we modify the start position. The
5928 keyframe flag should probably be ignored when no change is requested but
5929 until we can claim this is all documented properly and all demuxer
5930 implement this, avoid setting the flag.
5931 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
5933 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
5935 * gst/rtsp-server/rtsp-thread-pool.c:
5936 thread-pool: Unref source after mainloop has quit to avoid races in GLib
5937 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
5939 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
5941 * gst/rtsp-server/rtsp-stream.c:
5942 stream: handle NULL seqnum and rtptime arguments
5944 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
5946 * gst/rtsp-server/rtsp-thread-pool.c:
5947 * tests/check/gst/threadpool.c:
5948 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
5949 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
5951 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
5953 * gst/rtsp-server/rtsp-stream.c:
5954 stream: add fallback for missing stats property
5955 Use a fallback when the payloader does not have a stats property
5956 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
5958 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
5961 Automatic update of common submodule
5962 From f7bc1c3 to 1a07da9
5964 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
5966 * gst/rtsp-server/rtsp-stream.c:
5967 stream: don't leak stats structure
5968 Don't leak the stats structure and deal with NULL stats.
5970 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
5972 * gst/rtsp-server/rtsp-stream.c:
5973 stream: Get rtpinfo properties atomically from payloader
5974 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
5976 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
5978 * gst/rtsp-server/rtsp-media.c:
5979 media: refactor state change functions and signals
5980 Make functions to set the target state and the pipeline state and emit
5981 the signals from those functions.
5983 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
5985 * gst/rtsp-server/rtsp-media.c:
5986 * gst/rtsp-server/rtsp-media.h:
5987 media: add signal to notify of pending state changes
5989 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
5991 * gst/rtsp-server/rtsp-media.c:
5992 * gst/rtsp-server/rtsp-stream.c:
5993 rtsp-server: support build against last stable release
5994 Until 1.2.3 is out with the new get_type function and we
5997 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
5999 * gst/rtsp-server/rtsp-stream.c:
6000 stream: fix compilation
6002 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
6004 * gst/rtsp-server/rtsp-media.c:
6005 * gst/rtsp-server/rtsp-media.h:
6006 * gst/rtsp-server/rtsp-stream.c:
6007 * gst/rtsp-server/rtsp-stream.h:
6008 stream: add property to configure profiles
6010 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
6012 * gst/rtsp-server/rtsp-client.c:
6013 client: let stream check supported transport
6014 Delegate the check if a transport is allowed to the stream.
6015 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
6017 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
6019 * gst/rtsp-server/rtsp-stream.c:
6020 * gst/rtsp-server/rtsp-stream.h:
6021 stream: add method to check supported transport
6022 Add a method to check if a transport is supported
6024 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
6027 configure.ac: Only check for gstreamer-check, not check
6028 We include check in gstreamer-check since quite some time now.
6030 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
6032 * gst/rtsp-server/rtsp-session-media.c:
6033 * gst/rtsp-server/rtsp-stream-transport.c:
6034 * gst/rtsp-server/rtsp-stream.c:
6035 * gst/rtsp-server/rtsp-stream.h:
6036 stream: return clock-rate from get_rtpinfo
6037 And use it to correct the rtptime to the requested start-time.
6038 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
6040 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
6042 * gst/rtsp-server/rtsp-session-media.c:
6043 * gst/rtsp-server/rtsp-stream-transport.c:
6044 * gst/rtsp-server/rtsp-stream-transport.h:
6045 session-media: calculate start-time
6047 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
6049 * gst/rtsp-server/rtsp-stream-transport.c:
6050 * gst/rtsp-server/rtsp-stream.c:
6051 * gst/rtsp-server/rtsp-stream.h:
6052 stream: also return the running-time
6053 Return the running-time in the rtpinfo as well.
6055 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
6057 * gst/rtsp-server/rtsp-client.c:
6058 * gst/rtsp-server/rtsp-session-media.c:
6059 * gst/rtsp-server/rtsp-session-media.h:
6060 * gst/rtsp-server/rtsp-stream-transport.c:
6061 * gst/rtsp-server/rtsp-stream-transport.h:
6062 session-media: let the session-media make the RTPInfo
6063 Add method to create the RTPInfo for a stream-transport.
6064 Add method to create the RTPInfo for all stream-transports in a
6066 Use the session-media RTPInfo code in client. This allows us to refactor
6067 another method to link the TCP callbacks.
6069 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
6071 mount-points: sort sequence before g_sequence_lookup
6072 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
6073 sort sequence if dirty, otherwise lookup will fail.
6074 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
6076 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
6079 configure: rename package from gst-rtsp to gst-rtsp-server
6080 To match git module name and avoid confusion with the
6081 rtsp lib in gst-plugins-base and rtsp plugin in -good.
6083 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
6086 configure: bump core/base/good requirement to 1.2.0
6087 Bump to released stable version and make implicit
6088 requirements explicit.
6090 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
6095 Fix broken gettext setup which is not used anyway
6097 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
6100 Automatic update of common submodule
6101 From dbedaa0 to d48bed3
6103 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
6105 * gst/rtsp-server/rtsp-client.c:
6106 * gst/rtsp-server/rtsp-media.c:
6107 * gst/rtsp-server/rtsp-media.h:
6108 media: add setup_sdp vmethod
6109 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
6110 gst_rtsp_media_setup_sdp.
6111 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
6113 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
6115 * gst/rtsp-server/rtsp-stream.c:
6116 rtsp-stream: Check return value of sscanf
6117 streamid is only valid if sscanf matched something.
6119 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
6121 * gst/rtsp-server/rtsp-client.c:
6122 rtsp-client: Fix iteration
6123 Wouldn't even enter the code block otherwise (i++ was used as the check
6124 and not the postfix).
6126 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
6128 * gst/rtsp-server/rtsp-client.c:
6129 * gst/rtsp-server/rtsp-client.h:
6130 client: add vmethod to configure media and streams
6131 Implement a vmethod that can be used to configure the media and the
6132 streams based on the current context. Handle the blocksize handling in
6133 the default handler.
6134 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
6136 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
6139 Make git ignore more unit test binaries
6141 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
6143 * gst/rtsp-server/rtsp-address-pool.h:
6144 * gst/rtsp-server/rtsp-auth.h:
6145 * gst/rtsp-server/rtsp-client.h:
6146 * gst/rtsp-server/rtsp-context.h:
6147 * gst/rtsp-server/rtsp-media-factory-uri.h:
6148 * gst/rtsp-server/rtsp-media-factory.h:
6149 * gst/rtsp-server/rtsp-media.h:
6150 * gst/rtsp-server/rtsp-mount-points.h:
6151 * gst/rtsp-server/rtsp-server.h:
6152 * gst/rtsp-server/rtsp-session-media.h:
6153 * gst/rtsp-server/rtsp-session-pool.h:
6154 * gst/rtsp-server/rtsp-session.h:
6155 * gst/rtsp-server/rtsp-stream-transport.h:
6156 * gst/rtsp-server/rtsp-stream.h:
6157 * gst/rtsp-server/rtsp-thread-pool.h:
6158 * gst/rtsp-server/rtsp-token.h:
6159 rtsp-server: add padding to many public structures
6160 Not mini objects though, since they are not subclassable
6161 anyway, nor kept on the stack or inlined in a structure.
6163 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
6165 media: add new create_rtpbin vmethod
6166 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
6167 https://bugzilla.gnome.org/show_bug.cgi?id=719734
6169 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
6171 * tests/check/gst/media.c:
6172 tests: fix memory leak, free test's thread pool
6173 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
6175 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
6177 * gst/rtsp-server/rtsp-stream-transport.c:
6178 stream-transport: free url in finalize
6180 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
6182 * gst/rtsp-server/rtsp-media.c:
6183 media: also do state change in suspended state
6185 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
6187 * gst/rtsp-server/rtsp-client.c:
6188 * gst/rtsp-server/rtsp-media.c:
6189 media: also handle prepare and range in suspended state
6190 When we are suspended, we are already prepared.
6191 We can get the range in the suspended state.
6193 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
6195 * tests/check/Makefile.am:
6196 * tests/check/gst/sessionmedia.c:
6197 check: add test for uri in setup
6198 Added unit tests for the new functionality in GstRTSPStreamTransport.
6199 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
6201 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
6203 * gst/rtsp-server/rtsp-client.c:
6204 client: store setup uri and use in PLAY response
6205 Store the uri used when doing the setup and use that in the PLAY
6207 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
6209 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
6211 * gst/rtsp-server/rtsp-stream-transport.c:
6212 * gst/rtsp-server/rtsp-stream-transport.h:
6213 stream-transport: add method to get/set url
6215 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
6217 * gst/rtsp-server/rtsp-client.c:
6218 client: suspend after SDP and unsuspend before PLAYING
6219 Based on patches by Ognyan Tonchev <ognyan@axis.com>
6220 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
6222 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
6224 * gst/rtsp-server/rtsp-media-factory.c:
6225 * gst/rtsp-server/rtsp-media-factory.h:
6226 * gst/rtsp-server/rtsp-media.c:
6227 * gst/rtsp-server/rtsp-media.h:
6228 * gst/rtsp-server/rtsp-session-media.c:
6229 * gst/rtsp-server/rtsp-session.c:
6230 * tests/check/gst/media.c:
6231 * tests/check/gst/mediafactory.c:
6232 media: add suspend modes
6233 Add support for different suspend modes. The stream is suspended right after
6234 producing the SDP and after PAUSE. Different suspend modes are available that
6235 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
6236 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
6237 state and RESET will bring the pipeline to the NULL state.
6238 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
6239 this means that the pipeline needs to be prerolled again.
6240 Base on patches by Ognyan Tonchev <ognyan@axis.com>
6241 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
6243 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
6245 * gst/rtsp-server/rtsp-media.c:
6246 media: start live streams in blocked state
6247 Start live streams in the blocked state and make them preroll using the
6248 messages. This ensure that no data is played by the sink until we explicitly
6249 unblock the stream right before going to PLAYING.
6250 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
6252 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
6254 * gst/rtsp-server/rtsp-media.c:
6255 media: refactor starting and waiting for preroll
6256 Based on patches from Ognyan Tonchev <ognyan@axis.com>
6257 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
6259 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
6261 * gst/rtsp-server/rtsp-stream.c:
6262 * gst/rtsp-server/rtsp-stream.h:
6263 stream: add API to block streams
6264 Add an API to block on the streams and make it post a message.
6265 Based on patch by Ognyan Tonchev <ognyan@axis.com>
6266 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
6268 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
6270 * docs/libs/Makefile.am:
6271 docs: Specify the override file
6272 Even if it's empty (for now) it avoids make distcheck complaining
6274 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
6276 * gst/rtsp-server/rtsp-media.c:
6277 media: move default implementations to where they are used
6279 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
6281 * gst/rtsp-server/rtsp-media.c:
6282 media: take the right lock in gst_rtsp_media_set_pipeline_state()
6283 We need to take the state_lock when calling this method.
6285 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
6287 * gst/rtsp-server/rtsp-media.c:
6288 media: handle add-added on non-bins too
6289 Handle dynamic payloaders that are not bins, as used in the unit-test.
6291 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
6293 * gst/rtsp-server/rtsp-media-factory.c:
6294 * gst/rtsp-server/rtsp-media-factory.h:
6295 * gst/rtsp-server/rtsp-media.c:
6296 rtsp-media/-factory: Fix request pad name comments
6297 These must be escaped for gtk-doc to parse the comments without warnings.
6299 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
6301 rtsp-media: remove transports if media is in error status
6302 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
6303 trying to change to GST_STATE_NULL and media is in error status, we
6304 remove all transports.
6305 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
6307 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
6309 * gst/rtsp-server/rtsp-media.c:
6310 rtsp-media: use element metadata to find payloader
6311 Use the element metadata to find the payloader instead of checking
6313 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
6315 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
6317 rtsp-stream: add getter for payload type
6318 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
6319 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
6320 element and create the stream with this one instead of the dynpay%d
6322 https://bugzilla.gnome.org/show_bug.cgi?id=712396
6324 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
6326 * gst/rtsp-server/rtsp-client.c:
6327 * gst/rtsp-server/rtsp-context.h:
6328 * gst/rtsp-server/rtsp-media.c:
6329 * gst/rtsp-server/rtsp-mount-points.c:
6330 * gst/rtsp-server/rtsp-server.c:
6331 * gst/rtsp-server/rtsp-token.c:
6332 rtsp-*: Refer to NULL as a constant in comments
6334 https://bugzilla.gnome.org/show_bug.cgi?id=714988
6336 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
6338 rtsp-*: Fix type name typos in comments
6339 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
6340 * rtsp-auth: Refer to part of constant name as text
6341 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
6342 * rtsp-session-media: Fix GstRTSPSessionMedia typo
6343 * rtsp-stream: Fix typo when refering to GstBin
6344 https://bugzilla.gnome.org/show_bug.cgi?id=714988
6346 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
6349 * docs/libs/gst-rtsp-server-docs.sgml:
6350 * docs/libs/gst-rtsp-server-sections.txt:
6351 docs: Improve documentation
6352 * Include annotation-glossary to quiet gtk-doc
6353 * Rename remaining ClientState -> Context
6354 * Rename object hierarchy file
6355 * Remove stale chapter references
6356 * Add missing function and object references
6357 * Include missing GstRTSPAddressPoolResult
6358 https://bugzilla.gnome.org/show_bug.cgi?id=714988
6360 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
6362 * gst/rtsp-server/rtsp-client.c:
6363 * gst/rtsp-server/rtsp-server.c:
6364 * gst/rtsp-server/rtsp-session-pool.c:
6365 * gst/rtsp-server/rtsp-session.c:
6366 * gst/rtsp-server/rtsp-stream.c:
6367 rtsp-server: sprinkle some allow-none annotations for g-i
6369 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
6371 * gst/rtsp-server/rtsp-stream.c:
6372 * gst/rtsp-server/rtsp-stream.h:
6373 stream: add method to filter transports
6374 Add a method to safely iterate and collect the stream transports
6375 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
6377 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
6379 * gst/rtsp-server/rtsp-client.c:
6380 * gst/rtsp-server/rtsp-server.c:
6381 * gst/rtsp-server/rtsp-session-pool.c:
6382 * gst/rtsp-server/rtsp-session.c:
6383 rtsp: allow NULL func in filters
6384 Passing a null function make the filters return a list of
6387 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
6389 * gst/rtsp-server/rtsp-address-pool.c:
6390 * tests/check/gst/addresspool.c:
6391 address-pool: fix address increment
6392 Use a guint instead of guint8 to increment the address. It's still not
6393 completely correct because a guint might not be able to hold the complete
6394 address range, but that's an enhacement for later.
6395 Add unit test to test improved behaviour.
6396 https://bugzilla.gnome.org/show_bug.cgi?id=708237
6398 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
6400 * gst/rtsp-server/rtsp-client.c:
6401 * tests/check/gst/client.c:
6402 client: allow absolute path in requests
6403 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
6405 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
6407 * gst/rtsp-server/rtsp-client.c:
6408 * gst/rtsp-server/rtsp-client.h:
6409 client: make make_path_from_uri a vmethod
6411 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
6413 * docs/libs/gst-rtsp-server-sections.txt:
6414 * gst/rtsp-server/rtsp-stream.c:
6415 * gst/rtsp-server/rtsp-stream.h:
6416 * tests/check/Makefile.am:
6417 * tests/check/gst/stream.c:
6418 stream: Add functions to get rtp and rtcp sockets
6419 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
6421 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
6423 * gst/rtsp-server/rtsp-context.c:
6424 * gst/rtsp-server/rtsp-context.h:
6425 context: defing a GType for the context
6426 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
6428 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6430 * gst/rtsp-server/Makefile.am:
6431 * gst/rtsp-server/rtsp-auth.c:
6432 * gst/rtsp-server/rtsp-context.c:
6433 * gst/rtsp-server/rtsp-media.c:
6434 * gst/rtsp-server/rtsp-mount-points.c:
6435 * gst/rtsp-server/rtsp-server.h:
6436 * gst/rtsp-server/rtsp-session-media.c:
6437 * gst/rtsp-server/rtsp-session.c:
6438 * gst/rtsp-server/rtsp-stream.c:
6439 Fixed several GIR warnings
6441 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
6443 * gst/rtsp-server/rtsp-auth.c:
6446 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6448 * tests/check/Makefile.am:
6449 * tests/check/gst/token.c:
6450 tests: Add unit tests for token
6451 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
6453 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6455 * gst/rtsp-server/rtsp-token.c:
6456 token: Validate args for gst_rtsp_token_is_allowed
6457 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
6459 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6461 * gst/rtsp-server/rtsp-token.c:
6462 token: Fix bug when creating empty token
6463 We always want to have a valid GstStructure in the token.
6464 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
6466 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
6468 * gst/rtsp-server/rtsp-thread-pool.c:
6469 thread-pool: avoid race in shutdown
6470 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
6471 don't actually stop the mainloop ever. Solve this race by adding an idle source
6472 to the mainloop that calls the _quit. This way we immediately exit the mainloop
6473 if quit was called before we started it.
6475 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6477 * tests/check/Makefile.am:
6478 * tests/check/gst/permissions.c:
6479 tests: Add unit tests for permissions
6480 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
6482 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6484 * tests/check/gst/mediafactory.c:
6485 tests: Test mediafactory permissions
6486 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
6488 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6490 * gst/rtsp-server/rtsp-permissions.c:
6491 permissions: Fix refcounting when adding/removing roles
6492 Previously a role that was removed was unreffed twice, and when
6493 replacing an existing role the replaced role was freed while still being
6494 referenced. Both bugs are now fixed.
6495 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
6497 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6499 * tests/check/gst/media.c:
6500 * tests/check/gst/mediafactory.c:
6501 * tests/check/gst/rtspserver.c:
6502 tests: Check gst_rtsp_url_parse return value
6503 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
6505 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
6508 Automatic update of common submodule
6509 From 865aa20 to dbedaa0
6511 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
6513 * gst/rtsp-server/rtsp-server.c:
6514 rtsp-server: Fix socket leak
6515 https://bugzilla.gnome.org/show_bug.cgi?id=710088
6517 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
6519 * gst/rtsp-server/rtsp-session-pool.c:
6520 rtsp-session-pool: Make sure session IDs are properly URI-escaped
6521 https://bugzilla.gnome.org/show_bug.cgi?id=643812
6523 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
6525 * examples/.gitignore:
6526 * examples/test-video.c:
6527 examples: fix compilation when WITH_AUTH is defined
6528 https://bugzilla.gnome.org/show_bug.cgi?id=710228
6530 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
6533 gitignore: Add new test binary
6535 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
6537 * tests/check/Makefile.am:
6538 * tests/check/gst/threadpool.c:
6539 thread-pool: Add unit test for the thread pools
6540 https://bugzilla.gnome.org/show_bug.cgi?id=710228
6542 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
6544 * gst/rtsp-server/rtsp-thread-pool.c:
6545 thread-pool: Fix thread leak when reusing threads
6546 https://bugzilla.gnome.org/show_bug.cgi?id=709730
6548 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
6550 * gst/rtsp-server/rtsp-server.c:
6551 * tests/check/gst/rtspserver.c:
6552 tests: fixed racy behavior in rtspserver tests
6553 https://bugzilla.gnome.org/show_bug.cgi?id=710078
6555 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6557 * tests/check/gst/addresspool.c:
6558 tests: Improve address pool unit tests
6559 Add a range with mixed IPV4 and IPV6 addresses to pool.
6560 Get an IPV4 address from an IPV6-only pool.
6561 Get an IPV6 address from an IPV4-only pool.
6562 Reserve a IPV6 address from an IPV4-only pool.
6563 Check for unicast addresses in multicast-only pool.
6564 Check for unicast addresses in uni-/multicast-mixed pool.
6565 https://bugzilla.gnome.org/show_bug.cgi?id=710128
6567 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6569 * gst/rtsp-server/rtsp-client.c:
6570 client: append query string in PAUSE/PLAY/TEARDOWN as well
6572 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
6574 * gst/rtsp-server/rtsp-client.c:
6575 client: Add query to control path
6576 If the SETUP url contains a query it must be appended to the control
6577 path so that it matches any already created stream in the media. The
6578 query will also be appended to the session media path.
6580 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6582 * gst/rtsp-server/rtsp-media.c:
6583 rtsp-media: remove old line
6585 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
6587 * gst/rtsp-server/rtsp-stream.c:
6588 stream: Correct control comparison
6589 https://bugzilla.gnome.org/show_bug.cgi?id=709176
6591 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6593 * gst/rtsp-server/rtsp-media.c:
6594 media: Check dynamically if the pipeline supports seeking
6595 We should not depend on whether or not the pipeline state change
6596 returned NO_PREROLL or not. A media could dynamically change its
6597 element and switch from seekable to non seekable so it's best to test
6598 the seekable nature of the pipeline dynamically when we try to do a seek.
6600 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6602 * gst/rtsp-server/rtsp-media.c:
6603 media: Return FALSE if seeking is not supported
6605 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6607 * gst/rtsp-server/rtsp-media.c:
6608 rtsp-media: don't seek accurate by default
6609 Accurate seeking is perhaps a little overkill in the most common situation and
6610 causes some formats (mp3) over slow media to seek extremely slowly.
6612 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
6614 * tests/check/gst/rtspserver.c:
6615 tests: fix unit test
6616 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
6618 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
6620 * gst/rtsp-server/rtsp-client.c:
6621 client: Reply 400 if media cannot be constructed
6622 Reply 400 Bad Request instead of 503 Service Unavailable if media
6623 cannot be constructed in SETUP.
6624 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
6626 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
6628 * gst/rtsp-server/rtsp-client.c:
6629 client: Send setup reply once only
6630 If find_media() failed in handle_setup_request() two replies was sent.
6631 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
6633 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
6636 Automatic update of common submodule
6637 From 6b03ba7 to 865aa20
6639 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
6641 * gst/rtsp-server/rtsp-server.c:
6642 server: Emit client-connected signal earlier
6643 Emit client-connected before the client ref is given to a GSource,
6644 otherwise client-connected can be emitted after the client object has
6647 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
6649 * gst/rtsp-server/rtsp-address-pool.c:
6650 * gst/rtsp-server/rtsp-address-pool.h:
6651 * gst/rtsp-server/rtsp-stream.c:
6652 * tests/check/gst/addresspool.c:
6653 addresspool: return reason of failure
6654 Let gst_rtsp_address_pool_reserve_address() return the reason why
6655 the address could not be reserved.
6656 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
6658 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
6661 autogen.sh: Sync behaviour with other GStreamer modules
6662 Allows building from outside of tree amongst other things
6664 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
6667 Automatic update of common submodule
6668 From b613661 to 6b03ba7
6670 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
6673 Automatic update of common submodule
6674 From 74a6857 to b613661
6676 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
6679 Automatic update of common submodule
6680 From 01a7a46 to 74a6857
6682 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
6684 * gst/rtsp-server/rtsp-client.c:
6685 client: Do not read beyond end of path string
6686 If the setup was done without a control url, make sure we don't try to read the
6687 non-existing control string and crash.
6689 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6691 * gst/rtsp-server/rtsp-client.c:
6692 client: Fix RTPInfo header
6693 Refactor the method to make the content_base.
6694 Use the content-base and the control url to construct the RTPInfo
6697 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6699 * gst/rtsp-server/rtsp-client.c:
6700 client: map url to path only in describe
6701 Only map the request url to a path in the DESCRIBE method. The SDP then
6702 contains the base and control urls that should be used to SETUP/PAUSE/
6703 PLAY/TEARDOWN the media.
6705 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6707 * gst/rtsp-server/rtsp-client.c:
6708 Revert "client: map URL to path in requests"
6709 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
6710 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
6711 contains the base and control urls which are used in the SETUP, PLAY,
6712 PAUSE and TEARDOWN requests.
6714 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6716 * gst/rtsp-server/rtsp-client.c:
6717 client: map URL to path in requests
6719 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6721 * gst/rtsp-server/rtsp-client.c:
6722 * gst/rtsp-server/rtsp-mount-points.c:
6723 * gst/rtsp-server/rtsp-mount-points.h:
6724 mount-points: make vmethod to make path from uri
6725 Make a vmethod to transform an url into a path. The path is then used to lookup
6726 the factory. This makes it possible to also use other bits of the url, such as
6727 the query parameters, to locate the factory.
6729 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
6731 * gst/rtsp-server/rtsp-thread-pool.c:
6732 * gst/rtsp-server/rtsp-thread-pool.h:
6733 thread-pool: Add cleanup to wait for the threadpool to finish
6734 Also fix race condition if two threads are asking for the first
6735 thread from the thread pool at once. This would case two internal
6736 GThreadPools to be created.
6737 https://bugzilla.gnome.org/show_bug.cgi?id=707753
6739 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
6741 * gst/rtsp-server/rtsp-client.c:
6742 * tests/check/gst/client.c:
6743 client: free threadpool
6744 https://bugzilla.gnome.org/show_bug.cgi?id=707638
6746 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
6748 * tests/check/gst/mountpoints.c:
6749 mountpoints tests: unref matched factories
6750 https://bugzilla.gnome.org/show_bug.cgi?id=707638
6752 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
6754 * tests/check/gst/media.c:
6755 media tests: unref thread pool and caps
6756 https://bugzilla.gnome.org/show_bug.cgi?id=707638
6758 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
6760 * gst/rtsp-server/rtsp-auth.c:
6761 * gst/rtsp-server/rtsp-media-factory.c:
6762 * gst/rtsp-server/rtsp-media.c:
6763 auth, media, media-factory: unref permissions
6764 https://bugzilla.gnome.org/show_bug.cgi?id=707638
6766 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6768 * examples/Makefile.am:
6769 Makefile: add rule for appsrc example
6771 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6773 * examples/test-appsrc.c:
6774 tests: add appsrc example
6775 Add an example on how to use appsrc to feed the server pipeline with data.
6777 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
6779 * gst/rtsp-server/rtsp-client.c:
6780 rtsp-client: remove query part from content-base string
6781 Make sure that after the control url has been resolved, it's
6782 not a part of the query-string.
6783 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
6785 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6787 * gst/rtsp-server/rtsp-client.c:
6788 client: don't check url in response
6789 There is no url or method in the response to check
6791 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6793 * gst/rtsp-server/rtsp-client.c:
6794 * gst/rtsp-server/rtsp-client.h:
6795 Add handle-response signal for when we receive a GET_PARAMETER response
6797 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6799 * gst/rtsp-server/rtsp-server.c:
6800 Fix gst_rtsp_server_client_filter, using wrong variable type
6802 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
6804 * gst/rtsp-server/rtsp-media-factory-uri.c:
6805 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
6806 For AAC we need to check for framed=true instead of parsed=true.
6807 https://bugzilla.gnome.org/show_bug.cgi?id=701384
6809 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6811 * gst/rtsp-server/rtsp-stream.c:
6812 stream: optimize pipeline for protocols
6813 When TCP is not an allowed protocol for the stream, avoid creating the
6814 appsrc/appsink/queue and tee elements.
6816 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6818 * gst/rtsp-server/rtsp-media.c:
6819 media: set protocols on streams
6821 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6823 * gst/rtsp-server/rtsp-client.c:
6824 client: use protocols supported by stream
6826 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6828 * gst/rtsp-server/rtsp-media-factory.c:
6829 * gst/rtsp-server/rtsp-media.c:
6830 * gst/rtsp-server/rtsp-stream.c:
6831 media-factory: allow all protocols
6833 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6835 * gst/rtsp-server/rtsp-media.c:
6836 media: configure protocols in new streams
6838 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6840 * gst/rtsp-server/rtsp-stream.c:
6841 * gst/rtsp-server/rtsp-stream.h:
6842 stream: add protocols property
6844 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6846 * gst/rtsp-server/rtsp-media.c:
6847 rtsp-media: send state in "new-state" signal
6848 https://bugzilla.gnome.org/show_bug.cgi?id=705110
6850 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
6853 build: add subdir-objects to AM_INIT_AUTOMAKE
6854 Fixes warnings with automake 1.14
6855 https://bugzilla.gnome.org/show_bug.cgi?id=705350
6857 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6859 * docs/libs/gst-rtsp-server-sections.txt:
6860 * gst/rtsp-server/rtsp-client.c:
6861 * gst/rtsp-server/rtsp-server.c:
6862 * gst/rtsp-server/rtsp-server.h:
6863 server: add method to iterate clients of server
6865 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6867 * gst/rtsp-server/rtsp-media.c:
6868 * gst/rtsp-server/rtsp-media.h:
6869 Add vmethod for rtsp-media subclass to access rtpbin
6871 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6873 * gst/rtsp-server/rtsp-client.h:
6874 small documentation fix
6876 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6878 * gst/rtsp-server/rtsp-client.c:
6879 Do not take range header if range is invalid
6881 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6883 * docs/libs/gst-rtsp-server-sections.txt:
6884 * gst/rtsp-server/rtsp-media.c:
6885 media: add docs for new method
6887 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6889 * gst/rtsp-server/rtsp-media.c:
6890 * gst/rtsp-server/rtsp-media.h:
6891 Add API to rtsp-media set the pipeline's state
6893 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6895 * gst/rtsp-server/rtsp-media.c:
6896 Update current position/duration when gst_rtsp_media_get_range_string is called
6898 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6900 * examples/test-cgroups.c:
6901 tests: add some more docs
6903 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6905 * examples/test-cgroups.c:
6906 * gst/rtsp-server/Makefile.am:
6907 * gst/rtsp-server/rtsp-auth.c:
6908 * gst/rtsp-server/rtsp-auth.h:
6909 * gst/rtsp-server/rtsp-client.c:
6910 * gst/rtsp-server/rtsp-client.h:
6911 * gst/rtsp-server/rtsp-context.c:
6912 * gst/rtsp-server/rtsp-context.h:
6913 * gst/rtsp-server/rtsp-params.c:
6914 * gst/rtsp-server/rtsp-params.h:
6915 * gst/rtsp-server/rtsp-server.c:
6916 * gst/rtsp-server/rtsp-thread-pool.c:
6917 * gst/rtsp-server/rtsp-thread-pool.h:
6918 * tests/check/gst/client.c:
6919 ClientState -> Context
6920 Rename the clientstate to context and put the code in a separate file.
6922 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6924 * examples/test-auth.c:
6925 * gst/rtsp-server/rtsp-auth.c:
6926 * gst/rtsp-server/rtsp-auth.h:
6927 auth: add support for default token
6928 The default token is used when the user is not authenticated and can be used to
6929 give minimal permissions.
6931 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6933 * examples/test-auth.c:
6934 * gst/rtsp-server/rtsp-auth.c:
6935 auth: use defines when possible
6937 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6939 * gst/rtsp-server/rtsp-address-pool.c:
6940 address-pool: improve docs
6942 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6944 * gst/rtsp-server/rtsp-permissions.c:
6945 permissions: add the role to the copy
6947 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
6949 * gst/rtsp-server/rtsp-permissions.c:
6950 permissions: Also copy the roles
6952 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
6954 * gst/rtsp-server/rtsp-permissions.c:
6955 permissions: Make it build
6957 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6959 * gst/rtsp-server/rtsp-address-pool.h:
6962 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6964 * docs/libs/gst-rtsp-server-sections.txt:
6965 * gst/rtsp-server/rtsp-auth.c:
6966 * gst/rtsp-server/rtsp-auth.h:
6967 * gst/rtsp-server/rtsp-media.c:
6968 * gst/rtsp-server/rtsp-session-media.c:
6969 * gst/rtsp-server/rtsp-stream-transport.c:
6970 * gst/rtsp-server/rtsp-stream-transport.h:
6971 * gst/rtsp-server/rtsp-stream.c:
6972 * tests/check/gst/client.c:
6975 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6977 * docs/libs/gst-rtsp-server-sections.txt:
6978 * gst/rtsp-server/rtsp-address-pool.c:
6979 * gst/rtsp-server/rtsp-address-pool.h:
6980 * tests/check/gst/addresspool.c:
6981 * tests/check/gst/rtspserver.c:
6982 address-pool: cleanups
6983 Remove redundant method, improve docs.
6985 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6987 * docs/libs/gst-rtsp-server-sections.txt:
6988 * gst/rtsp-server/rtsp-auth.h:
6989 * gst/rtsp-server/rtsp-permissions.c:
6990 * gst/rtsp-server/rtsp-permissions.h:
6991 * gst/rtsp-server/rtsp-token.c:
6994 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6996 * gst/rtsp-server/rtsp-permissions.c:
6997 permissions: implement _remove_role
6999 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7001 * gst/rtsp-server/rtsp-permissions.c:
7002 permissions: update docs
7004 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7006 * tests/check/gst/client.c:
7007 tests: simplify tests
7008 Client settings are now disabled by default so we don't need an auth
7009 module to disable them.
7011 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7013 * gst/rtsp-server/rtsp-auth.c:
7014 auth: add default authorizations
7015 When no auth module is specified, use our table of defaults to look up the
7016 default value of the check instead of always allowing everything. This was
7017 we can disallow client settings by default.
7019 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7022 README: update readme
7024 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7026 * gst/rtsp-server/rtsp-thread-pool.c:
7027 * gst/rtsp-server/rtsp-thread-pool.h:
7028 thread-pool: add more docs
7030 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7032 * gst/rtsp-server/rtsp-thread-pool.c:
7033 * gst/rtsp-server/rtsp-thread-pool.h:
7034 thread-pool: fix race in thread reuse
7035 If we try to reuse a thread right after we made it stop, we end up using a
7036 stopped thread. Catch this case and only reuse threads that are not stopping.
7038 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7040 * gst/rtsp-server/rtsp-server.c:
7041 server: add small debug
7043 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7045 * tests/check/gst/client.c:
7047 Add some permissions to media so we can use the auth and enable
7050 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7052 * gst/rtsp-server/rtsp-client.c:
7053 client: support pushed context in handle_request
7054 If we already have a pushed state, reuse it and add our own things. This makes
7055 it easier to write tests.
7057 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7059 * gst/rtsp-server/rtsp-auth.c:
7060 auth: don't auth on methods
7061 Don't authorize on methods anymore but on the resources that we
7062 try to access, this is more flexible.
7063 Move the authorization checks to where they are needed and let the
7064 check return the response on error.
7066 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7068 * gst/rtsp-server/rtsp-mount-points.c:
7069 mount-points: add some debug
7071 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7073 * tests/check/gst/client.c:
7074 tests: almost fix test
7076 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7078 * gst/rtsp-server/rtsp-auth.c:
7079 * gst/rtsp-server/rtsp-auth.h:
7080 * gst/rtsp-server/rtsp-client.c:
7081 * gst/rtsp-server/rtsp-client.h:
7082 * gst/rtsp-server/rtsp-server.c:
7083 * gst/rtsp-server/rtsp-server.h:
7084 auth: let the auth module check client_settings
7085 Let the auth module decide if client settings are allowed for the
7088 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7090 * gst/rtsp-server/rtsp-token.c:
7091 * gst/rtsp-server/rtsp-token.h:
7092 token: add method to check boolean permission
7094 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7096 * examples/test-auth.c:
7097 * examples/test-cgroups.c:
7098 * gst/rtsp-server/rtsp-token.c:
7099 * gst/rtsp-server/rtsp-token.h:
7100 token: simplify token constructor
7101 Use variable arguments to make easier API.
7103 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7105 * examples/test-auth.c:
7106 * examples/test-cgroups.c:
7107 * gst/rtsp-server/rtsp-media-factory.c:
7108 * gst/rtsp-server/rtsp-media-factory.h:
7109 media-factory: add convenience API for factory
7111 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7113 * examples/test-auth.c:
7114 * examples/test-cgroups.c:
7115 * gst/rtsp-server/rtsp-permissions.c:
7116 * gst/rtsp-server/rtsp-permissions.h:
7117 permissions: simplify API a little
7118 Avoid passing GstStructure in the add_role method, use varargs instead
7119 to construct the structure behind the scenes. We can then also use the
7120 structure name as the role and simplify some more logic.
7122 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7124 * gst/rtsp-server/rtsp-auth.c:
7127 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7129 * gst/rtsp-server/rtsp-auth.c:
7130 * gst/rtsp-server/rtsp-auth.h:
7131 * gst/rtsp-server/rtsp-client.c:
7132 auth: handle unauthorized response
7133 Move handling of the unauthorized response to the auth module, it can add
7134 the appropriate headers to request authorization for the required method
7135 much better than the client.
7137 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7139 * gst/rtsp-server/rtsp-client.c:
7140 * gst/rtsp-server/rtsp-client.h:
7141 client: allow for sending any message, not only requests
7142 Change the _send_request() method to _send_message() so that we
7143 can both send requests and replies.
7145 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7147 * docs/libs/gst-rtsp-server-sections.txt:
7148 * gst/rtsp-server/rtsp-server.h:
7151 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7153 * examples/test-video.c:
7154 * gst/rtsp-server/rtsp-auth.c:
7155 * gst/rtsp-server/rtsp-auth.h:
7156 * gst/rtsp-server/rtsp-server.c:
7157 * gst/rtsp-server/rtsp-server.h:
7158 auth: move TLS handling to auth module
7159 Remove the TLS settings on the server and move it to the auth module because
7160 that is where security related bits go.
7162 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7164 * gst/rtsp-server/rtsp-client.c:
7165 * gst/rtsp-server/rtsp-client.h:
7166 client: add state push/pop
7168 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7170 * gst/rtsp-server/rtsp-client.c:
7171 * gst/rtsp-server/rtsp-client.h:
7172 client: add connection to state
7174 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7176 * gst/rtsp-server/rtsp-mount-points.c:
7177 mount-points: fix debug
7179 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7181 * tests/check/gst/media.c:
7182 tests: fix media test
7184 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7186 * gst/rtsp-server/rtsp-thread-pool.c:
7187 thread-pool: we don't require a state
7189 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7191 * gst/rtsp-server/rtsp-server.c:
7192 server: let context ref the server
7193 So that we don't risk losing the server object early anc crash.
7195 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7197 * tests/check/gst/client.c:
7198 tests: fix client test
7200 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7203 * docs/libs/gst-rtsp-server-docs.sgml:
7204 * docs/libs/gst-rtsp-server-sections.txt:
7205 * gst/rtsp-server/rtsp-address-pool.c:
7206 * gst/rtsp-server/rtsp-auth.c:
7207 * gst/rtsp-server/rtsp-client.c:
7208 * gst/rtsp-server/rtsp-client.h:
7209 * gst/rtsp-server/rtsp-media-factory-uri.c:
7210 * gst/rtsp-server/rtsp-media-factory.c:
7211 * gst/rtsp-server/rtsp-media-factory.h:
7212 * gst/rtsp-server/rtsp-media.c:
7213 * gst/rtsp-server/rtsp-mount-points.c:
7214 * gst/rtsp-server/rtsp-params.c:
7215 * gst/rtsp-server/rtsp-permissions.c:
7216 * gst/rtsp-server/rtsp-sdp.c:
7217 * gst/rtsp-server/rtsp-server.c:
7218 * gst/rtsp-server/rtsp-server.h:
7219 * gst/rtsp-server/rtsp-session-media.c:
7220 * gst/rtsp-server/rtsp-session-pool.c:
7221 * gst/rtsp-server/rtsp-session.c:
7222 * gst/rtsp-server/rtsp-stream-transport.c:
7223 * gst/rtsp-server/rtsp-stream.c:
7224 * gst/rtsp-server/rtsp-thread-pool.c:
7225 * gst/rtsp-server/rtsp-token.c:
7228 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7230 * gst/rtsp-server/rtsp-session-pool.c:
7231 * gst/rtsp-server/rtsp-session-pool.h:
7232 session-pool: make vmethod to create a session
7233 Make a vmethod to create a sessions so that subclasses can create
7234 custom session objects
7236 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7238 * gst/rtsp-server/rtsp-auth.c:
7239 * gst/rtsp-server/rtsp-media-factory.h:
7240 * gst/rtsp-server/rtsp-media.h:
7241 * gst/rtsp-server/rtsp-mount-points.h:
7242 * gst/rtsp-server/rtsp-session-pool.h:
7243 * gst/rtsp-server/rtsp-stream.h:
7246 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7248 * docs/libs/gst-rtsp-server-docs.sgml:
7249 * docs/libs/gst-rtsp-server-sections.txt:
7250 * gst/rtsp-server/rtsp-address-pool.c:
7251 * gst/rtsp-server/rtsp-address-pool.h:
7252 * gst/rtsp-server/rtsp-auth.c:
7253 * gst/rtsp-server/rtsp-client.h:
7254 * gst/rtsp-server/rtsp-media-factory.h:
7255 * gst/rtsp-server/rtsp-media.c:
7256 * gst/rtsp-server/rtsp-media.h:
7257 * gst/rtsp-server/rtsp-permissions.c:
7258 * gst/rtsp-server/rtsp-permissions.h:
7259 * gst/rtsp-server/rtsp-server.h:
7260 * gst/rtsp-server/rtsp-session-media.c:
7261 * gst/rtsp-server/rtsp-session-media.h:
7262 * gst/rtsp-server/rtsp-session-pool.h:
7263 * gst/rtsp-server/rtsp-session.h:
7264 * gst/rtsp-server/rtsp-stream-transport.h:
7265 * gst/rtsp-server/rtsp-stream.c:
7266 * gst/rtsp-server/rtsp-thread-pool.h:
7269 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7272 * examples/Makefile.am:
7273 configure: compile cgroup example conditionally
7274 Only compile the cgroup example when we have libcgroup
7276 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7279 * examples/Makefile.am:
7280 * examples/test-cgroups.c:
7281 examples: add cgroups example
7283 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7285 * tests/check/gst/rtspserver.c:
7286 tests: fix compilation
7288 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7290 * gst/rtsp-server/rtsp-thread-pool.c:
7291 thread-pool: fix vmethod invocation
7293 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7295 * gst/rtsp-server/rtsp-thread-pool.c:
7296 * gst/rtsp-server/rtsp-thread-pool.h:
7297 thread-pool: store thread type in thread
7299 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7301 * gst/rtsp-server/rtsp-client.c:
7302 client: pass thread from pool to media _prepare
7303 Get a thread from the configured threadpool and pass it to the prepare method of
7306 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7308 * gst/rtsp-server/rtsp-media.c:
7309 * gst/rtsp-server/rtsp-media.h:
7310 media: Accept a thread in _prepare
7311 Remove out own threadpool handling and use the provided thread and
7312 maincontext for the bus messages and the state changes.
7314 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7316 * gst/rtsp-server/rtsp-server.c:
7317 server: configure client thread pool
7319 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7321 * gst/rtsp-server/rtsp-client.c:
7322 * gst/rtsp-server/rtsp-client.h:
7323 client: add method to configure thread pool
7325 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7327 * gst/rtsp-server/rtsp-client.h:
7328 * gst/rtsp-server/rtsp-server.c:
7329 * gst/rtsp-server/rtsp-server.h:
7330 server: use thread pool
7331 Use the thread pool instead of doing our own thing.
7333 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7335 * gst/rtsp-server/Makefile.am:
7336 * gst/rtsp-server/rtsp-thread-pool.c:
7337 * gst/rtsp-server/rtsp-thread-pool.h:
7338 thread-pool: add object to manage threads
7339 Add an object to manage the client and media threads.
7341 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7343 * gst/rtsp-server/rtsp-auth.c:
7344 auth: debug authorization check
7346 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7348 * gst/rtsp-server/rtsp-media.c:
7349 media: start media pipeline in context
7350 Start the media pipeline in the provided context (or our default one
7351 when NULL). This makes sure that we run the bus thread in this context and that
7352 all media threads are children of this context.
7354 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7356 * gst/rtsp-server/rtsp-media-factory.c:
7357 factory: pass permissions to media by default
7359 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7361 * examples/test-auth.c:
7362 test: add permissions to auth test
7363 Ass some permissions to the media factory in the test.
7365 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7367 * gst/rtsp-server/rtsp-auth.c:
7368 * gst/rtsp-server/rtsp-auth.h:
7369 * gst/rtsp-server/rtsp-client.c:
7370 auth: simplify auth checks
7371 Remove client from methods, it's now in the state
7372 Perform the check specified by the string, use the information from the
7373 thread local context.
7375 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7377 * gst/rtsp-server/rtsp-client.c:
7378 * gst/rtsp-server/rtsp-client.h:
7379 client: add state to current thread
7380 Add the client to the ClientState object.
7381 Place the ClientState on the current thread.
7383 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7385 * gst/rtsp-server/rtsp-media-factory.c:
7386 * gst/rtsp-server/rtsp-media-factory.h:
7387 * gst/rtsp-server/rtsp-media.c:
7388 * gst/rtsp-server/rtsp-media.h:
7389 media: make it possible to set permissions
7390 Make it possible to set permissions on media and media factory objects
7392 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7394 * gst/rtsp-server/Makefile.am:
7395 * gst/rtsp-server/rtsp-permissions.c:
7396 * gst/rtsp-server/rtsp-permissions.h:
7397 permissions: add permissions object
7398 Add a mini object to store permissions based on a role.
7400 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7402 * examples/test-auth.c:
7403 * gst/rtsp-server/rtsp-auth.c:
7404 * gst/rtsp-server/rtsp-auth.h:
7405 * gst/rtsp-server/rtsp-client.c:
7406 auth: add auth checks
7407 Add an enum with auth checks and implement the checks in the auth object.
7408 Perform the checks from the client.
7410 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7412 * examples/test-auth.c:
7413 * gst/rtsp-server/rtsp-auth.c:
7414 * gst/rtsp-server/rtsp-auth.h:
7415 * gst/rtsp-server/rtsp-client.h:
7416 auth: use the token after authentication
7417 After we authenticated a user, keep the Token around in the state.
7419 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7421 * gst/rtsp-server/rtsp-client.c:
7422 * gst/rtsp-server/rtsp-media.c:
7423 * gst/rtsp-server/rtsp-media.h:
7424 * tests/check/gst/media.c:
7425 media: add optional context for bus messages
7426 Add an optional mainloop to _prepare that will handle the bus messages instead
7427 of always using the shared mainloop.
7429 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7431 * gst/rtsp-server/Makefile.am:
7432 * gst/rtsp-server/rtsp-token.c:
7433 * gst/rtsp-server/rtsp-token.h:
7434 token: add authorization token
7435 Add a simply miniobject that contains the authorizations. The object contains a
7436 GstStructure that hold all authorization fields. When a user is authenticated,
7437 the auth module will create a Token for the user. The token is then used to
7438 check what operations the user is allowed to do and various other configuration
7441 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7443 * examples/test-auth.c:
7444 * gst/rtsp-server/rtsp-auth.c:
7445 * gst/rtsp-server/rtsp-auth.h:
7446 * gst/rtsp-server/rtsp-client.c:
7447 * gst/rtsp-server/rtsp-client.h:
7448 * gst/rtsp-server/rtsp-media-factory.c:
7449 * gst/rtsp-server/rtsp-media-factory.h:
7450 * gst/rtsp-server/rtsp-media.c:
7451 * gst/rtsp-server/rtsp-media.h:
7452 auth: remove auth from media and factory
7453 Remove the auth object from media and factory. We want to have the RTSPClient
7454 authenticate and authorize resources, there is no need to place another auth
7455 manager on the media/factory.
7457 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7459 * examples/test-auth.c:
7460 * gst/rtsp-server/rtsp-auth.c:
7461 * gst/rtsp-server/rtsp-auth.h:
7462 * gst/rtsp-server/rtsp-client.h:
7463 auth: add support for multiple basic auth tokens
7464 Make it possible to add multiple basic authorisation tokens to one authorization
7465 object. Associate with each token an authorization group that will define what
7466 capabilities are allowed.
7468 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7470 * gst/rtsp-server/rtsp-client.c:
7471 client: error out on non-aggregate control
7472 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
7474 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7476 * gst/rtsp-server/rtsp-client.c:
7477 client: rework setup request a little
7478 Cache the media in DESCRIBE based on the longest matching path with the uri
7479 that we can find in the mount points.
7480 Rework the setup request a little to get the media from the session or from
7481 the longest matching path, this way we can derive the control string as
7482 everything after the path instead of hardcoding it.
7483 Find the stream based on the control string and only open a session when all
7486 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7488 * gst/rtsp-server/rtsp-media.c:
7489 * gst/rtsp-server/rtsp-media.h:
7490 media: add method to find a stream by control url
7492 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7494 * gst/rtsp-server/rtsp-stream.c:
7495 * gst/rtsp-server/rtsp-stream.h:
7496 stream: add method to check control url of stream
7498 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7500 * gst/rtsp-server/rtsp-client.c:
7501 * gst/rtsp-server/rtsp-session-media.c:
7502 * gst/rtsp-server/rtsp-session-media.h:
7503 * gst/rtsp-server/rtsp-session.c:
7504 * gst/rtsp-server/rtsp-session.h:
7505 session: use path matching for session media
7506 Use a path string instead of a uri to lookup session media in the sessions. Also
7507 use path matching to find the largest possible path that matches.
7509 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7511 * gst/rtsp-server/rtsp-client.c:
7512 * gst/rtsp-server/rtsp-mount-points.c:
7513 * gst/rtsp-server/rtsp-mount-points.h:
7514 * tests/check/gst/mountpoints.c:
7515 mount-points: remove useless vmethod
7516 Making lookups in the mount points should not be done with a URL, if there is a
7517 mapping to be done from URL to mount points, we'll need to do it somewhere
7520 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7522 * gst/rtsp-server/rtsp-mount-points.c:
7523 * gst/rtsp-server/rtsp-mount-points.h:
7524 * tests/check/gst/mountpoints.c:
7525 mount-points: improve mount point searching
7526 Use a GSequence to keep track of the mount points.
7527 Match a URL to the longest matching registered mount point. This should be the
7528 URL to perform aggreagate control and the remainder is the stream specific
7530 Add some unit tests for this.
7532 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
7534 * gst/rtsp-server/Makefile.am:
7535 rtsp-server: Allow building of static library
7537 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7539 * tests/check/gst/mediafactory.c:
7540 tests: fix compilation
7542 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7544 * gst/rtsp-server/rtsp-sdp.c:
7545 sdp: get control string from stream
7546 Use the control string as configured in the stream.
7548 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7550 * gst/rtsp-server/rtsp-stream.c:
7551 * gst/rtsp-server/rtsp-stream.h:
7552 stream: add methods and property to set control string
7554 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7556 * gst/rtsp-server/rtsp-client.c:
7558 Rename variables for clarity
7559 Keep media in state when we can
7561 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7563 * gst/rtsp-server/rtsp-client.c:
7564 * gst/rtsp-server/rtsp-stream.c:
7565 * gst/rtsp-server/rtsp-stream.h:
7566 stream: add more support for IPv6
7567 Rename _get_address to _get_multicast_address in GstRTSPStream to
7568 make it clear that this function only deals with multicast.
7569 Make it possible to have both an IPv4 and IPv6 multicast address on
7570 a stream. Give the client an IPv4 or IPv6 address depending on the
7571 address it used to connect to the server.
7572 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
7574 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7576 * gst/rtsp-server/rtsp-client.c:
7579 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7581 * gst/rtsp-server/rtsp-stream.c:
7582 stream: handle failed port allocation
7583 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
7584 can't allocate any family at all. Also keep track of what port families we
7586 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
7588 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7590 * gst/rtsp-server/rtsp-stream.c:
7591 stream: improve docs
7593 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7595 * gst/rtsp-server/rtsp-stream-transport.c:
7596 stream-transport: remove old if 0 block
7598 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
7600 * tests/check/gst/client.c:
7602 gst_rtsp_client_get_uri() has been removed
7603 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
7605 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7607 * gst/rtsp-server/rtsp-client.c:
7608 * gst/rtsp-server/rtsp-client.h:
7609 client: add method to filter managed sessions
7610 Add a method to filter the sessions managed by this client connection.
7611 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
7613 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7615 * gst/rtsp-server/rtsp-client.c:
7616 * gst/rtsp-server/rtsp-client.h:
7617 client: remove _get_uri() method
7618 Remove the get_uri() method on the client. A client has no uri, the uri
7619 property is an internal property to manage the last cached media for
7622 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7624 * gst/rtsp-server/rtsp-media-factory.h:
7625 media-factory: fix typo
7627 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
7629 * gst/rtsp-server/rtsp-media.c:
7630 rtsp-media: Do not leak the query in default_query_stop
7631 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
7633 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7635 * gst/rtsp-server/rtsp-media.c:
7636 media: don't unlock when conversion fails
7637 Don't unlock the state lock when conversion fails because it was not locked.
7639 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
7641 * gst/rtsp-server/rtsp-media.c:
7642 * gst/rtsp-server/rtsp-media.h:
7643 Add query_position and query_stop vmethods to rtsp-media
7645 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
7647 * gst/rtsp-server/rtsp-media.c:
7648 Fix typo in property install for rtsp-media's time-provider
7650 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7652 * gst/rtsp-server/rtsp-client.c:
7653 * gst/rtsp-server/rtsp-client.h:
7654 client: clean some variables
7655 Clean some variables and add some guards to _send_request()
7657 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
7659 * gst/rtsp-server/rtsp-client.c:
7660 * gst/rtsp-server/rtsp-client.h:
7661 Add gst_rtsp_client_send_request API
7662 This makes it possible to send arbitrary messages to a client, such as
7663 SET_PARAMETER or GET_PARAMETER
7665 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7667 * gst/rtsp-server/rtsp-media.c:
7668 * gst/rtsp-server/rtsp-media.h:
7669 media: add _get_element() method
7670 Add method to get the element used when creating the media.
7671 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
7673 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7675 * gst/rtsp-server/rtsp-media.c:
7678 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
7680 * gst/rtsp-server/rtsp-stream.c:
7681 * gst/rtsp-server/rtsp-stream.h:
7682 stream: allow access to the rtp session
7683 https://bugzilla.gnome.org/show_bug.cgi?id=703004
7685 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
7687 * gst/rtsp-server/rtsp-stream.c:
7688 * gst/rtsp-server/rtsp-stream.h:
7689 dscp qos support in gst-rtsp-stream
7690 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
7692 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7694 * tests/check/gst/rtspserver.c:
7696 Actually do what the comment says. Also keep the old code around, not sure what
7697 should happen when you get a 454 from a TEARDOWN, does it close the connection?
7698 it currently doesn't.
7700 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7702 * gst/rtsp-server/rtsp-client.c:
7703 client: also watch newly created session
7704 When we newly created a session, start watching it immediately instead of
7705 on the next request.
7707 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
7709 * tests/check/gst/client.c:
7710 tests: add unit test for new-session
7711 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
7713 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7715 * gst/rtsp-server/rtsp-client.c:
7716 client: emit new-session when new session is created
7717 Only emit new-session when we created a new session for a client, not when a
7718 client picked up a previous session.
7719 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
7721 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
7723 * gst/rtsp-server/rtsp-client.c:
7724 client: handle asterisk as path in requests
7725 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
7727 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7729 * gst/rtsp-server/rtsp-media.c:
7730 media: handle segment query format mismatch
7731 It's possible that the segment query returns with a different format than what
7732 we asked for, handle this case also.
7734 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
7736 * gst/rtsp-server/rtsp-media.c:
7737 media: use segment stop in collect_media_stats
7738 Use segment stop instead of duration as range end point.
7739 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
7741 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
7743 * gst/rtsp-server/rtsp-media.c:
7744 * tests/check/gst/media.c:
7745 rtsp-media: Do not leak the element in take_pipeline
7746 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
7748 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
7750 * gst/rtsp-server/rtsp-client.c:
7751 * gst/rtsp-server/rtsp-client.h:
7752 rtsp-client: Make configure_client_transport virtual
7753 This patch makes configure_client_transport virtual. The functionality is
7754 needed to handle some weird clients sending multicast transport settings as url
7756 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
7758 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
7760 * gst/rtsp-server/rtsp-client.c:
7761 * gst/rtsp-server/rtsp-client.h:
7762 rtsp-client: Make param_set and param_get virtual
7763 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
7765 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
7767 * gst/rtsp-server/rtsp-client.c:
7768 * gst/rtsp-server/rtsp-media.c:
7769 * gst/rtsp-server/rtsp-media.h:
7770 media: convert_range replaces get_range_times
7771 get_range_times worked for handling UTC ranges for seeks, but we also
7772 need to convert back from NPT to the requested unit in
7773 get_range_string. convert_range is now used for both.
7774 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
7776 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7778 * gst/rtsp-server/rtsp-client.c:
7779 * gst/rtsp-server/rtsp-sdp.c:
7780 * gst/rtsp-server/rtsp-sdp.h:
7781 sdp: cleanup sdp info
7782 We don't need to pass the proto, we can more easily check a boolean.
7783 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
7785 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
7787 * gst/rtsp-server/rtsp-sdp.c:
7788 use 0.0.0.0 or :: for c= line instead of server address
7790 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
7792 * gst/rtsp-server/rtsp-client.c:
7793 use local address, not remote, in SDP
7794 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
7796 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7799 Automatic update of common submodule
7800 From 098c0d7 to 01a7a46
7802 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
7804 * gst/rtsp-server/rtsp-media.c:
7805 * gst/rtsp-server/rtsp-media.h:
7806 media: possibility to override range time conversion
7807 Make it possible to override the conversion from GstRTSPTimeRange to
7808 GstClockTimes, that is done before seeking on the media
7809 pipeline. Overriding can be useful for UTC ranges, where the default
7810 conversion gives nanoseconds since 1900.
7811 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
7813 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
7815 * gst/rtsp-server/rtsp-server.c:
7816 * gst/rtsp-server/rtsp-server.h:
7817 rtsp-server: Expose the use_client_settings API
7818 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
7820 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
7822 * gst/rtsp-server/rtsp-client.c:
7823 * gst/rtsp-server/rtsp-stream.c:
7824 * gst/rtsp-server/rtsp-stream.h:
7825 rtspstream: handle both ipv4 and ipv6 clients
7826 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
7828 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7830 * gst/rtsp-server/rtsp-sdp.c:
7831 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
7832 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
7833 We already have a way to place extra attributes in the SDP by using a string
7834 property with prefix x- or a- in the caps.
7836 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7838 * gst/rtsp-server/rtsp-sdp.c:
7839 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
7840 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
7841 We already have a way to place extra attributes in the SDP, just make a string
7842 property in the payloader with a- or x- prefix.
7844 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7846 * gst/rtsp-server/rtsp-sdp.c:
7847 rtsp: place a- and x- properties as attributes
7848 application/x-rtp has properties with a- and x- prefixes that should be
7849 placed as attributes in the SDP for the media instead of being added to the
7852 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7854 * examples/Makefile.am:
7855 * examples/test-video.c:
7856 example: add TLS example
7858 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7860 * gst/rtsp-server/rtsp-server.c:
7861 * gst/rtsp-server/rtsp-server.h:
7862 server: add support for TLS
7863 Add methods to set and get a TLS certificate.
7864 Add vmethod to configure a new connection. By default, configure the TLS
7865 certificate in a new connection if needed.
7867 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7869 * gst/rtsp-server/rtsp-server.c:
7870 * gst/rtsp-server/rtsp-server.h:
7871 server: remove accept_client vmethod
7872 This vmethod is not very useful so remove it.
7874 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7876 * gst/rtsp-server/rtsp-server.c:
7877 server: don't crash on NULL GError
7879 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
7881 * gst/rtsp-server/rtsp-session-pool.c:
7882 rtsp-session-pool: corrected session timeout detection
7883 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
7885 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7887 * gst/rtsp-server/rtsp-client.c:
7888 client: improve debug
7890 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7892 * gst/rtsp-server/rtsp-client.c:
7893 * gst/rtsp-server/rtsp-client.h:
7894 * gst/rtsp-server/rtsp-server.c:
7895 server: refactor connection setup
7896 Let the server accept the socket connection and construct a GstRTSPConnection
7897 from it. Remove the code from the client and let the client only deal with
7898 a fully configure GstRTSPConnection object.
7899 We will need this later when the server will configure the connection for
7902 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7904 * gst/rtsp-server/rtsp-stream.c:
7905 stream: keep the transport object alive
7906 Keep the transport object alive while we have it as qdata on the
7909 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
7911 * gst/rtsp-server/rtsp-client.c:
7912 * gst/rtsp-server/rtsp-server.c:
7913 rtsp-server: Do not crash on nmapping of server
7914 * generate error when gst_rtsp_connection_accept fails
7915 * do not stop accepting incoming connections because
7916 accepting a client fails
7917 https://bugzilla.gnome.org/show_bug.cgi?id=701072
7919 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
7921 * gst/rtsp-server/rtsp-client.c:
7922 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
7923 https://bugzilla.gnome.org/show_bug.cgi?id=700953
7925 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
7927 * gst/rtsp-server/rtsp-sdp.c:
7928 rtsp-sdp: Parse framerate caps field and set SDP attribute
7929 The SDP attribute and its format is described in RFC4566.
7930 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
7932 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
7934 * gst/rtsp-server/rtsp-sdp.c:
7935 rtsp-sdp: Parse width/height from caps and set SDP attribute
7936 The SDP attribute and its format is described in RFC6064.
7937 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
7939 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
7941 * gst/rtsp-server/rtsp-sdp.c:
7942 * tests/check/gst/client.c:
7943 rtsp-sdp: add bandwidth line
7944 https://bugzilla.gnome.org/show_bug.cgi?id=699220
7946 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7949 Automatic update of common submodule
7950 From 5edcd85 to 098c0d7
7952 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
7954 * tests/check/gst/media.c:
7955 tests: add dynamic payloader prepare/unprepare check
7957 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7959 * gst/rtsp-server/rtsp-media.c:
7960 media: release lock when removing fakesink
7962 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7964 * gst/rtsp-server/rtsp-stream.c:
7965 stream: set elements to NULL before removing
7966 When removing a stream, set the elements to NULL first. This avoids
7967 element-is-not-in-NULL-state errors when we dispose the elements.
7969 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
7972 Automatic update of common submodule
7973 From 3cb3d3c to 5edcd85
7975 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7977 * gst/rtsp-server/rtsp-media.c:
7978 * gst/rtsp-server/rtsp-media.h:
7979 media: listen to pad-removed signals
7980 Listen to the pad-removed signal and remove the stream associated with the
7982 Add signal to be notified of the removed pad.
7983 Remove the fakesink in unprepare()
7984 Fix signatures of the signal methods
7986 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7988 * examples/test-sdp.c:
7989 tests: add example of reusable pipelines
7991 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
7993 * gst/rtsp-server/rtsp-stream.c:
7994 * gst/rtsp-server/rtsp-stream.h:
7995 stream: add method to get the srcpad
7997 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
7999 * tests/check/gst/media.c:
8000 check: add media prepare/unprepare test
8001 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
8003 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
8005 * gst/rtsp-server/rtsp-media.c:
8006 media: disconnect from signal handlers in unprepare()
8007 We connected to the pad-added and no-more-pads signals in prepare() so
8008 we need to disconnect from them in unprepare().
8009 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
8011 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
8013 * gst/rtsp-server/rtsp-media.c:
8014 media: don't free streams array
8015 Don't free the streams array in the unprepare() method, they were not
8017 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
8019 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
8021 * gst/rtsp-server/rtsp-media.c:
8022 media: don't unref the pipeline in unprepare
8023 Unprepare() should undo what prepare() does. Because the pipeline is
8024 not created in prepare(), we should not unref it in unprepare()
8026 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
8028 * gst/rtsp-server/rtsp-stream.c:
8029 stream: clear session and caps for reuse
8030 Set the session and caps to NULL after unref otherwise we might unref
8032 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
8034 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
8036 * gst/rtsp-server/rtsp-client.c:
8037 client: send out teardown signal before tearing down
8038 The advantage is that in the signal handler you get direct access to
8039 information about what streams are about to get torn down (in the
8040 GstRTSPClientState).
8041 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
8043 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
8045 * gst/rtsp-server/rtsp-client.c:
8046 * gst/rtsp-server/rtsp-client.h:
8047 client: expose connection
8048 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
8050 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
8053 Automatic update of common submodule
8054 From aed87ae to 3cb3d3c
8056 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8058 * gst/rtsp-server/rtsp-media.c:
8059 * gst/rtsp-server/rtsp-media.h:
8060 * gst/rtsp-server/rtsp-session-media.c:
8061 * gst/rtsp-server/rtsp-session-media.h:
8062 media: add method to get the base_time of the pipeline
8063 Together with a shared clock, this base-time could eventually be sent to
8064 the client so that it can reconstruct the exact running-time of the clock
8067 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8069 * gst/rtsp-server/Makefile.am:
8070 * gst/rtsp-server/rtsp-media.c:
8071 * gst/rtsp-server/rtsp-media.h:
8072 * gst/rtsp-server/rtsp-sdp.c:
8073 media: add GstNetTimeProvider support
8074 Add a property to let the media provide a GstNetTimeProvider for its clock.
8075 Make methods to get the clock and nettimeprovider
8076 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
8077 provider and also the current time of the clock. This should make it possible
8078 for (GStreamer) clients to slave their clock to the server clock.
8080 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
8083 Automatic update of common submodule
8084 From 04c7a1e to aed87ae
8086 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8088 * gst/rtsp-server/rtsp-media.c:
8089 media: wait for buffering to complete
8090 Wait for buffering to complete before changing the state to the target state.
8092 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8094 * gst/rtsp-server/rtsp-media.c:
8095 media: small cleanup
8097 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
8099 * tests/check/gst/rtspserver.c:
8100 tests: remove extra unref in test_setup_non_existing_stream
8101 The unref is not needed anymore, teardown runs without it.
8102 https://bugzilla.gnome.org/show_bug.cgi?id=696542
8104 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
8106 * tests/check/gst/rtspserver.c:
8107 tests: GSocketService cleanup in test_bind_already_in_use
8108 Use g_socket_service_stop so the rtspserver test stops listening for
8109 incoming connections in test_bind_already_in_use.
8110 https://bugzilla.gnome.org/show_bug.cgi?id=696541
8112 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
8114 * gst/rtsp-server/rtsp-media-factory.c:
8115 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
8116 Instead use a GWeakRef which is safe to use
8117 This is a known GLib bug, see:
8118 https://bugzilla.gnome.org/show_bug.cgi?id=667145
8120 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
8122 * gst/rtsp-server/rtsp-client.c:
8123 * gst/rtsp-server/rtsp-media.c:
8124 * gst/rtsp-server/rtsp-media.h:
8125 * gst/rtsp-server/rtsp-sdp.c:
8126 * tests/check/gst/media.c:
8127 * tests/check/gst/rtspserver.c:
8128 rtsp-media/client: Reply to PLAY request with same type of Range
8129 Remember the type of Range from the PLAY request and use the same type for
8132 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
8134 * gst/rtsp-server/rtsp-client.c:
8135 * gst/rtsp-server/rtsp-client.h:
8136 * tests/check/gst/client.c:
8137 rtsp-client: expose uri
8139 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
8141 * tests/check/gst/mediafactory.c:
8142 tests: Hold ref while creating second media
8143 To test if the media aren't shared, make sure we keep the first one while creating a second
8144 otherwise the same memory address may be reused.
8146 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
8149 configure: remove out-of-date comment
8151 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
8154 .gitignore: ignore more build files
8156 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
8158 * tests/check/Makefile.am:
8159 tests: use right _LIBS variable for gst-plugins-base libs
8161 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8163 * tests/check/Makefile.am:
8164 check: add librtp to libs
8166 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
8168 * tests/check/gst/rtspserver.c:
8169 tests: Add test to check selecting a port the server will send from
8171 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
8173 * tests/check/gst/rtspserver.c:
8174 tests: Make sure packets are actually received
8176 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
8178 * gst/rtsp-server/rtsp-stream.c:
8179 stream: Select unicast address from pool if appropriate
8181 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
8183 * gst/rtsp-server/rtsp-stream.c:
8184 stream: Properties are always there in Gst 1.0
8186 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
8188 * tests/check/gst/addresspool.c:
8189 tests: Add tests for unicast addresses in pool
8191 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
8193 * gst/rtsp-server/rtsp-address-pool.c:
8194 * tests/check/gst/addresspool.c:
8195 address-pool: Verify that multicast addresses are used for multicast and vice-versa
8197 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
8199 * docs/libs/gst-rtsp-server-sections.txt:
8200 * gst/rtsp-server/rtsp-address-pool.c:
8201 * gst/rtsp-server/rtsp-address-pool.h:
8202 * gst/rtsp-server/rtsp-stream.c:
8203 * tests/check/gst/addresspool.c:
8204 address-pool: Add unicast addresses
8206 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
8209 * gst/rtsp-server/rtsp-server.c:
8210 * tests/check/gst/rtspserver.c:
8211 rtsp-server: Limit the number of threads per server instance
8212 If we exceed the maximum, just round robin the clients over the existing
8215 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
8217 * gst/rtsp-server/rtsp-server.c:
8218 rtsp-server: No need to store the GMainContext in the client context
8220 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
8222 * tests/check/gst/rtspserver.c:
8223 tests: Add test for client disconnection
8225 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
8227 * tests/check/gst/rtspserver.c:
8228 tests: Test client and session timeouts with multiple threads
8230 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
8232 * gst/rtsp-server/rtsp-address-pool.c:
8233 * gst/rtsp-server/rtsp-auth.c:
8234 * gst/rtsp-server/rtsp-client.c:
8235 * gst/rtsp-server/rtsp-media-factory-uri.c:
8236 * gst/rtsp-server/rtsp-media-factory.c:
8237 * gst/rtsp-server/rtsp-media.c:
8238 * gst/rtsp-server/rtsp-mount-points.c:
8239 * gst/rtsp-server/rtsp-server.c:
8240 * gst/rtsp-server/rtsp-session-media.c:
8241 * gst/rtsp-server/rtsp-session-pool.c:
8242 * gst/rtsp-server/rtsp-session.c:
8243 Document locking and its order
8245 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
8247 * tests/check/gst/rtspserver.c:
8248 tests: Test that slow DESCRIBE don't block other clients
8250 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
8252 * tests/check/gst/client.c:
8253 tests: Add tests for client-requested multicast address
8255 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
8257 * docs/libs/gst-rtsp-server-sections.txt:
8258 docs: Put the various functions in the right sections
8260 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
8262 * docs/libs/gst-rtsp-server-docs.sgml:
8263 * docs/libs/gst-rtsp-server-sections.txt:
8264 * gst/rtsp-server/rtsp-address-pool.c:
8265 * gst/rtsp-server/rtsp-address-pool.h:
8266 docs: Generate docs for GstRTSPAddressPool
8268 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
8270 * gst/rtsp-server/rtsp-client.c:
8271 * gst/rtsp-server/rtsp-stream.c:
8272 * gst/rtsp-server/rtsp-stream.h:
8273 client: Check client provided addresses against the address pool
8275 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
8277 * gst/rtsp-server/rtsp-address-pool.c:
8278 * gst/rtsp-server/rtsp-address-pool.h:
8279 * tests/check/gst/addresspool.c:
8280 address-pool: Add API to request a specific address from the pool
8281 Also add relevant unit tests.
8283 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
8285 * tests/check/gst/mediafactory.c:
8286 tests: Check the passing around of a RTSPAddressPool
8287 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
8288 way down to the stream.
8290 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
8292 * tests/check/gst/addresspool.c:
8293 tests: Add more tests for the address pool
8295 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
8297 * gst/rtsp-server/rtsp-address-pool.c:
8298 address-pool: Fix off by one error
8299 When splitting a port range, the port after a skip is not part of range.
8301 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
8304 Automatic update of common submodule
8305 From 2de221c to 04c7a1e
8307 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
8310 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
8311 AM_CONFIG_HEADER was removed in automake 1.13
8312 https://bugzilla.gnome.org/show_bug.cgi?id=693368
8314 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
8317 Automatic update of common submodule
8318 From a942293 to 2de221c
8320 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8322 * gst/rtsp-server/rtsp-client.c:
8323 client: make sure the watch exists while sending data
8324 Protect the send_func with a lock. This allows us to wait for sending
8325 to complete before changing the send_func and user_data. We add an
8326 extra ref to the watch to make sure that it remains valid during
8328 When closing the connection, set the send_func to NULL
8329 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
8331 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8333 * tests/check/Makefile.am:
8334 tests: use GST_*_1_0 environment variables everywhere
8335 The _1_0 suffixed environment variables override the
8336 non-suffixed ones, so if we're in an environment that
8337 sets the _1_0 suffixed ones, such as jhbuild, we need
8338 to set those to make sure ours actually always get
8341 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8344 Automatic update of common submodule
8345 From acb04d9 to a942293
8347 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8349 * gst/rtsp-server/rtsp-client.c:
8350 rtsp-client: set the client backlog
8351 Set the client backlog to a reasonable default
8353 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
8355 * gst/rtsp-server/rtsp-media.c:
8356 rtsp-media: Make the element a constructor parameter
8357 https://bugzilla.gnome.org/show_bug.cgi?id=689594
8359 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8361 * docs/libs/Makefile.am:
8362 docs: Link with gcov library when gcov is enabled
8363 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
8365 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8367 * gst/rtsp-server/rtsp-media.c:
8368 media: match prepare with unprepare
8369 Really unprepare when there were an equal amount of prepare calls.
8371 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8373 * gst/rtsp-server/rtsp-media.c:
8374 media: media has to be unprepared in finalize
8375 Because unprepare takes away the last ref on the media.
8377 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8379 * gst/rtsp-server/rtsp-client.c:
8380 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
8381 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
8382 We can't use the refcount to trigger unprepare because it is the unprepare call
8383 that removes the last refcount after all messages are consumed. What we should
8384 probably do is make a prepared refcount and only unprepare when the refcount
8387 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8389 * gst/rtsp-server/rtsp-media.c:
8390 media: let the source unref the last media ref
8391 the last ref to the media is held by the source so we don't need to add more ref
8392 and unrefs, we simply destroy the media when the source is gone.
8394 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8396 * gst/rtsp-server/rtsp-media.c:
8397 media: improve debug
8399 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8401 * gst/rtsp-server/rtsp-media.c:
8403 Make sure we are in the right state when collecting the position and duration.
8404 Only make ourselves PREPARED when we were previously PREPARING.
8406 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8408 * gst/rtsp-server/rtsp-media.c:
8409 media: use g_object_ref/unref for GObjects
8411 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
8413 * gst/rtsp-server/rtsp-client.c:
8414 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
8415 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
8416 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
8417 isn't being used anymore.
8419 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
8421 * gst/rtsp-server/rtsp-media.c:
8422 Fix compiler warning
8424 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
8426 * gst/rtsp-server/rtsp-media-factory-uri.c:
8427 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
8429 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8431 * gst/rtsp-server/rtsp-session-media.h:
8434 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8436 * gst/rtsp-server/rtsp-media.c:
8437 * tests/check/gst/media.c:
8438 media: avoid element leak
8440 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8442 * gst/rtsp-server/rtsp-media.c:
8443 media: require an element in media constructor
8445 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8447 * gst/rtsp-server/rtsp-client.c:
8448 Revert "client: TEARDOWN brings that state to Init again"
8449 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
8450 The object is already disposed, there is no point in setting the state.
8452 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8454 * gst/rtsp-server/rtsp-client.c:
8455 client: TEARDOWN brings that state to Init again
8457 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8459 * docs/libs/gst-rtsp-server-sections.txt:
8460 * examples/test-auth.c:
8461 * gst/rtsp-server/rtsp-auth.c:
8462 * gst/rtsp-server/rtsp-auth.h:
8463 * gst/rtsp-server/rtsp-client.c:
8464 * gst/rtsp-server/rtsp-client.h:
8465 * gst/rtsp-server/rtsp-media-factory-uri.c:
8466 * gst/rtsp-server/rtsp-media-factory-uri.h:
8467 * gst/rtsp-server/rtsp-media-factory.c:
8468 * gst/rtsp-server/rtsp-media-factory.h:
8469 * gst/rtsp-server/rtsp-media.c:
8470 * gst/rtsp-server/rtsp-media.h:
8471 * gst/rtsp-server/rtsp-mount-points.c:
8472 * gst/rtsp-server/rtsp-mount-points.h:
8473 * gst/rtsp-server/rtsp-sdp.c:
8474 * gst/rtsp-server/rtsp-server.c:
8475 * gst/rtsp-server/rtsp-server.h:
8476 * gst/rtsp-server/rtsp-session-media.c:
8477 * gst/rtsp-server/rtsp-session-media.h:
8478 * gst/rtsp-server/rtsp-session-pool.c:
8479 * gst/rtsp-server/rtsp-session-pool.h:
8480 * gst/rtsp-server/rtsp-session.c:
8481 * gst/rtsp-server/rtsp-session.h:
8482 * gst/rtsp-server/rtsp-stream-transport.c:
8483 * gst/rtsp-server/rtsp-stream-transport.h:
8484 * gst/rtsp-server/rtsp-stream.c:
8485 * gst/rtsp-server/rtsp-stream.h:
8486 * tests/check/gst/media.c:
8487 rtsp: make object details private
8488 Make all object details private
8489 Add methods to access private bits
8491 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8493 * tests/check/Makefile.am:
8494 * tests/check/gst/media.c:
8495 tests: add media tests
8497 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8499 * gst/rtsp-server/rtsp-media.c:
8500 media: check if prepared for some methods
8501 Check that the media object is prepared before doing seek and getting the
8502 current position etc.
8503 Add some g_return checks.
8505 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8507 * tests/check/Makefile.am:
8508 * tests/check/gst/mediafactory.c:
8509 tests: add mediafactory test
8511 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8513 * gst/rtsp-server/rtsp-stream.c:
8514 stream: improve debug
8516 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8518 * gst/rtsp-server/rtsp-media.c:
8519 * gst/rtsp-server/rtsp-media.h:
8520 media: unref pipeline in finalize to avoid leaking it
8522 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8524 * gst/rtsp-server/rtsp-media-factory-uri.c:
8525 * gst/rtsp-server/rtsp-media.c:
8526 rtsp: use gst_object_unref on GstObjects
8528 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8530 * gst/rtsp-server/rtsp-media-factory.c:
8531 media-factory: require an url
8533 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8535 * examples/test-uri.c:
8536 examples: fix include
8538 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8540 * gst/rtsp-server/rtsp-server.h:
8541 server: remove unused include
8543 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8545 * tests/check/Makefile.am:
8546 * tests/check/gst/mountpoints.c:
8547 tests: add test for mountpoints
8549 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8551 * gst/rtsp-server/rtsp-client.c:
8552 client: fix factory leak
8553 Keep the factory in the state object only for authorization checks and make
8554 sure we unref it on failure. Also don't keep invalid objects in the state
8557 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8559 * gst/rtsp-server/rtsp-mount-points.c:
8560 mounts: add g_return_if guards
8562 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8564 * tests/check/gst/client.c:
8565 tests: add more tests
8567 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8569 * gst/rtsp-server/rtsp-client.c:
8570 client: improve debug
8572 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8574 * gst/rtsp-server/rtsp-client.c:
8575 client: improve debug and fix leaks
8576 Cleanup the uri and session when there is a bad request.
8578 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8583 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8585 * tests/check/gst/client.c:
8586 test: add test for session in options request
8588 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8590 * gst/rtsp-server/rtsp-client.c:
8591 client: use 454 when session can't be found
8592 We should use 454 when a session can't be found because there was no session
8593 pool configured in the server. This is not a server configuration problem
8594 because the server on which the request is done might not be the same one that
8595 will keep the sessions for us and so it does not need to support sessions.
8597 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8599 * gst/rtsp-server/rtsp-client.c:
8600 client: only free connection when there is one
8601 It's possible that the client doesn't have a connection when we try to free it.
8603 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8605 * tests/check/Makefile.am:
8606 * tests/check/gst/client.c:
8607 tests: add unit test for the client object
8609 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8611 * gst/rtsp-server/rtsp-client.c:
8612 client: small cleanup
8614 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8616 * gst/rtsp-server/rtsp-client.h:
8617 client: remove unused include
8619 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8621 * gst/rtsp-server/rtsp-client.c:
8622 client: fix compilation
8624 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8626 * gst/rtsp-server/rtsp-client.c:
8627 client: call destroy without the lock
8629 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8631 * gst/rtsp-server/rtsp-client.c:
8632 * gst/rtsp-server/rtsp-client.h:
8633 client: make the client usable without a socket
8634 Make a method to let the client handle a message and a callback when the client
8635 wants us to send a response message back. This makes it possible to also use the
8636 client object without the sockets, which should make it easier to test.
8638 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8640 * gst/rtsp-server/rtsp-client.c:
8641 * gst/rtsp-server/rtsp-client.h:
8642 client: small cleanup
8644 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8646 * docs/libs/gst-rtsp-server-sections.txt:
8647 * gst/rtsp-server/rtsp-client.c:
8648 * gst/rtsp-server/rtsp-client.h:
8649 * gst/rtsp-server/rtsp-server.c:
8650 client: remove reference to server
8651 We don't need to keep a ref to the server
8653 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8655 * gst/rtsp-server/rtsp-client.c:
8656 * gst/rtsp-server/rtsp-client.h:
8658 Also add some g_return_if()
8660 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8662 * gst/rtsp-server/rtsp-client.c:
8663 client: log more errors
8665 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8667 * gst/rtsp-server/rtsp-client.c:
8668 client: fix compilation
8670 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8672 * gst/rtsp-server/rtsp-client.c:
8673 * gst/rtsp-server/rtsp-client.h:
8674 client: add generic close-after-send support
8675 Add a property to send_response() to close the connection after the response has
8676 been sent to the client.
8678 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8681 * docs/libs/gst-rtsp-server-docs.sgml:
8682 * docs/libs/gst-rtsp-server-sections.txt:
8683 * docs/libs/gst-rtsp-server.types:
8684 * examples/test-auth.c:
8685 * examples/test-launch.c:
8686 * examples/test-mp4.c:
8687 * examples/test-multicast.c:
8688 * examples/test-multicast2.c:
8689 * examples/test-ogg.c:
8690 * examples/test-readme.c:
8691 * examples/test-sdp.c:
8692 * examples/test-uri.c:
8693 * examples/test-video.c:
8694 * gst/rtsp-server/Makefile.am:
8695 * gst/rtsp-server/rtsp-auth.h:
8696 * gst/rtsp-server/rtsp-client.c:
8697 * gst/rtsp-server/rtsp-client.h:
8698 * gst/rtsp-server/rtsp-media-mapping.c:
8699 * gst/rtsp-server/rtsp-media-mapping.h:
8700 * gst/rtsp-server/rtsp-mount-points.c:
8701 * gst/rtsp-server/rtsp-mount-points.h:
8702 * gst/rtsp-server/rtsp-server.c:
8703 * gst/rtsp-server/rtsp-server.h:
8704 * gst/rtsp-server/rtsp-session-media.c:
8705 * gst/rtsp-server/rtsp-session-pool.c:
8706 * gst/rtsp-server/rtsp-session-pool.h:
8707 * tests/check/gst/rtspserver.c:
8708 MediaMapping -> MountPoints
8709 Describes better what the object manages.
8711 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8714 configure: bump required version of -base
8716 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8718 * gst/rtsp-server/rtsp-media.c:
8721 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8723 * gst/rtsp-server/rtsp-media.c:
8724 * gst/rtsp-server/rtsp-media.h:
8725 media: support more Range formats
8726 Use the new -base methods to convert the Range string into a seek start and stop
8729 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8731 * examples/test-launch.c:
8732 examples: fix whitespace
8734 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8736 * examples/test-auth.c:
8737 test-auth: add example of how to remove sessions
8738 Add an example of the session filter api.
8740 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8742 * examples/test-uri.c:
8743 test-uri: remove mapping example
8745 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8747 * examples/test-uri.c:
8748 test-uri: fix callback signature
8750 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8752 * gst/rtsp-server/rtsp-media-factory.c:
8753 factory: keep ref to factory while media active
8754 While the media from a factory is alive, keep a ref to the factory.
8755 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
8757 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8759 * gst/rtsp-server/rtsp-media-factory-uri.c:
8760 factory-uri: add some debug
8762 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8764 * gst/rtsp-server/rtsp-stream.c:
8765 stream: set udp sources to PLAYING
8766 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
8767 so that it doesn't cause our pipeline to produce ASYNC-DONE.
8769 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8771 * gst/rtsp-server/rtsp-media-factory-uri.c:
8772 factory-uri: take ref to factory
8773 Take a ref to the factory that we place in our list.
8775 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8777 * tests/Makefile.am:
8778 * tests/test-reuse.c:
8779 test: add test for server reuse
8780 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
8782 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
8784 * gst/rtsp-server/rtsp-server.c:
8785 server: start and stop multiple times
8786 Stop listening on the RTSP port when the GSource is removed, so clients
8787 can't connect and the server can be started again.
8788 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
8790 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8792 * gst/rtsp-server/rtsp-server.c:
8793 server: fix small leak
8795 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8797 * gst/rtsp-server/rtsp-media.c:
8798 media: unref source in finish_unprepare
8799 The source is created in prepare, unref it in finish_unprepare.
8800 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
8802 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
8804 * gst/rtsp-server/rtsp-client.c:
8805 * gst/rtsp-server/rtsp-media.c:
8806 rtsp-media: remove bus watch before finalizing
8807 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
8808 * An extra media ref is added for the bus watch. This extra ref is unreffed by
8809 the GDestroyNotify function.
8810 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
8811 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
8812 gst_rtsp_media_unprepare before unreffing the media.
8813 This way, the bus watch will be removed before the media is finalized.
8814 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
8816 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
8818 * gst/rtsp-server/rtsp-client.c:
8819 * gst/rtsp-server/rtsp-client.h:
8820 client: wait until the TEARDOWN response is sent to close the connection
8821 Responses can be sent async so we need to wait until the TEARDOWN response has
8822 been written before we close the connection to the client. This avoids the risk
8823 of writing/polling closed sockets.
8824 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
8826 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
8828 * gst/rtsp-server/rtsp-stream.c:
8829 rtsp-stream: plug socket leak
8830 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
8832 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
8835 Automatic update of common submodule
8836 From 6bb6951 to a72faea
8838 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
8840 * gst/rtsp-server/rtsp-media-factory-uri.c:
8841 rtsp-server: don't use deprecated API
8843 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
8845 * gst/rtsp-server/rtsp-client.c:
8846 rtsp-client: fix unused-but-set-variable compiler warning
8847 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
8849 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8852 * docs/libs/gst-rtsp-server-sections.txt:
8853 * gst/rtsp-server/rtsp-client.c:
8856 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8858 * examples/Makefile.am:
8859 * examples/test-multicast2.c:
8860 examples: add another multicast example
8861 Add an example for how to configure separate multicast ranges for each media
8864 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8866 * examples/test-multicast.c:
8869 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8871 * gst/rtsp-server/rtsp-client.c:
8872 * gst/rtsp-server/rtsp-media.c:
8873 * gst/rtsp-server/rtsp-session-media.c:
8874 * gst/rtsp-server/rtsp-session-media.h:
8875 * gst/rtsp-server/rtsp-stream-transport.c:
8876 * gst/rtsp-server/rtsp-stream-transport.h:
8877 stream: use the address managed by the stream
8878 Use the address managed by the stream for multicast. This allows us to have 1
8879 multicast address for each stream.
8880 Because the address is now managed by the stream we don't have to pass it around
8882 Set the address pool on the streams.
8884 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8886 * gst/rtsp-server/rtsp-client.c:
8887 * gst/rtsp-server/rtsp-media.c:
8888 * gst/rtsp-server/rtsp-stream.c:
8891 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8893 * gst/rtsp-server/rtsp-media.c:
8894 * gst/rtsp-server/rtsp-media.h:
8895 media: add signal for new streams
8896 This allows applications to listen for new streams and configure properties on
8897 them, like the address pool.
8899 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8901 * gst/rtsp-server/rtsp-media.c:
8902 media: configure address pool in new streams
8904 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8906 * gst/rtsp-server/rtsp-stream.c:
8907 * gst/rtsp-server/rtsp-stream.h:
8908 stream: add methods to deal with address pool
8909 Add methods to get and set the address pool for the stream
8910 Add method to allocate and get the multicast addresses for this stream.
8912 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8914 * docs/libs/gst-rtsp-server-sections.txt:
8915 * gst/rtsp-server/rtsp-media.c:
8916 * gst/rtsp-server/rtsp-media.h:
8917 media: remove MTU property
8918 It is a stream property
8920 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8922 * gst/rtsp-server/rtsp-client.c:
8923 client: set blocksize only on stream
8924 Set the blocksize only on the current stream.
8926 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8928 * gst/rtsp-server/rtsp-stream.c:
8929 stream: share src and sink sockets
8930 the allocated socket is in the used-socket property, not socket.
8932 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8934 * gst/rtsp-server/rtsp-address-pool.c:
8935 * gst/rtsp-server/rtsp-address-pool.h:
8936 * gst/rtsp-server/rtsp-client.c:
8937 * gst/rtsp-server/rtsp-session-media.c:
8938 * gst/rtsp-server/rtsp-session-media.h:
8939 * gst/rtsp-server/rtsp-stream-transport.c:
8940 * gst/rtsp-server/rtsp-stream-transport.h:
8941 * tests/check/gst/addresspool.c:
8942 rtsp: make address-pool return an address object
8943 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
8944 store more info in the structure and allows us to more easily return the address
8945 to the right pool when no longer needed.
8946 Pass the address to the StreamTransport so that we can return it to the pool
8947 when the stream transport is freed or changed.
8949 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8951 * examples/Makefile.am:
8952 * examples/test-multicast.c:
8953 examples: add multicast example
8954 Show how to set up the multicast address pool so that media can be
8955 server with multicast.
8957 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8959 * gst/rtsp-server/rtsp-client.c:
8960 * gst/rtsp-server/rtsp-media-factory.c:
8961 * gst/rtsp-server/rtsp-media-factory.h:
8962 * gst/rtsp-server/rtsp-media.c:
8963 * gst/rtsp-server/rtsp-media.h:
8964 rtsp: use AddressPool
8965 Remove the multicast_group property.
8966 Use the configured addresspool to allocate multicast addresses.
8968 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8970 * gst/rtsp-server/rtsp-address-pool.c:
8971 * gst/rtsp-server/rtsp-address-pool.h:
8972 address-pool: add clear method
8974 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8976 * gst/rtsp-server/rtsp-address-pool.c:
8977 address-pool: small cleanups
8979 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8981 * tests/check/Makefile.am:
8982 * tests/check/gst/addresspool.c:
8983 tests: add addresspool unit test
8985 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8987 * gst/rtsp-server/Makefile.am:
8988 * gst/rtsp-server/rtsp-address-pool.c:
8989 * gst/rtsp-server/rtsp-address-pool.h:
8990 address-pool: add object to manage multicast addresses
8991 Make an object that can manage a rage of multicast addresses and ports.
8993 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8995 * gst/rtsp-server/rtsp-server.c:
8996 server: set default max-threads property
8998 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9000 * gst/rtsp-server/rtsp-media.c:
9001 media: wait for concurrent _prepare
9002 If a prepare is busy, wait for the result.
9004 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9006 * gst/rtsp-server/rtsp-media.c:
9007 media: add lock around message handler
9008 We don't want to dispatch messages while we are still processing the result of
9011 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9013 * gst/rtsp-server/rtsp-media.c:
9014 * gst/rtsp-server/rtsp-media.h:
9015 media: add lock to protect state changes
9017 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9019 * gst/rtsp-server/rtsp-stream.c:
9020 * gst/rtsp-server/rtsp-stream.h:
9023 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9025 * gst/rtsp-server/rtsp-stream-transport.c:
9026 * gst/rtsp-server/rtsp-stream-transport.h:
9027 * gst/rtsp-server/rtsp-stream.c:
9028 stream-transport: add keep-alive method
9030 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9032 * gst/rtsp-server/rtsp-stream-transport.c:
9033 * gst/rtsp-server/rtsp-stream-transport.h:
9034 * gst/rtsp-server/rtsp-stream.c:
9035 stream-transport: add method to handle RTP/RTCP
9036 Call new methods instead of poking into the structures directly.
9038 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9040 * gst/rtsp-server/rtsp-session-media.c:
9041 * gst/rtsp-server/rtsp-session-media.h:
9042 session-media: add locking
9044 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9046 * gst/rtsp-server/rtsp-session.c:
9047 * gst/rtsp-server/rtsp-session.h:
9048 session: add locking
9050 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9052 * gst/rtsp-server/rtsp-server.c:
9053 server: free old socket
9055 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9057 * gst/rtsp-server/rtsp-media-mapping.c:
9058 * gst/rtsp-server/rtsp-media-mapping.h:
9059 mapping: add locking
9061 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9063 * gst/rtsp-server/rtsp-media-factory.c:
9064 media-factory: add locking
9066 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9068 * gst/rtsp-server/rtsp-auth.c:
9069 * gst/rtsp-server/rtsp-auth.h:
9072 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9074 * gst/rtsp-server/rtsp-server.c:
9075 * gst/rtsp-server/rtsp-server.h:
9076 server: add max-thread property
9078 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9080 * gst/rtsp-server/rtsp-server.c:
9081 * gst/rtsp-server/rtsp-server.h:
9082 server: use a threadpool for the mainloops
9084 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9086 * gst/rtsp-server/rtsp-client.c:
9087 * gst/rtsp-server/rtsp-client.h:
9088 client: rename method
9089 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
9090 don't really create the client from the socket, we use the socket for the
9093 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9095 * gst/rtsp-server/rtsp-client.c:
9096 * gst/rtsp-server/rtsp-client.h:
9097 * gst/rtsp-server/rtsp-server.c:
9098 server: rework maincontext handling in clients
9099 Make a separate method to attach a client to a MainContext.
9100 Let the server decide in what GMainContext the client will operate and give this
9101 context to the client in attach. Then the server can later decide to use a
9102 separate thread for each client or just use the mainthread.
9104 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9106 * gst/rtsp-server/rtsp-client.c:
9107 * gst/rtsp-server/rtsp-session.c:
9108 * gst/rtsp-server/rtsp-session.h:
9109 session: move session header code in session object
9111 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
9115 * examples/test-auth.c:
9116 * examples/test-launch.c:
9117 * examples/test-mp4.c:
9118 * examples/test-ogg.c:
9119 * examples/test-readme.c:
9120 * examples/test-sdp.c:
9121 * examples/test-uri.c:
9122 * examples/test-video.c:
9123 * gst/rtsp-server/rtsp-auth.c:
9124 * gst/rtsp-server/rtsp-auth.h:
9125 * gst/rtsp-server/rtsp-client.c:
9126 * gst/rtsp-server/rtsp-client.h:
9127 * gst/rtsp-server/rtsp-media-factory-uri.c:
9128 * gst/rtsp-server/rtsp-media-factory-uri.h:
9129 * gst/rtsp-server/rtsp-media-factory.c:
9130 * gst/rtsp-server/rtsp-media-factory.h:
9131 * gst/rtsp-server/rtsp-media-mapping.c:
9132 * gst/rtsp-server/rtsp-media-mapping.h:
9133 * gst/rtsp-server/rtsp-media.c:
9134 * gst/rtsp-server/rtsp-media.h:
9135 * gst/rtsp-server/rtsp-params.c:
9136 * gst/rtsp-server/rtsp-params.h:
9137 * gst/rtsp-server/rtsp-sdp.c:
9138 * gst/rtsp-server/rtsp-sdp.h:
9139 * gst/rtsp-server/rtsp-server.c:
9140 * gst/rtsp-server/rtsp-server.h:
9141 * gst/rtsp-server/rtsp-session-media.c:
9142 * gst/rtsp-server/rtsp-session-media.h:
9143 * gst/rtsp-server/rtsp-session-pool.c:
9144 * gst/rtsp-server/rtsp-session-pool.h:
9145 * gst/rtsp-server/rtsp-session.c:
9146 * gst/rtsp-server/rtsp-session.h:
9147 * gst/rtsp-server/rtsp-stream-transport.c:
9148 * gst/rtsp-server/rtsp-stream-transport.h:
9149 * gst/rtsp-server/rtsp-stream.c:
9150 * gst/rtsp-server/rtsp-stream.h:
9151 * tests/check/gst/rtspserver.c:
9152 * tests/test-cleanup.c:
9155 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9157 * gst/rtsp-server/rtsp-media.c:
9158 * gst/rtsp-server/rtsp-session-media.c:
9159 * gst/rtsp-server/rtsp-session.c:
9160 rtsp-server: added annotations to indicate type of ownership transfer of return values
9161 https://bugzilla.gnome.org/show_bug.cgi?id=680777
9163 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
9166 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
9168 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
9171 * bindings/Makefile.am:
9172 * bindings/vala/Makefile.am:
9173 * bindings/vala/gst-rtsp-server-0.10.deps:
9174 * bindings/vala/gst-rtsp-server-0.10.vapi:
9175 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
9176 * bindings/vala/packages/gst-rtsp-server-0.10.files:
9177 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9178 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9179 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
9181 bindings: remove vala bindings
9182 They'll be reunited with the other GStreamer bindings
9183 https://bugzilla.gnome.org/show_bug.cgi?id=680777
9185 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9187 * gst/rtsp-server/rtsp-client.c:
9188 * gst/rtsp-server/rtsp-session-media.c:
9189 * gst/rtsp-server/rtsp-session-media.h:
9190 * gst/rtsp-server/rtsp-stream-transport.c:
9191 * gst/rtsp-server/rtsp-stream-transport.h:
9192 rtsp: only create transport when needed
9193 Only create the StreamTransport when configured.
9195 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9197 * gst/rtsp-server/rtsp-client.c:
9198 client: small cleanup
9200 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9202 * gst/rtsp-server/rtsp-client.c:
9203 * gst/rtsp-server/rtsp-client.h:
9204 * gst/rtsp-server/rtsp-stream-transport.c:
9205 * gst/rtsp-server/rtsp-stream-transport.h:
9206 rtsp: refactor configuration of transport
9207 Move the configuration of the transport to a place where it makes
9210 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9212 * gst/rtsp-server/rtsp-client.c:
9213 client: refactor transport parsing
9215 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9217 * gst/rtsp-server/rtsp-client.c:
9218 client: refuse to change the MTU on shared media
9219 If we change the MTU of chared media, it changes for all clients.
9220 We don't want to set the MTU to something large for clients that
9223 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9225 * examples/test-mp4.c:
9226 * gst/rtsp-server/rtsp-media.c:
9227 small fixes to docs and debug
9229 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9231 * gst/rtsp-server/rtsp-stream.c:
9232 stream: transports must already have been removed
9234 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9236 * gst/rtsp-server/rtsp-media.c:
9237 * gst/rtsp-server/rtsp-stream.c:
9238 * gst/rtsp-server/rtsp-stream.h:
9239 stream: improve join and leave of the pipeline
9241 Do the cleanup properly
9244 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9246 * gst/rtsp-server/rtsp-media.c:
9247 media: move unprepare below default implementation
9248 Makes it easier to find the default implementation
9250 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9252 * gst/rtsp-server/rtsp-media.c:
9253 media: signal unprepared when we actually finish
9255 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9257 * gst/rtsp-server/rtsp-media.c:
9258 media: no need to unlock, unprepare does that when needed
9260 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9262 * docs/libs/gst-rtsp-server-sections.txt:
9263 * gst/rtsp-server/rtsp-media-factory.h:
9264 * gst/rtsp-server/rtsp-media-mapping.c:
9265 * gst/rtsp-server/rtsp-media.h:
9266 * gst/rtsp-server/rtsp-params.c:
9267 * gst/rtsp-server/rtsp-server.c:
9268 * gst/rtsp-server/rtsp-session-pool.h:
9269 * gst/rtsp-server/rtsp-session.c:
9270 * gst/rtsp-server/rtsp-session.h:
9271 * gst/rtsp-server/rtsp-stream-transport.h:
9272 * gst/rtsp-server/rtsp-stream.h:
9275 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9277 * gst/rtsp-server/rtsp-client.c:
9278 * gst/rtsp-server/rtsp-media-mapping.h:
9279 * gst/rtsp-server/rtsp-media.c:
9280 * gst/rtsp-server/rtsp-media.h:
9281 * gst/rtsp-server/rtsp-server.h:
9282 * gst/rtsp-server/rtsp-stream.c:
9283 * gst/rtsp-server/rtsp-stream.h:
9284 rtsp: fix MTU setting
9285 Fix setting of the MTU. There is no need for a vmethod.
9287 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9292 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9295 configure: bump version number after refactoring
9297 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9299 * gst/rtsp-server/Makefile.am:
9300 * gst/rtsp-server/rtsp-client.c:
9301 * gst/rtsp-server/rtsp-client.h:
9302 * gst/rtsp-server/rtsp-media-factory-uri.c:
9303 * gst/rtsp-server/rtsp-media-factory.c:
9304 * gst/rtsp-server/rtsp-media-factory.h:
9305 * gst/rtsp-server/rtsp-media.c:
9306 * gst/rtsp-server/rtsp-media.h:
9307 * gst/rtsp-server/rtsp-sdp.c:
9308 * gst/rtsp-server/rtsp-session-media.c:
9309 * gst/rtsp-server/rtsp-session-media.h:
9310 * gst/rtsp-server/rtsp-session.c:
9311 * gst/rtsp-server/rtsp-session.h:
9312 * gst/rtsp-server/rtsp-stream-transport.c:
9313 * gst/rtsp-server/rtsp-stream-transport.h:
9314 * gst/rtsp-server/rtsp-stream.c:
9315 * gst/rtsp-server/rtsp-stream.h:
9316 rtsp: massive refactoring
9317 Make GObjects from the remaining simple structures.
9318 Remove GstRTSPSessionStream, it's not needed.
9319 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
9320 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
9321 a GstRTSPStream should be transported to a client.
9322 Rename GstRTSPMediaFactory::get_element -> create_element because that
9323 more accurately describes what it does.
9324 Make nice methods instead of poking in the structures.
9325 Move some methods inside the relevant object source code.
9326 Use GPtrArray to store objects instead of plain arrays, it is more
9327 natural and allows us to more easily clean up.
9328 Move the allocation of udp ports to the Stream object. The Stream object
9329 contains the elements needed to stream the media to a client.
9330 Improve the prepare and unprepare methods. Unprepare should now undo
9331 everything prepare did. Improve also async unprepare when doing EOS on
9332 shutdown. Make sure we always unprepare correctly.
9334 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
9336 * gst/rtsp-server/rtsp-client.c:
9337 rtsp-client: Unref server address clients connected to
9338 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
9340 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
9342 * gst/rtsp-server/rtsp-server.c:
9343 rtsp-server: don't ref server socket if it is NULL
9344 Fixes test_bind_already_in_use unit test again after commit 6a497440.
9345 https://bugzilla.gnome.org/show_bug.cgi?id=686644
9347 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
9349 * tests/check/Makefile.am:
9350 tests: Add libgio link dependency
9351 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
9353 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9355 * gst/rtsp-server/rtsp-media-mapping.c:
9356 * gst/rtsp-server/rtsp-media-mapping.h:
9357 rtsp-media-mapping: rename find_media vfunc to find_factory
9358 The virtual method and class method should have the same name
9359 so it is correctly represented in GIR file
9360 https://bugzilla.gnome.org/show_bug.cgi?id=680777
9362 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9364 * gst/rtsp-server/rtsp-auth.c:
9365 * gst/rtsp-server/rtsp-client.c:
9366 * gst/rtsp-server/rtsp-media-factory-uri.c:
9367 * gst/rtsp-server/rtsp-media-factory.c:
9368 * gst/rtsp-server/rtsp-media-mapping.c:
9369 * gst/rtsp-server/rtsp-media.c:
9370 * gst/rtsp-server/rtsp-server.c:
9371 * gst/rtsp-server/rtsp-session-pool.c:
9372 * gst/rtsp-server/rtsp-session.c:
9373 rtsp-server: fixed comments and GIR annotations
9374 https://bugzilla.gnome.org/show_bug.cgi?id=680777
9376 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
9378 * gst/rtsp-server/rtsp-media-mapping.c:
9379 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
9381 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
9383 * gst/rtsp-server/rtsp-server.c:
9384 rtsp-server: allow binding on port 0 (binds on a random port)
9386 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
9388 * gst/rtsp-server/rtsp-server.c:
9389 * gst/rtsp-server/rtsp-server.h:
9390 rtsp-server: add bound-port property
9391 bound-port can be used to retrieve the port number when the server is bound on
9392 port 0, which binds on a random port.
9394 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
9396 * gst/rtsp-server/rtsp-media-factory.c:
9397 * gst/rtsp-server/rtsp-media-factory.h:
9398 rtsp-media-factory: make ::get_element overridable by GI bindings
9399 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
9400 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
9401 as the invoker for ::get_element(), making it overridable by GI generated
9404 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
9406 * gst/rtsp-server/rtsp-media-factory-uri.c:
9407 rtsp-media-factory-uri: don't autoplug parsers in a loop
9408 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
9411 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
9413 * gst/rtsp-server/Makefile.am:
9414 Explicitly link against gio. Fix link error on mac.
9416 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
9418 * gst/rtsp-server/rtsp-session.c:
9419 session: add ttl to the transport header in SETUP
9420 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
9422 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
9424 * gst/rtsp-server/rtsp-client.c:
9425 * gst/rtsp-server/rtsp-client.h:
9426 * gst/rtsp-server/rtsp-media.c:
9427 client: Use client transport settings for multicast if allowed.
9428 This patch makes it possible for the client to send transport settings for
9429 multicast (destination && ttl). Client settings must be explicitly allowed or
9430 the server will use its own settings.
9431 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
9433 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
9436 Automatic update of common submodule
9437 From 6c0b52c to 6bb6951
9439 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
9441 * gst/rtsp-server/rtsp-client.c:
9442 rtsp-client: do not destroy the rtsp watch
9443 Don't destroy the client watch while dispatching. The rtsp watch is
9444 automatically destroyed after the rtsp watch function closed() has
9446 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
9448 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
9451 Automatic update of common submodule
9452 From 4f962f7 to 6c0b52c
9454 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
9456 * gst/rtsp-server/rtsp-media.c:
9457 media: fix check for seekability
9459 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9461 * gst/rtsp-server/rtsp-client.c:
9462 client: use more GIO
9463 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
9465 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9467 * gst/rtsp-server/rtsp-server.c:
9468 server: remove obsolete includes
9470 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
9472 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
9473 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
9474 be available in "on_new_ssrc". The transports are added in
9475 gst_rtsp_media_set_state when going to PLAYING state. However,
9476 "on_new_ssrc" might be called before this happens.
9477 https://bugzilla.gnome.org/show_bug.cgi?id=683304
9479 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
9481 * gst/rtsp-server/rtsp-client.c:
9482 * gst/rtsp-server/rtsp-client.h:
9483 rtsp-client: add signals for rtsp requests (fixes #683287)
9485 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
9487 * gst/rtsp-server/rtsp-client.c:
9488 * gst/rtsp-server/rtsp-client.h:
9489 add new-session signal to rtsp-client (fixes #683058)
9491 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
9494 Automatic update of common submodule
9495 From 668acee to 4f962f7
9497 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
9499 * gst/rtsp-server/rtsp-server.c:
9500 * tests/check/gst/rtspserver.c:
9501 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
9502 Do not assume that *error is set in g_socket_address_enumerator_next.
9503 Added test_bind_already_in_use unit-test.
9504 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
9506 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
9509 Automatic update of common submodule
9510 From 94ccf4c to 668acee
9512 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
9514 * gst/rtsp-server/rtsp-client.c:
9515 * gst/rtsp-server/rtsp-client.h:
9516 rtsp-client: make create_sdp virtual method
9517 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
9519 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9522 Automatic update of common submodule
9523 From 98e386f to 94ccf4c
9525 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9527 * gst/rtsp-server/rtsp-client.c:
9530 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
9532 * gst/rtsp-server/rtsp-client.c:
9533 * gst/rtsp-server/rtsp-client.h:
9534 * gst/rtsp-server/rtsp-server.c:
9535 * gst/rtsp-server/rtsp-server.h:
9536 rtsp-server: use an existing socket to establish HTTP tunnel
9537 Make it possible to transfer a socket from an HTTP server to be used as
9538 an RTSP over HTTP tunnel.
9540 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
9542 * gst/rtsp-server/rtsp-client.c:
9543 * gst/rtsp-server/rtsp-media.c:
9544 * gst/rtsp-server/rtsp-media.h:
9545 rtsp: Handle the blocksize parameter
9546 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
9548 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
9550 * tests/check/Makefile.am:
9551 * tests/check/gst/rtspserver.c:
9552 Have unit test get header from source dir, not installed dir
9553 This makes compilation of unit tests work in a build directory other
9554 than the source directory.
9555 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
9557 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
9559 * gst/rtsp-server/rtsp-media.c:
9560 rtsp-media: update for gst_element_make_from_uri() changes
9562 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
9565 * tests/Makefile.am:
9566 * tests/check/Makefile.am:
9567 * tests/check/gst/rtspserver.c:
9569 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
9571 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
9573 * gst/rtsp-server/rtsp-media.c:
9574 rtsp-media: don't collect media stats when going to NULL
9575 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
9577 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9579 * gst/rtsp-server/rtsp-client.c:
9580 client: don't leak transports
9582 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
9584 * gst/rtsp-server/rtsp-client.c:
9585 rtsp-client: free transport on no_stream in SETUP handler
9587 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
9589 * gst/rtsp-server/rtsp-client.c:
9590 rtsp-client: changed session media iteration
9591 In client_unlink_session: now don't iterate in session->medias
9592 list where items are removed by gst_rtsp_session_release_media.
9593 Instead, repeatedly remove the first item.
9595 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
9597 * gst/rtsp-server/rtsp-client.c:
9598 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
9599 GstRTSPSessionMedia is not a GObject type. When the
9600 GstRTSPSession is freed, it will free the media.
9602 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
9604 * gst/rtsp-server/rtsp-media-factory.c:
9605 factory: plug pad leak in collect_streams
9606 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
9607 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
9608 will take one reference, and the other reference will otherwise
9611 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
9614 configure: suppress some warnings when debug is disabled
9615 Warnings about unused variables should be suppressed if core has the
9616 debug system disabled.
9617 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
9619 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9621 * docs/libs/Makefile.am:
9622 docs: fix build in uninstalled setup
9623 Include gst-plugins-base libs properly.
9625 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
9627 * docs/libs/gst-rtsp-server.types:
9628 docs: include headers defining rtsp-server object types
9629 Fixes compiler warnings during docs build.
9630 https://bugzilla.gnome.org/show_bug.cgi?id=676824
9632 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
9635 configure: Add warning flags for compiler when configuring
9636 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
9638 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
9641 Automatic update of common submodule
9642 From 03a0e57 to 98e386f
9644 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
9647 Automatic update of common submodule
9648 From 1fab359 to 03a0e57
9650 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
9652 * gst/rtsp-server/rtsp-client.c:
9653 client: fix GSocketAddress leak in gst_rtsp_client_accept
9654 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
9656 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
9659 Automatic update of common submodule
9660 From f1b5a96 to 1fab359
9662 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9665 Automatic update of common submodule
9666 From 92b7266 to f1b5a96
9668 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9671 Automatic update of common submodule
9672 From ec1c4a8 to 92b7266
9674 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9677 Automatic update of common submodule
9678 From 3429ba6 to ec1c4a8
9680 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
9682 * gst/rtsp-server/rtsp-auth.c:
9683 * gst/rtsp-server/rtsp-client.c:
9684 * gst/rtsp-server/rtsp-media-factory-uri.c:
9685 * gst/rtsp-server/rtsp-server.c:
9686 rtsp: fix compiler warnings
9687 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
9689 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9692 Automatic update of common submodule
9693 From dc70203 to 3429ba6
9695 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9697 * gst/rtsp-server/rtsp-client.c:
9698 * gst/rtsp-server/rtsp-media-factory.c:
9699 * gst/rtsp-server/rtsp-media-factory.h:
9700 * gst/rtsp-server/rtsp-media.c:
9701 * gst/rtsp-server/rtsp-media.h:
9702 * gst/rtsp-server/rtsp-server.c:
9703 * gst/rtsp-server/rtsp-server.h:
9704 * gst/rtsp-server/rtsp-session-pool.c:
9705 * gst/rtsp-server/rtsp-session-pool.h:
9706 rtsp-server: port to new thread API
9708 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9711 Automatic update of common submodule
9712 From 6db25be to dc70203
9714 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9716 * gst/rtsp-server/rtsp-auth.c:
9717 * gst/rtsp-server/rtsp-auth.h:
9718 * gst/rtsp-server/rtsp-client.c:
9719 rtsp-server: Fix compilation and compiler warnings
9721 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9725 * gst/rtsp-server/Makefile.am:
9726 configure: Modernize autotools setup a bit
9727 Also we now only create tar.bz2 and tar.xz tarballs.
9729 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9732 Automatic update of common submodule
9733 From 464fe15 to 6db25be
9735 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9738 Automatic update of common submodule
9739 From 7fda524 to 464fe15
9741 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9744 * docs/libs/Makefile.am:
9745 * docs/version.entities.in:
9747 * gst/rtsp-server/Makefile.am:
9748 * pkgconfig/Makefile.am:
9749 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
9750 * pkgconfig/gstreamer-rtsp-server.pc.in:
9751 * tests/Makefile.am:
9752 rtsp-server: Update versioning
9754 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9756 Merge remote-tracking branch 'origin/0.10'
9758 gst/rtsp-server/rtsp-session-pool.c
9760 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9762 * gst/rtsp-server/rtsp-session-pool.c:
9763 rtsp-server: Don't use deprecated GLib API
9765 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9767 Replace master with 0.11
9769 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9771 Merge branch 'master' into 0.11
9773 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9775 Merge branch 'master' into 0.11
9777 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
9780 A couple minor typo fixes
9782 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9784 * gst/rtsp-server/rtsp-media.c:
9785 media: fix state of the appqueue
9787 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9789 * gst/rtsp-server/rtsp-media-factory-uri.c:
9790 factory: use videoconvert
9792 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9794 * gst/rtsp-server/rtsp-media-factory-uri.c:
9795 factory: change to new style caps
9797 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9799 * gst/rtsp-server/rtsp-client.c:
9800 * gst/rtsp-server/rtsp-client.h:
9801 * gst/rtsp-server/rtsp-media-factory-uri.c:
9802 * gst/rtsp-server/rtsp-media.c:
9803 * gst/rtsp-server/rtsp-server.c:
9804 * gst/rtsp-server/rtsp-server.h:
9805 * gst/rtsp-server/rtsp-session-pool.c:
9806 rtsp-server: port to GIO
9809 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9812 configure: fix build
9814 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9817 docs: fix for gst_rtsp_server_set_port() -> _set_service()
9818 https://bugzilla.gnome.org/show_bug.cgi?id=666548
9820 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9823 * examples/Makefile.am:
9824 First rule of gst-rtsp-server club: don't talk about gst-phonon
9826 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9829 * pkgconfig/Makefile.am:
9830 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
9831 * pkgconfig/gstreamer-rtsp-server.pc.in:
9832 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
9833 For consistency with all other modules.
9835 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9837 * gst/rtsp-server/rtsp-client.c:
9838 rtsp-client: update for new map API
9840 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9843 * bindings/Makefile.am:
9844 * bindings/python/Makefile.am:
9845 * bindings/python/arg-types.py:
9846 * bindings/python/codegen/Makefile.am:
9847 * bindings/python/codegen/__init__.py:
9848 * bindings/python/codegen/argtypes.py:
9849 * bindings/python/codegen/code-coverage.py:
9850 * bindings/python/codegen/codegen.py:
9851 * bindings/python/codegen/definitions.py:
9852 * bindings/python/codegen/defsparser.py:
9853 * bindings/python/codegen/docextract.py:
9854 * bindings/python/codegen/docgen.py:
9855 * bindings/python/codegen/fileprefix.override:
9856 * bindings/python/codegen/fileprefixmodule.c:
9857 * bindings/python/codegen/h2def.py:
9858 * bindings/python/codegen/mergedefs.py:
9859 * bindings/python/codegen/mkskel.py:
9860 * bindings/python/codegen/override.py:
9861 * bindings/python/codegen/reversewrapper.py:
9862 * bindings/python/codegen/scmexpr.py:
9863 * bindings/python/rtspserver-types.defs:
9864 * bindings/python/rtspserver.defs:
9865 * bindings/python/rtspserver.override:
9866 * bindings/python/rtspservermodule.c:
9867 * bindings/python/test.py:
9869 python: remove pygst-based python bindings
9870 pygi is the future, apparently.
9872 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
9875 Automatic update of common submodule
9876 From c463bc0 to 7fda524
9878 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9881 Automatic update of common submodule
9882 From 2a59016 to c463bc0
9884 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9887 Automatic update of common submodule
9888 From 0807187 to 2a59016
9890 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9893 Automatic update of common submodule
9894 From 11f0cd5 to 0807187
9896 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9898 * examples/test-auth.c:
9899 example: update for new caps
9901 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9903 * examples/test-video.c:
9904 * gst/rtsp-server/rtsp-client.c:
9905 * gst/rtsp-server/rtsp-media-factory-uri.c:
9906 * gst/rtsp-server/rtsp-media.c:
9907 * gst/rtsp-server/rtsp-media.h:
9908 * gst/rtsp-server/rtsp-session.c:
9909 * gst/rtsp-server/rtsp-session.h:
9910 rtsp-server: port some more to 0.11
9912 Remove bufferlist stuff
9914 Add queue before appsink now that preroll-queue-len is gone.
9915 Update for request pad changes.
9917 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9919 Merge branch 'master' into 0.11
9921 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
9923 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9924 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
9925 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
9927 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
9929 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9930 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
9931 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
9933 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9935 Merge branch 'master' into 0.11
9937 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9939 * gst/rtsp-server/rtsp-media.c:
9940 * gst/rtsp-server/rtsp-media.h:
9941 media: add a seekable boolean
9942 Maintain the seekable state with a new variable instead of reusing the
9945 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
9947 * gst/rtsp-server/rtsp-media.c:
9948 Disallow seek in live media
9950 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9952 Merge branch 'master' into 0.11
9954 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
9956 * gst/rtsp-server/rtsp-server.c:
9957 #ifdef statements for windows socket creation were missing
9959 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
9962 Automatic update of common submodule
9963 From a39eb83 to 11f0cd5
9965 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
9968 Automatic update of common submodule
9969 From 605cd9a to a39eb83
9971 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9973 Merge branch 'master' into 0.11
9975 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9977 * gst/rtsp-server/rtsp-client.c:
9978 client: use method to access property
9980 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9982 * gst/rtsp-server/rtsp-media-factory.c:
9983 * gst/rtsp-server/rtsp-media-factory.h:
9984 media-factory: add protocols property
9985 Add a property to configure the allowed protocols in the media created from the
9988 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9990 * gst/rtsp-server/rtsp-media-factory.c:
9991 * gst/rtsp-server/rtsp-media-factory.h:
9992 media-factory: add media-configure signal
9993 Add signal to allow the application to configure the media after it was created
9996 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9998 * gst/rtsp-server/rtsp-client.c:
9999 client: use method to access property
10001 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10003 * gst/rtsp-server/rtsp-media-factory.c:
10004 * gst/rtsp-server/rtsp-media-factory.h:
10005 media-factory: add protocols property
10006 Add a property to configure the allowed protocols in the media created from the
10009 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10011 * gst/rtsp-server/rtsp-media-factory.c:
10012 * gst/rtsp-server/rtsp-media-factory.h:
10013 media-factory: add media-configure signal
10014 Add signal to allow the application to configure the media after it was created
10017 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10019 Merge branch 'master' into 0.11
10021 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10023 * gst/rtsp-server/rtsp-client.c:
10024 client: use media multicast group
10026 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10028 * gst/rtsp-server/rtsp-media-factory.h:
10029 * gst/rtsp-server/rtsp-server.h:
10030 * gst/rtsp-server/rtsp-session-pool.h:
10031 * gst/rtsp-server/rtsp-session.h:
10034 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
10036 * gst/rtsp-server/rtsp-client.c:
10037 * gst/rtsp-server/rtsp-sdp.h:
10038 sdp: copy and free the server ip address
10039 Copy and free the server ip address to make memory management easier later.
10041 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10043 * gst/rtsp-server/rtsp-media-factory.c:
10044 media-factory: configure multicast in media
10046 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10048 * gst/rtsp-server/rtsp-media.c:
10049 * gst/rtsp-server/rtsp-media.h:
10050 media: add property for multicast group
10051 Add a property to configure the multicast group in the media.
10052 Based on patches from Marc Leeman and Robert Krakora.
10054 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10056 * gst/rtsp-server/rtsp-media-factory.c:
10057 * gst/rtsp-server/rtsp-media-factory.h:
10058 media-factory: add property for multicast group
10059 Add a property to configure the multicast group in the media factory.
10060 Based on patches from Marc Leeman and Robert Krakora.
10062 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10064 * gst/rtsp-server/rtsp-client.c:
10065 client: do configuration of transport in one place
10066 Move the configuration of the transport destination address to where we also
10067 configure the other bits.
10069 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10071 * gst/rtsp-server/rtsp-client.c:
10072 client: use media multicast group
10074 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10076 * gst/rtsp-server/rtsp-media-factory.h:
10077 * gst/rtsp-server/rtsp-server.h:
10078 * gst/rtsp-server/rtsp-session-pool.h:
10079 * gst/rtsp-server/rtsp-session.h:
10082 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
10084 * gst/rtsp-server/rtsp-client.c:
10085 * gst/rtsp-server/rtsp-sdp.h:
10086 sdp: copy and free the server ip address
10087 Copy and free the server ip address to make memory management easier later.
10089 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10091 * gst/rtsp-server/rtsp-media-factory.c:
10092 media-factory: configure multicast in media
10094 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10096 * gst/rtsp-server/rtsp-media.c:
10097 * gst/rtsp-server/rtsp-media.h:
10098 media: add property for multicast group
10099 Add a property to configure the multicast group in the media.
10100 Based on patches from Marc Leeman and Robert Krakora.
10102 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10104 * gst/rtsp-server/rtsp-media-factory.c:
10105 * gst/rtsp-server/rtsp-media-factory.h:
10106 media-factory: add property for multicast group
10107 Add a property to configure the multicast group in the media factory.
10108 Based on patches from Marc Leeman and Robert Krakora.
10110 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10112 * gst/rtsp-server/rtsp-client.c:
10113 client: do configuration of transport in one place
10114 Move the configuration of the transport destination address to where we also
10115 configure the other bits.
10117 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10119 Merge branch 'master' into 0.11
10121 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
10123 * gst/rtsp-server/rtsp-client.c:
10124 client: destroy pipeline on client disconnect with no prior TEARDOWN.
10125 The problem occurs when the client abruptly closes the connection without
10126 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
10127 server is where the pipeline gets torn down. Since this handler is not called,
10128 the pipeline remains and is up and running. Subsequent clients get their own
10129 pipelines and if the do not issue TEARDOWNs then those pipelines will also
10130 remain up and running. This is a resource leak.
10132 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10134 Merge branch 'master' into 0.11
10136 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
10138 * gst/rtsp-server/rtsp-media-factory.c:
10139 * gst/rtsp-server/rtsp-media-factory.h:
10140 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
10141 For example, it can be used to retrieve source elements like appsrc, in a more
10142 convenient way than subclassing get_element.
10144 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10146 Merge branch 'master' into 0.11
10148 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
10150 * gst/rtsp-server/rtsp-server.c:
10151 rtsp-server: hold on to reference while using object
10153 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10155 * gst/rtsp-server/rtsp-media.c:
10158 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10161 configure: use unstable api
10163 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
10165 * gst/rtsp-server/rtsp-client.c:
10166 client: fix reference counting
10168 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
10170 * gst/rtsp-server/rtsp-client.c:
10171 * gst/rtsp-server/rtsp-media.c:
10172 fix compiler warnings about unused variables
10174 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
10176 * examples/test-launch.c:
10177 * examples/test-readme.c:
10178 * examples/test-uri.c:
10179 * examples/test-video.c:
10180 examples: tell rtsp uri when ready
10182 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
10185 Automatic update of common submodule
10186 From 69b981f to 605cd9a
10188 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10190 * gst/rtsp-server/rtsp-client.c:
10191 client: update for buffer API change
10193 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
10195 * gst/rtsp-server/Makefile.am:
10196 Makefile.am: 0.10 => @GST_MAJORMINOR@
10198 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
10200 * gst/rtsp-server/rtsp-media-factory-uri.c:
10201 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
10203 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
10205 * gst/rtsp-server/.gitignore:
10206 .gitignore: 0.10 => 0.11
10208 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
10210 * gst/rtsp-server/Makefile.am:
10211 Makefile.am: 0.10 => @GST_MAJORMINOR@
10213 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10215 Merge branch 'master' into 0.11
10217 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
10220 Automatic update of common submodule
10221 From 9e5bbd5 to 69b981f
10223 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
10226 Automatic update of common submodule
10227 From fd35073 to 9e5bbd5
10229 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
10232 Automatic update of common submodule
10233 From 46dfcea to fd35073
10235 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10237 * gst/rtsp-server/rtsp-media-factory-uri.c:
10238 * gst/rtsp-server/rtsp-media.c:
10239 media: port to new caps API
10241 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10243 Merge branch 'master' into 0.11
10245 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
10247 * bindings/vala/gst-rtsp-server-0.10.vapi:
10248 Updated Vala bindings.
10249 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
10251 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
10253 * gst/rtsp-server/rtsp-server.c:
10254 * gst/rtsp-server/rtsp-server.h:
10255 Add a signal for newly connected clients.
10256 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
10258 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
10260 * bindings/python/rtspserver.override:
10261 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
10263 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10265 * gst/rtsp-server/Makefile.am:
10266 * gst/rtsp-server/rtsp-client.c:
10267 * gst/rtsp-server/rtsp-funnel.c:
10268 * gst/rtsp-server/rtsp-funnel.h:
10269 * gst/rtsp-server/rtsp-media.c:
10270 rtsp-server: port to 0.11
10272 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10277 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10279 Merge branch 'master' into 0.11
10284 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10287 Automatic update of common submodule
10288 From c3cafe1 to 46dfcea
10290 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
10292 * bindings/python/Makefile.am:
10293 * bindings/python/rtspserver.defs:
10294 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
10296 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
10298 * bindings/python/arg-types.py:
10299 python bindings: add GstRTSPUrlParam
10300 Needed to implement MediaFactory virtual proxies
10302 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
10304 * bindings/python/arg-types.py:
10305 python bindings: fix returning GstRTSPUrl types
10307 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
10309 * bindings/python/arg-types.py:
10310 python bindings: add arg type for GstRTSPUrl
10312 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
10314 * bindings/python/rtspserver.defs:
10315 python bindings: fix the definition of MediaFactory.collect_stream
10317 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
10320 Automatic update of common submodule
10321 From 1ccbe09 to c3cafe1
10323 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10326 Automatic update of common submodule
10327 From 193b717 to 1ccbe09
10329 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
10332 Automatic update of common submodule
10333 From b77e2bf to 193b717
10335 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10338 build: Include lcov.mak to allow test coverage report generation
10340 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10343 Automatic update of common submodule
10344 From d8814b6 to b77e2bf
10346 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10349 Automatic update of common submodule
10350 From 6aaa286 to d8814b6
10352 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
10355 Automatic update of common submodule
10356 From 6aec6b9 to 6aaa286
10358 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
10361 autogen: wingo signed comment
10363 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
10365 * gst/rtsp-server/rtsp-session-pool.c:
10366 session: use full charset for RTSP session ID
10367 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
10368 session ID more difficult.
10369 https://bugzilla.gnome.org/show_bug.cgi?id=643812
10371 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10373 * gst/rtsp-server/Makefile.am:
10374 rtsp-server: Don't install the funnel header
10376 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
10379 Automatic update of common submodule
10380 From 1de7f6a to 6aec6b9
10382 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10385 configure: require core/base 0.10.31
10386 Needed at least for gst_plugin_feature_rank_compare_func().
10388 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
10391 Automatic update of common submodule
10392 From f94d739 to 1de7f6a
10394 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10396 * gst/rtsp-server/rtsp-media.c:
10397 media: remove more unused code
10399 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10401 * gst/rtsp-server/rtsp-media.c:
10402 * gst/rtsp-server/rtsp-media.h:
10403 media: remove duplicate filtering
10404 Remove the duplicate filtering code now that we have a released -good version.
10405 Give a warning instead.
10407 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10409 * gst/rtsp-server/rtsp-media-factory.c:
10410 * gst/rtsp-server/rtsp-media.c:
10411 media: fix default buffer size
10413 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10415 * gst/rtsp-server/rtsp-media-factory.c:
10416 * gst/rtsp-server/rtsp-media-factory.h:
10417 media-factory: add property to configure the buffer-size
10418 Add a property to configure the kernel UDP buffer size.
10420 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10422 * gst/rtsp-server/rtsp-media.c:
10423 * gst/rtsp-server/rtsp-media.h:
10424 media: add property to configure kernel buffer sizes
10425 Add a property to configure the kernel UDP buffer size.
10427 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10430 configure: set PYGOBJECT_REQ before using it
10431 https://bugzilla.gnome.org/show_bug.cgi?id=640641
10433 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10435 * docs/Makefile.am:
10436 docs: recursive into sub-directories on 'make upload'
10438 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10440 * docs/libs/gst-rtsp-server-docs.sgml:
10441 * docs/version.entities.in:
10442 docs: mention full version these docs are for, not just major-minor
10444 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10447 back to development
10449 === release 0.10.8 ===
10451 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10456 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10458 * gst/rtsp-server/rtsp-server.c:
10459 rtsp-server: clarify docs a little
10461 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10463 * gst/rtsp-server/rtsp-media.c:
10464 media: init debug category before starting thread
10466 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10468 * gst/rtsp-server/rtsp-auth.c:
10469 auth: add realm to make it more spec compliant
10471 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10473 * gst/rtsp-server/rtsp-server.c:
10474 * gst/rtsp-server/rtsp-server.h:
10475 server: add locking
10477 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10479 * examples/test-video.c:
10480 example: improve example docs a little
10482 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10484 * gst/rtsp-server/rtsp-server.c:
10485 server: ensure the watch has a ref to the server
10487 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10489 * gst/rtsp-server/rtsp-server.c:
10490 server: simpify channel function
10492 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10494 * gst/rtsp-server/rtsp-server.c:
10495 * gst/rtsp-server/rtsp-server.h:
10496 server: simplify management of channel and source
10497 We don't need to keep around the channel and source objects. Let the mainloop
10498 and the source manage the source and channel respectively.
10500 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10506 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10508 * tests/.gitignore:
10509 * tests/Makefile.am:
10510 * tests/test-cleanup.c:
10511 tests: add tests directory and cleanup test
10513 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10515 * gst/rtsp-server/rtsp-media-factory-uri.c:
10516 * gst/rtsp-server/rtsp-media-factory.c:
10517 * gst/rtsp-server/rtsp-media-mapping.c:
10518 * gst/rtsp-server/rtsp-media.c:
10519 * gst/rtsp-server/rtsp-session-pool.c:
10520 * gst/rtsp-server/rtsp-session.c:
10521 server: improve debugging in various objects
10523 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10525 * gst/rtsp-server/rtsp-server.c:
10526 server: chain up to the parent finalize
10528 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
10530 * bindings/python/rtspserver-types.defs:
10531 * bindings/python/rtspserver.defs:
10532 * bindings/python/rtspserver.override:
10533 * bindings/python/test.py:
10534 gst-rtsp-server: update python bindings
10536 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10538 * gst/rtsp-server/rtsp-client.c:
10539 client: use the response from the clientstate
10540 Create the response object only once and store in the client state.
10541 Make all methods use the state response,
10543 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10545 * gst/rtsp-server/rtsp-server.c:
10546 server: use signal to keep track of clients
10547 Keep track of all the clients that the server creates and remove them when they
10548 fire the 'closed' signal.
10550 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10552 * gst/rtsp-server/rtsp-client.c:
10553 * gst/rtsp-server/rtsp-client.h:
10554 client: emit signal when closing
10556 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10558 * examples/.gitignore:
10559 * examples/Makefile.am:
10560 * examples/test-auth.c:
10561 * examples/test-video.c:
10562 * gst/rtsp-server/rtsp-auth.c:
10563 * gst/rtsp-server/rtsp-auth.h:
10564 * gst/rtsp-server/rtsp-client.c:
10565 * gst/rtsp-server/rtsp-media-factory.c:
10566 * gst/rtsp-server/rtsp-media.c:
10567 * gst/rtsp-server/rtsp-media.h:
10568 * gst/rtsp-server/rtsp-session-pool.h:
10569 * gst/rtsp-server/rtsp-session.h:
10570 media: enable per factory authorisations
10571 Allow for adding a GstRTSPAuth on the factory and media level and check
10572 permissions when accessing the factory.
10573 Add hints to the auth methods for future more fine grained authorisation.
10574 Add example application for per factory authentication.
10576 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10578 * gst/rtsp-server/rtsp-auth.c:
10579 * gst/rtsp-server/rtsp-auth.h:
10580 * gst/rtsp-server/rtsp-client.c:
10581 * gst/rtsp-server/rtsp-client.h:
10582 * gst/rtsp-server/rtsp-params.c:
10583 * gst/rtsp-server/rtsp-params.h:
10584 rtsp-server: Pass ClientState structure arround
10585 Pass the collected information for the ongoing request in a GstRTSPClientState
10586 structure that we can then pass around to simplify the method arguments. This
10587 will also be handy when we implement logging functionality.
10589 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10591 * gst/rtsp-server/rtsp-media-factory.c:
10592 * gst/rtsp-server/rtsp-media-factory.h:
10593 media-factory: add methods to configure authorisation
10595 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10597 * gst/rtsp-server/rtsp-client.c:
10598 client: unref auth in finalize
10600 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10602 * gst/rtsp-server/rtsp-server.c:
10603 server: unref auth in finalize
10605 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10607 * docs/libs/gst-rtsp-server-docs.sgml:
10608 * docs/libs/gst-rtsp-server-sections.txt:
10609 * docs/libs/gst-rtsp-server.types:
10610 docs: add more docs
10612 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10614 * gst/rtsp-server/rtsp-server.c:
10615 * gst/rtsp-server/rtsp-server.h:
10616 server: separate create and accept
10617 Create separate create and accept methods so that subclasses can create custom
10619 Configure the server in the client object and prepare for keeping track of
10622 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10624 * gst/rtsp-server/rtsp-client.c:
10625 * gst/rtsp-server/rtsp-client.h:
10626 client: add support for setting the server.
10627 Add support for keeping a ref to the server that started this client
10630 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10632 * gst/rtsp-server/rtsp-auth.c:
10633 auth: fix memleak and add some docs
10634 Fix a memleak of the basic auth token.
10635 Add docs for the helper function
10637 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10639 * gst/rtsp-server/rtsp-auth.c:
10640 * gst/rtsp-server/rtsp-auth.h:
10641 * gst/rtsp-server/rtsp-client.c:
10642 client: delegate setup of auth to the manager
10643 Delegate the configuration of the authentication tokens to the manager object
10646 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10648 * examples/test-video.c:
10649 * gst/rtsp-server/Makefile.am:
10650 * gst/rtsp-server/rtsp-auth.c:
10651 * gst/rtsp-server/rtsp-auth.h:
10652 * gst/rtsp-server/rtsp-client.c:
10653 * gst/rtsp-server/rtsp-client.h:
10654 * gst/rtsp-server/rtsp-server.c:
10655 * gst/rtsp-server/rtsp-server.h:
10656 auth: add authentication object
10657 Add an object that can check the authorization of requests.
10658 Implement basic authentication.
10659 Add example authentication to test-video
10661 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10663 * gst/rtsp-server/rtsp-server.c:
10664 * gst/rtsp-server/rtsp-server.h:
10665 server: move includes back
10666 the includes are needed for sockaddr_in.
10668 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10670 * gst/rtsp-server/rtsp-client.c:
10671 * gst/rtsp-server/rtsp-client.h:
10672 * gst/rtsp-server/rtsp-server.c:
10673 * gst/rtsp-server/rtsp-server.h:
10674 rtsp: move network includes where they are needed
10676 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
10678 * gst/rtsp-server/rtsp-media.h:
10679 rtsp-media.h: Minor corrections in comments.
10682 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
10685 Automatic update of common submodule
10686 From e572c87 to f94d739
10688 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
10692 * docs/libs/.gitignore:
10693 * examples/.gitignore:
10694 * gst/rtsp-server/.gitignore:
10697 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
10699 * docs/libs/Makefile.am:
10700 docs: We don't build ps/pdf for API reference docs
10702 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10705 Automatic update of common submodule
10706 From ccbaa85 to e572c87
10708 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10711 Automatic update of common submodule
10712 From 46445ad to ccbaa85
10714 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10716 * gst/rtsp-server/Makefile.am:
10717 * gst/rtsp-server/rtsp-funnel.c:
10718 * gst/rtsp-server/rtsp-funnel.h:
10719 * gst/rtsp-server/rtsp-media.c:
10720 funnel: rename fsfunnel to rtspfunnel
10721 Rename the funnel to avoid conflicts with the farsight one.
10723 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10725 * gst/rtsp-server/Makefile.am:
10726 * gst/rtsp-server/fs-funnel.c:
10727 * gst/rtsp-server/fs-funnel.h:
10728 * gst/rtsp-server/rtsp-media.c:
10729 rtsp-media: add and use fsfunnel
10730 Add a copy of fsfunnel to the build because input-selector removed the (broken)
10731 select-all property that we need.
10733 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10735 * gst/rtsp-server/Makefile.am:
10736 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
10737 Use PKG_CONFIG_PATH specified at configure time (if any) as well
10738 for the g-ir-compiler, rather than just assuming the env var has
10741 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10748 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
10750 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10753 * gst/rtsp-server/Makefile.am:
10754 gobject-introspection: fix g-i build for uninstalled setup
10755 Requires gst-plugins-base git (> 0.10.31.2).
10757 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10759 * examples/test-uri.c:
10760 examples: add some more options and comments
10762 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10764 * gst/rtsp-server/rtsp-media-factory-uri.c:
10765 factory-uri: use right property type
10767 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10769 * gst/rtsp-server/rtsp-media-factory-uri.c:
10770 factory-uri: attempt to configure buffer-lists
10771 Attempt to configure buffer lists in the payloader for improved performance.
10773 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10775 * gst/rtsp-server/rtsp-media.c:
10776 media: attempt to configure bigger UDP buffers
10777 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
10778 send buffers with high bitrate streams.
10780 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
10782 * gst/rtsp-server/rtsp-client.c:
10783 client: use the socket length from getsockname
10784 Use the length returned by getsockname to perform the getnameinfo call because
10785 the size can depend on the socket type and platform.
10788 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10790 * docs/libs/gst-rtsp-server-docs.sgml:
10791 * docs/libs/gst-rtsp-server-sections.txt:
10792 docs: add uri factory to the docs
10794 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10796 * gst/rtsp-server/rtsp-client.c:
10797 * gst/rtsp-server/rtsp-media.h:
10800 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10802 * gst/rtsp-server/rtsp-client.c:
10803 * gst/rtsp-server/rtsp-media.c:
10804 * gst/rtsp-server/rtsp-media.h:
10805 * gst/rtsp-server/rtsp-session.c:
10806 * gst/rtsp-server/rtsp-session.h:
10807 rtsp-server: add support for buffer lists
10808 Add support for sending bufferlists received from appsink.
10811 2010-12-28 18:35:01 +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 media: make method to retrieve the play range
10818 Make a method to retrieve the playback range so that we can conditionally create
10819 a different range for the SDP and the PLAY requests.
10821 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10823 * gst/rtsp-server/rtsp-media.c:
10824 * gst/rtsp-server/rtsp-media.h:
10825 media: add signal to notify of state changes
10827 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10829 * gst/rtsp-server/rtsp-client.h:
10830 client: cleanup headers
10832 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10834 * gst/rtsp-server/rtsp-client.c:
10837 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10839 * gst/rtsp-server/rtsp-media-factory-uri.c:
10840 * gst/rtsp-server/rtsp-media-factory-uri.h:
10841 factory-uri: add support for gstpay
10842 Add an option to prefer gstpay over decoder + raw payloader.
10844 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10846 * gst/rtsp-server/rtsp-media-factory-uri.c:
10847 * gst/rtsp-server/rtsp-media-factory-uri.h:
10848 factory-uri: rework the autoplugger.
10849 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
10852 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10854 * gst/rtsp-server/rtsp-media-factory-uri.c:
10855 factory-uri: use better factory filter
10856 Make better payloader filter based on autoplug rank and RTP use case.
10858 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
10861 Automatic update of common submodule
10862 From 169462a to 46445ad
10864 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10866 * gst/rtsp-server/rtsp-server.c:
10867 server: set SO_REUSEADDR before bind
10868 Set the SO_REUSEADDR _before_ bind() to make it actually work.
10870 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10872 * gst/rtsp-server/rtsp-media.c:
10873 * gst/rtsp-server/rtsp-media.h:
10874 media: emit prepared signal when prepared
10875 Make a 'prepared' signal and emit it when we successfully prepared the element.
10876 This signal can be used to configure the media object after it has been prepared
10879 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
10882 Automatic update of common submodule
10883 From 011bcc8 to 169462a
10885 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
10887 python an optional dependency
10888 * configure.ac: Move up valgrind and g-i checks. Make the python
10889 dependency optional, as it was before.
10891 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10893 Merge branch 'master' into 0.11
10898 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10900 * gst/rtsp-server/rtsp-media.c:
10901 media: update range when active clients changed
10902 When we changed the number of active clients, update the current range
10903 information because we want the second client connecting to a shared resource
10904 continue from where the stream currently.
10906 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10908 * gst/rtsp-server/rtsp-media-factory-uri.c:
10909 * gst/rtsp-server/rtsp-media-factory-uri.h:
10910 factory-uri: add colorspace and fix pt
10911 Rework the way we pass data to the autoplugger.
10912 When we have raw caps, plug a converter element to make pluggin to raw
10913 payloaders more successful.
10914 Make sure all dynamically plugged payloaders have a unique payload types.
10916 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10918 * examples/Makefile.am:
10919 * examples/test-uri.c:
10920 example: add example of the uri factory
10922 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10924 * gst/rtsp-server/Makefile.am:
10925 * gst/rtsp-server/rtsp-media-factory-uri.c:
10926 * gst/rtsp-server/rtsp-media-factory-uri.h:
10927 * gst/rtsp-server/rtsp-server.h:
10928 factory-uri: add a factory to stream any URI
10929 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
10932 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10934 * gst/rtsp-server/rtsp-media.c:
10935 * gst/rtsp-server/rtsp-media.h:
10936 media: ignore spurious ASYNC_DONE messages
10937 When we are dynamically adding pads, the addition of the udpsrc elements will
10938 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
10939 the real ASYNC_DONE when everything is prerolled.
10941 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10943 * gst/rtsp-server/rtsp-media-factory.c:
10944 * gst/rtsp-server/rtsp-media-factory.h:
10945 media-factory: make lock macro
10947 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
10949 * gst/rtsp-server/rtsp-client.c:
10950 rtsp-server: Remove unused variable and dead assignment
10952 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
10954 * examples/test-launch.c:
10955 * examples/test-mp4.c:
10956 * examples/test-ogg.c:
10957 * examples/test-readme.c:
10958 * examples/test-sdp.c:
10959 * examples/test-video.c:
10960 examples: Run gst-indent
10962 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
10964 * gst/rtsp-server/rtsp-client.c:
10965 * gst/rtsp-server/rtsp-media-factory.c:
10966 * gst/rtsp-server/rtsp-media-mapping.c:
10967 * gst/rtsp-server/rtsp-media.c:
10968 * gst/rtsp-server/rtsp-params.c:
10969 * gst/rtsp-server/rtsp-sdp.c:
10970 * gst/rtsp-server/rtsp-server.c:
10971 * gst/rtsp-server/rtsp-session-pool.c:
10972 * gst/rtsp-server/rtsp-session.c:
10973 rtsp-server: Run gst-indent
10974 Since it wasn't using the upstream common previously, there was no
10975 indentation check before commiting.
10977 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
10979 * gst/rtsp-server/rtsp-media-mapping.h:
10980 * gst/rtsp-server/rtsp-media.c:
10981 * gst/rtsp-server/rtsp-media.h:
10982 * gst/rtsp-server/rtsp-sdp.c:
10983 * gst/rtsp-server/rtsp-session-pool.h:
10984 * gst/rtsp-server/rtsp-session.c:
10985 * gst/rtsp-server/rtsp-session.h:
10986 rtsp-server: Some more doc fixups
10988 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
10991 Makefile: Add cruft-cleaning support
10993 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
10997 * docs/Makefile.am:
10998 * docs/libs/Makefile.am:
10999 * docs/libs/gst-rtsp-server-docs.sgml:
11000 * docs/libs/gst-rtsp-server-sections.txt:
11001 * docs/libs/gst-rtsp-server.types:
11002 * docs/version.entities.in:
11003 docs: Add gtk-doc build system
11005 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
11007 * gst/rtsp-server/Makefile.am:
11008 Makefile.am: Use standard GIR make behaviour
11010 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
11014 autogen/configure: Bring more in sync to standard gst module behaviour
11016 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11018 * gst/rtsp-server/rtsp-media.c:
11019 media: warn and fail when gstrtpbin is not found
11021 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11024 configure: open 0.11 branch
11026 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
11030 Add common submodule
11032 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
11034 * common/ChangeLog:
11035 * common/Makefile.am:
11036 * common/c-to-xml.py:
11037 * common/check.mak:
11038 * common/coverage/coverage-report-entry.pl:
11039 * common/coverage/coverage-report.pl:
11040 * common/coverage/coverage-report.xsl:
11041 * common/coverage/lcov.mak:
11042 * common/gettext.patch:
11043 * common/glib-gen.mak:
11044 * common/gst-autogen.sh:
11045 * common/gst-xmlinspect.py:
11047 * common/gstdoc-scangobj:
11048 * common/gtk-doc-plugins.mak:
11049 * common/gtk-doc.mak:
11050 * common/m4/.gitignore:
11051 * common/m4/Makefile.am:
11052 * common/m4/README:
11053 * common/m4/as-ac-expand.m4:
11054 * common/m4/as-auto-alt.m4:
11055 * common/m4/as-compiler-flag.m4:
11056 * common/m4/as-compiler.m4:
11057 * common/m4/as-docbook.m4:
11058 * common/m4/as-libtool-tags.m4:
11059 * common/m4/as-libtool.m4:
11060 * common/m4/as-python.m4:
11061 * common/m4/as-scrub-include.m4:
11062 * common/m4/as-version.m4:
11063 * common/m4/ax_create_stdint_h.m4:
11064 * common/m4/check.m4:
11065 * common/m4/glib-gettext.m4:
11066 * common/m4/gst-arch.m4:
11067 * common/m4/gst-args.m4:
11068 * common/m4/gst-check.m4:
11069 * common/m4/gst-debuginfo.m4:
11070 * common/m4/gst-default.m4:
11071 * common/m4/gst-doc.m4:
11072 * common/m4/gst-error.m4:
11073 * common/m4/gst-feature.m4:
11074 * common/m4/gst-function.m4:
11075 * common/m4/gst-gettext.m4:
11076 * common/m4/gst-glib2.m4:
11077 * common/m4/gst-libxml2.m4:
11078 * common/m4/gst-plugindir.m4:
11079 * common/m4/gst-valgrind.m4:
11080 * common/m4/gtk-doc.m4:
11081 * common/m4/introspection.m4:
11082 * common/m4/pkg.m4:
11083 * common/mangle-tmpl.py:
11084 * common/plugins.xsl:
11086 * common/release.mak:
11087 * common/scangobj-merge.py:
11088 * common/upload.mak:
11089 common: Remove static version
11091 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
11093 * common/m4/introspection.m4:
11094 Update introspection.m4 to match usage
11096 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11100 Remove old stuff from the README
11102 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11105 back to development
11107 === release 0.10.7 ===
11109 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11114 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11116 * examples/test-ogg.c:
11117 test-ogg: remove parsers
11118 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
11119 buffers with timestamps. Using the parsers also seems to break things.
11121 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11123 * bindings/vala/gst-rtsp-server-0.10.vapi:
11124 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11125 Updated Vala bindings
11127 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11129 * common/m4/introspection.m4:
11131 * gst/rtsp-server/Makefile.am:
11132 Added initial gobject-introspection support
11134 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11136 * gst/rtsp-server/rtsp-media-factory.c:
11137 media-factory: don't use host for shared hash key
11138 When we generate the key to share made between connections, don't include the
11139 host used to connect so that we can share media even if between clients that
11140 connected with localhost and ones with the ip address.
11142 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11144 * bindings/vala/Makefile.am:
11145 build: fix distcheck
11147 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11149 * bindings/vala/gst-rtsp-server-0.10.vapi:
11150 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11151 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11152 Update Vala bindings
11154 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11156 * bindings/vala/Makefile.am:
11158 Fix configure checks and installation location for Vala bindings
11161 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11164 back to development
11166 === release 0.10.6 ===
11168 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11171 configure: release 0.10.6
11173 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11175 * gst/rtsp-server/rtsp-media.c:
11176 media: help the compiler a little
11178 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11180 * gst/rtsp-server/rtsp-media.c:
11181 * gst/rtsp-server/rtsp-media.h:
11182 * gst/rtsp-server/rtsp-session.c:
11183 media: cleanup media transport before freeing
11184 Cleanup the media transport data before freeing. In particular, remove the qdata
11185 from the rtpsource object.
11187 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11189 * gst/rtsp-server/rtsp-media-factory.c:
11190 * gst/rtsp-server/rtsp-media-factory.h:
11191 * gst/rtsp-server/rtsp-media.c:
11192 * gst/rtsp-server/rtsp-media.h:
11193 media-factory: add eos-shutdown property
11194 Add an eos-shutdown property that will send an EOS to the pipeline before
11195 shutting it down. This allows for nice cleanup in case of a muxer.
11198 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11200 * gst/rtsp-server/rtsp-media.c:
11201 * gst/rtsp-server/rtsp-media.h:
11202 media: use multiudpsink send-duplicates when we can
11203 If we have a new enough multiudpsink with the send-duplicates property, use this
11204 instead of doing our own filtering. Our custom filtering code should eventually
11205 be removed when we can depend on a released -good.
11207 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11209 * gst/rtsp-server/rtsp-media.c:
11210 media: don't leak destinations
11211 Refactor and cleanup the destinations array when the stream is destroyed.
11213 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11215 * gst/rtsp-server/rtsp-media.c:
11216 * gst/rtsp-server/rtsp-media.h:
11217 media: don't add udp addresses multiple times
11218 Keep track of the udp addresses we added to udpsink and never add the same udp
11219 destination twice. This avoids duplicate packets when using multicast.
11221 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11223 * gst/rtsp-server/rtsp-server.c:
11224 server: disable use of SO_LINGER
11225 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
11226 server close()s the connection.
11228 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11230 * gst/rtsp-server/rtsp-server.c:
11231 server: use 5 second linger period in SO_LINGER
11232 Wait 5 seconds before clearing the send buffers and reseting the connection with
11233 the client when we do a close. This should be enough time to get the message to
11237 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
11239 * gst/rtsp-server/rtsp-server.c:
11240 server: use SO_LINGER
11241 SO_LINGER on the socket will make sure that any pending data on the socket is
11242 flushed ASAP and that the socket connection is reset. This makes sure that the
11243 socket can be reused immediately.
11246 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11249 README: add blurb about shared media factories
11251 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
11253 * gst/rtsp-server/rtsp-media.c:
11254 Add stdlib.h for atoi()
11256 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11258 * bindings/python/Makefile.am:
11259 * bindings/vala/Makefile.am:
11260 build: distcheck fixes
11261 Fix 'make distcheck', somewhat (it still fails because it tries to
11262 install files into /usr/share/vala/vapi/ irrespective of the
11263 configured prefix).
11265 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11268 configure: bump core/base requirements to released version
11269 Makes things less confusing for people.
11271 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11274 configure: fail if GStreamer core/base requirements are not met
11276 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11278 * gst/rtsp-server/rtsp-client.c:
11279 client: improve client cleanups
11280 Make sure the session does not timeout when using TCP. We need to do this
11281 because quicktime player does not send RTCP for some reason in tunneled
11283 Refactor some cleanup code.
11286 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11288 * gst/rtsp-server/rtsp-session.c:
11289 * gst/rtsp-server/rtsp-session.h:
11290 session: add support for prevent session timeouts
11291 Add an atomix counter to prevent session timeouts when we are, for example,
11292 streaming over TCP.
11294 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11296 * gst/rtsp-server/rtsp-client.c:
11297 client: fix unlink on session timeouts
11298 When our session times out, make sure we unlink all streams in this
11300 Remove the tunnelid when closing the connection.
11302 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11304 * gst/rtsp-server/rtsp-session.c:
11305 session: small cleanups
11307 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11309 * gst/rtsp-server/rtsp-client.c:
11310 client: handle lost_tunnel callbacks
11311 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
11312 hashtable so that we can reuse it for when the client reopens the POST
11314 Close the connection after a TEARDOWN.
11315 Make sure or watchid is cleared when the watch is removed.
11318 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11320 * gst/rtsp-server/rtsp-client.c:
11321 * gst/rtsp-server/rtsp-media.c:
11322 * gst/rtsp-server/rtsp-sdp.c:
11323 rtsp-server: add more support for multicast
11325 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11328 * gst/rtsp-server/rtsp-media.c:
11329 * gst/rtsp-server/rtsp-media.h:
11330 media: allow configuration of allowed lower transport
11332 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11334 * gst/rtsp-server/rtsp-client.h:
11335 * gst/rtsp-server/rtsp-media.c:
11336 * gst/rtsp-server/rtsp-media.h:
11337 * gst/rtsp-server/rtsp-sdp.c:
11338 * gst/rtsp-server/rtsp-sdp.h:
11339 * gst/rtsp-server/rtsp-server.c:
11340 rtsp: keep track of server ip and ipv6
11341 Keep track of how the client connected to the server and setup the udp ports
11342 with the same protocol.
11343 Copy the server ip address in the SDP so that clients can send RTCP back to
11346 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11348 * gst/rtsp-server/rtsp-session.c:
11351 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11353 * gst/rtsp-server/rtsp-client.c:
11354 client: use right size for malloc
11356 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11358 * gst/rtsp-server/rtsp-server.c:
11359 server: comment ipv6 server listening address
11361 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11363 * gst/rtsp-server/rtsp-media.c:
11364 media: allow for ipv6 sockets
11366 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11368 * gst/rtsp-server/rtsp-server.c:
11369 * gst/rtsp-server/rtsp-server.h:
11370 server: rework server part
11371 Allow setting a bind address, make sure we can deal with ipv6.
11372 Remove the port property and change with the service property.
11374 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11376 * gst/rtsp-server/rtsp-media.h:
11377 media: update comments a little
11379 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11381 * gst/rtsp-server/rtsp-client.c:
11382 client: make content-base better
11383 Use the URI formatting functions to make a content-base. Also make sure that
11384 there is a trailing / at the end.
11386 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11388 * gst/rtsp-server/rtsp-client.c:
11389 client: guard against invalid paths
11391 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11393 * examples/test-video.c:
11394 test: catch server bind errors
11396 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
11398 * gst/rtsp-server/rtsp-media.c:
11399 rtspmedia: emit "unprepared" if _prepare fails.
11400 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
11401 media object is removed from its factory's cache.
11403 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11405 * gst/rtsp-server/rtsp-media.c:
11406 media: collect media position when seek completes
11408 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
11410 * gst/rtsp-server/rtsp-client.c:
11411 client: call unlink_streams in client finalize
11414 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11416 * gst/rtsp-server/rtsp-media.c:
11417 media: limit the time to wait to something huge
11418 Avoid waiting forever but limit the timeout to 20 seconds.
11420 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11422 * gst/rtsp-server/rtsp-sdp.c:
11423 sdp: reindent and check for prepared status
11425 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11427 * gst/rtsp-server/rtsp-media.c:
11428 * gst/rtsp-server/rtsp-media.h:
11429 * gst/rtsp-server/rtsp-session.c:
11430 media: avoid doing _get_state() for state changes
11431 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
11432 until the media is prerolled or in error. This avoids doing a blocking call of
11433 gst_element_get_state() that can cause lockups when there is an error.
11436 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11438 * gst/rtsp-server/rtsp-media.c:
11441 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11443 * gst/rtsp-server/rtsp-media-factory.c:
11444 media-factory: better error handling
11445 Improve the error handling a bit.
11447 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11449 * gst/rtsp-server/rtsp-client.c:
11450 client: rework transport parsing
11451 Rework the transport parsing code so that we can ignore transports we don't
11452 support instead of just picking the first one we can parse.
11453 Configure a (for now hardcoded) destination for multicast transports.
11455 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11457 * gst/rtsp-server/rtsp-media.c:
11458 media: set multicast sink parameters
11459 Disable loop and automatic multicast join on the udpsink elements.
11460 Add some more debug info.
11461 Reset some state variables in the right place.
11462 Use the right port numbers for multicast.
11464 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11466 * gst/rtsp-server/rtsp-session.c:
11467 session: handle transport setup correctly
11468 Handle UDP, MCAST and TCP transport negotiation more correctly.
11469 Store the server session SSRC in the transport.
11471 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11473 * gst/rtsp-server/rtsp-client.c:
11474 rtsp-client: implement error_full
11475 Implement error_full to avoid some segfaults when the rtspconnection calls it.
11478 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11481 * gst/rtsp-server/rtsp-client.c:
11482 * gst/rtsp-server/rtsp-server.c:
11483 docs: update docs and comments
11485 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
11487 * gst/rtsp-server/rtsp-sdp.c:
11488 sdp: make server work better when behind a proxy
11490 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11492 * gst/rtsp-server/rtsp-client.c:
11493 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
11495 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11497 * gst/rtsp-server/rtsp-client.c:
11498 * gst/rtsp-server/rtsp-media-factory.c:
11499 * gst/rtsp-server/rtsp-media-mapping.c:
11500 * gst/rtsp-server/rtsp-media.c:
11501 * gst/rtsp-server/rtsp-server.c:
11502 * gst/rtsp-server/rtsp-session-pool.c:
11503 * gst/rtsp-server/rtsp-session.c:
11504 Use GStreamer's debugging subsystem
11506 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11508 * gst/rtsp-server/rtsp-media-factory.c:
11509 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
11511 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11514 back to development
11516 === release 0.10.5 ===
11518 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11523 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11526 configure: bump required versions
11528 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
11530 * gst/rtsp-server/rtsp-client.c:
11531 client: call weak-unref on client->sessions from finalize
11534 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11536 * gst/rtsp-server/rtsp-media.c:
11537 media: Fixed crasher where caps got unref'ed too often
11539 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11542 * pkgconfig/.gitignore:
11543 * pkgconfig/Makefile.am:
11544 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
11545 Added pkg-config file to use gst-rtsp-server uninstalled
11547 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11549 * gst/rtsp-server/rtsp-media.c:
11550 media: add some docs
11552 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
11554 * gst/rtsp-server/rtsp-client.c:
11555 rtsp: Use gst_rtsp_watch_send_message().
11556 Use gst_rtsp_watch_send_message() since the old API which used
11557 gst_rtsp_watch_queue_message() has been deprecated.
11559 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11562 back to development
11564 === release 0.10.4 ===
11566 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11571 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11573 * gst/rtsp-server/rtsp-client.c:
11574 * gst/rtsp-server/rtsp-session.c:
11575 * gst/rtsp-server/rtsp-session.h:
11576 rtsp: allocate channels in TCP mode
11577 When the client does not provide us with channels in TCP mode, allocate channels
11580 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11582 * gst/rtsp-server/rtsp-client.c:
11583 client: don't crash when tunnelid is missing
11584 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
11585 don't crash but return an error response to the client.
11588 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11590 * bindings/vala/gst-rtsp-server-0.10.vapi:
11591 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11592 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11593 bindings: update vala bindings with new method
11595 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11597 * gst/rtsp-server/rtsp-session-pool.c:
11598 * gst/rtsp-server/rtsp-session-pool.h:
11599 sessionpool: add function to filter sessions
11600 Add generic function to retrieve/remove sessions.
11602 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11605 configure: bump core/base requirements to release
11607 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11609 * gst/rtsp-server/rtsp-media.c:
11610 media: fix indentation
11612 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11614 * gst/rtsp-server/rtsp-media.c:
11615 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
11617 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11619 * gst/rtsp-server/rtsp-media.c:
11620 set state and remove elements of media in for loop
11622 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
11624 * bindings/vala/gst-rtsp-server-0.10.vapi:
11625 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11626 Added gst_rtsp_media_remove_elements function to Vala bindings
11628 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
11630 * gst/rtsp-server/rtsp-media.c:
11631 * gst/rtsp-server/rtsp-media.h:
11632 Added gst_rtsp_media_remove_elements function
11634 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
11636 * gst/rtsp-server/rtsp-media.c:
11637 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
11639 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11641 * bindings/vala/gst-rtsp-server-0.10.vapi:
11642 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11643 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11644 Updated Vala bindings
11646 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11648 * gst/rtsp-server/rtsp-media.c:
11649 * gst/rtsp-server/rtsp-media.h:
11650 Added vmethod unprepare to GstRTSPMedia
11651 The default implementation sets the state of the pipeline to GST_STATE_NULL
11653 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11655 * gst/rtsp-server/rtsp-media-factory.c:
11656 * gst/rtsp-server/rtsp-media-factory.h:
11657 Made collect_streams function public
11659 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11661 * gst/rtsp-server/rtsp-media-factory.c:
11662 * gst/rtsp-server/rtsp-media-factory.h:
11663 * gst/rtsp-server/rtsp-media.c:
11664 Added vmethod create_pipeline to GstRTSPMediaFactory
11665 The pipeline is created in this method and the GstRTSPMedia's element is added to it
11667 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11669 * gst/rtsp-server/rtsp-client.c:
11670 client: use g_source_destroy()
11671 We need to use g_source_destroy() because we might have added the source to a
11672 different main context than the default one.
11674 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11676 * gst/rtsp-server/Makefile.am:
11677 * gst/rtsp-server/rtsp-client.c:
11678 * gst/rtsp-server/rtsp-params.c:
11679 * gst/rtsp-server/rtsp-params.h:
11680 rtsp: prepare for handling GET/SET_PARAMETER
11681 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
11683 Fix return codes of handlers.
11685 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11687 * gst/rtsp-server/rtsp-media.c:
11688 media: don't leak session pads
11690 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11692 * gst/rtsp-server/rtsp-media.c:
11693 media: clean up the messages a bit
11695 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11697 * gst/rtsp-server/rtsp-sdp.c:
11698 sdp: warn and skip streams without media
11700 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11702 * bindings/vala/gst-rtsp-server-0.10.vapi:
11703 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11704 vala: Fixed typo in header file of RTSPMediaStream
11706 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11708 * gst/rtsp-server/rtsp-media.c:
11710 Fix a debug message
11711 Make dumping RTCP stats configurable
11713 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11715 * gst/rtsp-server/rtsp-media.c:
11716 media: be less verbose and leak less
11718 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11720 * gst/rtsp-server/rtsp-media.c:
11721 media: don't leak the destination address
11723 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11725 * gst/rtsp-server/rtsp-client.c:
11726 * gst/rtsp-server/rtsp-media.c:
11727 * gst/rtsp-server/rtsp-media.h:
11728 * gst/rtsp-server/rtsp-session.c:
11729 * gst/rtsp-server/rtsp-session.h:
11730 rtsp: use RTCP to keep the session alive
11731 Use the RTCP rtcp-from stats field to find the associated session and use this
11732 to keep the session alive.
11734 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11736 * gst/rtsp-server/rtsp-session.c:
11737 session: add 5sec to the real session timeout
11738 Allow the session to live 5sec longer before really timing out. This should give
11739 clients some extra time to keep the session active.
11741 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11743 * gst/rtsp-server/rtsp-client.c:
11744 client: replay OK to GET/SET_PARAMETER
11745 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
11746 so that we return OK for those requests.
11748 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11750 * gst/rtsp-server/rtsp-media.c:
11751 * gst/rtsp-server/rtsp-media.h:
11752 media: keep track of active transports
11753 Keep track of which transport is active to avoid closing the connection too
11755 Remove the destination transport also when going to NULL.
11756 Print some stats about the SDES and other RTCP messages we receive from the
11759 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11761 * examples/.gitignore:
11762 * examples/Makefile.am:
11763 * examples/test-sdp.c:
11764 example: add SDP relay example
11766 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11768 * gst/rtsp-server/rtsp-media.c:
11769 media: also count active TCP connections
11771 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11773 * gst/rtsp-server/rtsp-media-factory.c:
11774 * gst/rtsp-server/rtsp-media.c:
11775 * gst/rtsp-server/rtsp-media.h:
11776 rtsp: add support for dynamic elements
11777 Add support for dynamic elements.
11778 Don't set live pipelines back to paused.
11780 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11782 * gst/rtsp-server/rtsp-sdp.c:
11783 sdp: don't add encoding name when absent in caps
11785 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11787 * gst/rtsp-server/rtsp-client.c:
11788 client: warn when we can't do RTP-Info
11790 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11792 * gst/rtsp-server/rtsp-media-factory.c:
11793 factory: factor out the stream construction
11795 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11797 * gst/rtsp-server/rtsp-client.c:
11798 client: only add RTP-Info when we have the info
11799 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
11802 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11805 back to development
11807 === release 0.10.3 ===
11809 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11813 - Fixes a bug where it put the wrong verion in pkgconfig
11814 - Link RTP and RTCP sources
11816 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11818 * gst/rtsp-server/rtsp-media.c:
11819 * gst/rtsp-server/rtsp-media.h:
11820 media: link the RTP udpsrc to the session manager
11821 Link the RTP udpsrc and the appsrc to the session manager so that they don't
11822 shut down when the client sends a packet to open firewalls.
11824 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11826 * pkgconfig/gst-rtsp-server.pc.in:
11827 Don't use hard-coded version number in pkg-config file
11829 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11832 back to development
11834 === release 0.10.2 ===
11836 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11841 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11844 * common/m4/.gitignore:
11845 * examples/.gitignore:
11846 * pkgconfig/.gitignore:
11847 add some .gitignore files
11849 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11851 * gst/rtsp-server/rtsp-media.c:
11852 media: seek to key frames
11854 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11856 * gst/rtsp-server/rtsp-media.c:
11857 media: emit the unprepared signal by id
11858 Emit the unprepared signal by id instead of name and set the media as
11861 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11863 * gst/rtsp-server/rtsp-media.c:
11864 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
11866 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11868 * gst/rtsp-server/rtsp-server.c:
11869 Added finalize function to GstRTPSPServer to unref session pool and media mapping
11871 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11873 * bindings/vala/gst-rtsp-server-0.10.vapi:
11874 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11875 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11876 Updated vala bindings
11878 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11880 * gst/rtsp-server/Makefile.am:
11881 * gst/rtsp-server/rtsp-client.c:
11882 * gst/rtsp-server/rtsp-media.c:
11883 server: use appsink and appsrc with the API
11884 Use the appsink/appsrc API instead of the signals for higher
11887 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11889 * examples/test-ogg.c:
11890 tests: set the payload type correctly
11892 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11894 * gst/rtsp-server/rtsp-media-factory.c:
11895 factory: connect to the unprepare signal
11896 Connect to the unprepare signal for non-reusable media so that we can remove
11897 them from the cache.
11899 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11901 * gst/rtsp-server/rtsp-media.c:
11902 * gst/rtsp-server/rtsp-media.h:
11903 media: add signal to notify of unprepare
11905 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11907 * gst/rtsp-server/rtsp-media.c:
11908 * gst/rtsp-server/rtsp-media.h:
11909 media: more work on making the media shared
11910 Add a reusable flag to medias, indicating that they can be reused after a state
11914 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11916 * examples/test-readme.c:
11917 examples: mark the example as shared for testing
11919 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11921 * gst/rtsp-server/rtsp-media.c:
11922 * gst/rtsp-server/rtsp-media.h:
11923 client: support shared media
11924 Always perform the state actions even if the target state of the pipeline is
11925 already correct, we still want to add/remove the transports when we are dealing
11927 Keep a counter of the number of active transports for a media so that we can use
11928 this to perform a state change when needed.
11929 Perform a state change of the pipeline only when the first transport was added
11930 or when there are no active transports.
11932 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11934 * gst/rtsp-server/rtsp-client.c:
11935 client: fix refcounting crasher
11936 Don't need to remove the weak refs in the finalize methods, they are already
11937 removed in the dispose.
11938 Don't register the callback with a DestroyNofity.
11940 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11942 * gst/rtsp-server/rtsp-client.c:
11943 Fix rtsp client refcount management in TCP mode.
11944 Don't unref a client ref we never had. Fixes an unref
11945 of an already-free client object after a client
11946 teardown request for me.
11948 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11950 * gst/rtsp-server/rtsp-session.c:
11951 docs: fix typo in API docs
11953 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11955 * gst/rtsp-server/rtsp-media.c:
11956 More seeking fixes.
11957 Keep the udp sources in playing even if we go to paused. unlock the sources when
11959 Add some more debug info.
11960 Only seek when we need to.
11961 Keep track of the position when we go to paused.
11963 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11965 * gst/rtsp-server/rtsp-client.c:
11966 * gst/rtsp-server/rtsp-media.c:
11967 * gst/rtsp-server/rtsp-media.h:
11968 Add beginnings of seeking.
11969 Parse the Range header and perform a seek on the pipeline for the requested
11970 position. It's disabled currently until I figure out what's going wrong.
11972 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11974 * gst/rtsp-server/rtsp-client.c:
11975 allow pause requests for now.
11978 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11980 * gst/rtsp-server/rtsp-client.c:
11981 Remove weak ref on the session in teardown
11982 We need to remove our weakref from the session when we do a teardown because
11983 else we close the TCP connection prematurely.
11985 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11987 * gst/rtsp-server/rtsp-client.c:
11988 * gst/rtsp-server/rtsp-client.h:
11989 * gst/rtsp-server/rtsp-session-pool.c:
11990 Do some more session cleanup
11991 Make session timeout kill the TCP connection that currently watches the
11993 Remove the client timeout property.
11995 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11997 * gst/rtsp-server/rtsp-client.c:
11998 * gst/rtsp-server/rtsp-client.h:
11999 * gst/rtsp-server/rtsp-media.c:
12000 * gst/rtsp-server/rtsp-media.h:
12001 * gst/rtsp-server/rtsp-server.c:
12002 * gst/rtsp-server/rtsp-session.c:
12003 * gst/rtsp-server/rtsp-session.h:
12005 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
12008 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12010 * examples/Makefile.am:
12011 * examples/test-launch.c:
12012 Add example server that takes launch lines
12013 Add an example server that streams any -launch line.
12015 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12017 * examples/test-readme.c:
12018 * gst/rtsp-server/rtsp-client.c:
12019 * gst/rtsp-server/rtsp-media.c:
12020 * gst/rtsp-server/rtsp-media.h:
12021 Add support for live streams
12022 Add support for live streams and ranges
12023 Start on handling TCP data transfer.
12025 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12027 * gst/rtsp-server/rtsp-media.c:
12028 Free the pipeline before other things
12031 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12033 * gst/rtsp-server/rtsp-client.c:
12034 Only free the pending tunnel if there is one
12037 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12039 * gst/rtsp-server/rtsp-client.c:
12040 * gst/rtsp-server/rtsp-client.h:
12041 * gst/rtsp-server/rtsp-media.c:
12042 rtsp-server: Add support for tunneling
12043 Add support for tunneling over HTTP.
12044 Use new connection methods to retrieve the url.
12045 Dispatch messages based on the message type instead of blindly
12046 assuming it's always a request.
12047 Keep track of the watch id so that we can remove it later.
12048 Set the media pipeline to NULL before unreffing the pipeline.
12050 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12052 * gst/rtsp-server/rtsp-client.c:
12053 * gst/rtsp-server/rtsp-client.h:
12054 Fix for channel -> watch rename in gstreamer
12055 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
12057 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12059 * gst/rtsp-server/rtsp-client.c:
12060 * gst/rtsp-server/rtsp-client.h:
12062 Use the async RTSP channels instead of spawning a new thread for each client.
12063 If a sessionid is specified in a request, fail if we don't have the session.
12065 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12067 * gst/rtsp-server/rtsp-media.c:
12068 Add better debug info
12069 Add some better debug info.
12071 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12073 * examples/test-video.c:
12075 Add support for session timeouts in the example.
12077 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12079 * gst/rtsp-server/rtsp-session-pool.c:
12080 * gst/rtsp-server/rtsp-session-pool.h:
12081 Pass GTimeVal around for performance reasons
12082 Get the current time only once and pass it around so that sessions don't have to
12083 get the current time anymore.
12084 Add experimental support for a GSource that dispatches when the session needs to
12087 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12089 * gst/rtsp-server/rtsp-session.c:
12090 * gst/rtsp-server/rtsp-session.h:
12091 Add better support for session timeouts
12092 Add a method to request the number of milliseconds when a session will timeout.
12094 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12096 * gst/rtsp-server/rtsp-media.c:
12097 * gst/rtsp-server/rtsp-media.h:
12098 Add suport for RTP manager monitoring
12099 Add the first stage in monitoring the rtp manager.
12100 Make sure we don't update the state to something we don't want.
12102 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12104 * gst/rtsp-server/rtsp-client.c:
12105 Add support for session keepalive
12106 Get and update the session timeout for all requests. get the session as early as
12109 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12111 * gst/rtsp-server/rtsp-media-factory.h:
12112 * gst/rtsp-server/rtsp-media.c:
12113 * gst/rtsp-server/rtsp-media.h:
12114 Handle media bus messages
12115 Handle media bus messages in a custom mainloop and dispatch them to the
12116 RTSPMedia objects. Let the default implementation handle some common messages.
12118 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12120 * gst/rtsp-server/rtsp-client.c:
12121 * gst/rtsp-server/rtsp-session-pool.c:
12122 * gst/rtsp-server/rtsp-session.c:
12123 Some more session timeout handling
12124 Move the session header setting code to a central place so that we always add
12125 the timeout parameter too.
12126 Handle timeouts by running the session cleanup code.
12127 Stop media before cleaning up.
12129 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12131 * gst/rtsp-server/rtsp-client.c:
12132 * gst/rtsp-server/rtsp-client.h:
12133 Add timeout property
12134 Add a timeout property ot the client and make the other properties into GObject
12137 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12139 * gst/rtsp-server/rtsp-session-pool.c:
12140 Use getters and setters in property code
12141 Use the getters and setters for the timeout property instead of locking
12144 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12146 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
12148 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12150 * gst/rtsp-server/rtsp-session-pool.c:
12151 * gst/rtsp-server/rtsp-session-pool.h:
12152 * gst/rtsp-server/rtsp-session.c:
12153 * gst/rtsp-server/rtsp-session.h:
12154 Add more timeout stuff
12155 Add method to check if a session is expired.
12156 Add method to perform cleanup on a session pool.
12158 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12160 * gst/rtsp-server/rtsp-client.c:
12161 * gst/rtsp-server/rtsp-session-pool.c:
12162 * gst/rtsp-server/rtsp-session-pool.h:
12163 * gst/rtsp-server/rtsp-session.c:
12164 * gst/rtsp-server/rtsp-session.h:
12165 Add beginnings of session timeouts and limits
12166 Add the timeout value to the Session header for unusual timeout values.
12167 Allow us to configure a limit to the amount of active sessions in a pool. Set a
12168 limit on the amount of retry we do after a sessionid collision.
12169 Add properties to the sessionid and the timeout of a session. Keep track of
12170 creation time and last access time for sessions.
12172 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12174 * gst/rtsp-server/rtsp-client.c:
12175 * gst/rtsp-server/rtsp-media.c:
12176 * gst/rtsp-server/rtsp-media.h:
12177 * gst/rtsp-server/rtsp-sdp.c:
12178 * gst/rtsp-server/rtsp-session-pool.c:
12179 * gst/rtsp-server/rtsp-session.c:
12180 * gst/rtsp-server/rtsp-session.h:
12181 Cleanup of sessions and more
12182 Fix the refcounting of media and sessions in the client. Properly clean up the
12183 session data when the client performs a teardown.
12184 Add Server header to responses.
12185 Allow for multiple uri setups in one session.
12186 Add Range header to the PLAY response and add the range attribute to the SDP
12188 Fix the session pool remove method, it used the wrong key in the hashtable. Also
12189 give the ownership of the sessionid to the session object.
12191 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12193 * gst/rtsp-server/rtsp-server.c:
12194 * gst/rtsp-server/rtsp-server.h:
12196 Rename the 'server_port' variable to simply 'port'.
12198 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12201 * gst/rtsp-server/rtsp-client.c:
12202 * gst/rtsp-server/rtsp-media.c:
12203 * gst/rtsp-server/rtsp-media.h:
12204 * gst/rtsp-server/rtsp-session.c:
12205 * gst/rtsp-server/rtsp-session.h:
12206 Rework the way we handle transports for streams
12207 Make the media accept an array of transports for the streams that we have
12208 configured for the play/pause requests.
12209 Implement server states for a client and its media.
12210 Require 0.10.22.1 (git HEAD) of gstreamer.
12212 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12214 * gst/rtsp-server/rtsp-client.c:
12215 * gst/rtsp-server/rtsp-media-factory.c:
12216 Drop const from functions dealing with urls
12217 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
12218 have the right const in them.
12220 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12222 * gst/rtsp-server/rtsp-client.c:
12223 * gst/rtsp-server/rtsp-media.c:
12224 * gst/rtsp-server/rtsp-sdp.c:
12228 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12230 * gst/rtsp-server/rtsp-client.c:
12231 * gst/rtsp-server/rtsp-media-factory.c:
12232 * gst/rtsp-server/rtsp-media.c:
12233 * gst/rtsp-server/rtsp-media.h:
12235 Don't keep a reference to the GstRTSPMedia in the stream.
12236 Free more things when freeing the GstRTSPMedia.
12238 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12241 * gst/rtsp-server/rtsp-media-factory.c:
12242 * gst/rtsp-server/rtsp-media-factory.h:
12243 * gst/rtsp-server/rtsp-media.c:
12244 * gst/rtsp-server/rtsp-media.h:
12245 * gst/rtsp-server/rtsp-server.c:
12246 * gst/rtsp-server/rtsp-server.h:
12247 More docs and small cleanups
12248 Add some more docs and update the README
12249 Cleanup some method names.
12250 Remove an unneeded idx field in the GstRTSPMediaStream
12252 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12255 * examples/Makefile.am:
12256 * examples/test-readme.c:
12257 Add a README and more example code
12258 Add a README file that contains a small introduction on how to use the server
12259 along with the example code explained in the readme.
12261 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12263 * gst/rtsp-server/rtsp-media.c:
12264 * gst/rtsp-server/rtsp-server.c:
12265 Fix some leaks and change default port
12266 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
12267 we finished the initial preroll. If we keep them locked, setting the pipeline to
12268 NULL will not stop and clean up the sources correctly.
12269 Change the default RTSP port to 8554 aka the official alternative RTSP port.
12271 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12273 * gst/rtsp-server/rtsp-session.c:
12274 * gst/rtsp-server/rtsp-session.h:
12275 Cleanups to the session object
12276 Remove some unneeded variables in the session state of a stream such as the
12277 owner media and the server transport.
12278 Get the configuration of a media stream in a session based on the media_stream
12279 in the original object instead of our cached index.
12280 Free more data in the finalize method.
12282 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12284 * gst/rtsp-server/rtsp-client.c:
12285 * gst/rtsp-server/rtsp-client.h:
12286 Cleanups and reuse media from DESCRIBE
12287 Handle thread create errors.
12288 Rename some internal methods to better match what they actually do.
12289 Handle misconfiguration of session_pool and media_mapping gracefully.
12290 Cache the DESCRIBE media and uri in the client connection and reuse them when
12291 we receive a SETUP request in the same connection for the same uri.
12292 Cleanup the client connection object.
12294 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12296 * gst/rtsp-server/rtsp-media-factory.c:
12297 * gst/rtsp-server/rtsp-media-factory.h:
12298 * gst/rtsp-server/rtsp-media.c:
12299 * gst/rtsp-server/rtsp-media.h:
12300 Add shared properties to media and factory
12301 Add the shared property to media.
12302 Implement some simple caching in the factory depending on if the media is shared
12305 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12307 * gst/rtsp-server/rtsp-client.c:
12308 Add a little comment
12309 Add some comment about the content-base header.
12311 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12313 * examples/Makefile.am:
12314 * examples/test-mp4.c:
12315 * examples/test-ogg.c:
12316 * examples/test-video.c:
12317 * gst/rtsp-server/Makefile.am:
12318 * gst/rtsp-server/rtsp-client.c:
12319 * gst/rtsp-server/rtsp-client.h:
12320 * gst/rtsp-server/rtsp-media-factory.c:
12321 * gst/rtsp-server/rtsp-media-factory.h:
12322 * gst/rtsp-server/rtsp-media.c:
12323 * gst/rtsp-server/rtsp-media.h:
12324 * gst/rtsp-server/rtsp-sdp.c:
12325 * gst/rtsp-server/rtsp-sdp.h:
12326 * gst/rtsp-server/rtsp-server.c:
12327 * gst/rtsp-server/rtsp-server.h:
12328 * gst/rtsp-server/rtsp-session.c:
12329 * gst/rtsp-server/rtsp-session.h:
12330 Reorganize things, prepare for media sharing
12331 Added various other test server examples
12332 Move the SDP message generation to a separate helper.
12333 Refactor common code for finding the session.
12334 Add content-base for realplayer compatibility
12335 Clean up request uris before processing for better vlc compatibility.
12336 Move prerolling and pipeline construction to the RTSPMedia object.
12337 Use multiudpsink for future pipeline reuse.
12339 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12342 Back to development
12345 === release 0.10.1 ===
12347 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12350 Make 0.10.1 release
12353 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12355 * bindings/vala/Makefile.am:
12357 Add more directories and files to the dist.
12359 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
12361 * bindings/python/Makefile.am:
12362 * bindings/python/rtspserver.override:
12363 Fixed compile error of python bindings
12365 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
12367 * bindings/vala/gst-rtsp-server-0.10.vapi:
12368 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
12369 Marked values as nullable accordingly
12371 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
12373 * bindings/vala/gst-rtsp-server-0.10.vapi:
12374 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
12375 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
12376 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
12377 Updated Vala bindings
12379 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12381 * gst/rtsp-server/rtsp-client.c:
12382 * gst/rtsp-server/rtsp-media-mapping.c:
12383 * gst/rtsp-server/rtsp-media-mapping.h:
12384 * gst/rtsp-server/rtsp-media.h:
12385 * gst/rtsp-server/rtsp-session-pool.h:
12386 Cleanups and doc updates
12387 Add some more documentation and do some minor cleanups here and there.
12389 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12391 * gst/rtsp-server/rtsp-client.c:
12392 * gst/rtsp-server/rtsp-media-factory.c:
12393 * gst/rtsp-server/rtsp-media-factory.h:
12394 * gst/rtsp-server/rtsp-media.c:
12395 * gst/rtsp-server/rtsp-media.h:
12396 * gst/rtsp-server/rtsp-session.c:
12397 * gst/rtsp-server/rtsp-session.h:
12399 Rename GstRTSPMediaBin to GstRTSPMedia
12400 Parse the request url into a GstRTSPUri object and pass this object to the
12401 various handlers and methods that require the uri.
12403 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12407 Add some more docs and remove some old code from the example.
12409 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12411 * gst/rtsp-server/rtsp-client.c:
12412 Handle state change failures better
12413 Handle state change failures better when changing the state of the pipeline to
12416 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12418 * gst/rtsp-server/rtsp-media-factory.c:
12419 * gst/rtsp-server/rtsp-media-factory.h:
12420 Make element creation more extendible
12421 Add get_element vmethod to the default MediaFactory so that subclasses can just
12422 override that method and still use the default logic for making a MediaBin from
12425 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12428 * gst/rtsp-server/Makefile.am:
12429 * gst/rtsp-server/rtsp-client.c:
12430 * gst/rtsp-server/rtsp-client.h:
12431 * gst/rtsp-server/rtsp-media-factory.c:
12432 * gst/rtsp-server/rtsp-media-factory.h:
12433 * gst/rtsp-server/rtsp-media-mapping.c:
12434 * gst/rtsp-server/rtsp-media-mapping.h:
12435 * gst/rtsp-server/rtsp-media.c:
12436 * gst/rtsp-server/rtsp-media.h:
12437 * gst/rtsp-server/rtsp-server.c:
12438 * gst/rtsp-server/rtsp-server.h:
12439 * gst/rtsp-server/rtsp-session.c:
12440 * gst/rtsp-server/rtsp-session.h:
12441 Make the server handle arbitrary pipelines
12442 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
12443 The GstMediaBin object has a handle to a bin with elements and to a list of
12444 GstMediaStream objects that this bin produces.
12445 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
12446 with methods to register and remove those mappings.
12447 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
12448 used by the server instance.
12449 Modify the example application so that it shows how to create custom pipelines
12450 attached to a specific mount point.
12451 Various misc cleanps.
12453 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12455 * gst/rtsp-server/rtsp-server.c:
12456 * gst/rtsp-server/rtsp-server.h:
12457 Allow setting a custom media factory for a server
12459 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12461 * gst/rtsp-server/rtsp-client.c:
12462 * gst/rtsp-server/rtsp-client.h:
12463 Allow setting a custom media factory for a client.
12465 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12467 * gst/rtsp-server/Makefile.am:
12468 Add Makefile entry for the media factory
12470 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12472 * gst/rtsp-server/rtsp-media-factory.c:
12473 * gst/rtsp-server/rtsp-media-factory.h:
12474 Add media factory to map urls to media pipeline objects.
12476 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12478 * gst/rtsp-server/rtsp-media.c:
12479 * gst/rtsp-server/rtsp-media.h:
12480 Add comments. Remove unused field
12482 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12484 * gst/rtsp-server/rtsp-session-pool.c:
12485 * gst/rtsp-server/rtsp-session-pool.h:
12486 Allow custom session pools to override the session id allocation algorithms Add some comments.
12488 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12490 * gst/rtsp-server/rtsp-session.h:
12493 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12495 * gst/rtsp-server/rtsp-client.c:
12496 * gst/rtsp-server/rtsp-client.h:
12497 Move the connection code in one place Add some comments
12499 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12501 * gst/rtsp-server/rtsp-server.c:
12502 * gst/rtsp-server/rtsp-server.h:
12503 Make vmethod to create and accept new clients. Add some docs.
12505 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12507 * gst/rtsp-server/rtsp-server.c:
12508 * gst/rtsp-server/rtsp-server.h:
12509 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
12511 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12513 * gst/rtsp-server/rtsp-client.c:
12514 * gst/rtsp-server/rtsp-client.h:
12515 Name the parameters more appropriately.
12517 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12519 * gst/rtsp-server/rtsp-session-pool.c:
12520 Do some more cleanup of the session pool.
12522 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12524 * gst/rtsp-server/Makefile.am:
12525 * gst/rtsp-server/rtsp-client.c:
12526 Check if return value of gst_rtsp_session_get_media is not NULL
12528 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12530 * gst/rtsp-server/Makefile.am:
12531 Install rtsp-session and rtsp-session-pool headers
12533 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12538 * bindings/python/Makefile.am:
12539 * bindings/python/arg-types.py:
12540 * bindings/python/codegen/Makefile.am:
12541 * bindings/python/codegen/__init__.py:
12542 * bindings/python/codegen/argtypes.py:
12543 * bindings/python/codegen/code-coverage.py:
12544 * bindings/python/codegen/codegen.py:
12545 * bindings/python/codegen/definitions.py:
12546 * bindings/python/codegen/defsparser.py:
12547 * bindings/python/codegen/docextract.py:
12548 * bindings/python/codegen/docgen.py:
12549 * bindings/python/codegen/fileprefix.override:
12550 * bindings/python/codegen/fileprefixmodule.c:
12551 * bindings/python/codegen/h2def.py:
12552 * bindings/python/codegen/mergedefs.py:
12553 * bindings/python/codegen/mkskel.py:
12554 * bindings/python/codegen/override.py:
12555 * bindings/python/codegen/reversewrapper.py:
12556 * bindings/python/codegen/scmexpr.py:
12557 * bindings/python/rtspserver-types.defs:
12558 * bindings/python/rtspserver.defs:
12559 * bindings/python/rtspserver.override:
12560 * bindings/python/rtspservermodule.c:
12562 Add python bindings.
12564 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12566 * bindings/Makefile.am:
12568 Don't go into python dir when requirements for python bindings are missing
12570 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12572 * bindings/Makefile.am:
12573 * bindings/vala/Makefile.am:
12575 Install Vala bindings if vala is available
12577 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
12579 * bindings/vala/gst-rtsp-server-0.10.deps:
12580 * bindings/vala/gst-rtsp-server-0.10.vapi:
12581 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
12582 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
12583 * bindings/vala/packages/gst-rtsp-server-0.10.files:
12584 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
12585 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
12586 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
12587 Regenerated Vala bindings
12589 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
12591 * bindings/vala/gst-rtsp-server.vapi:
12592 * bindings/vala/packages/gst-rtsp-server.metadata:
12593 Fixed typo in included headers for vala bindings
12595 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12599 * pkgconfig/Makefile.am:
12600 * pkgconfig/gst-rtsp-server.pc.in:
12601 Added pkgconfig file
12603 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
12605 * bindings/vala/gst-rtsp-server.vapi:
12606 * bindings/vala/packages/gst-rtsp-server.excludes:
12607 * bindings/vala/packages/gst-rtsp-server.gi:
12608 * bindings/vala/packages/gst-rtsp-server.metadata:
12609 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
12611 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
12613 * bindings/vala/gst-rtsp-server.vapi:
12614 * bindings/vala/packages/gst-rtsp-server.deps:
12615 * bindings/vala/packages/gst-rtsp-server.files:
12616 * bindings/vala/packages/gst-rtsp-server.gi:
12617 * bindings/vala/packages/gst-rtsp-server.metadata:
12618 * bindings/vala/packages/gst-rtsp-server.namespace:
12619 Added Vala bindings
12621 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
12623 * gst/rtsp-server/rtsp-session.c:
12624 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
12626 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
12628 * examples/Makefile.am:
12629 * gst/rtsp-server/Makefile.am:
12630 Put GStreamer version in library name
12632 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12634 * examples/Makefile.am:
12635 * gst/rtsp-server/Makefile.am:
12636 Fix some issues to pass distcheck
12638 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12640 * gst/rtsp-server/rtsp-server.c:
12641 Added port property to GstRTSPServer class.
12643 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12648 * examples/Makefile.am:
12651 * gst/rtsp-server/Makefile.am:
12652 * gst/rtsp-server/rtsp-client.c:
12653 * gst/rtsp-server/rtsp-client.h:
12654 * gst/rtsp-server/rtsp-media.c:
12655 * gst/rtsp-server/rtsp-media.h:
12656 * gst/rtsp-server/rtsp-server.c:
12657 * gst/rtsp-server/rtsp-server.h:
12658 * gst/rtsp-server/rtsp-session-pool.c:
12659 * gst/rtsp-server/rtsp-session-pool.h:
12660 * gst/rtsp-server/rtsp-session.c:
12661 * gst/rtsp-server/rtsp-session.h:
12663 Split in library and example program
12665 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
12667 * src/rtsp-client.h:
12668 Removed obsolete variable
12670 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
12672 * src/rtsp-client.c:
12673 * src/rtsp-client.h:
12674 Removed pipeline variable GstRTSPClient, because it's only used in one function
12676 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12678 * src/rtsp-media.c:
12679 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
12681 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
12683 * src/rtsp-session.c:
12684 Initialize some more vars.
12686 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
12688 * src/rtsp-session.c:
12689 Initialize variable to avoid compiler warning.
12691 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
12694 Add a reasonable generic .gitignore
projects / platform / upstream / gstreamer.git / blob