3 2019-09-23 11:17:41 +0100 Tim-Philipp Müller <tim@centricular.com>
9 * gst-rtsp-server.doap:
13 2019-08-30 14:00:52 +0200 Kristofer Björkström <kristofb@axis.com>
15 * gst/rtsp-server/rtsp-client.c:
16 rtsp-client: RTP Info must exist in PLAY response
17 If RTP Info is missing. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR
20 2019-06-25 13:19:44 +0100 Tim-Philipp Müller <tim@centricular.com>
22 * gst/rtsp-server/rtsp-onvif-media-factory.c:
23 * gst/rtsp-server/rtsp-onvif-media.c:
24 onvif-media: fix "void function returning a value" compiler warning
26 2019-04-23 14:38:05 +0300 Sebastian Dröge <sebastian@centricular.com>
28 * gst/rtsp-server/rtsp-media-factory.c:
29 * gst/rtsp-server/rtsp-media.c:
30 * gst/rtsp-server/rtsp-stream-transport.c:
31 * gst/rtsp-server/rtsp-stream.c:
32 rtsp-server: Add various missing Since: 1.16 markers
34 2019-04-23 15:01:32 +0300 Sebastian Dröge <sebastian@centricular.com>
36 * gst/rtsp-server/rtsp-media.c:
37 * gst/rtsp-server/rtsp-sdp.c:
38 * gst/rtsp-server/rtsp-session-media.c:
39 * gst/rtsp-server/rtsp-stream.c:
40 rtsp-server: Add various Since: 1.14 markers
42 2019-04-23 15:09:34 +0300 Sebastian Dröge <sebastian@centricular.com>
44 * gst/rtsp-server/rtsp-auth.c:
45 * gst/rtsp-server/rtsp-client.h:
46 rtsp-server: Fix various Since markers
48 2019-05-02 12:35:34 +0100 Tim-Philipp Müller <tim@centricular.com>
51 ci: use template from 1.16 branch
53 === release 1.16.0 ===
55 2019-04-19 00:34:54 +0100 Tim-Philipp Müller <tim@centricular.com>
61 * gst-rtsp-server.doap:
65 2019-04-15 20:33:01 +0300 Sebastian Dröge <sebastian@centricular.com>
67 * gst/rtsp-sink/gstrtspclientsink.c:
68 rtspclientsink: Notify the stream transport about each written message
69 Otherwise it will never try to send us the next one: it tries to keep
70 exactly one message in-flight all the time.
71 In gst-rtsp-server this is done asynchronously via the GstRTSPWatch but
72 in the client sink we always write data out synchronously.
74 2019-04-02 08:05:03 +0200 Göran Jönsson <goranjn@axis.com>
76 * gst/rtsp-server/rtsp-stream.c:
77 rtsp_server: Free thread pool before clean transport cache
78 If not waiting for free thread pool before clean transport caches, there
79 can be a crash if a thread is executing in transport list loop in
80 function send_tcp_message.
81 Also add a check if priv->send_pool in on_message_sent to avoid that a
82 new thread is pushed during wait of free thread pool. This is possible
83 since when waiting for free thread pool mutex have to be unlocked.
85 === release 1.15.90 ===
87 2019-04-11 00:35:55 +0100 Tim-Philipp Müller <tim@centricular.com>
93 * gst-rtsp-server.doap:
97 2019-04-10 10:32:53 +0200 Ulf Olsson <ulfo@axis.com>
99 * gst/rtsp-server/rtsp-stream.c:
100 rtsp-stream: Add support for GCM (RFC 7714)
103 2019-03-28 00:27:37 +0100 Erlend Eriksen <erlend_ne@hotmail.com>
105 * gst/rtsp-server/rtsp-session-pool.c:
106 session pool: fix missing klass-> in klass->create_session
108 2019-03-23 19:16:17 +0000 Tim-Philipp Müller <tim@centricular.com>
111 g-i: pass --quiet to g-ir-scanner
112 This suppresses the annoying 'g-ir-scanner: link: cc ..' output
113 that we get even if everything works just fine.
114 We still get g-ir-scanner warnings and compiler warnings if
117 2019-03-23 19:15:48 +0000 Tim-Philipp Müller <tim@centricular.com>
120 g-i: silence 'nested extern' compiler warnings when building scanner binary
121 We need a nested extern in our init section for the scanner binary
122 so we can call gst_init to make sure GStreamer types are initialised
123 (they are not all lazy init via get_type functions, but some are in
124 exported variables). There doesn't seem to be any other mechanism to
125 achieve this, so just remove that warning, it's not important at all.
127 2019-03-21 11:49:10 +0000 Tim-Philipp Müller <tim@centricular.com>
130 meson: pass -Wno-unused to compiler if gstreamer debug system is disabled
132 2019-03-14 07:37:26 +0100 Göran Jönsson <goranjn@axis.com>
134 * gst/rtsp-server/rtsp-media.c:
135 * tests/check/gst/media.c:
136 rtsp-media: Handle set state when preparing.
137 Handle the situation when a call to gst_rtsp_media_set_state is done
138 when media status is preparing.
139 Also add unit test for this scenario.
140 The unit test simulate on a media level when two clients share a (live)
142 Both clients have done SETUP and got responses. Now client 1 is doing
143 play and client 2 is just closing the connection.
144 Then without patch there are a problem when
145 client1 is calling gst_rtsp_media_unsuspend in handle_play_request.
146 And client2 is doing closing connection we can end up in a call
147 to gst_rtsp_media_set_state when
148 priv->status == GST_RTSP_MEDIA_STATUS_PREPARING and all the logic for
149 shut down media is jumped over .
150 With this patch and this scenario we wait until
151 priv->status == GST_RTSP_MEDIA_STATUS_PREPARED and then continue to
152 execute after that and now we will execute the logic for
155 2019-03-04 09:13:30 +0000 Tim-Philipp Müller <tim@centricular.com>
163 === release 1.15.2 ===
165 2019-02-26 11:58:53 +0000 Tim-Philipp Müller <tim@centricular.com>
171 * gst-rtsp-server.doap:
175 2019-02-19 09:45:08 +0100 Göran Jönsson <goranjn@axis.com>
177 * gst/rtsp-server/rtsp-media.c:
178 * tests/check/gst/client.c:
179 rtsp-media: Fix multicast use case with common media
188 2019-01-16 12:59:11 +0100 Göran Jönsson <goranjn@axis.com>
190 * gst/rtsp-server/rtsp-client.c:
191 * gst/rtsp-server/rtsp-stream.c:
192 * gst/rtsp-server/rtsp-stream.h:
193 rtsp-server: remove recursive behavior
194 Introduce a threadpool to send rtp and rtcp to avoid recursive behavior.
196 2019-01-25 14:22:42 +0200 Sebastian Dröge <sebastian@centricular.com>
198 * gst/rtsp-server/rtsp-client.c:
199 rtsp-client: Only allow to set either a send_func or send_messages_func but not both
200 And route all messages through the send_func if no send_messages_func
202 We otherwise break backwards compatibility.
204 2018-09-17 22:18:46 +0300 Sebastian Dröge <sebastian@centricular.com>
206 * docs/libs/gst-rtsp-server-sections.txt:
207 * gst/rtsp-server/rtsp-client.c:
208 * gst/rtsp-server/rtsp-client.h:
209 * gst/rtsp-server/rtsp-stream.c:
210 rtsp-client: Add support for sending buffer lists directly
211 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
213 2018-06-27 12:17:07 +0200 Sebastian Dröge <sebastian@centricular.com>
215 * docs/libs/gst-rtsp-server-sections.txt:
216 * gst/rtsp-server/rtsp-client.c:
217 * gst/rtsp-server/rtsp-media.c:
218 * gst/rtsp-server/rtsp-stream-transport.c:
219 * gst/rtsp-server/rtsp-stream-transport.h:
220 * gst/rtsp-server/rtsp-stream.c:
221 * gst/rtsp-sink/gstrtspclientsink.c:
222 rtsp-server: Add support for buffer lists
223 This adds new functions for passing buffer lists through the different
224 layers without breaking API/ABI, and enables the appsink to actually
225 provide buffer lists.
226 This should already reduce CPU usage and potentially context switches a
227 bit by passing a whole buffer list from the appsink instead of
228 individual buffers. As a next step it would be necessary to
229 a) Add support for a vector of data for the GstRTSPMessage body
230 b) Add support for sending multiple messages at once to the
231 GstRTSPWatch and let it be handled internally
232 c) Adding API to GOutputStream that works like writev()
233 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
235 2018-12-04 14:12:04 +0100 Benjamin Berg <bberg@redhat.com>
237 * gst/rtsp-server/rtsp-client.c:
238 client: Fix crash in close handler
239 The close handler could trigger a crash because it invalidated the
240 watch_context while still leaving a source attached to it which would be
241 cleaned up at a later point.
243 2019-01-29 14:42:35 +0100 Edward Hervey <edward@centricular.com>
245 * gst/rtsp-server/rtsp-stream.c:
246 rtsp-stream: Use cached address when allocating sockets
247 If an address/port was previously decided upon (ex: multicast in the
248 SDP), then use that instead of re-creating another one
249 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/57
251 2018-12-27 11:28:17 +0100 Lars Wiréen <larswi@axis.com>
253 * gst/rtsp-server/rtsp-media.c:
254 rtsp-media: Fix race codition in finish_unprepare
255 The previous fix for race condition around finish_unprepare where the
256 function could be called twice assumed that the status wouldn't change
257 during execution of the function. This assumption is incorrect as the
258 state may change, for example if an error message arrives from the
260 Instead a flag keeping track on whether the finish_unprepare function
261 is currently executing is introduced and checked.
262 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/59
264 === release 1.15.1 ===
266 2019-01-17 02:26:48 +0000 Tim-Philipp Müller <tim@centricular.com>
272 * gst-rtsp-server.doap:
276 2018-12-05 15:07:25 +0100 Patricia Muscalu <patricia@axis.com>
278 * gst/rtsp-server/rtsp-stream.c:
279 Add source elements to the pipeline before activation
280 In plug_src we changed the element state before adding it to
281 the owner container. This prevented the pipeline from intercepting
282 a GST_STREAM_STATUS_TYPE_CREATE message from the pad in order
283 to assign a custom task pool.
284 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/53
286 2018-12-05 17:24:59 -0300 Thibault Saunier <tsaunier@igalia.com>
289 Automatic update of common submodule
290 From ed78bee to 59cb678
292 2018-11-20 19:12:09 +0100 Ingo Randolf <ingo.randolf@servus.at>
294 * examples/test-appsrc.c:
295 examples: test-appsrc: fix coding style error
297 2018-11-20 11:07:48 +0100 Ingo Randolf <ingo.randolf@servus.at>
299 * examples/test-appsrc.c:
300 examples: test-appsrc: fix buffer leak
302 2018-11-17 19:19:54 +0100 Patricia Muscalu <patricia@axis.com>
304 * gst/rtsp-server/rtsp-media.c:
305 rtsp-media: Update priv->blocked when linked streams are unblocked.
306 Media is considered to be blocked when all streams that belong to
307 that media are blocked.
308 This patch solves the problem of inconsistent updates of
309 priv->blocked that are not synchronized with the media state.
311 2018-11-17 18:18:27 +0100 Patricia Muscalu <patricia@axis.com>
313 * gst/rtsp-server/rtsp-media.c:
314 rtsp-media: Don't block streams before seeking
315 Before the seek operation is performed on media, it's required that
316 its pipeline is prepared <=> the pipeline is in the PAUSED state.
317 At this stage, all transport parts (transport sinks) have been successfully
318 added to the pipeline and there is no need for blocking the streams.
320 2018-11-17 16:11:53 +0100 Patricia Muscalu <patricia@axis.com>
322 * tests/check/gst/rtspserver.c:
323 tests: rtspserver: Add shared media test case for TCP
325 2018-11-06 18:21:54 +0100 Linus Svensson <linussn@axis.com>
327 * gst/rtsp-server/rtsp-stream.c:
328 rtsp-stream: Use seqnum-offset for rtpinfo
329 The sequence number in the rtpinfo is supposed to be the first RTP
330 sequence number. The "seqnum" property on a payloader is supposed to be
331 the number from the last processed RTP packet. The sequence number for
332 payloaders that inherit gstrtpbasepayload will not be correct in case of
333 buffer lists. In order to fix the seqnum property on the payloaders
334 gst-rtsp-server must get the sequence number for rtpinfo elsewhere and
335 "seqnum-offset" from the "stats" property contains the value of the
336 very first RTP packet in a stream. The server will, however, try to look
337 at the last simple in the sink element and only use properties on the
338 payloader in case there no sink elements yet, and by looking at the last
339 sample of the sink gives the server full control of which RTP packet it
340 looks at. If the payloader does not have the "stats" property, "seqnum"
341 is still used since "seqnum-offset" is only present in as part of
342 "stats" and this is still an issue not solved with this patch.
343 Needed for gst-plugins-base!17
345 2018-11-06 18:10:56 +0100 Linus Svensson <linussn@axis.com>
347 * gst/rtsp-server/rtsp-stream.c:
348 rtsp-stream: Plug memory leak
349 Attaching a GSource to a context will increase the refcount. The idle
350 source will never be free'd since the initial reference is never
353 2018-11-12 16:06:39 +0200 Jordan Petridis <jordan@centricular.com>
356 Add Gitlab CI configuration
357 This commit adds a .gitlab-ci.yml file, which uses a feature
358 to fetch the config from a centralized repository. The intent is
359 to have all the gstreamer modules use the same configuration.
360 The configuration is currently hosted at the gst-ci repository
361 under the gitlab/ci_template.yml path.
362 Part of https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/29
364 2018-11-05 05:56:35 +0000 Matthew Waters <matthew@centricular.com>
367 * gst-rtsp-server.doap:
368 Update git locations to gitlab
370 2018-11-01 14:20:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
372 * gst/rtsp-server/meson.build:
373 meson: add new onvif types
375 2018-11-01 12:49:51 +0200 Sebastian Dröge <sebastian@centricular.com>
377 * gst/rtsp-server/meson.build:
378 Add ONVIF subclass headers to the installed headers in meson.build too
380 2018-11-01 11:29:01 +0200 Sebastian Dröge <sebastian@centricular.com>
382 * gst/rtsp-server/rtsp-server-object.h:
383 * gst/rtsp-server/rtsp-server.h:
384 rtsp-server: Declare GstRTSPServer struct before anything else
385 It's needed by all kinds of other headers, including the ones that are
386 required for defining the GstRTSPServer struct itself and its API.
388 2018-11-01 10:23:02 +0200 Sebastian Dröge <sebastian@centricular.com>
390 * gst/rtsp-server/rtsp-onvif-client.h:
391 * gst/rtsp-server/rtsp-onvif-media-factory.h:
392 * gst/rtsp-server/rtsp-onvif-media.h:
393 * gst/rtsp-server/rtsp-onvif-server.h:
394 Mark all ONVIF-specific subclasses as Since 1.14
396 2018-11-01 10:18:22 +0200 Sebastian Dröge <sebastian@centricular.com>
398 * gst/rtsp-server/Makefile.am:
399 * gst/rtsp-server/meson.build:
400 * gst/rtsp-server/rtsp-context.h:
401 * gst/rtsp-server/rtsp-onvif-server.c:
402 * gst/rtsp-server/rtsp-onvif-server.h:
403 * gst/rtsp-server/rtsp-server-object.h:
404 * gst/rtsp-server/rtsp-server-prelude.h:
405 * gst/rtsp-server/rtsp-server.c:
406 * gst/rtsp-server/rtsp-server.h:
407 * gst/rtsp-server/rtsp-session.h:
408 Include ONVIF types from single-include rtsp-server.h
409 ... by actually making it a single-include header and moving everything
410 related to the GstRTSPServer type to rtsp-server-object.h instead.
411 Otherwise there are too many circular includes.
412 https://bugzilla.gnome.org/show_bug.cgi?id=797361
414 2018-10-18 07:25:05 +0200 Göran Jönsson <goranjn@axis.com>
416 * gst/rtsp-server/rtsp-client.c:
417 * gst/rtsp-server/rtsp-latency-bin.c:
418 * gst/rtsp-server/rtsp-stream.c:
419 * gst/rtsp-server/rtsp-stream.h:
420 rtsp-stream: use idle source in on_message_sent
421 When the underlying layers are running on_message_sent, this sometimes
422 causes the underlying layer to send more data, which will cause the
423 underlying layer to run callback on_message_sent again. This can go on
425 To break this chain, we introduce an idle source that takes care of
426 sending data if there are more to send when running callback
427 https://bugzilla.gnome.org/show_bug.cgi?id=797289
429 2018-10-20 16:14:53 +0200 Edward Hervey <edward@centricular.com>
431 * gst/rtsp-server/rtsp-client.c:
432 rtsp-client: Remove timeout GSource on cleanup
433 Avoids ending up with races where a timeout would still be around
434 *after* a client was gone. This could happen rather easily in
435 RTSP-over-HTTP mode on a local connection, where each RTSP message
436 would be sent as a different HTTP connection with the same tunnelid.
437 If not properly removed, that timeout would then try to free again
438 a client (and its contents).
440 2018-10-04 14:31:59 +0100 Tim-Philipp Müller <tim@centricular.com>
442 * gst/rtsp-server/Makefile.am:
443 autotools: fix distcheck
445 2018-09-12 11:55:15 +0200 Ognyan Tonchev <ognyan@axis.com>
447 * gst/rtsp-server/Makefile.am:
448 * gst/rtsp-server/meson.build:
449 * gst/rtsp-server/rtsp-latency-bin.c:
450 * gst/rtsp-server/rtsp-latency-bin.h:
451 * gst/rtsp-server/rtsp-onvif-media.c:
452 onvif: encapsulate onvif part into a bin
453 ...and thus do not let onvif affect pipelines latency
454 https://bugzilla.gnome.org/show_bug.cgi?id=797174
456 2018-09-27 19:57:13 +0200 Patricia Muscalu <patricia@dovakhiin.com>
458 * tests/check/gst/client.c:
459 tests: client: Avoid bind() failures in tests
460 https://bugzilla.gnome.org/show_bug.cgi?id=797059
462 2018-09-06 16:17:33 +0200 Patricia Muscalu <patricia@axis.com>
464 * gst/rtsp-server/rtsp-media-factory.c:
465 * gst/rtsp-server/rtsp-media-factory.h:
466 * gst/rtsp-server/rtsp-media.c:
467 * gst/rtsp-server/rtsp-media.h:
468 * gst/rtsp-server/rtsp-stream.c:
469 * gst/rtsp-server/rtsp-stream.h:
470 * tests/check/gst/client.c:
471 * tests/check/gst/mediafactory.c:
472 New property for socket binding to mcast addresses
473 By default the multicast sockets are bound to INADDR_ANY,
474 as it's not allowed to bind sockets to multicast addresses
475 in Windows. This default behaviour can be changed by setting
476 bind-mcast-address property on the media-factory object.
477 https://bugzilla.gnome.org/show_bug.cgi?id=797059
479 2018-09-24 09:36:21 +0100 Tim-Philipp Müller <tim@centricular.com>
482 * gst/rtsp-server/Makefile.am:
483 * gst/rtsp-server/meson.build:
484 * gst/rtsp-server/rtsp-address-pool.c:
485 * gst/rtsp-server/rtsp-auth.c:
486 * gst/rtsp-server/rtsp-client.c:
487 * gst/rtsp-server/rtsp-context.c:
488 * gst/rtsp-server/rtsp-media-factory-uri.c:
489 * gst/rtsp-server/rtsp-media-factory.c:
490 * gst/rtsp-server/rtsp-media.c:
491 * gst/rtsp-server/rtsp-mount-points.c:
492 * gst/rtsp-server/rtsp-params.c:
493 * gst/rtsp-server/rtsp-permissions.c:
494 * gst/rtsp-server/rtsp-sdp.c:
495 * gst/rtsp-server/rtsp-server-prelude.h:
496 * gst/rtsp-server/rtsp-server.c:
497 * gst/rtsp-server/rtsp-session-media.c:
498 * gst/rtsp-server/rtsp-session-pool.c:
499 * gst/rtsp-server/rtsp-session.c:
500 * gst/rtsp-server/rtsp-stream-transport.c:
501 * gst/rtsp-server/rtsp-stream.c:
502 * gst/rtsp-server/rtsp-thread-pool.c:
503 * gst/rtsp-server/rtsp-token.c:
505 libs: fix API export/import and 'inconsistent linkage' on MSVC
506 Export rtsp-server library API in headers when we're building the
507 library itself, otherwise import the API from the headers.
508 This fixes linker warnings on Windows when building with MSVC.
509 Fix up some missing config.h includes when building the lib which
510 is needed to get the export api define from config.h
511 https://bugzilla.gnome.org/show_bug.cgi?id=797185
513 2018-09-19 14:31:56 +0200 Edward Hervey <edward@centricular.com>
515 * gst/rtsp-server/rtsp-media-factory.c:
516 rtsp-media-factory: Add missing break statements
517 This resulted in warnings/assertions whenever one accessed the
518 max-mcast-ttl property.
522 2018-09-19 12:21:30 +0100 Tim-Philipp Müller <tim@centricular.com>
526 meson: add gobject-cast-checks, glib-asserts, glib-checks options
528 2018-09-19 12:17:57 +0100 Tim-Philipp Müller <tim@centricular.com>
532 * tests/check/meson.build:
533 meson: add option to disable build of rtspclientsink plugin
535 2018-09-19 12:10:14 +0100 Tim-Philipp Müller <tim@centricular.com>
538 meson: re-arrange options
540 2018-09-01 11:21:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
544 * tests/check/meson.build:
546 meson: Use feature option for tests option
547 This was somehow missed the last time around.
549 2018-08-31 14:42:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
551 * gst/rtsp-server/meson.build:
553 meson: Maintain macOS ABI through dylib versioning
554 Requires Meson 0.48, but the feature will be ignored on older versions
555 so it's safe to add it without bumping the requirement.
557 https://github.com/mesonbuild/meson/blob/master/docs/markdown/Reference-manual.md#shared_library
559 2018-08-31 17:20:47 +1000 Matthew Waters <matthew@centricular.com>
561 * gst/rtsp-sink/meson.build:
563 meson: add pkg-config file for the rtspclientsink plugin
565 2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
567 * gst/rtsp-server/rtsp-client.c:
568 * tests/check/gst/client.c:
569 rtsp-client: Avoid reuse of channel numbers for interleaved
570 If a (strange) client would reuse interleaved channel numbers in
571 multiple SETUP requests, we should not accept them. The channel
572 numbers are used for looking up stream transports in the
573 priv->transports hash table, and transports disappear from the table
574 if channel numbers are reused.
575 RFC 7826 (RTSP 2.0), Section 18.54, clarifies that it is OK for the
576 server to change the channel numbers suggested by the client.
577 https://bugzilla.gnome.org/show_bug.cgi?id=796988
579 2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
581 * tests/check/gst/client.c:
582 rtsp-client: Add unit test of SETUP for RTSP/RTP/TCP
583 Allow regex for matching transport header against expected pattern.
584 https://bugzilla.gnome.org/show_bug.cgi?id=796988
586 2018-08-15 18:57:27 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
588 * tests/check/meson.build:
589 meson: There is no gstreamer-plugins-good-1.0.pc
590 There is no installed version of that, only an uninstalled version.
592 2018-08-14 14:31:55 +0300 Sebastian Dröge <sebastian@centricular.com>
594 * gst/rtsp-server/rtsp-client.c:
595 * tests/check/gst/stream.c:
596 Fix indentation again
598 2018-07-26 12:01:16 +0200 Patricia Muscalu <patricia@axis.com>
600 * gst/rtsp-server/rtsp-client.c:
601 * gst/rtsp-server/rtsp-stream.c:
602 * gst/rtsp-server/rtsp-stream.h:
603 * tests/check/gst/client.c:
604 * tests/check/gst/stream.c:
605 stream: Added a list of multicast client addresses
606 When media is shared, the same media stream can be sent
607 to multiple multicast groups. Currently, there is no API
608 to retrieve multicast addresses from the stream.
609 When calling gst_rtsp_stream_get_multicast_address() function,
610 only the first multicast address is returned.
611 With this patch, each multicast destination requested in SETUP
612 will be stored in an internal list (call to
613 gst_rtsp_stream_add_multicast_client_address()).
614 The list of multicast groups requested by the clients can be
615 retrieved by calling gst_rtsp_stream_get_multicast_client_addresses().
616 There still exist some problems with the current implementation
617 in the multicast case:
618 1) The receiving part is currently only configured with
619 regard to the first multicast client (see
620 https://bugzilla.gnome.org/show_bug.cgi?id=796917).
621 2) Secondly, of security reasons, some constraints should be
622 put on the requested multicast destinations (see
623 https://bugzilla.gnome.org/show_bug.cgi?id=796916).
624 Change-Id: I6b060746e472a0734cc2fd828ffe4ea2956733ea
625 https://bugzilla.gnome.org/show_bug.cgi?id=793441
627 2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
629 * gst/rtsp-server/rtsp-client.c:
630 * gst/rtsp-server/rtsp-stream.c:
631 * gst/rtsp-server/rtsp-stream.h:
632 * tests/check/gst/client.c:
633 stream: Choose the maximum ttl value provided by multicast clients
634 The maximum ttl value provided so far by the multicast clients
635 will be chosen and reported in the response to the current
637 Change-Id: I5408646e3b5a0a224d907ae215bdea60c4f1905f
638 https://bugzilla.gnome.org/show_bug.cgi?id=793441
640 2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
642 * gst/rtsp-server/rtsp-stream.c:
643 * tests/check/gst/client.c:
644 rtsp-stream: Don't require address pool in the transport specific case
645 If "transport.client-settings" parameter is set to true, the client is
646 allowed to specify destination, ports and ttl.
647 There is no need for pre-configured address pool.
648 Change-Id: I6ae578fb5164d78e8ec1e2ee82dc4eaacd0912d1
649 https://bugzilla.gnome.org/show_bug.cgi?id=793441
651 2018-07-24 14:02:40 +0200 Patricia Muscalu <patricia@axis.com>
653 * gst/rtsp-server/rtsp-client.c:
654 * tests/check/gst/client.c:
655 client: Don't reserve multicast address in the client setting case
656 When two multicast clients request specific transport
657 configurations, and "transport.client-settings" parameter is
658 set to true, it's wrong to actually require that these two
659 clients request the same multicast group.
660 Removed test_client_multicast_invalid_transport_specific test
661 cases as they wrongly require that the requested destination
662 address is supposed to be present in the address pool, also in
663 the case when "transport.client-settings" parameter is set to true.
664 Change-Id: I4580182ef35996caf644686d6139f72ec599c9fa
665 https://bugzilla.gnome.org/show_bug.cgi?id=793441
667 2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
669 * gst/rtsp-server/rtsp-media-factory.c:
670 * gst/rtsp-server/rtsp-media-factory.h:
671 * gst/rtsp-server/rtsp-media.c:
672 * gst/rtsp-server/rtsp-media.h:
673 * gst/rtsp-server/rtsp-stream.c:
674 * gst/rtsp-server/rtsp-stream.h:
675 * tests/check/gst/mediafactory.c:
676 Add new API for setting/getting maximum multicast ttl value
677 Change-Id: I5ef4758188c14785e17fb8fbf42a3dc0cb054233
678 https://bugzilla.gnome.org/show_bug.cgi?id=793441
680 2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
682 * gst/rtsp-server/rtsp-stream.c:
683 rtsp-stream: avoid duplicating the first multicast client
684 In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
685 clients were dynamically added and removed to the multicast
686 udp sinks, as such we should no longer add a first client in
687 set_multicast_socket_for_udpsink
688 https://bugzilla.gnome.org/show_bug.cgi?id=793441
690 2018-08-14 14:25:53 +0300 Sebastian Dröge <sebastian@centricular.com>
692 * gst/rtsp-server/rtsp-stream.c:
693 Revert "rtsp-stream: avoid duplicating the first multicast client"
694 This reverts commit 33570944401747f44d8ebfec535350651413fb92.
695 Commits where accidentially squashed together
697 2018-08-14 14:25:42 +0300 Sebastian Dröge <sebastian@centricular.com>
699 * gst/rtsp-server/rtsp-client.c:
700 * gst/rtsp-server/rtsp-media-factory.c:
701 * gst/rtsp-server/rtsp-media-factory.h:
702 * gst/rtsp-server/rtsp-media.c:
703 * gst/rtsp-server/rtsp-media.h:
704 * gst/rtsp-server/rtsp-stream.c:
705 * gst/rtsp-server/rtsp-stream.h:
706 * tests/check/gst/client.c:
707 * tests/check/gst/mediafactory.c:
708 Revert "Add new API for setting/getting maximum multicast ttl value"
709 This reverts commit 7f0ae77e400fb8a0462a76a5dd2e63e12c4a2e52.
710 Commits where accidentially squashed together
712 2018-08-14 14:25:37 +0300 Sebastian Dröge <sebastian@centricular.com>
714 * gst/rtsp-server/rtsp-stream.c:
715 * tests/check/gst/client.c:
716 Revert "rtsp-stream: Don't require address pool in the transport specific case"
717 This reverts commit a9db3e7f092cfeb5475e9aa24b1e91906c141d52.
718 Commits where accidentially squashed together
720 2018-08-14 14:25:14 +0300 Sebastian Dröge <sebastian@centricular.com>
722 * gst/rtsp-server/rtsp-client.c:
723 * gst/rtsp-server/rtsp-stream.c:
724 * gst/rtsp-server/rtsp-stream.h:
725 * tests/check/gst/client.c:
726 * tests/check/gst/stream.c:
727 Revert "stream: Choose the maximum ttl value provided by multicast clients"
728 This reverts commit 499e437e501215849d24cdaa157e0edf4de097d0.
729 Commits where accidentially squashed together
731 2018-08-14 14:10:56 +0300 Sebastian Dröge <sebastian@centricular.com>
733 * examples/test-auth-digest.c:
734 examples: Fix indentation
736 2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
738 * gst/rtsp-server/rtsp-client.c:
739 * gst/rtsp-server/rtsp-stream.c:
740 * gst/rtsp-server/rtsp-stream.h:
741 * tests/check/gst/client.c:
742 * tests/check/gst/stream.c:
743 stream: Choose the maximum ttl value provided by multicast clients
744 The maximum ttl value provided so far by the multicast clients
745 will be chosen and reported in the response to the current
747 https://bugzilla.gnome.org/show_bug.cgi?id=793441
749 2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
751 * gst/rtsp-server/rtsp-stream.c:
752 * tests/check/gst/client.c:
753 rtsp-stream: Don't require address pool in the transport specific case
754 If "transport.client-settings" parameter is set to true, the client is
755 allowed to specify destination, ports and ttl.
756 There is no need for pre-configured address pool.
757 https://bugzilla.gnome.org/show_bug.cgi?id=793441
759 2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
761 * gst/rtsp-server/rtsp-client.c:
762 * gst/rtsp-server/rtsp-media-factory.c:
763 * gst/rtsp-server/rtsp-media-factory.h:
764 * gst/rtsp-server/rtsp-media.c:
765 * gst/rtsp-server/rtsp-media.h:
766 * gst/rtsp-server/rtsp-stream.c:
767 * gst/rtsp-server/rtsp-stream.h:
768 * tests/check/gst/client.c:
769 * tests/check/gst/mediafactory.c:
770 Add new API for setting/getting maximum multicast ttl value
771 https://bugzilla.gnome.org/show_bug.cgi?id=793441
773 2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
775 * gst/rtsp-server/rtsp-stream.c:
776 rtsp-stream: avoid duplicating the first multicast client
777 In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
778 clients were dynamically added and removed to the multicast
779 udp sinks, as such we should no longer add a first client in
780 set_multicast_socket_for_udpsink
781 https://bugzilla.gnome.org/show_bug.cgi?id=793441
783 2018-08-06 15:33:04 -0400 Thibault Saunier <tsaunier@igalia.com>
785 * gst/rtsp-server/Makefile.am:
786 rtsp-server: Add gstreamer-base gir dir in autotools
788 2018-07-25 19:54:55 +0200 Mathieu Duponchelle <mathieu@centricular.com>
790 * gst/rtsp-server/rtsp-client.c:
791 * gst/rtsp-server/rtsp-stream.c:
792 rtsp-client: always allocate both IPV4 and IPV6 sockets
793 multiudpsink does not support setting the socket* properties
794 after it has started, which meant that rtsp-server could no
795 longer serve on both IPV4 and IPV6 sockets since the patches
796 from https://bugzilla.gnome.org/show_bug.cgi?id=757488 were
798 When first connecting an IPV6 client then an IPV4 client,
799 multiudpsink fell back to using the IPV6 socket.
800 When first connecting an IPV4 client, then an IPV6 client,
801 multiudpsink errored out, released the IPV4 socket, then
802 crashed when trying to send a message on NULL nevertheless,
803 that is however a separate issue.
804 This could probably be fixed by handling the setting of
805 sockets in multiudpsink after it has started, that will
806 however be a much more significant effort.
807 For now, this commit simply partially reverts the behaviour
808 of rtsp-stream: it will continue to only create the udpsinks
809 when needed, as was the case since the patches were merged,
810 it will however when creating them, always allocate both
811 sockets and set them on the sink before it starts, as was
812 the case prior to the patches.
813 Transport configuration will only error out if the allocation
814 of UDP sockets fails for the actual client's family, this
815 also downgrades the GST_ERRORs in alloc_ports_one_family
816 to GST_WARNINGs, as failing to allocate is no longer
818 https://bugzilla.gnome.org/show_bug.cgi?id=796875
820 2018-07-25 17:22:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
824 meson: Convert common options to feature options
825 These are necessary for gst-build to set options correctly. The
826 remaining automagic option is cgroup support in examples.
827 https://bugzilla.gnome.org/show_bug.cgi?id=795107
829 2018-07-23 18:03:51 +0300 Sebastian Dröge <sebastian@centricular.com>
831 * gst/rtsp-server/rtsp-stream.c:
832 rtsp-stream: Slightly simplify locking
834 2018-06-28 11:22:21 +0200 David Svensson Fors <davidsf@axis.com>
836 * gst/rtsp-server/rtsp-client.c:
837 * gst/rtsp-server/rtsp-stream-transport.c:
838 * gst/rtsp-server/rtsp-stream-transport.h:
839 * gst/rtsp-server/rtsp-stream.c:
840 Limit queued TCP data messages to one per stream
841 Before, the watch backlog size in GstRTSPClient was changed
842 dynamically between unlimited and a fixed size, trying to avoid both
843 unlimited memory usage and deadlocks while waiting for place in the
844 queue. (Some of the deadlocks were described in a long comment in
846 In the previous commit, we changed to a fixed backlog size of 100.
847 This is possible, because we now handle RTP/RTCP data messages differently
848 from RTSP request/response messages.
849 The data messages are messages tunneled over TCP. We allow at most one
850 queued data message per stream in GstRTSPClient at a time, and
851 successfully sent data messages are acked by sending a "message-sent"
852 callback from the GstStreamTransport. Until that ack comes, the
853 GstRTSPStream does not call pull_sample() on its appsink, and
854 therefore the streaming thread in the pipeline will not be blocked
855 inside GstRTSPClient, waiting for a place in the queue.
856 pull_sample() is called when we have both an ack and a "new-sample"
857 signal from the appsink. Then, we know there is a buffer to write.
858 RTSP request/response messages are not acked in the same way as data
859 messages. The rest of the 100 places in the queue are used for
860 them. If the queue becomes full of request/response messages, we
861 return an error and close the connection to the client.
862 Change-Id: I275310bc90a219ceb2473c098261acc78be84c97
864 2018-06-28 11:22:13 +0200 David Svensson Fors <davidsf@axis.com>
866 * gst/rtsp-server/rtsp-client.c:
867 rtsp-client: Use fixed backlog size
868 Change to using a fixed backlog size WATCH_BACKLOG_SIZE.
869 Preparation for the next commit, which changes to a different way of
870 avoiding both deadlocks and unlimited memory usage with the watch
873 2018-07-16 21:57:08 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
875 * gst/rtsp-server/rtsp-media.c:
876 rtsp-media: unref clock (if set) when finalizing
877 https://bugzilla.gnome.org/show_bug.cgi?id=796814
879 2018-07-16 21:56:44 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
881 * docs/libs/gst-rtsp-server-sections.txt:
882 rtsp-media: add gst_rtsp_media_*_set_clock to docs
883 https://bugzilla.gnome.org/show_bug.cgi?id=796814
885 2018-07-12 19:01:54 +0100 Tim-Philipp Müller <tim@centricular.com>
887 * gst/rtsp-server/rtsp-media-factory.c:
888 media-factory: unref old clock when setting new clock
889 https://bugzilla.gnome.org/show_bug.cgi?id=796724
891 2018-06-29 15:20:57 -0700 Brendan Shanks <brendan.shanks@teradek.com>
893 * gst/rtsp-server/rtsp-media-factory.c:
894 media-factory: unref clock in finalize
895 https://bugzilla.gnome.org/show_bug.cgi?id=796724
897 2018-07-12 18:57:21 +0100 Tim-Philipp Müller <tim@centricular.com>
899 * gst/rtsp-server/rtsp-onvif-media.c:
900 rtsp-onvif-media: fix g-ir-scanner warnings
902 2018-07-10 23:56:23 +0100 Tim-Philipp Müller <tim@centricular.com>
905 .gitignore: add another example binary
907 2018-07-10 23:55:20 +0100 Tim-Philipp Müller <tim@centricular.com>
909 * examples/meson.build:
910 meson: add new test-appsrc2 example to meson build
912 2018-07-10 23:53:41 +0100 Tim-Philipp Müller <tim@centricular.com>
914 * examples/Makefile.am:
915 examples: fix build of new test-appsrc2 example
916 Need to link against libgstapp-1.0.
918 2018-07-11 01:25:51 +1000 Jan Schmidt <jan@centricular.com>
920 * examples/.gitignore:
921 * examples/Makefile.am:
922 * examples/test-appsrc2.c:
923 examples: Add test-appsrc2
924 Add an example of feeding both audio and video into an RTSP
927 2016-01-08 18:12:14 -0500 Louis-Francis Ratté-Boulianne <lfrb@collabora.com>
929 * gst/rtsp-server/rtsp-client.c:
930 client: Strip transport parts as whitespaces could be around commas
931 https://bugzilla.gnome.org/show_bug.cgi?id=758428
933 2018-06-27 08:30:42 +0200 Göran Jönsson <goranjn@axis.com>
935 * gst/rtsp-server/rtsp-stream.c:
936 rtsp-stream: avoid pushing data on unlinked udpsrc pad during setup
937 Fix race when setting up source elements.
938 Since we set the source element(s) to PLAYING state before hooking
939 them up to the downstream funnel, it's possible for the source element
940 to receive packets before we actually get to linking it to the funnel,
941 in which case buffers would be pushed out on an unlinked pad, causing
942 it to error out and stop receiving more data.
943 We fix this by blocking the source's srcpad until we have linked it.
944 https://bugzilla.gnome.org/show_bug.cgi?id=796160
946 2018-03-21 10:56:51 +0100 Ognyan Tonchev <ognyan@axis.com>
948 * gst/rtsp-server/rtsp-stream.c:
949 rtsp-stream: Fix mismatch between allowed and configured protocols
950 https://bugzilla.gnome.org/show_bug.cgi?id=796679
952 2017-02-01 09:44:50 +0100 Ulf Olsson <ulfo@axis.com>
954 * gst/rtsp-server/rtsp-stream.c:
955 rtsp-stream: Emit a signal when the SRTP decoder is created
956 https://bugzilla.gnome.org/show_bug.cgi?id=778080
958 2018-03-13 11:10:35 +0100 Patricia Muscalu <patricia@axis.com>
960 * gst/rtsp-server/rtsp-stream.c:
961 rtsp-stream: Don't require presence of sinks in _get_*_socket()
962 Transport specific sink elements are added to the pipeline
963 in PLAY request and sockets are already created in SETUP so
964 it's actually wrong to require the presence of sinks in
965 _get_*_socket() functions.
966 https://bugzilla.gnome.org/show_bug.cgi?id=793441
968 2018-02-14 10:41:02 +0100 Patricia Muscalu <patricia@axis.com>
970 * gst/rtsp-server/rtsp-stream.c:
971 rtsp-stream: Update transport for multicast clients as well
972 If a multicast client requests different transport settings
973 than the existing one make sure that this new transport
974 configuruation is propagated to the multicast udp sink.
975 https://bugzilla.gnome.org/show_bug.cgi?id=793441
977 2018-02-13 11:04:36 +0100 Patricia Muscalu <patricia@axis.com>
979 * gst/rtsp-server/rtsp-stream.c:
980 rtsp-stream: Set the multicast TTL parameter on multicast udp sinks
981 And not on unicast udp sinks
982 https://bugzilla.gnome.org/show_bug.cgi?id=793441
984 2018-06-24 12:44:26 +0200 Tim-Philipp Müller <tim@centricular.com>
986 * gst/rtsp-server/rtsp-address-pool.c:
987 * gst/rtsp-server/rtsp-auth.c:
988 * gst/rtsp-server/rtsp-client.c:
989 * gst/rtsp-server/rtsp-media-factory-uri.c:
990 * gst/rtsp-server/rtsp-media-factory.c:
991 * gst/rtsp-server/rtsp-media.c:
992 * gst/rtsp-server/rtsp-mount-points.c:
993 * gst/rtsp-server/rtsp-server.c:
994 * gst/rtsp-server/rtsp-session-media.c:
995 * gst/rtsp-server/rtsp-session-pool.c:
996 * gst/rtsp-server/rtsp-session.c:
997 * gst/rtsp-server/rtsp-stream-transport.c:
998 * gst/rtsp-server/rtsp-stream.c:
999 * gst/rtsp-server/rtsp-thread-pool.c:
1000 Update for g_type_class_add_private() deprecation in recent GLib
1002 2018-06-24 12:45:49 +0200 Tim-Philipp Müller <tim@centricular.com>
1004 * gst/rtsp-server/rtsp-auth.c:
1005 * gst/rtsp-server/rtsp-media.c:
1006 * gst/rtsp-server/rtsp-sdp.c:
1007 * gst/rtsp-server/rtsp-stream.c:
1010 2018-06-22 23:17:08 +1000 Jan Schmidt <jan@centricular.com>
1012 * examples/Makefile.am:
1013 * examples/test-video-disconnect.c:
1014 examples: Add test-video-disconnect example
1015 Simple example which cuts off all clients 10 seconds
1016 after the first one connects.
1018 2018-06-20 04:37:11 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1020 * docs/libs/gst-rtsp-server-sections.txt:
1021 * examples/test-auth-digest.c:
1022 * gst/rtsp-server/rtsp-auth.c:
1023 * gst/rtsp-server/rtsp-auth.h:
1024 rtsp-auth: Add support for parsing .htdigest files
1025 Passwords are usually not stored in clear text, but instead
1026 stored already hashed in a .htdigest file.
1027 Add support for parsing such files, add API to allow setting
1028 a custom realm in RTSPAuth, and update the digest example.
1029 https://bugzilla.gnome.org/show_bug.cgi?id=796637
1031 2018-06-19 14:53:02 +1000 Matthew Waters <matthew@centricular.com>
1033 * gst/rtsp-sink/gstrtspclientsink.c:
1034 * gst/rtsp-sink/gstrtspclientsink.h:
1035 rtspclientsink: fix waiting for multiple streams
1036 We were previously only ever waiting for a single stream to notify it's
1037 blocked status through GstRTSPStreamBlocking. Actually count streams to
1039 Fixes rtspclientsink sending SDP's without out some of the input
1041 https://bugzilla.gnome.org/show_bug.cgi?id=796624
1043 2018-06-20 04:30:04 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1045 * docs/libs/gst-rtsp-server-sections.txt:
1046 docs: add missing auth methods
1048 2018-06-20 00:10:18 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1050 * gst/rtsp-server/rtsp-stream.c:
1051 rtsp-stream: only create funnel if it didn't exist already.
1052 This precented using multiple protocols for the same stream.
1053 https://bugzilla.gnome.org/show_bug.cgi?id=796634
1055 2018-06-20 01:35:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1057 * examples/meson.build:
1058 meson: build auth-digest example
1060 2018-06-05 08:44:44 +0200 Patricia Muscalu <patricia@axis.com>
1062 * gst/rtsp-server/rtsp-client.c:
1063 * gst/rtsp-server/rtsp-media.c:
1064 * gst/rtsp-server/rtsp-sdp.c:
1065 * gst/rtsp-server/rtsp-session-media.c:
1066 * gst/rtsp-server/rtsp-stream-transport.c:
1067 Get payloader stats only for the sending streams
1068 Get/set payloader properties only for streams that actually
1069 contain a payloader element.
1070 https://bugzilla.gnome.org/show_bug.cgi?id=796523
1072 2018-05-18 14:53:49 +0200 Edward Hervey <edward@centricular.com>
1074 * gst/rtsp-server/Makefile.am:
1075 Makefile: Don't hardcode libtool for g-i build
1076 Similar to the other commits in core/base/bad
1078 2018-05-08 14:13:31 +0200 Johan Bjäreholt <johanbj@axis.com>
1080 * gst/rtsp-server/rtsp-onvif-media-factory.h:
1081 rtsp-onvif-media-factory: export gst_rtsp_onvif_media_factory_requires_backchannel
1082 https://bugzilla.gnome.org/show_bug.cgi?id=796229
1084 2018-05-09 04:09:02 +1000 Jan Schmidt <jan@centricular.com>
1086 * gst/rtsp-sink/gstrtspclientsink.c:
1087 rtspclientsink: Don't deadlock in preroll on early close
1088 If the connection is closed very early, the flushing
1089 marker might not get set and rtspclientsink can get
1090 deadlocked waiting for preroll forever.
1091 https://bugzilla.gnome.org/show_bug.cgi?id=786961
1093 2018-05-05 19:51:52 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
1096 * meson_options.txt:
1097 meson: Update option names to omit disable_ and with- prefixes
1098 Also yield common options to the outer project (gst-build in our case)
1099 so that they don't have to be set manually.
1101 2018-04-25 11:00:32 +0100 Tim-Philipp Müller <tim@centricular.com>
1104 meson: use -Wl,-Bsymbolic-functions where supported
1105 Just like the autotools build.
1107 2018-04-22 20:09:01 +0100 Tim-Philipp Müller <tim@centricular.com>
1110 * tests/check/Makefile.am:
1111 configure: check for -good and -bad plugins only in uninstalled setup
1112 Avoids confusing configure messages looking or a -good .pc file
1114 Also use plugindir variables that common macros set while at it.
1115 https://bugzilla.gnome.org/show_bug.cgi?id=795466
1117 2018-04-17 11:03:11 +0200 Joakim Johansson <joakimj@axis.com>
1119 * gst/rtsp-server/rtsp-client.c:
1120 rtsp-client: Fix session timeout
1121 When streaming data over TCP then is not the keep-alive
1122 functionality working.
1123 The reason is that the function do_send_data have changed
1124 to boolean but the code is still checking the received result
1125 from send_func with GST_RTSP_OK.
1126 The result is that a successful send_func will always lead to
1127 that do_send_data is returning false and the keep-alive will
1129 https://bugzilla.gnome.org/show_bug.cgi?id=795321
1131 2018-04-02 22:49:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1133 * docs/libs/gst-rtsp-server-sections.txt:
1134 * gst/rtsp-server/rtsp-media.c:
1135 * gst/rtsp-server/rtsp-sdp.c:
1136 * gst/rtsp-server/rtsp-stream.c:
1137 * gst/rtsp-server/rtsp-stream.h:
1138 * gst/rtsp-sink/gstrtspclientsink.c:
1139 * gst/rtsp-sink/gstrtspclientsink.h:
1140 Implement support for ULP Forward Error Correction
1141 In this initial commit, interface is only exposed for RECORD,
1142 further work will be needed in rtspsrc to support this for
1144 https://bugzilla.gnome.org/show_bug.cgi?id=794911
1146 2018-04-17 17:47:30 +0300 Sebastian Dröge <sebastian@centricular.com>
1148 * gst/rtsp-server/rtsp-onvif-media.c:
1149 Revert "rtsp-server: Switch around sendonly/recvonly attributes"
1150 This reverts commit 3d275b1345b76151418e3f56ed014d9089ac1a57.
1151 While RFC 3264 (SDP) says that sendonly/recvonly are from the point of view of
1152 the requester, the actual RTSP RFCs (RFC 2326 / 7826) disagree and say
1153 the opposite, just like the ONVIF standard.
1154 Let's follow those RFCs as we're doing RTSP here, and add a property at
1155 a later time if needed to switch to the SDP RFC behaviour.
1156 https://bugzilla.gnome.org/show_bug.cgi?id=793964
1158 2018-04-16 10:53:52 +0100 Tim-Philipp Müller <tim@centricular.com>
1161 Automatic update of common submodule
1162 From 3fa2c9e to ed78bee
1164 2018-04-04 10:06:06 +0300 Sebastian Dröge <sebastian@centricular.com>
1166 * gst/rtsp-server/rtsp-client.c:
1167 * gst/rtsp-server/rtsp-media-factory.c:
1168 * gst/rtsp-server/rtsp-media.c:
1169 * gst/rtsp-server/rtsp-stream.c:
1170 * tests/check/gst/rtspclientsink.c:
1171 gst: Run everything through gst-indent again
1173 2018-04-03 08:57:47 +0200 Branko Subasic <branko@axis.com>
1175 * gst/rtsp-server/rtsp-media.c:
1176 * tests/check/gst/media.c:
1177 rtsp-media: query the position on active streams if media is complete
1178 If the media is complete, i.e. one or more streams have been configured
1179 with sinks, then we want to query the position on those streams only.
1180 A query on an incomplete stream may return a position that originates from
1182 https://bugzilla.gnome.org/show_bug.cgi?id=794964
1184 2018-04-02 12:35:04 +0100 Tim-Philipp Müller <tim@centricular.com>
1186 * gst/rtsp-sink/gstrtspclientsink.c:
1187 rtspclientsink: make sure not to use freed string
1188 Set transport string to NULL after freeing it, so that
1189 at worst we get a NULL pointer if constructing a new
1190 transport string fails (which shouldn't really fail here).
1191 Also check return value of that, just in case.
1194 2018-03-30 23:34:01 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1196 * gst/rtsp-server/rtsp-client.c:
1197 rtsp-client: do not free string passed to take_header
1199 2018-03-30 23:10:10 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1201 * gst/rtsp-server/rtsp-stream.c:
1202 rtsp-stream: do not take lock in request_aux_receiver
1203 Added it right before pushing the previous commit, it is
1204 incorrect and deadlocks because this function gets called
1205 from the join_bin thread, which already holds the lock,
1206 that's the reason why request_aux_sender didn't take the
1209 2018-03-29 22:49:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1211 * docs/libs/gst-rtsp-server-sections.txt:
1212 * gst/rtsp-server/rtsp-media-factory.c:
1213 * gst/rtsp-server/rtsp-media-factory.h:
1214 * gst/rtsp-server/rtsp-media.c:
1215 * gst/rtsp-server/rtsp-media.h:
1216 * gst/rtsp-server/rtsp-stream.c:
1217 * gst/rtsp-server/rtsp-stream.h:
1218 rtsp-server: add API to enable retransmission requests
1219 "do-retransmission" was previously set when rtx-time != 0,
1220 which made no sense as do-retransmission is used to enable
1221 the sending of retransmission requests, where as rtx-time
1222 is used by the peer to enable storing of buffers in order
1223 to respond to retransmission requests.
1224 rtsp-media now also provides a callback for the
1225 request-aux-receiver signal.
1226 https://bugzilla.gnome.org/show_bug.cgi?id=794822
1228 2018-03-29 16:18:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1230 * gst/rtsp-sink/gstrtspclientsink.c:
1231 rtspclientsink: add rtx ssrc to mikey's crypto sessions
1232 https://bugzilla.gnome.org/show_bug.cgi?id=794813
1234 2018-03-29 16:15:45 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1236 * gst/rtsp-sink/gstrtspclientsink.c:
1237 rtspclientsink: Handle the KeyMgmt header in ANNOUNCE response
1238 This in order to be able to decrypt the RTCP backchannel
1239 https://bugzilla.gnome.org/show_bug.cgi?id=794813
1241 2018-03-29 16:12:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1243 * gst/rtsp-server/rtsp-client.c:
1244 rtsp-client: Send KeyMgmt header in ANNOUNCE response
1245 When sending back an encrypted RTCP back channel, it is useful
1246 for the client to know the encryption key.
1247 https://bugzilla.gnome.org/show_bug.cgi?id=794813
1249 2018-03-29 16:06:31 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1251 * gst/rtsp-server/rtsp-client.c:
1252 * gst/rtsp-server/rtsp-stream.c:
1253 * gst/rtsp-server/rtsp-stream.h:
1254 rtsp-stream: extract handle_keymgmt from rtsp-client
1255 rtspclientsink will also need to parse KeyMgmt headers
1256 sent by the server to decrypt the RTCP backchannel stream
1257 https://bugzilla.gnome.org/show_bug.cgi?id=794813
1259 2018-03-29 02:51:02 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1261 * gst/rtsp-sink/gstrtspclientsink.c:
1262 * tests/check/gst/rtspclientsink.c:
1263 rtspclientsink: Fix client ports for the RTCP backchannel
1264 This was broken since the work for delayed transport creation
1265 was merged: the creation of the transports string depends on
1266 calling stream_get_server_port, which only starts returning
1267 something meaningful after a call to stream_allocate_udp_sockets
1268 has been made, this function expects a transport that we parse
1269 from the transport string ...
1270 Significant refactoring is in order, but does not look entirely
1271 trivial, for now we put a band aid on and create a second transport
1272 string after the stream has been completed, to pass it in
1273 the request headers instead of the previous, incomplete one.
1274 https://bugzilla.gnome.org/show_bug.cgi?id=794789
1276 2018-02-15 13:26:16 +0100 Göran Jönsson <goranjn@axis.com>
1278 * gst/rtsp-server/rtsp-client.c:
1279 rtsp-client:Error handling when equal http session cookie
1280 There are some clients that are sending same session cookie on random
1282 https://bugzilla.gnome.org/show_bug.cgi?id=753616
1284 2018-03-20 16:21:37 +0200 Sebastian Dröge <sebastian@centricular.com>
1286 * gst/rtsp-server/rtsp-media-factory-uri.c:
1287 rtsp-media-factory-uri: Fix compilation with latest GLib
1288 rtsp-media-factory-uri.c: In function ‘rtsp_media_factory_uri_create_element’:
1289 rtsp-media-factory-uri.c:621:17: error: assignment from incompatible pointer type [-Werror=incompatible-pointer-types]
1290 data->factory = g_object_ref (factory);
1293 2018-03-20 10:21:36 +0000 Tim-Philipp Müller <tim@centricular.com>
1301 === release 1.14.0 ===
1303 2018-03-19 20:27:04 +0000 Tim-Philipp Müller <tim@centricular.com>
1309 * gst-rtsp-server.doap:
1313 === release 1.13.91 ===
1315 2018-03-13 19:28:33 +0000 Tim-Philipp Müller <tim@centricular.com>
1321 * gst-rtsp-server.doap:
1325 2018-03-13 13:30:41 +0000 Tim-Philipp Müller <tim@centricular.com>
1327 * gst/rtsp-server/Makefile.am:
1328 * gst/rtsp-server/meson.build:
1329 * gst/rtsp-server/rtsp-address-pool.h:
1330 * gst/rtsp-server/rtsp-auth.h:
1331 * gst/rtsp-server/rtsp-client.h:
1332 * gst/rtsp-server/rtsp-context.h:
1333 * gst/rtsp-server/rtsp-media-factory-uri.h:
1334 * gst/rtsp-server/rtsp-media-factory.h:
1335 * gst/rtsp-server/rtsp-media.h:
1336 * gst/rtsp-server/rtsp-mount-points.h:
1337 * gst/rtsp-server/rtsp-onvif-client.h:
1338 * gst/rtsp-server/rtsp-onvif-media-factory.h:
1339 * gst/rtsp-server/rtsp-onvif-media.h:
1340 * gst/rtsp-server/rtsp-onvif-server.h:
1341 * gst/rtsp-server/rtsp-params.h:
1342 * gst/rtsp-server/rtsp-permissions.h:
1343 * gst/rtsp-server/rtsp-sdp.h:
1344 * gst/rtsp-server/rtsp-server-prelude.h:
1345 * gst/rtsp-server/rtsp-server.h:
1346 * gst/rtsp-server/rtsp-session-media.h:
1347 * gst/rtsp-server/rtsp-session-pool.h:
1348 * gst/rtsp-server/rtsp-session.h:
1349 * gst/rtsp-server/rtsp-stream-transport.h:
1350 * gst/rtsp-server/rtsp-stream.h:
1351 * gst/rtsp-server/rtsp-thread-pool.h:
1352 * gst/rtsp-server/rtsp-token.h:
1353 rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API
1354 We need different export decorators for the different libs.
1355 For now no actual change though, just rename before the release,
1356 and add prelude headers to define the new decorator to GST_EXPORT.
1358 2018-03-07 12:20:05 +0200 Sebastian Dröge <sebastian@centricular.com>
1360 * gst/rtsp-server/rtsp-onvif-media-factory.c:
1361 rtsp-onvif-media-factory: Document that backchannel pipelines must end with async=false sinks
1362 https://bugzilla.gnome.org/show_bug.cgi?id=794143
1364 === release 1.13.90 ===
1366 2018-03-03 22:49:34 +0000 Tim-Philipp Müller <tim@centricular.com>
1372 * gst-rtsp-server.doap:
1376 2018-03-02 16:24:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1378 * gst/rtsp-server/rtsp-media-factory.c:
1379 * gst/rtsp-server/rtsp-permissions.c:
1380 permissions: add Since tags and example for new API
1382 2018-03-02 01:36:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1384 * docs/libs/gst-rtsp-server-sections.txt:
1385 * gst/rtsp-server/rtsp-media-factory.c:
1386 * gst/rtsp-server/rtsp-media-factory.h:
1387 * gst/rtsp-server/rtsp-permissions.c:
1388 * gst/rtsp-server/rtsp-permissions.h:
1389 * tests/check/gst/permissions.c:
1390 permissions: more bindings-friendly API
1391 https://bugzilla.gnome.org/show_bug.cgi?id=793975
1393 2018-03-01 19:28:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1396 meson: enable more warnings
1398 2018-02-28 21:12:43 +0200 Sebastian Dröge <sebastian@centricular.com>
1400 * gst/rtsp-server/rtsp-client.c:
1401 rtsp-client: Place netaddress meta on packets received via TCP
1402 This allows us to later map signals from rtpbin/rtpsource back to the
1403 corresponding stream transport, and allows to do keep-alive based on
1404 RTCP packets in case of TCP media transport.
1405 https://bugzilla.gnome.org/show_bug.cgi?id=789646
1407 2018-02-27 20:34:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1409 * gst/rtsp-sink/gstrtspclientsink.c:
1410 rtspclientsink: if OPEN failed, unqueue next command
1411 As READY_TO_PAUSED can no longer return async, the RECORD
1412 command will be queued before the OPEN command fails
1413 (for example in case the server could not be connected),
1414 and record then waits for ever.
1415 https://bugzilla.gnome.org/show_bug.cgi?id=793896
1417 2018-02-26 22:59:17 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1419 * gst/rtsp-sink/gstrtspclientsink.c:
1420 rtspclientsink: fix retrieval of custom payloader caps
1421 If a bin is passed as the custom payloader, the caps of
1422 its factory will be empty, the correct way to obtain the caps
1423 is to query its sinkpad.
1425 2018-02-26 22:59:00 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1427 * gst/rtsp-sink/gstrtspclientsink.c:
1428 rtspclientsink: fix extra unref of custom payloader
1430 2018-02-26 22:57:39 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1432 * gst/rtsp-sink/gstrtspclientsink.c:
1433 rspclientsink: fix recent code indentation
1435 2018-02-26 20:27:57 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1437 * gst/rtsp-sink/gstrtspclientsink.c:
1438 rtspclientsink: add missing get_type prototype
1440 2018-02-24 03:52:15 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1442 * gst/rtsp-sink/gstrtspclientsink.c:
1443 rtspclientsink: allow setting payloader as pad property
1444 This was a FIXME item, and can be quite useful, also
1445 allowing to specify payloader properties from the command
1446 line, which is always nice.
1447 https://bugzilla.gnome.org/show_bug.cgi?id=793776
1449 2018-02-26 14:16:54 +0100 Carlos Rafael Giani <dv@pseudoterminal.org>
1451 * gst/rtsp-server/rtsp-media.c:
1452 rtsp-media: Replace g_print() log line
1453 https://bugzilla.gnome.org/show_bug.cgi?id=793838
1455 2018-02-22 20:17:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1457 * gst/rtsp-server/rtsp-media.c:
1458 * tests/check/gst/rtspclientsink.c:
1459 rtsp-media: fix RECORD getting stuck
1460 The test_record case was working because async=false had
1461 been added in https://bugzilla.gnome.org/show_bug.cgi?id=757488
1462 but that was incorrect, as it should not be needed.
1463 Removing async=false made the test fail as expected, this is
1464 fixed by not trying to preroll when preparing the media for
1465 RECORD, as start_prepare is called upon receiving ANNOUNCE,
1466 and our peer will not start sending media until it has received
1467 a response to that request, and sent and received a response
1468 to RECORD as well, thus obviously preventing preroll.
1469 https://bugzilla.gnome.org/show_bug.cgi?id=793738
1471 2018-02-23 03:26:21 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1473 * gst/rtsp-server/rtsp-auth.c:
1474 rtsp-auth: fix set_tls_authentication_mode annotation
1476 2018-02-19 11:57:29 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
1478 * gst/rtsp-server/rtsp-onvif-media.c:
1479 rtp-server: remove redefined variable
1480 res is a boolean variable which is defined in the function scope and
1481 redefined, with no reason, in the loop scope. This patch removes the
1483 https://bugzilla.gnome.org/show_bug.cgi?id=793592
1485 2018-02-05 11:49:07 +0100 Ognyan Tonchev <ognyan@axis.com>
1487 * gst/rtsp-server/rtsp-media.c:
1488 * gst/rtsp-server/rtsp-stream.c:
1489 * gst/rtsp-server/rtsp-stream.h:
1490 stream: Add functions for checking if stream is receiver or sender
1491 ...and replace all checks for RECORD in GstRTSPMedia which are really
1492 for "sender-only". This way the code becomes more generic and introducing
1493 support for onvif-backchannel later on will require no changes in
1496 2017-10-21 14:06:30 +0200 Ognyan Tonchev <ognyan@axis.com>
1498 * gst/rtsp-server/rtsp-onvif-media-factory.c:
1499 * gst/rtsp-server/rtsp-onvif-media-factory.h:
1500 onvif: Make requires_backchannel() public
1501 ...in order to let subclasses building the onvif part of the pipeline
1502 check whether backchannel shall be included or not.
1504 2018-01-22 12:46:34 +0200 Sebastian Dröge <sebastian@centricular.com>
1506 * gst/rtsp-server/rtsp-onvif-media.c:
1507 rtsp-server: Switch around sendonly/recvonly attributes
1508 They are wrong in the ONVIF streaming spec. The backchannel should be
1509 recvonly and the normal media should be sendonly: direction is always
1510 from the point of view of the SDP offerer (the server) according to
1513 2017-09-25 19:41:05 +0300 Sebastian Dröge <sebastian@centricular.com>
1515 * docs/libs/gst-rtsp-server-docs.sgml:
1516 * docs/libs/gst-rtsp-server-sections.txt:
1517 * examples/.gitignore:
1518 * examples/Makefile.am:
1519 * examples/test-onvif-backchannel.c:
1520 * gst/rtsp-server/Makefile.am:
1521 * gst/rtsp-server/rtsp-media.h:
1522 * gst/rtsp-server/rtsp-onvif-client.c:
1523 * gst/rtsp-server/rtsp-onvif-client.h:
1524 * gst/rtsp-server/rtsp-onvif-media-factory.c:
1525 * gst/rtsp-server/rtsp-onvif-media-factory.h:
1526 * gst/rtsp-server/rtsp-onvif-media.c:
1527 * gst/rtsp-server/rtsp-onvif-media.h:
1528 * gst/rtsp-server/rtsp-onvif-server.c:
1529 * gst/rtsp-server/rtsp-onvif-server.h:
1530 * gst/rtsp-server/rtsp-sdp.c:
1531 * gst/rtsp-server/rtsp-sdp.h:
1532 rtsp: Add support for ONVIF backchannel
1533 This adds a new RTSP server, client, media-factory and media subclass
1534 for handling the specifics of the backchannel. Ideally this later can be
1535 extended with other ONVIF specific features.
1537 2017-10-12 21:00:16 +0300 Sebastian Dröge <sebastian@centricular.com>
1539 * gst/rtsp-server/rtsp-media.c:
1540 rtsp-media: Add support for sending+receiving medias
1541 We need to add an appsrc/appsink in that case because otherwise the
1542 media bin will be a sink and a source for rtpbin, causing a pipeline
1544 https://bugzilla.gnome.org/show_bug.cgi?id=788950
1546 2018-02-15 19:44:28 +0000 Tim-Philipp Müller <tim@centricular.com>
1552 === release 1.13.1 ===
1554 2018-02-15 17:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
1558 * gst-rtsp-server.doap:
1562 2018-02-14 17:11:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1564 * gst/rtsp-server/rtsp-session-pool.c:
1565 session-pool: remove nullable return annotation
1566 create_watch can only return NULL from the API guards, no
1569 2018-02-13 18:59:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1571 * gst/rtsp-server/rtsp-media-factory.c:
1572 * gst/rtsp-server/rtsp-media.c:
1573 set_clock functions: Add nullable annotations
1575 2018-02-10 00:07:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1577 * gst/rtsp-server/rtsp-auth.c:
1578 * gst/rtsp-server/rtsp-client.c:
1579 * gst/rtsp-server/rtsp-media-factory.c:
1580 * gst/rtsp-server/rtsp-media.c:
1581 * gst/rtsp-server/rtsp-mount-points.c:
1582 * gst/rtsp-server/rtsp-server.c:
1583 * gst/rtsp-server/rtsp-session-media.c:
1584 * gst/rtsp-server/rtsp-session-pool.c:
1585 * gst/rtsp-server/rtsp-session.c:
1586 * gst/rtsp-server/rtsp-stream-transport.c:
1587 * gst/rtsp-server/rtsp-stream.c:
1588 * gst/rtsp-server/rtsp-thread-pool.c:
1589 All around: add annotations and API guards
1591 2018-02-12 19:12:35 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1593 * tests/test-cleanup.c:
1594 test-cleanup: bind any port
1595 The meson test suite runs tests in parallel, trying to bind
1596 a single port made the test fail.
1598 2018-02-08 19:15:10 +0000 Tim-Philipp Müller <tim@centricular.com>
1601 meson: make version numbers ints and fix int/string comparison
1602 WARNING: Trying to compare values of different types (str, int).
1603 The result of this is undefined and will become a hard error
1604 in a future Meson release.
1606 2018-02-06 18:00:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1608 * gst/rtsp-server/rtsp-context.c:
1609 gst_rtsp_context_get_current: add (skip) annotation
1610 The return value type is defined with G_DEFINE_POINTER_TYPE,
1611 and gi emits the following warning:
1612 Invalid non-constant return of bare structure or union; register as
1613 boxed type or (skip)
1615 2018-02-06 17:58:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1617 * gst/rtsp-server/rtsp-client.c:
1618 rtsp-client: add type annotations
1619 gi doesn't seem to be able to figure out the type of the
1620 signal parameters when defined with G_DEFINE_POINTER_TYPE
1622 2018-02-04 12:24:09 +0100 Tim-Philipp Müller <tim@centricular.com>
1625 autotools: use -fno-strict-aliasing where supported
1626 https://bugzilla.gnome.org/show_bug.cgi?id=769183
1628 2018-01-30 20:35:21 +0000 Tim-Philipp Müller <tim@centricular.com>
1631 meson: use -fno-strict-aliasing where supported
1632 https://bugzilla.gnome.org/show_bug.cgi?id=769183
1634 2018-01-25 12:09:03 +0000 Tim-Philipp Müller <tim@centricular.com>
1636 * gst/rtsp-server/rtsp-mount-points.c:
1637 mount-points: bail out of loop again when matching mount points
1638 Previous patch led to us iterating the entire sequence. Bail out
1639 of the loop again if we have a match but are moving away from it.
1640 https://bugzilla.gnome.org/show_bug.cgi?id=771555
1642 2018-01-25 12:06:57 +0000 Tim-Philipp Müller <tim@centricular.com>
1644 * tests/check/gst/mountpoints.c:
1645 tests: mountpoints: add more checks for mount point path matching
1646 https://bugzilla.gnome.org/show_bug.cgi?id=771555
1648 2016-09-16 20:41:19 +0000 Andrew Bott <andrew.bott@blackmoth.com>
1650 * gst/rtsp-server/rtsp-mount-points.c:
1651 mount-points: fix matching of paths where there's also an entry with a common prefix
1652 e.g. with the following mount points
1656 _match() would not match /raw/video and /raw/snapshot correctly.
1657 https://bugzilla.gnome.org/show_bug.cgi?id=771555
1659 2018-01-18 23:53:20 +0000 Tim-Philipp Müller <tim@centricular.com>
1661 * docs/libs/gst-rtsp-server-sections.txt:
1662 * gst/rtsp-server/rtsp-permissions.c:
1663 * gst/rtsp-server/rtsp-permissions.h:
1664 * tests/check/gst/permissions.c:
1665 permissions: add some new API to make this usable from bindings
1666 https://bugzilla.gnome.org/show_bug.cgi?id=787073
1668 2018-01-18 11:32:32 +0000 Tim-Philipp Müller <tim@centricular.com>
1670 * gst/rtsp-server/rtsp-token.c:
1671 rtsp-token: annotate constructors for bindings
1672 This maps _new_empty() to _new(), which also makes RTSPToken()
1673 work properly now. Since this API wasn't usable from bindings
1674 before, this should hopefully be fine.
1675 https://bugzilla.gnome.org/show_bug.cgi?id=787073
1677 2018-01-18 11:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
1679 * docs/libs/gst-rtsp-server-sections.txt:
1680 * gst/rtsp-server/rtsp-token.c:
1681 * gst/rtsp-server/rtsp-token.h:
1682 * tests/check/gst/token.c:
1683 rtsp-token: add some API to set fields from bindings
1684 The existing functions are all vararg-based and as such
1685 not usable from bindings.
1686 https://bugzilla.gnome.org/show_bug.cgi?id=787073
1688 2018-01-13 15:02:28 +0000 Tim-Philipp Müller <tim@centricular.com>
1690 * tests/check/gst/rtspclientsink.c:
1691 * tests/check/gst/rtspserver.c:
1692 * tests/check/gst/sessionpool.c:
1693 * tests/check/gst/stream.c:
1694 tests: fix indentation
1697 2018-01-13 14:58:55 +0000 Tim-Philipp Müller <tim@centricular.com>
1699 * tests/check/gst/rtspserver.c:
1700 tests: rtspserver: fix another ref leak
1701 Even if this didn't show up in valgrind.
1703 2018-01-13 14:58:00 +0000 Tim-Philipp Müller <tim@centricular.com>
1705 * tests/check/gst/rtspclientsink.c:
1706 tests: rtspclientsink: fix leak
1708 2018-01-02 14:19:31 +0100 Branko Subasic <branko@axis.com>
1710 * tests/check/gst/rtspserver.c:
1711 test: rtspserver: plug memory leak in test_no_session_timeout
1712 In test_no_session_timeout, unref the rtsp session object when the
1714 https://bugzilla.gnome.org/show_bug.cgi?id=792127
1716 2017-12-20 14:17:02 +0100 Edward Hervey <edward@centricular.com>
1718 * gst/rtsp-sink/gstrtspclientsink.c:
1719 rtpsclientsink: Initialize and clear newly added mutex and cond
1720 While it *did* work, glib would automatically create new mutex and cond
1721 ... which never got freed
1723 2017-12-19 11:34:37 +0200 Sebastian Dröge <sebastian@centricular.com>
1725 * gst/rtsp-server/rtsp-stream.c:
1726 rtsp-stream: Set multicast TTL on the multicast sockets
1727 And not if we do unicast UDP.
1728 https://bugzilla.gnome.org/show_bug.cgi?id=791743
1730 2017-12-19 11:14:48 +0200 Sebastian Dröge <sebastian@centricular.com>
1732 * gst/rtsp-server/rtsp-stream.c:
1733 rtsp-stream: Decide based on the sockets, not the addresses if we already allocated a socket
1734 In the multicast case (as in test-multicast, not test-multicast2), the
1735 address could be allocated/reserved (and thus set) already without
1736 allocating the actual socket. We need to allocate the socket here still
1737 instead of just claiming that it was already allocated.
1738 See https://bugzilla.gnome.org/show_bug.cgi?id=791743#c2
1740 2017-12-16 21:46:53 +0100 Patricia Muscalu <patricia@dovakhiin.com>
1742 * gst/rtsp-sink/gstrtspclientsink.c:
1743 * gst/rtsp-sink/gstrtspclientsink.h:
1744 rtspclientsink: Use the new rtsp-stream API
1745 https://bugzilla.gnome.org/show_bug.cgi?id=790412
1747 2017-12-16 21:01:43 +0100 Patricia Muscalu <patricia@dovakhiin.com>
1749 * gst/rtsp-sink/gstrtspclientsink.c:
1750 * gst/rtsp-sink/gstrtspclientsink.h:
1751 rtspclientsink: Wait until OPEN has been scheduled
1752 Make sure that the sink thread has started opening connection
1753 to the server before continuing.
1754 https://bugzilla.gnome.org/show_bug.cgi?id=790412
1756 2017-12-14 14:53:35 +1100 Matthew Waters <matthew@centricular.com>
1759 Automatic update of common submodule
1760 From e8c7a71 to 3fa2c9e
1762 2017-12-07 16:08:29 +0100 Edward Hervey <edward@centricular.com>
1764 * gst/rtsp-server/rtsp-media.c:
1765 * gst/rtsp-server/rtsp-session-media.c:
1766 * gst/rtsp-server/rtsp-stream.c:
1767 rtsp-server: Minor doc fixes
1770 2017-12-06 20:47:22 +0000 Tim-Philipp Müller <tim@centricular.com>
1773 * tests/Makefile.am:
1774 tests: disable all tests when --disable-tests is used
1775 Move conditional subdir include into top level.
1776 Based on patch by: Joel Holdsworth
1777 https://bugzilla.gnome.org/show_bug.cgi?id=757703
1779 2017-12-06 20:42:39 +0000 Tim-Philipp Müller <tim@centricular.com>
1782 * meson_options.txt:
1783 * tests/meson.build:
1784 meson: build more tests and add options to disable tests and examples
1786 2017-11-26 13:26:39 -0300 Thibault Saunier <tsaunier@gnome.org>
1788 * gst/rtsp-server/rtsp-session.c:
1789 Fix build when -Werror=deprecated-declarations is on
1790 As gst_rtsp_session_next_timeout is deprecated.
1792 ../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]
1793 res = (gst_rtsp_session_next_timeout (session, now) == 0);
1795 ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-session.c:685:1: note: declared here
1796 gst_rtsp_session_next_timeout (GstRTSPSession * session, GTimeVal * now)
1797 ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~
1800 2017-11-27 20:18:24 +1100 Matthew Waters <matthew@centricular.com>
1803 Automatic update of common submodule
1804 From 3f4aa96 to e8c7a71
1806 2017-11-25 20:34:16 +0100 Patricia Muscalu <patricia@dovakhiin.com>
1808 * tests/check/gst/media.c:
1809 check/media: Add seekability test case: not all streams are active
1810 Media contains two streams but only one is complete and prepared
1812 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1814 2017-11-25 20:32:02 +0100 Patricia Muscalu <patricia@dovakhiin.com>
1816 * gst/rtsp-server/rtsp-stream.c:
1817 rtsp-stream: Do not reset 'blocking' if stream is already blocked
1818 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1820 2017-11-25 20:45:44 +0100 Patricia Muscalu <patricia@dovakhiin.com>
1822 * gst/rtsp-server/rtsp-media.c:
1823 rtsp-media: Fix missing lock in gst_rtsp_media_seekable()
1824 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1826 2017-11-26 16:29:49 +0000 Tim-Philipp Müller <tim@centricular.com>
1829 meson: remove vs_module_defs_dir variable which is no longer needed
1831 2017-11-26 14:46:05 +0000 Tim-Philipp Müller <tim@centricular.com>
1833 * gst/rtsp-server/rtsp-session.h:
1836 2017-11-26 12:53:42 +0000 Tim-Philipp Müller <tim@centricular.com>
1839 * gst/rtsp-server/meson.build:
1841 * win32/common/libgstrtspserver.def:
1842 win32: remove .def file with exports
1843 They're no longer needed, symbol exporting is now explicit
1844 via GST_EXPORT in all cases (autotools, meson, incl. MSVC).
1846 2017-11-26 12:28:40 +0000 Tim-Philipp Müller <tim@centricular.com>
1849 autotools: stop controlling symbol visibility with -export-symbols-regex
1850 Instead, use -fvisibility=hidden and explicit exports via GST_EXPORT.
1851 This should result in consistent behaviour for the autotools and
1854 2017-11-26 12:47:08 +0000 Tim-Philipp Müller <tim@centricular.com>
1856 * gst/rtsp-server/rtsp-media.h:
1857 * gst/rtsp-server/rtsp-server.h:
1858 * gst/rtsp-server/rtsp-session.c:
1859 * gst/rtsp-server/rtsp-session.h:
1860 rtsp-server: add missing GST_EXPORT and export deprecated funcs
1862 2017-11-25 07:53:30 +0100 Edward Hervey <edward@centricular.com>
1864 * tests/check/gst/media.c:
1865 check: Add seekability testing on medias
1866 Make sure that once GstRTSPMedia are prepared they returned
1867 the expected seekability results
1868 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1870 2017-11-24 17:34:31 +0100 Edward Hervey <edward@centricular.com>
1872 * docs/libs/gst-rtsp-server-sections.txt:
1873 * gst/rtsp-server/rtsp-media.c:
1874 * gst/rtsp-server/rtsp-stream.c:
1875 * gst/rtsp-server/rtsp-stream.h:
1876 * win32/common/libgstrtspserver.def:
1877 rtsp-media: Enable seeking query before pipeline is complete
1878 SDP are now provided *before* the pipeline is fully complete. In order
1879 to know whether a media is seekable or not therefore requires asking
1880 the invididual streams.
1881 API: gst_rtsp_stream_seekable
1882 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1884 2017-11-23 20:34:03 +0100 Patricia Muscalu <patricia@axis.com>
1886 * gst/rtsp-server/rtsp-media.c:
1887 rtsp-media: Fix handling in default_unsuspend()
1888 Handle the case when streams are not blocked and media
1889 is suspended from PAUSED.
1890 Change-Id: I2f3d222ea7b9b20a0732ea5dc81a32d17ab75040
1891 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1893 2017-11-23 18:51:21 +0100 Patricia Muscalu <patricia@axis.com>
1895 * tests/check/gst/media.c:
1896 check/media: Fix thread pool leak.
1897 Change-Id: I0f92b1caca0ee518ae64a7dacfbd28a214c3eea1
1898 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1900 2017-11-23 18:39:44 +0100 Patricia Muscalu <patricia@axis.com>
1902 * gst/rtsp-server/rtsp-media.c:
1903 rtsp-media: Removed fakesink elements
1904 There is not need of adding fakesink elements to the media
1905 pipeline in the dynamic-payloader case.
1906 The media pipeline itself is dynamically updated with
1907 the receiver and sender parts that are based on the client
1908 transport information known after SETUP has been received.
1909 Change-Id: I4e88c9b500c04030669822f0d03b1842913f6cb9
1910 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1912 2017-11-23 09:10:54 +0100 Patricia Muscalu <patricia@axis.com>
1914 * gst/rtsp-server/rtsp-media.c:
1915 rtsp-media: Corrected ASYNC_DONE handling
1916 Media is complete when all the transport based parts are
1917 added to the media pipeline. At this point ASYNC_DONE is
1918 posted by the media pipeline and media is ready to enter
1920 Change-Id: I50fb8dfed88ebaf057d9a35fca2d7f0a70e9d1fa
1921 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1923 2017-11-22 12:24:38 +0100 Edward Hervey <bilboed@bilboed.com>
1925 * tests/check/gst/media.c:
1926 check/media: Check that prepared media can provide a SDP
1927 Whenever a RTSPMedia is prepared, it should be able to provide a SDP
1929 2017-11-21 09:53:19 +0100 Edward Hervey <edward@centricular.com>
1931 * gst/rtsp-server/rtsp-client.c:
1932 rtsp-client: Don't leak addr
1935 2017-11-21 09:53:08 +0100 Edward Hervey <bilboed@bilboed.com>
1937 * gst/rtsp-server/rtsp-client.c:
1938 * gst/rtsp-server/rtsp-session-media.c:
1939 * gst/rtsp-server/rtsp-stream.c:
1942 2017-11-20 18:30:19 +0100 Edward Hervey <bilboed@bilboed.com>
1944 * gst/rtsp-server/rtsp-media.c:
1945 rtsp-media: Don't unblock with remaining dynamic payloaders
1946 If we still have some dynamic paylaoders which haven't posted
1947 no-more-pads yet, don't go to PREPARED if one of the streams
1949 The risk was that we would end up not exposing/using all specified
1951 The downside is that if you have _multiple_ _live_ _dynamic_ payloaders
1952 then it will take a bit more time to start. But only if those 3
1953 conditions are present.
1954 https://bugzilla.gnome.org/show_bug.cgi?id=769521
1956 2017-11-20 16:49:29 +0100 Edward Hervey <edward@centricular.com>
1958 * gst/rtsp-server/rtsp-media.c:
1961 2017-11-20 16:48:55 +0100 Edward Hervey <edward@centricular.com>
1963 * gst/rtsp-server/rtsp-media.c:
1964 rtsp-media: Don't set float on a gint64 variable
1965 Just use 0. Fixes 'undefined' behaviour from clang
1967 2017-11-20 18:29:02 +0100 Edward Hervey <edward@centricular.com>
1969 * gst/rtsp-server/rtsp-media.c:
1970 rtsp-media: Fix previous commit
1971 We only want to count dynamic payloaders
1973 2017-11-20 09:32:07 +0100 Edward Hervey <edward@centricular.com>
1975 * gst/rtsp-server/rtsp-media.c:
1976 * tests/check/gst/media.c:
1977 rtsp-media: Handle multiple dynamic elements
1978 If we have more than one dynamic payloader in the pipeline, we need
1979 to wait until the *last* one emits 'no-more-pads' before switching
1981 Failure to do so would result in a race where some of the streams
1982 wouldn't properly be prepared
1983 https://bugzilla.gnome.org/show_bug.cgi?id=769521
1985 2017-11-16 12:18:20 +0200 Sebastian Dröge <sebastian@centricular.com>
1987 * win32/common/libgstrtspserver.def:
1988 win32: Fix exported symbols list
1990 2017-11-15 19:52:29 +0200 Sebastian Dröge <sebastian@centricular.com>
1992 * gst/rtsp-server/rtsp-stream.c:
1993 rtsp-stream: Only update the RTP udpsink if it actually exists
1994 For send-only streams it does not exist, but the RTCP udpsink might.
1996 2017-11-15 18:15:53 +0200 Sebastian Dröge <sebastian@centricular.com>
1998 * win32/common/libgstrtspserver.def:
1999 win32: Update exports
2001 2017-10-23 09:49:09 +0200 Patricia Muscalu <patricia@axis.com>
2003 * gst/rtsp-server/rtsp-media.c:
2004 * gst/rtsp-server/rtsp-stream.c:
2005 * gst/rtsp-server/rtsp-stream.h:
2006 rtsp-media: seek on media pipelines that are complete
2007 Make sure that a seek is performed on pipelines that
2008 contain at least one sink element.
2009 Change-Id: Icf398e10add3191d104b1289de612412da326819
2010 https://bugzilla.gnome.org/show_bug.cgi?id=788340
2012 2017-10-17 10:44:33 +0200 Patricia Muscalu <patricia@axis.com>
2014 * gst/rtsp-server/rtsp-client.c:
2015 * gst/rtsp-server/rtsp-media.c:
2016 * gst/rtsp-server/rtsp-media.h:
2017 * gst/rtsp-server/rtsp-stream.c:
2018 * gst/rtsp-server/rtsp-stream.h:
2019 * tests/check/gst/client.c:
2020 * tests/check/gst/media.c:
2021 * tests/check/gst/rtspserver.c:
2022 * tests/check/gst/stream.c:
2023 Dynamically reconfigure pipeline in PLAY based on transports
2024 The initial pipeline does not contain specific transport
2025 elements. The receiver and the sender parts are added
2027 If the media is shared, the streams are dynamically
2028 reconfigured after each PLAY.
2029 https://bugzilla.gnome.org/show_bug.cgi?id=788340
2031 2017-10-16 12:40:57 +0200 Patricia Muscalu <patricia@axis.com>
2033 * gst/rtsp-server/rtsp-stream.c:
2034 rtsp-stream: obtain stream position from pad
2035 If no sinks have been added yet, obtain the current and
2036 the stop position of the stream from the send_src pad.
2037 Change-Id: Iacd4ab4bdc69f6b49370d06012880ce48a7d595a
2038 https://bugzilla.gnome.org/show_bug.cgi?id=788340
2040 2017-10-16 11:35:10 +0200 Patricia Muscalu <patricia@axis.com>
2042 * gst/rtsp-server/rtsp-session-media.c:
2043 * gst/rtsp-server/rtsp-session-media.h:
2044 rtsp-session-media: add function to get a list of transports
2045 Change-Id: I817e10624da0f3200f24d1b232cff481099278e3
2046 https://bugzilla.gnome.org/show_bug.cgi?id=788340
2048 2017-10-16 11:15:55 +0200 Patricia Muscalu <patricia@axis.com>
2050 * gst/rtsp-server/rtsp-stream.c:
2051 * gst/rtsp-server/rtsp-stream.h:
2052 rtsp-stream: add functions to get rtp and rtcp multicast sockets
2053 Change-Id: Iddfe6e0bd250cb0159096d5eba9e4202d22b56db
2054 https://bugzilla.gnome.org/show_bug.cgi?id=788340
2056 2017-10-20 12:21:48 +0200 Patricia Muscalu <patricia@axis.com>
2058 * gst/rtsp-server/rtsp-stream.c:
2059 stream: set async=sync=false only for RTCP appsink
2060 Change-Id: I929a218a9adf4759f61322b6f2063aacc5595f90
2061 https://bugzilla.gnome.org/show_bug.cgi?id=788340
2063 2017-10-16 10:10:17 +0200 Patricia Muscalu <patricia@axis.com>
2065 * gst/rtsp-server/rtsp-media.c:
2066 rtsp-media: return minimum value in query position case
2067 The minimum position should be returned as we are interested
2068 in the whole interval.
2069 Change-Id: I30e297fc040c995ae40c25dee8ff56321612fe2b
2070 https://bugzilla.gnome.org/show_bug.cgi?id=788340
2072 2017-08-09 11:52:38 +0200 Jonathan Karlsson <jonakn@axis.com>
2074 * gst/rtsp-server/rtsp-session.c:
2075 * tests/check/gst/rtspserver.c:
2076 rtsp-session: Handle the case when timeout=0
2077 According to the documentation, a timeout of value 0 means
2078 that the session never timeouts. This adds handling of that.
2079 If timeout=0 we just return with a -1 from
2080 gst_rtsp_session_next_timeout_usec ().
2081 https://bugzilla.gnome.org/show_bug.cgi?id=785058
2083 2017-07-17 17:15:22 +0300 Sebastian Dröge <sebastian@centricular.com>
2085 * gst/rtsp-sink/gstrtspclientsink.c:
2086 rtspclientsink: Add "accept-certificate" signal for manually checking a TLS certificate for validity
2087 https://bugzilla.gnome.org/show_bug.cgi?id=785024
2089 2017-10-26 14:43:19 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2091 * docs/libs/gst-rtsp-server-sections.txt:
2092 * gst/rtsp-server/rtsp-media-factory.c:
2093 docs: add media factory transport mode accessors
2094 and fix the documentation for the return value of the getter
2096 2017-10-09 12:43:01 +0200 Branko Subasic <branko@axis.com>
2098 * gst/rtsp-server/rtsp-client.c:
2099 rtsp-client: unref 'pipelined_requests' in finalize
2100 The hash table priv->pipelined_requests is not unref:ed in the
2101 finalize funktion. Make sure it is.
2102 https://bugzilla.gnome.org/show_bug.cgi?id=788704
2104 2017-10-09 14:44:40 +0200 Thibault Saunier <thibault.saunier@osg.samsung.com>
2106 * gst/rtsp-server/rtsp-media.c:
2107 rtsp-media: Initialize scalar variable
2110 2017-10-06 10:27:34 +0200 Edward Hervey <edward@centricular.com>
2112 * win32/common/libgstrtspserver.def:
2113 win32: Update export file
2115 2017-04-22 09:26:07 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
2117 * gst/rtsp-server/rtsp-client.c:
2118 * gst/rtsp-server/rtsp-media.c:
2119 * gst/rtsp-server/rtsp-media.h:
2120 Start support for RTSP 2.0
2121 This adds basic support for new 2.0 features, though the protocol is
2122 subposdely backward incompatible, most semantics are the sames.
2125 * version negotiation
2126 * pipelined requests support
2127 * Media-Properties support
2128 * Accept-Ranges support
2130 * gst_rtsp_media_seekable
2131 The RTSP methods that have been removed when using 2.0 now return
2133 https://bugzilla.gnome.org/show_bug.cgi?id=781446
2135 2017-06-02 15:37:54 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
2137 * gst/rtsp-server/rtsp-stream.c:
2138 stream: Use stream duration as stream-stop if segment was not configured with a stop
2139 Allowing client to know stream duration when no seeking happened.
2140 https://bugzilla.gnome.org/show_bug.cgi?id=783435
2142 2017-09-25 19:40:17 +0300 Sebastian Dröge <sebastian@centricular.com>
2144 * gst/rtsp-server/rtsp-media-factory.c:
2145 rtsp-media-factory: Don't cache any media if NULL was returned as key
2146 The docs already mentioned this, but we actually stored it in the hash
2147 table with key==NULL and leaked its reference forever.
2149 2017-09-18 19:31:31 +0200 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
2151 * gst/rtsp-sink/gstrtspclientsink.c:
2152 * gst/rtsp-sink/gstrtspclientsink.h:
2153 rtspclientsink: Use a mutex for protecting against concurrent send/receives
2154 This is a simple port of:
2155 * a722f6e8329032c6eda4865d6a07f4ba5981d7ea
2156 * c438545dc9e2f14f657bc0ef261fff726449867b
2157 * cd17c71dcea5c9310d21f1347c7520983e5869ac
2158 in gst-plugins-good.
2160 2017-08-31 13:24:15 +0530 Satya Prakash Gupta <sp.gupta@samsung.com>
2162 * gst/rtsp-server/rtsp-sdp.c:
2163 sdp: fix Memory leak in error case
2164 https://bugzilla.gnome.org/show_bug.cgi?id=787059
2166 2017-08-18 17:37:01 +0100 Tim-Philipp Müller <tim@centricular.com>
2168 * pkgconfig/meson.build:
2169 meson: don't install -uninstalled.pc file
2170 https://bugzilla.gnome.org/show_bug.cgi?id=786457
2172 2017-08-17 12:26:17 +0100 Tim-Philipp Müller <tim@centricular.com>
2175 Automatic update of common submodule
2176 From 48a5d85 to 3f4aa96
2178 2017-08-14 21:04:23 +0300 Sebastian Dröge <sebastian@centricular.com>
2180 * gst/rtsp-server/rtsp-client.c:
2181 rtsp-client: Fix typo in debug message
2183 2017-08-11 14:14:32 +0100 Tim-Philipp Müller <tim@centricular.com>
2186 meson: hide symbols by default unless explicitly exported
2188 2017-08-10 14:20:12 +0100 Tim-Philipp Müller <tim@centricular.com>
2190 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
2191 pkgconfig: remove -I@srcdir@/.. which duplicates abs_top_srcdir
2192 Fixes meson warning about undefined @srcdir@.
2194 2017-07-21 13:36:00 +0100 Tim-Philipp Müller <tim@centricular.com>
2196 * tests/meson.build:
2197 meson: skip tests on windows for now
2198 As we do in the other modules. As libgstcheck is currently not
2199 built on windows. Fixes "Fallback variable 'gst_check_dep' in
2200 the subproject 'gstreamer' does not exist"" Meson error.
2202 2017-06-22 07:25:07 -0700 Julien Isorce <julien.isorce@gmail.com>
2204 * gst/rtsp-server/rtsp-stream.c:
2205 rtsp-stream: fix connection delay due to wrong assumption on last-sample
2206 Commit 852cc09f542af5cadd79ffd7fe79d6475cf57e14 assumed that
2207 multiudpsink's last-sample always comes from the payloader. Which
2208 is wrong if auxiliary streams are multiplexed in the same stream.
2209 So check the buffer's ssrc against the caps'ssrc before to use its
2210 seqnum. If not the same ssrc just use the payloader as done prior
2211 the commit above or when there is no last-sample yet.
2212 https://bugzilla.gnome.org/show_bug.cgi?id=784094
2214 2017-06-23 16:19:04 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
2217 meson: Allow using glib as a subproject
2219 2017-06-26 09:55:49 +0100 Tim-Philipp Müller <tim@centricular.com>
2222 meson: fix with-package-name option
2223 https://bugzilla.gnome.org/show_bug.cgi?id=784082
2225 2017-06-09 20:16:28 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
2228 Distribute meson_options.txt
2230 2017-06-09 20:11:47 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
2233 And config.h.meson is no longer dist either
2235 2017-06-09 21:27:09 +0100 Tim-Philipp Müller <tim@centricular.com>
2239 meson: config.h.meson is no longer needed
2241 2017-06-07 13:04:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
2243 * tests/check/meson.build:
2244 * tests/meson.build:
2245 meson: Fix building tests and activate them again
2247 2017-06-07 12:55:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
2249 * tests/check/meson.build:
2250 meson: Do not use path separator in test names
2251 Avoiding warnings like:
2252 WARNING: Target "elements/audioamplify" has a path separator in its name.
2254 2017-05-20 15:07:31 +0100 Tim-Philipp Müller <tim@centricular.com>
2257 * meson_options.txt:
2258 meson: add options to set package name and origin
2259 https://bugzilla.gnome.org/show_bug.cgi?id=782172
2261 2017-05-18 10:35:18 +0100 Tim-Philipp Müller <tim@centricular.com>
2263 * gst/rtsp-server/rtsp-address-pool.h:
2264 * gst/rtsp-server/rtsp-auth.h:
2265 * gst/rtsp-server/rtsp-client.h:
2266 * gst/rtsp-server/rtsp-context.h:
2267 * gst/rtsp-server/rtsp-media-factory-uri.h:
2268 * gst/rtsp-server/rtsp-media-factory.h:
2269 * gst/rtsp-server/rtsp-media.h:
2270 * gst/rtsp-server/rtsp-mount-points.h:
2271 * gst/rtsp-server/rtsp-params.h:
2272 * gst/rtsp-server/rtsp-permissions.h:
2273 * gst/rtsp-server/rtsp-sdp.h:
2274 * gst/rtsp-server/rtsp-server.h:
2275 * gst/rtsp-server/rtsp-session-media.h:
2276 * gst/rtsp-server/rtsp-session-pool.h:
2277 * gst/rtsp-server/rtsp-session.h:
2278 * gst/rtsp-server/rtsp-stream-transport.h:
2279 * gst/rtsp-server/rtsp-stream.h:
2280 * gst/rtsp-server/rtsp-thread-pool.h:
2281 * gst/rtsp-server/rtsp-token.h:
2282 Mark symbols explicitly for export with GST_EXPORT
2284 2017-05-16 14:44:43 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
2287 * gst/rtsp-sink/Makefile.am:
2288 Remove plugin specific static build option
2289 Static and dynamic plugins now have the same interface. The standard
2290 --enable-static/--enable-shared toggle are sufficient.
2292 2017-05-04 18:59:14 +0300 Sebastian Dröge <sebastian@centricular.com>
2298 === release 1.12.0 ===
2300 2017-05-04 15:40:46 +0300 Sebastian Dröge <sebastian@centricular.com>
2306 * gst-rtsp-server.doap:
2310 === release 1.11.91 ===
2312 2017-04-27 17:42:02 +0300 Sebastian Dröge <sebastian@centricular.com>
2318 * gst-rtsp-server.doap:
2322 2017-04-24 20:30:37 +0100 Tim-Philipp Müller <tim@centricular.com>
2325 Automatic update of common submodule
2326 From 60aeef6 to 48a5d85
2328 2017-04-13 14:20:10 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
2330 * gst/rtsp-server/rtsp-media-factory.c:
2331 * gst/rtsp-server/rtsp-media.c:
2332 * gst/rtsp-server/rtsp-session.c:
2333 * gst/rtsp-server/rtsp-stream.c:
2334 gi: Fix some annotations and docstrings
2336 2017-04-13 13:52:26 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
2338 * gst/rtsp-server/meson.build:
2340 * meson_options.txt:
2343 2017-04-10 23:51:12 +0100 Tim-Philipp Müller <tim@centricular.com>
2347 Automatic update of common submodule
2348 From 39ac2f5 to 60aeef6
2350 === release 1.11.90 ===
2352 2017-04-07 16:35:03 +0300 Sebastian Dröge <sebastian@centricular.com>
2358 * gst-rtsp-server.doap:
2362 2017-03-27 18:19:33 +0100 Tim-Philipp Müller <tim@centricular.com>
2364 * examples/test-launch.c:
2365 examples: make test-launch pipeline shared by default as well
2367 2017-02-27 19:10:44 +0200 Sebastian Dröge <sebastian@centricular.com>
2369 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
2370 gstreamer-rtsp-server: Add both srcdir and builddir to the include path
2371 Just the build dir is not going to work for srcdir!=builddir.
2373 2017-02-24 15:59:54 +0200 Sebastian Dröge <sebastian@centricular.com>
2376 meson: Update version
2378 2017-02-24 15:37:49 +0200 Sebastian Dröge <sebastian@centricular.com>
2383 === release 1.11.2 ===
2385 2017-02-24 15:10:07 +0200 Sebastian Dröge <sebastian@centricular.com>
2391 * gst-rtsp-server.doap:
2394 2017-02-14 20:40:26 +0000 Tim-Philipp Müller <tim@centricular.com>
2397 meson: dist meson build files
2398 Ship meson build files in tarballs, so people who use tarballs
2399 in their builds can start playing with meson already.
2401 2017-02-07 23:39:37 +1100 Jan Schmidt <jan@centricular.com>
2403 * examples/test-record.c:
2404 examples/test-record: Add extra line to initial printout
2405 Add an example line of how to deliver a stream to the
2408 2017-01-19 14:57:19 +0200 Sebastian Dröge <sebastian@centricular.com>
2410 * gst/rtsp-server/rtsp-client.c:
2411 rtsp-client: Also handle the (S|G)ET_PARAMETER case of size==0 || !data as keep-alive
2412 If there is no Content-Length header, no body would be allocated and the
2413 '\0' would also not be appended to the body.
2415 2017-01-19 14:24:07 +0200 Sebastian Dröge <sebastian@centricular.com>
2417 * gst/rtsp-server/rtsp-client.c:
2418 rtsp-client: Fix handling of keep-alive GET_PARAMETER/SET_PARAMETER
2419 While they logically have 0 bytes length, GstRTSPConnection is appending
2420 a '\0' to everything making the size be 1 instead.
2422 2017-01-13 12:39:36 +0000 Tim-Philipp Müller <tim@centricular.com>
2427 2017-01-12 19:04:23 +0200 Sebastian Dröge <sebastian@centricular.com>
2429 * gst/rtsp-server/rtsp-session.c:
2430 rtsp-session: Only remove deprecated API if requested to do so, not just when disabling
2431 gst_rtsp_session_is_expired() and gst_rtsp_session_next_timeout() were
2434 2017-01-12 16:32:59 +0200 Sebastian Dröge <sebastian@centricular.com>
2439 === release 1.11.1 ===
2441 2017-01-12 16:14:46 +0200 Sebastian Dröge <sebastian@centricular.com>
2447 * gst-rtsp-server.doap:
2448 * win32/common/libgstrtspserver.def:
2451 2017-01-10 08:34:50 +0100 Patricia Muscalu <patricia@axis.com>
2453 * gst/rtsp-server/rtsp-stream.c:
2454 rtsp-stream: corrected if-statement in _get_server_port()
2455 This bug was accidentally introduced while fixing a segfault
2456 in _get_server_port() function.
2457 https://bugzilla.gnome.org/show_bug.cgi?id=776345
2459 2017-01-09 14:12:05 +0100 Patricia Muscalu <patricia@axis.com>
2461 * gst/rtsp-server/rtsp-stream.c:
2462 * tests/check/gst/stream.c:
2463 rtsp-stream: fixed segmenation fault in _get_server_port()
2464 Calling function gst_rtsp_stream_get_server_port() results in
2465 segmenation fault in the RTP/RTSP/TCP case.
2466 Port that the server will use to receive RTCP makes only
2467 sense in the UDP case, however the function should handle
2468 the TCP case in a nicer way.
2469 https://bugzilla.gnome.org/show_bug.cgi?id=776345
2471 2017-01-09 12:22:40 +0300 Aleksandr Slobodeniuk <alenuke@yandex.ru>
2473 * gst/rtsp-server/rtsp-media-factory.c:
2474 dosc: Fix a little typo
2475 https://bugzilla.gnome.org/show_bug.cgi?id=777037
2477 2017-01-04 16:20:54 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
2479 * pkgconfig/Makefile.am:
2480 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
2481 * pkgconfig/meson.build:
2482 meson: generate pkg-config -uninstalled pc files
2483 Generating those files is useful for users building the GStreamer stack
2484 using meson and having to link it to another project which is still
2485 using the autotools.
2486 https://bugzilla.gnome.org/show_bug.cgi?id=776810
2488 2017-01-04 16:11:08 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
2490 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
2491 pkgconfig: fix -uninstalled pc file
2492 pcfiledir was never defined so the paths were wrong.
2493 https://bugzilla.gnome.org/show_bug.cgi?id=776867
2495 2016-12-21 13:41:50 +0100 Patricia Muscalu <patricia@axis.com>
2497 * gst/rtsp-server/rtsp-stream.c:
2498 * tests/check/gst/rtspserver.c:
2499 rtsp-stream: Fixed TCP transport case
2500 Make sure that the appsink element is actually added to
2501 the bin before trying to link it with the elements in it.
2502 https://bugzilla.gnome.org/show_bug.cgi?id=776343
2504 2016-12-16 17:26:04 +0000 Tim-Philipp Müller <tim@centricular.com>
2510 Remove generated .spec file
2511 Likely extremely bitrotten, and we should not ship this anyway.
2513 2016-12-03 08:21:02 +0100 Edward Hervey <bilboed@bilboed.com>
2516 Automatic update of common submodule
2517 From f980fd9 to 39ac2f5
2519 2016-12-02 15:40:09 +0100 Edward Hervey <edward@centricular.com>
2521 * gst/rtsp-server/rtsp-media.c:
2522 media: Fix pt map caps
2523 Since decryption is handled within rtpbin, all outcoming stream
2524 caps will be application/x-rtp (i.e. regular rtp)
2525 Fixes RECORD with SRTP streams
2527 2016-12-02 15:38:04 +0100 Edward Hervey <edward@centricular.com>
2529 * gst/rtsp-server/rtsp-media-factory.c:
2530 media-factory: Create media objects with the proper transport mode
2531 The function called immediately afterwards (collect_streams()) will
2532 need it to work properly
2534 2016-12-02 14:36:50 +0200 Sebastian Dröge <sebastian@centricular.com>
2536 * gst/rtsp-server/rtsp-auth.c:
2537 rtsp-auth: Don't remove digest-auth nonces that already/still have a client connected
2539 2016-12-01 18:04:34 +0200 Sebastian Dröge <sebastian@centricular.com>
2541 * gst/rtsp-server/rtsp-media-factory.c:
2542 rtsp-media-factory: Don't create a pipeline for the media pipeline string
2543 We're going to put a pipeline into a pipeline otherwise, which is not
2546 2016-10-25 15:41:28 +0300 Kseniia Vasilchuk <vasilchukkseniia@gmail.com>
2548 * gst/rtsp-server/rtsp-media.c:
2549 media: Fix race condition around finish_unprepare() if called multiple time
2550 https://bugzilla.gnome.org/show_bug.cgi?id=755329
2552 2016-11-30 14:06:36 +1100 Jan Schmidt <jan@centricular.com>
2554 * gst/rtsp-sink/gstrtspclientsink.c:
2555 rtspclientsink: Don't leave stale pointer after unref
2556 Fix a warning on shutdown - don't keep a pointer to an
2557 alread-unreffed object.
2559 2016-11-26 11:24:50 +0000 Tim-Philipp Müller <tim@centricular.com>
2562 common: use https protocol for common submodule
2563 https://bugzilla.gnome.org/show_bug.cgi?id=775110
2565 2016-11-21 23:29:56 +1100 Matthew Waters <matthew@centricular.com>
2567 * gst/rtsp-server/rtsp-stream.c:
2568 stream: block the output of rtpbin instead of the source pipeline
2569 85c52e194bcb81928b96614be0ae47d59eccb1ce introduced a more correct
2570 detection of the srtp rollover counter to add to the SDP.
2571 Unfortunately, it was incomplete for live pipelines where the logic
2572 blocks the source bin before creating the SDP and thus would never have
2573 the necessary informaiton to create a correct SDP with srtp encryption.
2574 Move the pad blocks to rtpbin's output pads instead so that the
2575 necessary information can be created before we need the information for
2577 https://bugzilla.gnome.org/show_bug.cgi?id=770239
2579 2016-11-21 16:02:39 +0100 Dag Gullberg <dagg@axis.com>
2581 * gst/rtsp-server/rtsp-client.c:
2582 rtsp-client: add IDLE timeout, before session exists
2583 The RTSP server will not timeout an idle RTSP connection
2584 (note this is different from doing timeout on a RTSP
2586 At least for Apache this is a problem when running RTSP over
2587 HTTPS since it uses one of the threads (there is a rather
2588 limited number) that are available for handling requests.
2589 https://bugzilla.gnome.org/show_bug.cgi?id=771830
2591 2016-11-23 09:45:08 +0000 Tim-Philipp Müller <tim@centricular.com>
2596 2016-11-21 13:05:50 +0100 Göran Jönsson <goranjn@axis.com>
2598 * gst/rtsp-server/rtsp-stream.c:
2599 rtsp-stream: Set close-socket FALSE on UDP src:es
2600 With this RTSP server can use the sockets independent on the udpsrc
2602 When the udp src is finalized it will unref socket and when g_socket
2603 is finalized the socket will be closed.
2604 https://bugzilla.gnome.org/show_bug.cgi?id=765673
2606 2016-11-18 17:47:13 +0200 Sebastian Dröge <sebastian@centricular.com>
2608 * gst/rtsp-sink/gstrtspclientsink.c:
2609 rtspclientsink: Move to new helper function to parse authentication responses
2610 https://bugzilla.gnome.org/show_bug.cgi?id=774416
2612 2016-11-16 08:42:24 +0200 Sebastian Dröge <sebastian@centricular.com>
2614 * examples/Makefile.am:
2615 * examples/test-auth-digest.c:
2616 * gst/rtsp-server/rtsp-auth.c:
2617 * gst/rtsp-server/rtsp-auth.h:
2618 * win32/common/libgstrtspserver.def:
2619 rtsp-auth: Add support for Digest authentication
2620 https://bugzilla.gnome.org/show_bug.cgi?id=774416
2622 2016-11-17 09:41:53 -0800 Scott D Phillips <scott.d.phillips@intel.com>
2625 * gst/rtsp-server/meson.build:
2627 * tests/check/meson.build:
2629 * win32/common/libgstrtspserver.def:
2630 Enable building with MSVC
2631 https://bugzilla.gnome.org/show_bug.cgi?id=774640
2633 2016-11-18 20:23:14 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
2636 meson: gstreamer gst_check_dep does not exist on windows
2638 2016-11-17 09:43:37 -0800 Scott D Phillips <scott.d.phillips@intel.com>
2640 * gst/rtsp-server/rtsp-client.c:
2641 client: update do_send_message to match type GstRTSPClientSendFunc
2642 This type mismatch fails building with MSVC
2643 https://bugzilla.gnome.org/show_bug.cgi?id=774640
2645 2016-11-11 14:42:08 +0200 Sebastian Dröge <sebastian@centricular.com>
2647 * gst/rtsp-server/rtsp-sdp.c:
2648 rtsp-sdp: Fix indentation
2650 2016-11-10 05:16:00 +0000 Neha Arora <arora.neha@samsung.com>
2652 * gst/rtsp-server/rtsp-media.c:
2653 rtsp-media: Only signal "new-state" if the state has actually changed
2654 https://bugzilla.gnome.org/show_bug.cgi?id=774173
2656 2016-08-24 11:39:13 +0200 Branko Subasic <branko@axis.com>
2658 * gst/rtsp-server/rtsp-client.c:
2659 * gst/rtsp-server/rtsp-client.h:
2660 client: emit signal in the beginning of each rtsp request
2661 These signals let the application validate the requests, configure the
2662 media/stream in a certain way and also generate error status code in
2663 case of error or bad request.
2664 https://bugzilla.gnome.org/show_bug.cgi?id=758062
2666 2016-11-01 18:10:35 +0000 Tim-Philipp Müller <tim@centricular.com>
2669 meson: update version
2671 === release 1.11.0 ===
2673 2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com>
2678 === release 1.10.0 ===
2680 2016-11-01 18:06:46 +0200 Sebastian Dröge <sebastian@centricular.com>
2686 * gst-rtsp-server.doap:
2689 2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com>
2691 * tests/check/gst/rtspserver.c:
2692 * tests/check/gst/stream.c:
2693 tests: try to avoid using the same ports in different tests
2694 Causes problems with client multicast tests otherwise if
2695 tests are run in parallel.
2696 https://bugzilla.gnome.org/show_bug.cgi?id=773640
2698 2016-10-28 17:50:59 +0100 Tim-Philipp Müller <tim@centricular.com>
2700 * tests/check/gst/client.c:
2701 tests: client: use fail_unless_equals_foo() for better failure reporting
2703 2016-09-26 11:16:04 +0200 Göran Jönsson <goranjn@axis.com>
2705 * gst/rtsp-server/rtsp-client.c:
2706 rtsp-client: Session filter in unwatch session
2707 Call session filter with filter_session_media as paramer in
2708 client_unwatch_session if using drop_backlog = FALSE.
2709 In client_unwatch_session its allowed to grow the watchs backlog.
2710 If using drop_backlog = FALSE and the backlog is full it will cause
2711 a deadlock when setting session media state to NULL
2712 if the backlog is not allowed to grow.
2713 https://bugzilla.gnome.org/show_bug.cgi?id=771983
2715 2016-10-20 21:40:18 +0100 Tim-Philipp Müller <tim@centricular.com>
2718 meson: add fallbacks for gst modules
2721 2016-09-14 17:48:39 +0300 Nikita Bobkov <NikitaDBobkov@gmail.com>
2723 * gst/rtsp-server/rtsp-client.c:
2724 rtsp-client: Fix factory leaking in find_media() in error cases
2725 https://bugzilla.gnome.org/show_bug.cgi?id=771488
2727 2016-10-06 11:47:50 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2729 * gst/rtsp-server/rtsp-stream.c:
2730 stream: Fix randomly missing streams from SDP with dynamic elements
2731 When using dynamic elements, gst_rtsp_stream_join_bin() is called from
2732 "pad-added" signal. In that case priv->srcpad could already have its caps,
2733 and they'll be sent to priv->send_src[0] pad. That means that when it
2734 connects "notify::caps" signal, that pad could already have received its
2735 caps and the signal won't be emitted anymore.
2736 In that case priv->caps stay to NULL and when building the SDP that stream
2737 gets ignored. Leading to missing video or audio when playing in client side.
2738 https://bugzilla.gnome.org/show_bug.cgi?id=772478
2740 2016-09-30 11:42:08 +0100 Tim-Philipp Müller <tim@centricular.com>
2743 meson: update version
2745 === release 1.9.90 ===
2747 2016-09-30 13:04:12 +0300 Sebastian Dröge <sebastian@centricular.com>
2753 * gst-rtsp-server.doap:
2756 2016-09-17 13:17:19 +0100 Ian Jamison <ian.dev@arkver.com>
2758 * gst/rtsp-server/rtsp-media-factory.c:
2759 * gst/rtsp-server/rtsp-media.c:
2760 * gst/rtsp-server/rtsp-stream.c:
2761 rtsp-server: Hint that set_multicast_iface expects the name of the interface
2762 To prevent any possibly confusion with IPs or anything else.
2763 https://bugzilla.gnome.org/show_bug.cgi?id=771530
2765 2016-09-18 09:58:55 -0400 Sebastian Dröge <sebastian@centricular.com>
2767 * gst/rtsp-server/rtsp-media-factory.c:
2768 * gst/rtsp-server/rtsp-media.c:
2769 rtsp-media: Call g_free() instead of g_object_unref() on multicast-iface strings
2770 https://bugzilla.gnome.org/show_bug.cgi?id=763000#c5
2772 2016-09-14 11:31:15 +0200 Sebastian Dröge <sebastian@centricular.com>
2775 configure: Depend on gstreamer 1.9.2.1
2777 2016-09-10 20:52:31 +1000 Jan Schmidt <jan@centricular.com>
2781 Automatic update of common submodule
2782 From b18d820 to f980fd9
2784 2016-09-10 09:58:31 +1000 Jan Schmidt <jan@centricular.com>
2788 Automatic update of common submodule
2789 From 6f2d209 to b18d820
2791 2016-09-07 18:44:34 +0300 Sebastian Dröge <sebastian@centricular.com>
2793 * gst/rtsp-server/rtsp-stream.c:
2794 rtsp-stream: Remove unused _locked() variant of a function
2795 It was added during refactoring.
2797 2016-09-07 10:21:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2799 * gst/rtsp-server/rtsp-stream.c:
2800 stream: cosmetic cleanup
2801 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2803 2016-09-07 10:16:19 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2805 * gst/rtsp-server/rtsp-stream.c:
2806 stream: Compare IP addresses case insensitive in more places
2807 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2809 2016-09-07 10:12:18 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2812 * gst/rtsp-server/rtsp-stream.c:
2813 stream: Fix leaked joined_bin
2814 There is no need to keep a strong ref on it, and _leave_bin() was
2815 setting it to NULL before calling g_clear_object() so it was leaked.
2816 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2818 2016-09-06 19:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
2820 * gst/rtsp-server/rtsp-stream.c:
2821 rtsp-stream: Compare IP address strings case insensitive
2822 Otherwise IPv6 addresses might fail this comparision.
2824 2016-09-06 19:10:21 +0300 Sebastian Dröge <sebastian@centricular.com>
2826 * gst/rtsp-server/rtsp-stream.c:
2827 rtsp-stream: Bind multicast sockets to ANY as before
2828 https://bugzilla.gnome.org/show_bug.cgi?id=766612#c48
2830 2016-09-05 18:31:36 +0300 Kseniia <vasilchukkseniia@gmail.com>
2832 * gst/rtsp-server/rtsp-session.c:
2833 rtsp-session: Fix segfault when doing keep-alive after removing the session
2834 If keep-alive happens after removing the session but before finalizing the
2835 stream transport, we would segfault.
2836 https://bugzilla.gnome.org/show_bug.cgi?id=750544
2838 2016-09-05 18:04:50 +0300 Sebastian Dröge <sebastian@centricular.com>
2840 * gst/rtsp-server/rtsp-stream.c:
2841 rtsp-stream: Always create multicast UDP elements if the protocol flag is set
2842 Adding them later will cause deadlocks due to
2843 1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
2844 2) adding the multicast sink
2845 3) waiting for it to get data to preroll again
2846 3) never happens because the queues after the tee are full.
2848 2016-09-05 16:32:57 +0300 Sebastian Dröge <sebastian@centricular.com>
2850 * gst/rtsp-server/rtsp-stream.c:
2851 rtsp-stream: Fix up various multicast related issues
2853 2016-09-05 13:40:59 +0300 Sebastian Dröge <sebastian@centricular.com>
2855 * tests/check/gst/stream.c:
2856 tests: Fix compilation
2858 2016-07-28 15:33:05 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2860 * gst/rtsp-server/rtsp-client.c:
2861 * gst/rtsp-server/rtsp-stream.c:
2862 * tests/check/gst/stream.c:
2863 stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast
2864 This is basically reverting changes introduced in commit f62a9a7,
2865 because it was introducing various regressions:
2866 - It introduces a leak of udpsrc elements that got wrongly fixed by adding
2867 an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
2868 ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
2869 - If a mcast client connects, it creates a new socket in SETUP to try to respect
2870 the destination/port given by the client in the transport, and overrides the
2871 socket already set on the udpsink element. That means that if we already had a
2872 client connected, the source address on the udp packets it receives suddenly
2874 - If a 2nd mcast client connects, the destination/port in its transport is
2875 ignored but its transport wasn't updated.
2876 What this patch does:
2877 - Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
2878 - Always have a tee+queue when udp is enabled. This could be optimized
2879 again in a later patch, but is more complicated. If no unicast clients
2880 connects then those elements are useless, this could be also optimized
2882 - When mcast transport is added, it creates a new set of udpsrc/udpsink,
2883 seperated from those for unicast clients. Since we already support only
2884 one mcast address, we also create only one set of elements.
2885 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2887 2016-07-28 15:20:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2889 * gst/rtsp-server/rtsp-stream.c:
2890 stream: factor our plug_src function
2891 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2893 2016-07-21 21:46:16 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2895 * gst/rtsp-server/rtsp-stream.c:
2896 stream: factor out plug_sink function
2897 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2899 2016-07-20 23:05:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2901 * gst/rtsp-server/rtsp-stream.c:
2902 stream: small documentation clarification
2903 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2905 2016-07-20 15:35:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2907 * gst/rtsp-server/rtsp-stream.c:
2908 stream: rename addr_v4/6 to mcast_addr_v4/6 for clarity
2909 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2911 2016-07-14 11:10:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2913 * gst/rtsp-server/rtsp-stream.c:
2914 stream: Keep a ref on joined bin
2915 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2917 2016-07-20 15:11:32 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2919 * gst/rtsp-server/rtsp-stream.c:
2920 stream: code cleanup
2921 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2923 2016-07-20 23:18:23 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2925 * gst/rtsp-server/rtsp-stream.c:
2926 stream: small fix in error code path
2927 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2929 2016-07-20 20:09:57 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2931 * gst/rtsp-server/rtsp-stream.c:
2932 Revert "rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc"
2933 This partly reverts commit cba045e1b19fad6e689e10206f57903e15f1229a,
2934 but keeps unit tests.
2935 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2937 2016-09-01 12:33:00 +0300 Sebastian Dröge <sebastian@centricular.com>
2942 === release 1.9.2 ===
2944 2016-09-01 12:32:51 +0300 Sebastian Dröge <sebastian@centricular.com>
2950 * gst-rtsp-server.doap:
2953 2016-01-27 01:03:52 +0000 Tim-Philipp Müller <tim@centricular.com>
2956 * examples/meson.build:
2958 * gst/rtsp-server/meson.build:
2959 * gst/rtsp-sink/meson.build:
2961 * pkgconfig/meson.build:
2962 * tests/check/meson.build:
2963 * tests/meson.build:
2964 Add support for Meson as alternative/parallel build system
2965 https://github.com/mesonbuild/meson
2967 2016-08-26 21:56:13 +0200 Josep Torra <n770galaxy@gmail.com>
2970 * tests/check/Makefile.am:
2971 build: silence error about pthread for 'make check' in osx
2972 Fixes "clang: error: argument unused during compilation: '-pthread'"
2974 2015-09-25 15:04:00 +0000 Nikita Bobkov <NikitaDBobkov@gmail.com>
2976 * gst/rtsp-server/rtsp-client.c:
2977 rtsp-client: Fix leaking of media in error cases
2978 With additional fixes by Kseniya Vasilchuk <vasilchukkseniia@gmail.com>
2979 and myself to make the media refcounting a bit easier to follow.
2980 https://bugzilla.gnome.org/show_bug.cgi?id=755632
2982 2016-08-02 15:08:22 +0300 Sebastian Dröge <sebastian@centricular.com>
2984 * gst/rtsp-server/rtsp-client.c:
2985 rtsp-client: Fix leaking of session in error cases
2986 https://bugzilla.gnome.org/show_bug.cgi?id=755632
2988 2016-07-11 21:16:04 +0200 Stefan Sauer <ensonic@users.sf.net>
2991 Automatic update of common submodule
2992 From f363b32 to f49c55e
2994 2016-07-06 13:51:15 +0300 Sebastian Dröge <sebastian@centricular.com>
2999 === release 1.9.1 ===
3001 2016-07-06 13:28:12 +0300 Sebastian Dröge <sebastian@centricular.com>
3007 * gst-rtsp-server.doap:
3010 2016-06-24 02:02:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
3013 configure: Need to add -DGST_STATIC_COMPILATION when building only statically
3014 https://bugzilla.gnome.org/show_bug.cgi?id=767463
3016 2016-06-21 11:49:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
3019 Automatic update of common submodule
3020 From ac2f647 to f363b32
3022 2016-04-14 22:56:11 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
3024 * gst/rtsp-server/rtsp-sdp.c:
3025 * gst/rtsp-server/rtsp-sdp.h:
3026 * gst/rtsp-server/rtsp-stream.c:
3027 * gst/rtsp-server/rtsp-stream.h:
3028 sdp: add rollover counters for all sender SSRC
3029 We add different crypto sessions in MIKEY, one for each sender
3030 SSRC. Currently, all of them will have the same security policy, 0.
3031 The rollover counters are obtained from the srtpenc element using the
3033 https://bugzilla.gnome.org/show_bug.cgi?id=730539
3035 2016-06-07 20:44:42 +0100 Tim-Philipp Müller <tim@centricular.com>
3037 * gst/rtsp-server/rtsp-media-factory.h:
3038 * gst/rtsp-server/rtsp-server.h:
3039 docs: fix some typos
3041 2016-05-25 10:28:43 +0100 Tim-Philipp Müller <tim@centricular.com>
3043 * gst/rtsp-server/Makefile.am:
3044 g-i: pass compiler env to g-ir-scanner
3045 It's what introspection.mak does as well. Should
3046 fix spurious build failures on gnome-continuous
3047 (caused by g-ir-scanner getting compiler details
3048 via python which is broken in some environments
3049 so passing the compiler details bypasses that).
3051 2016-05-18 16:48:44 +0100 Ian <ian.arkver.dev@gmail.com>
3053 * gst/rtsp-server/rtsp-session.c:
3054 rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header
3055 This works with rtspsrc and live555, but fails with e.g. ffmpeg.
3056 https://bugzilla.gnome.org/show_bug.cgi?id=766619
3058 2016-03-07 14:48:38 +0100 Edward Hervey <bilboed@bilboed.com>
3060 * gst/rtsp-sink/gstrtspclientsink.c:
3061 rtspclientsink: Check return value of sscanf
3062 And just make sure we always have 0/0 if we have an error
3065 2016-04-25 08:55:25 -0400 Jake Foytik <jake.foytik@ipconfigure.com>
3067 * gst/rtsp-server/rtsp-stream.c:
3068 * tests/check/gst/rtspserver.c:
3069 * tests/check/gst/stream.c:
3070 rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
3071 - Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
3072 - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
3073 - Create unit test for shared media.
3074 https://bugzilla.gnome.org/show_bug.cgi?id=764744
3076 2016-04-11 10:55:23 +0300 Sebastian Dröge <sebastian@centricular.com>
3078 * gst/rtsp-server/rtsp-stream.c:
3079 rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
3080 For IPv6 addresses, binding to a multicast group does not work on Linux
3081 either. Always bind to ANY and then later join the multicast group.
3082 https://bugzilla.gnome.org/show_bug.cgi?id=764679
3084 2016-04-14 10:05:02 +0100 Julien Isorce <j.isorce@samsung.com>
3087 Automatic update of common submodule
3088 From 6f2d209 to ac2f647
3090 2016-04-06 10:09:46 +0200 Patricia Muscalu <patricia@axis.com>
3092 * gst/rtsp-server/rtsp-thread-pool.c:
3093 rtsp-thread-pool: explained why GSource is a part of ThreadImpl
3094 Clarified why it is necessary to add source information to
3095 GstRTSPThreadImpl. See the reported bug in GLib:
3096 https://bugzilla.gnome.org/show_bug.cgi?id=720186
3097 for more information.
3098 https://bugzilla.gnome.org/show_bug.cgi?id=761702
3100 2016-04-04 12:58:38 +0300 Sebastian Dröge <sebastian@centricular.com>
3102 * examples/Makefile.am:
3103 examples: Clean up CFLAGS/LDADD even more
3104 The internal .la should come first and is part of LDADD, as is
3107 2016-04-04 12:39:39 +0300 Sebastian Dröge <sebastian@centricular.com>
3109 * examples/Makefile.am:
3110 examples: Clean up CFLAGS/LDADD to link with the correct versions of all libraries
3112 2016-04-03 12:06:29 +0300 Sebastian Dröge <sebastian@centricular.com>
3114 * gst/rtsp-server/Makefile.am:
3115 rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS)
3117 2015-12-30 18:39:05 +0200 Sebastian Dröge <sebastian@centricular.com>
3119 * gst/rtsp-server/rtsp-client.c:
3120 * gst/rtsp-server/rtsp-media-factory.c:
3121 * gst/rtsp-server/rtsp-media-factory.h:
3122 * gst/rtsp-server/rtsp-media.c:
3123 * gst/rtsp-server/rtsp-media.h:
3124 * gst/rtsp-server/rtsp-sdp.c:
3125 * gst/rtsp-server/rtsp-stream.c:
3126 * gst/rtsp-server/rtsp-stream.h:
3127 rtsp-server: Implement clock signalling according to RFC7273
3128 For NTP and PTP clocks we signal the actual clock that is used and signal
3129 the direct media clock offset.
3130 For all other clocks we at least signal that it's the local sender clock.
3131 This allows receivers to know which clock was used to generate the media and
3132 its RTP timestamps. Receivers can then implement network synchronization,
3133 either absolute or at least relative by getting the sender clock rate directly
3134 via NTP/PTP instead of estimating it from RTP timestamps and packet receive
3136 https://bugzilla.gnome.org/show_bug.cgi?id=760005
3138 2016-03-02 19:42:58 +0200 Sebastian Dröge <sebastian@centricular.com>
3140 * gst/rtsp-sink/gstrtspclientsink.c:
3141 rtspclientsink: Add support for setting the multicast interface
3142 https://bugzilla.gnome.org/show_bug.cgi?id=763000
3144 2016-03-02 19:42:13 +0200 Sebastian Dröge <sebastian@centricular.com>
3146 * gst/rtsp-server/rtsp-media-factory.c:
3147 * gst/rtsp-server/rtsp-media-factory.h:
3148 * gst/rtsp-server/rtsp-media.c:
3149 * gst/rtsp-server/rtsp-media.h:
3150 * gst/rtsp-server/rtsp-stream.c:
3151 * gst/rtsp-server/rtsp-stream.h:
3152 rtsp-media: Add support for setting the multicast interface
3153 https://bugzilla.gnome.org/show_bug.cgi?id=763000
3155 2016-03-07 08:50:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
3157 * gst/rtsp-sink/gstrtspclientsink.c:
3158 rtspclientsink: use new gst_element_class_add_static_pad_template()
3159 https://bugzilla.gnome.org/show_bug.cgi?id=763196
3161 2016-03-24 13:33:43 +0200 Sebastian Dröge <sebastian@centricular.com>
3166 === release 1.8.0 ===
3168 2016-03-24 13:00:35 +0200 Sebastian Dröge <sebastian@centricular.com>
3174 * gst-rtsp-server.doap:
3177 2016-03-16 23:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
3179 * gst/rtsp-server/rtsp-stream.c:
3180 rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
3181 This would get us NO_PREROLL in the bin again and break seeking.
3182 Thanks to Carlos Rafael Giani for helping to debug this!
3183 https://bugzilla.gnome.org/show_bug.cgi?id=740509
3185 === release 1.7.91 ===
3187 2016-03-15 12:26:13 +0200 Sebastian Dröge <sebastian@centricular.com>
3193 * gst-rtsp-server.doap:
3196 2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
3198 * gst/rtsp-server/rtsp-stream.c:
3199 rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
3200 Without this, RECORD pipelines are broken because
3201 a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
3202 added later. Previously it was there earlier and due to NO_PREROLL caused the
3203 pipeline to preroll immediately
3204 b) the udpsrc for the pipeline is added later and never set to PLAYING state,
3205 as the corresponding code previously was only for PLAY pipelines.
3206 https://bugzilla.gnome.org/show_bug.cgi?id=763281
3208 2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
3210 * gst/rtsp-server/rtsp-stream.c:
3211 rtsp-stream: Fix typo in the docstring
3212 gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
3214 2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
3216 * gst/rtsp-server/rtsp-stream.c:
3217 rtsp-stream: Disable multicast loopback for all our sockets
3218 On Windows this is a receiver-side setting, on Linux a sender-side setting. As
3219 we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
3220 loopback setting on the socket... while udpsink does which unfortunately has
3221 no effect here on Windows but on Linux.
3222 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3224 2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
3226 * tests/check/gst/stream.c:
3227 stream tests: added new tests
3228 Test a case when the address pool only contains multicast addresses
3229 and the client is requesting unicast udp.
3230 Added tests for multicast ports allocation.
3231 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3233 2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
3235 * gst/rtsp-server/rtsp-stream.c:
3236 rtsp-stream: Only bind multicast sockets to ANY on Windows
3237 On Linux it is still needed to bind to the multicast address
3238 to filter out random other packets, while on Windows binding
3239 to multicast addresses just fails.
3241 2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
3243 * gst/rtsp-server/rtsp-stream.c:
3244 rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
3245 Otherwise we fail to allocate UDP ports if the pool only contains multicast
3246 addresses, which is something that used to work before. For unicast addresses
3247 if the pool contains none, we just allocate them as if there is no pool at
3249 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3251 2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
3253 * gst/rtsp-server/rtsp-client.c:
3254 * gst/rtsp-server/rtsp-stream.c:
3255 rtsp-server: Fix indentation
3257 2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
3259 * gst/rtsp-server/rtsp-stream.c:
3260 rtsp-stream: Don't bind the sockets to multicast addresses
3261 This works on Linux but fails completely on Windows. You're supposed
3262 to bind to ANY and then join the multicast group.
3263 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3265 === release 1.7.90 ===
3267 2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
3273 * gst-rtsp-server.doap:
3276 2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
3279 Automatic update of common submodule
3280 From b64f03f to 6f2d209
3282 2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
3284 * gst/rtsp-sink/gstrtspclientsink.c:
3285 * tests/check/gst/rtspclientsink.c:
3286 rtspsink: Fix some leaks in rtspclientsink and the unit test.
3287 https://bugzilla.gnome.org/show_bug.cgi?id=762525
3289 2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
3291 * tests/check/gst/media.c:
3292 * tests/check/gst/rtspclientsink.c:
3293 * tests/check/gst/rtspserver.c:
3294 * tests/check/gst/stream.c:
3295 tests: unit test fixes
3296 Removed port allocation test from the media suite.
3297 The port allocation failure is now in the stream suite.
3299 Make sure that the media is suspended after the DESCRIBE request
3300 before reconfiguring the UDP sinks.
3302 In the RECORD case we have to set async property to false
3303 for the appsink element in the test in order to make sure
3304 that the media pipeline doesn't hang in start_preroll().
3305 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3307 2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
3309 * gst/rtsp-server/rtsp-client.c:
3310 * gst/rtsp-server/rtsp-stream.c:
3311 * gst/rtsp-server/rtsp-stream.h:
3312 rtsp-stream: postpone UDP socket allocation until SETUP
3313 Postpone the allocation of the UDP sockets until we know
3314 what transport has been chosen by the client.
3315 Both unicast and multicast UDP sources are created in one
3317 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3319 2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
3321 * gst/rtsp-server/rtsp-stream.c:
3322 rtsp-stream: postpone the creation of the UDP sources
3323 Code refactoring: allocate the UDP ports after the sender and
3324 the reciver parts have been created.
3325 We postpone the creation of the UDP sources until the UDP
3326 ports have been allocated.
3327 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3329 2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
3331 * gst/rtsp-server/rtsp-stream.c:
3332 rtsp-stream: added function for setting UDP sources to PLAYING state
3333 Code refactoring: Introduced a function for setting UDP sources
3335 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3337 2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
3339 * gst/rtsp-server/rtsp-stream.c:
3340 rtsp-stream: added function for creating and configuring UDP sources
3341 Code refactoring: create and configure UDP sources in a separate function.
3342 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3344 2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
3346 * gst/rtsp-server/rtsp-stream.c:
3347 rtsp-stream: added function for RTP/RTCP socket configuration
3348 Code refactoring: configure RTP and RTCP sockets for UDP sinks
3349 in a separate function.
3350 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3352 2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
3354 * gst/rtsp-server/rtsp-stream.c:
3355 rtsp-stream: added function for creating and configuring UDP sinks
3356 Code refactoring: create and configure UDP sinks in a separate function.
3357 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3359 2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
3361 * gst/rtsp-server/rtsp-stream.c:
3362 rtsp-stream: added helper function for creating the sender/receiver parts
3363 Code refactoring: introduced helper function for creating
3364 the receiver and the sender parts of the streaming pipeline.
3365 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3367 2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
3372 === release 1.7.2 ===
3374 2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
3380 * gst-rtsp-server.doap:
3383 2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
3385 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
3386 uninstalled.pc: add support for non libtool build systems
3387 Currently the .la path is provided which requires to use libtool as
3388 mentioned in the GStreamer manual section-helloworld-compilerun.html.
3389 It is fine as long as the application is built using libtool.
3390 So currently it is not possible to compile a GStreamer application
3391 within gst-uninstalled with CMake or other build system different
3393 This patch allows to do the following in gst-uninstalled env:
3394 gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
3395 gstreamer-rtsp-server-1.0)
3396 Previously it required to prepend libtool --mode=link
3397 https://bugzilla.gnome.org/show_bug.cgi?id=720778
3399 2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
3401 * gst/rtsp-sink/gstrtspclientsink.c:
3402 rtspclientsink: remove check for impossible condition
3403 Goto error label checks stream to see if it needs to be unreferenced before
3404 returning, but this goto jumps happens before the stream is ever set, so it
3405 will always be NULL in this error label.
3408 2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
3410 * gst/rtsp-sink/gstrtspclientsink.c:
3411 rtspclientsink: clean switch statements
3412 Coverity demands for fallthrough statements to be clearly commented,
3413 to distinguish from accidental fall throughs. And it also needs all
3414 cases to finish with a break, even if the break is never going to be
3415 executed like in the case of a continue jump.
3419 2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
3421 * tests/check/Makefile.am:
3422 tests: extend the AM_TESTS_ENVIRONMENT from check.mak
3423 To get the CK_DEFAULT_TIMEOUT defined for all tests
3424 Also removes a 120 seconds timeout that was set as default
3425 explicitly in this module
3426 https://bugzilla.gnome.org/show_bug.cgi?id=761472
3428 2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
3432 Automatic update of common submodule
3433 From 86e4663 to b64f03f
3435 2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
3437 * gst/rtsp-server/rtsp-media.c:
3438 rtsp-media: fix state_lock not locked again when preroll fails
3439 https://bugzilla.gnome.org/show_bug.cgi?id=761399
3441 2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
3444 configure: Move plugin specific flags below all the others
3445 They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
3446 -no-undefined. And -no-undefined is required on Windows to build DLLs.
3448 2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
3450 * gst/rtsp-sink/gstrtspclientsink.c:
3451 rtspclientsink: Simplify slightly using new -base API
3452 Use the new Mikey and SDP API in the base plugins libs
3453 to simplify some code.
3454 https://bugzilla.gnome.org/show_bug.cgi?id=758180
3456 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3461 * gst/rtsp-sink/Makefile.am:
3462 * gst/rtsp-sink/gstrtspclientsink.c:
3463 * gst/rtsp-sink/gstrtspclientsink.h:
3464 * gst/rtsp-sink/plugin.c:
3465 * tests/check/Makefile.am:
3466 * tests/check/gst/rtspclientsink.c:
3467 rtspsink: Add rtspclientsink element
3468 Add an rtspclientsink element that accepts streams for which
3469 there is a registered payloader and sends them to
3470 an RTSP server using RECORD.
3471 Sending is synchronised to the pipeline clock. Payload-types
3472 are automatically selected. The 'new-payloader' signal is fired
3473 for custom configuration of payloaders when they are created.
3474 Can now stream a movie like this:
3476 ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
3477 decodebin name=depay1 ! audioconvert ! autoaudiosink )"
3479 gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
3480 queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
3481 https://bugzilla.gnome.org/show_bug.cgi?id=758180
3483 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3485 * gst/rtsp-server/rtsp-stream.c:
3486 * gst/rtsp-server/rtsp-stream.h:
3487 rtsp-stream: Add functions for using rtsp-stream from the client
3488 Add a boolean to indicate that the rtsp-stream is running on the
3489 'client' side of an RTSP connection, for sending streams via
3490 RECORD. In that case, the roles of the client/server ports
3491 in transport setup are swapped.
3492 https://bugzilla.gnome.org/show_bug.cgi?id=758180
3494 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3496 * gst/rtsp-server/rtsp-sdp.c:
3497 * gst/rtsp-server/rtsp-sdp.h:
3498 rtsp-sdp: Add gst_rtsp_sdp_from_stream()
3499 A new function that adds info from a GstRTSPStream into an SDP message.
3500 https://bugzilla.gnome.org/show_bug.cgi?id=758180
3502 2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
3504 * gst/rtsp-server/rtsp-media.c:
3505 rtsp-media: Fix mutex beeing unlocked while they should be locked
3506 https://bugzilla.gnome.org/show_bug.cgi?id=761226
3508 2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
3510 * gst/rtsp-server/rtsp-media-factory.c:
3511 rtsp-media-factory: add missing break in "clock" property setter
3514 2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
3516 * gst/rtsp-server/rtsp-stream.c:
3517 rtsp-stream: fixed assert during update transport
3518 When RTSP server trying update transport during multicast, it throws an
3519 assert. The assert is thrown because it is trying to get the parent of
3520 an non-existing funnel element.
3521 https://bugzilla.gnome.org/show_bug.cgi?id=760150
3523 2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
3525 * gst/rtsp-server/rtsp-permissions.h:
3526 * gst/rtsp-server/rtsp-thread-pool.h:
3527 * gst/rtsp-server/rtsp-token.h:
3528 docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
3529 gtk-doc can handle static inline functions just fine these days,
3530 there's no need for this stuff any more.
3532 2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
3534 * gst/rtsp-server/rtsp-media.c:
3535 * gst/rtsp-server/rtsp-sdp.c:
3536 sdp: replace duplicated codes to call new base sdp apis
3537 https://bugzilla.gnome.org/show_bug.cgi?id=745880
3539 2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
3541 * examples/test-netclock.c:
3542 test-netclock: Use the new API to configure a clock directly
3544 2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
3546 * gst/rtsp-server/rtsp-media-factory.c:
3547 * gst/rtsp-server/rtsp-media-factory.h:
3548 * gst/rtsp-server/rtsp-media.c:
3549 * gst/rtsp-server/rtsp-media.h:
3550 rtsp-media: Add API to directly configure a clock on the media pipelines
3552 2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
3554 * gst/rtsp-server/rtsp-media.c:
3555 rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
3557 2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
3559 * gst/rtsp-server/rtsp-media-factory.c:
3560 rtsp-media-factory: Add FIXME for 2.0
3562 2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
3564 * gst/rtsp-server/rtsp-stream.c:
3565 rtsp-stream: Fix indentation
3567 2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3569 * gst/rtsp-server/rtsp-media.c:
3570 rtsp-media: Do not prepare media after media times out
3571 Deferred calls to start_prepare() can be deferred past the point until
3572 which wait_preroll() and by proxy gst_rtsp_media_get_status() is
3573 prepared to wait. Previously there was no lock and no check for this
3574 situation. This meant that a media could be prepared and unprepared
3575 simultaneously by two different threads. Now a lock is in place and a
3576 suitable check is done.
3577 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
3579 2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
3581 * gst/rtsp-server/rtsp-client.c:
3582 * gst/rtsp-server/rtsp-media-factory.c:
3583 * gst/rtsp-server/rtsp-media-factory.h:
3584 * gst/rtsp-server/rtsp-media.c:
3585 * gst/rtsp-server/rtsp-media.h:
3586 rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
3587 Without TEARDOWN it might be desireable to keep the media running and continue
3588 sending data to the client, even if the RTSP connection itself is
3590 Only do this for session medias that have only UDP transports. If there's at
3591 least on TCP transport, it will stop working and cause problems when the
3592 connection is disconnected.
3593 https://bugzilla.gnome.org/show_bug.cgi?id=758999
3595 2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
3600 === release 1.7.1 ===
3602 2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
3608 * gst-rtsp-server.doap:
3611 2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
3614 configure: Make -Bsymbolic check work with clang.
3615 Update the -Bsymbolic check with the version glib has. This version
3617 https://bugzilla.gnome.org/show_bug.cgi?id=759713
3619 2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
3621 * gst/rtsp-server/rtsp-session-pool.c:
3622 rtsp-session-pool: Avoid dollar sign ($) in session ids
3623 Live555 in VLC strips off dollar signs and then gets very confused,
3624 we don't loose too much entropy by just skipping it.
3626 2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
3628 * gst/rtsp-server/rtsp-address-pool.h:
3629 * gst/rtsp-server/rtsp-auth.h:
3630 * gst/rtsp-server/rtsp-client.h:
3631 * gst/rtsp-server/rtsp-media-factory-uri.h:
3632 * gst/rtsp-server/rtsp-media-factory.h:
3633 * gst/rtsp-server/rtsp-media.h:
3634 * gst/rtsp-server/rtsp-mount-points.h:
3635 * gst/rtsp-server/rtsp-permissions.h:
3636 * gst/rtsp-server/rtsp-server.h:
3637 * gst/rtsp-server/rtsp-session-media.h:
3638 * gst/rtsp-server/rtsp-session-pool.h:
3639 * gst/rtsp-server/rtsp-session.h:
3640 * gst/rtsp-server/rtsp-stream-transport.h:
3641 * gst/rtsp-server/rtsp-stream.h:
3642 * gst/rtsp-server/rtsp-thread-pool.h:
3643 * gst/rtsp-server/rtsp-token.h:
3644 rtsp-server: Add g_autoptr() support to all types
3645 https://bugzilla.gnome.org/show_bug.cgi?id=754464
3647 2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
3649 * gst/rtsp-server/rtsp-stream.c:
3650 rtsp-stream: fixed valgrind error
3651 Fixed the valgrind error in unit test. The UDP source created during
3652 gst_rtsp_stream_join_bin() was not released while destroying the rtp
3654 https://bugzilla.gnome.org/show_bug.cgi?id=759010
3656 2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
3660 Automatic update of common submodule
3661 From b319909 to 86e4663
3663 2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
3665 * gst/rtsp-server/rtsp-client.c:
3666 rtsp-client: suspend media during setup request
3667 SETUP request from clients needs to suspend the media to clear the
3668 prerolled buffers. Otherwise it will not affect the prerolled buffer
3669 and the prerolled buffers will be incorrect (for example block-size
3670 from setup request will not affect the prerolled buffer unless the
3671 media is suspended).
3672 https://bugzilla.gnome.org/show_bug.cgi?id=758268
3674 2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
3676 * gst/rtsp-server/rtsp-stream.c:
3677 rtsp-stream: create stream pipeline based on transport
3678 Based on the protocol, create the rtsp stream pipeline. If only TCP or
3679 only UDP is set as the transport protocol, it will not add the extra tee
3680 or queue element to the pipeline. Both these elements will be added, if
3681 it supports both TCP and UDP protocols. This improves the pipeline
3682 performance when one protocol is present.
3683 https://bugzilla.gnome.org/show_bug.cgi?id=758179
3685 2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
3687 * gst/rtsp-server/rtsp-stream.c:
3688 rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
3689 Adding them when not needed will start some logic inside rtpbin that might be
3690 problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
3691 would start up a rtpjitterbuffer and behave in weird ways.
3692 We still set up the UDP sources for RTP receiving for a sender media to be
3693 able to receive any packets sent by the client for NAT traversal. They will
3694 all go to a fakesink though.
3695 Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
3696 NO_PREROLL, which will cause deadlocks when seeking the media as it will never
3697 receive ASYNC_DONE after a seek.
3698 https://bugzilla.gnome.org/show_bug.cgi?id=758319
3700 2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
3702 * gst/rtsp-server/rtsp-stream.c:
3703 rtsp-stream: Disable multicast loopback for the multicast udp sources too
3704 On POSIX this setting is for sender sockets, on Windows for receiver sockets.
3705 Previously we were only setting this for sender sockets, which caused looped
3706 back packets to be received on Windows if a multicast transport was used.
3708 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3710 * examples/test-record-auth.c:
3711 * examples/test-record.c:
3712 examples: Actually use the provided port in the record examples
3714 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3716 * examples/test-record-auth.c:
3717 test-record-auth: Add the option to build in TLS support
3719 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3721 * examples/test-auth.c:
3722 test-auth: Use an 'anonymous' user for unauthenticated default
3723 There's a comment on one of the resources that 'user' and 'admin'
3724 shouldn't even be able to see it, but they can if the default
3725 token is 'admin2', since that gives them access anyway.
3727 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3729 * examples/.gitignore:
3730 * examples/Makefile.am:
3731 * examples/test-record-auth.c:
3732 Add test-record-auth example
3734 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3736 * gst/rtsp-server/rtsp-client.c:
3737 * tests/check/gst/client.c:
3738 rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
3740 2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
3742 * gst/rtsp-server/rtsp-server.c:
3743 rtsp-server: Change the logic so we don't pop a NULL context
3744 When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
3745 will sometimes fail. This call is made before any context is pushed
3746 resulting in an attempt to pop a NULL context.
3747 https://bugzilla.gnome.org/show_bug.cgi?id=757949
3749 2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
3751 * tests/check/gst/rtspserver.c:
3752 rtspserver: Add udp-mcast transport SETUP test
3753 Refactor utility functions in the test file so they can handle
3754 more than UDP and TCP as lower transport.
3755 https://bugzilla.gnome.org/show_bug.cgi?id=756969
3757 2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
3759 * gst/rtsp-server/rtsp-stream.c:
3760 rtsp-stream: Always unref return value of gst_object_get_parent()
3761 Fixes a leak of a GstBin in the udp-mcast case.
3762 https://bugzilla.gnome.org/show_bug.cgi?id=756968
3764 2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
3767 Automatic update of common submodule
3768 From b99800a to b319909
3770 2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
3773 Use new GST_ENABLE_EXTRA_CHECKS #define
3774 https://bugzilla.gnome.org/show_bug.cgi?id=756870
3776 2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
3779 Automatic update of common submodule
3780 From 6babecd to b99800a
3782 2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
3785 Update GLib dependency to 2.40.0
3787 2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
3789 * examples/test-mp4.c:
3790 * gst/rtsp-server/rtsp-stream.c:
3791 stream: listen to sender ssrc signals
3792 https://bugzilla.gnome.org/show_bug.cgi?id=746747
3794 2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
3797 common: update for new suppression
3798 Makes check-valgrind pass with glib 2.46
3800 2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3802 * gst/rtsp-server/rtsp-media.c:
3803 rtsp-media: Take reference to media that will be prepared
3804 default_prepare() takes a transfer-none reference GstRTSPMedia object.
3805 Later on a g_idle_source_new() is created and a pointer to the media
3806 object is passed as user data. If the media is freed before the idle
3807 source is dispatched the media object pointer is invalid, but the idle
3808 source callback expects it to still be valid. To fix this a reference to
3809 the media object is taken when registering the source callback function
3810 and a corresponding release of the reference is done when the souce is
3812 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
3814 2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
3816 * examples/test-launch.c:
3817 * examples/test-mp4.c:
3818 * examples/test-ogg.c:
3819 * examples/test-record.c:
3820 * examples/test-uri.c:
3821 rtsp-server: Fix memory leaks when context parse fails
3822 When g_option_context_parse fails, context and error variables are not getting free'd
3823 which results in memory leaks. Free'ing the same.
3824 And replacing g_error_free with g_clear_error, which checks if the error being passed
3825 is not NULL and sets the variable to NULL on free'ing.
3826 https://bugzilla.gnome.org/show_bug.cgi?id=753863
3828 2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
3833 === release 1.6.0 ===
3835 2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
3841 * gst-rtsp-server.doap:
3844 === release 1.5.91 ===
3846 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
3852 * gst-rtsp-server.doap:
3855 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
3857 * docs/libs/gst-rtsp-server-sections.txt:
3858 * gst/rtsp-server/rtsp-stream.c:
3859 stream: fix docs for recently-added get/set_buffer_size API
3860 https://bugzilla.gnome.org/show_bug.cgi?id=749095
3862 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
3864 * gst/rtsp-server/rtsp-media.c:
3865 rtsp-media: Don't crash on encrypted RTX SDP
3866 In parse_keymgmt(), don't mutate the input string that's been passed
3867 as const, especially since we might need the original value again if
3868 the same key info applies to multiple streams (RTX, for example).
3869 https://bugzilla.gnome.org/show_bug.cgi?id=754753
3871 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
3873 * examples/test-mp4.c:
3874 test-mp4: Support filenames with spaces in them. Error out on too few arguments
3876 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
3878 * examples/test-record.c:
3879 test-record: Check parameter count and print out help
3880 If no launch pipeline was supplied, print out some help
3882 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
3884 * gst/rtsp-server/rtsp-media.c:
3885 * gst/rtsp-server/rtsp-stream.c:
3886 * gst/rtsp-server/rtsp-stream.h:
3887 rtsp-stream: Implement UDP buffer size setting.
3888 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
3890 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
3891 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
3893 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
3895 * gst/rtsp-server/rtsp-media.h:
3896 rtsp-media: Fix small typo causing gtk-doc to complain
3898 === release 1.5.90 ===
3900 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
3906 * gst-rtsp-server.doap:
3909 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
3911 * gst/rtsp-server/rtsp-media-factory.c:
3912 media-factory: get port number through gst_rtsp_url_get_port
3913 https://bugzilla.gnome.org/show_bug.cgi?id=753473
3915 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
3917 * tests/check/gst/media.c:
3918 media-test: Removing unnecessary assertion
3919 https://bugzilla.gnome.org/show_bug.cgi?id=753385
3921 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
3923 * gst/rtsp-server/rtsp-server.c:
3924 Document that source keeps a ref on server until it's destroyed
3925 https://bugzilla.gnome.org/show_bug.cgi?id=749227
3927 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
3929 * tests/check/gst/media.c:
3930 media-test: Test for multiple dynamic payload
3931 https://bugzilla.gnome.org/show_bug.cgi?id=753385
3933 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
3935 * gst/rtsp-server/rtsp-media.c:
3936 media: Only add fakesink once per pipeline
3937 The intention is to prevent going PLAYING state before pads are created.
3938 If there was mutilple dynamic payload, it would leak few fakesink and
3939 actually prevent from ever reaching playing state.
3940 https://bugzilla.gnome.org/show_bug.cgi?id=753385
3942 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
3944 * gst/rtsp-server/rtsp-media.c:
3945 Revert "rtsp-media: Only add 1 fakesink per pipeline"
3946 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
3948 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
3950 * gst/rtsp-server/rtsp-media.c:
3951 rtsp-media: Only add 1 fakesink per pipeline
3952 There should be only one fakesink per pipeline, not per dynpay. This
3953 would lead to element naming clash.
3955 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
3957 * gst/rtsp-server/rtsp-media.c:
3958 rtsp-media: assertion error due to wrong condition check
3959 In media to caps function, reserved_keys array is being used for variable i,
3960 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
3961 changed it to variable j
3962 https://bugzilla.gnome.org/show_bug.cgi?id=753009
3964 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
3966 * gst/rtsp-server/rtsp-media.c:
3967 rtsp-media: Strip keys from the fmtp that we use internally in our caps
3968 Skip keys from the fmtp, which we already use ourselves for the
3969 caps. Some software is adding random things like clock-rate into
3970 the fmtp, and we would otherwise here set a string-typed clock-rate
3971 in the caps... and thus fail to create valid RTP caps
3972 https://bugzilla.gnome.org/show_bug.cgi?id=753009
3974 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
3976 * gst/rtsp-server/rtsp-thread-pool.c:
3977 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
3978 https://bugzilla.gnome.org/show_bug.cgi?id=752640
3980 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
3983 Automatic update of common submodule
3984 From f74b2df to 9aed1d7
3986 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
3991 === release 1.5.2 ===
3993 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
3999 * gst-rtsp-server.doap:
4002 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
4004 * gst/rtsp-server/rtsp-client.c:
4005 * gst/rtsp-server/rtsp-client.h:
4006 * tests/check/gst/client.c:
4007 rtsp-client: allow application to decide what requirements are supported
4008 Add "check-requirements" signal and vfunc to allow application
4009 (and subclasses) to check the requirements.
4010 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
4011 https://bugzilla.gnome.org/show_bug.cgi?id=749417
4013 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
4016 Automatic update of common submodule
4017 From 6015d26 to f74b2df
4019 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
4021 * gst/rtsp-server/rtsp-media.c:
4022 rtsp-media: Always use real payloader when creating streams
4023 A bin that contains the real payloader might be used as payloader. In this
4024 case we have to get the real payloader for the various properties it provides.
4025 Example use cases for this are bins that payload some media and then have
4026 additional elements that add metadata or RTP extension headers to the stream.
4027 https://bugzilla.gnome.org/show_bug.cgi?id=750800
4029 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
4031 * examples/test-netclock-client.c:
4032 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
4034 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
4036 * examples/test-netclock-client.c:
4037 * examples/test-netclock.c:
4038 test-netclock: Use new ntp-time-source property on rtpbin
4039 Select the clock time to be used as NTP time source. This allows proper
4040 synchronization between receivers, independent of sharing base times, and just
4041 requires them to use the same clock.
4043 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
4045 * examples/test-netclock-client.c:
4046 * examples/test-netclock.c:
4047 test-netclock: Setting the same base time on sender and receiver is not necessary
4048 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
4050 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
4052 * gst/rtsp-server/rtsp-stream.c:
4053 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
4054 https://bugzilla.gnome.org/show_bug.cgi?id=750764
4056 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
4058 * docs/libs/gst-rtsp-server.types:
4059 docs: add missing types
4060 https://bugzilla.gnome.org/show_bug.cgi?id=750764
4062 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
4064 * docs/libs/gst-rtsp-server-sections.txt:
4065 docs: add missing apis
4066 https://bugzilla.gnome.org/show_bug.cgi?id=750764
4068 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
4070 * examples/test-netclock-client.c:
4071 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
4073 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4075 * docs/libs/gst-rtsp-server-sections.txt:
4076 * gst/rtsp-server/rtsp-auth.c:
4077 * gst/rtsp-server/rtsp-auth.h:
4078 GstRTSPAuth: Add client certificate authentication support
4079 https://bugzilla.gnome.org/show_bug.cgi?id=750471
4081 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
4083 * examples/test-netclock-client.c:
4084 test-netclock-client: Use new GstClock API to wait for clock synchronization
4086 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
4088 * examples/test-netclock-client.c:
4089 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
4090 A mainloop is needed to get glimagesink to display something on OSX, and
4091 the source-setup signal just makes things a little bit easier.
4093 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
4096 Automatic update of common submodule
4097 From d9a3353 to 6015d26
4099 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
4102 Automatic update of common submodule
4103 From d37af32 to d9a3353
4105 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
4108 Automatic update of common submodule
4109 From 21ba2e5 to d37af32
4111 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
4114 Automatic update of common submodule
4115 From c408583 to 21ba2e5
4117 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
4119 * docs/libs/Makefile.am:
4120 docs: remove variables that we define in the snippet from common
4121 This is syncing our Makefile.am with upstream gtkdoc.
4123 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
4126 Automatic update of common submodule
4127 From 44a3517 to c408583
4129 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
4134 === release 1.5.1 ===
4136 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
4142 * gst-rtsp-server.doap:
4145 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
4147 * gst/rtsp-server/rtsp-client.c:
4148 rtsp-client: No flush during Teardown.
4149 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
4150 backlog is empty it can happen that just a part of a message will be
4151 sent and rest is in backlog queue. If then flush during teardown
4152 just a part of message will be sent.This can lead to client miss
4153 teardown response since it expect to get the last part of message.
4154 The flushing during teardown was introduced to fix a deadlock that now
4155 is fixed more generally in handle_request by temporary setting backlog
4157 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
4159 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
4161 * tests/check/Makefile.am:
4162 tests: Use AM_TESTS_ENVIRONMENT
4163 Needed by the new automake test runner and the
4164 current version of the common submodule.
4166 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
4168 * gst/rtsp-server/rtsp-media.h:
4169 * gst/rtsp-server/rtsp-stream.h:
4170 rtsp-server: Use single-include rtsp header to make sure we get all definitions
4172 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
4174 * gst/rtsp-server/rtsp-media.c:
4175 rtsp-media: Mark some more functions static
4177 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
4179 * gst/rtsp-server/rtsp-media.c:
4180 rtsp-media: Only unblock the media in suspend() when actually changing the state
4181 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
4183 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
4185 * examples/test-video-rtx.c:
4186 examples: Use AVPF profile for the RTX example
4188 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
4190 * gst/rtsp-server/rtsp-sdp.c:
4191 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
4193 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
4195 * gst/rtsp-server/rtsp-stream.c:
4196 rtsp-stream: get valid clock-rate from last-sample
4197 clock-rate in last-sample's caps is integer, not unsigned.
4198 To get this value properly, variable needs to be type-casted to int.
4199 https://bugzilla.gnome.org/show_bug.cgi?id=747614
4201 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
4205 autogen.sh: only run autopoint if gettext requested in configure.ac
4206 Not just because there happens to be a po directory.
4207 https://bugzilla.gnome.org/show_bug.cgi?id=748058
4209 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
4212 Revert "configure.ac: uncomment gettext version setup"
4213 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
4214 We don't need a gettext setup here and there's no po
4215 directory either, so no reason why autopoint would be
4216 run in the first place.
4217 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
4219 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
4221 * examples/test-multicast.c:
4222 * examples/test-multicast2.c:
4223 * examples/test-sdp.c:
4224 * examples/test-video-rtx.c:
4225 * examples/test-video.c:
4226 * tests/test-cleanup.c:
4227 * tests/test-reuse.c:
4228 Fix timeout function signatures across tests and examples
4230 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
4232 * tests/check/Makefile.am:
4233 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
4234 Make sure the test environment is set up.
4235 https://bugzilla.gnome.org//show_bug.cgi?id=747624
4237 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
4240 configure: bump automake requirement to 1.14 and autoconf to 2.69
4241 This is only required for builds from git, people can still
4242 build tarballs if they only have older autotools.
4243 https://bugzilla.gnome.org//show_bug.cgi?id=747624
4245 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
4248 configure.ac: uncomment gettext version setup
4249 Fixes autogen.sh. It would run autopoint, which would complain
4250 that it could not find the gettext version in configure.ac.
4251 https://bugzilla.gnome.org/show_bug.cgi?id=748058
4253 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
4255 * examples/test-video-rtx.c:
4256 test-video-rtx: set exact payload type to PCMA payloader
4257 Setting wrong payload type causes failure to do retransmission through audio stream
4258 https://bugzilla.gnome.org/show_bug.cgi?id=747839
4260 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
4262 * gst/rtsp-server/rtsp-media.c:
4263 * gst/rtsp-server/rtsp-stream.c:
4264 * gst/rtsp-server/rtsp-stream.h:
4265 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
4266 Because of duplicated g_signal_connect for request-aux-sender signal,
4267 wrong stream pointer is passed to the signal handler.
4268 Instead of passing each stream, pass stream array and get the relevant stream.
4269 https://bugzilla.gnome.org/show_bug.cgi?id=747839
4271 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
4275 Update autogen.sh to latest version from common
4276 Fixes build after aclocal_check etc. helpers have been removed.
4278 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
4281 Automatic update of common submodule
4282 From bc76a8b to c8fb372
4284 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
4286 * gst/rtsp-server/rtsp-stream.c:
4287 rtsp-stream: Limit the queues to 1 buffer
4288 We only need them to be able to pre-roll, queueing up more data here
4289 is only going to harm latency and memory usage.
4291 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
4293 * gst/rtsp-server/rtsp-stream.c:
4294 rtsp-stream: Update comment and ASCII art to the latest code
4295 We have a queue in front of the udpsink too to prevent the pipeline from
4298 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4300 * gst/rtsp-server/rtsp-stream.c:
4301 rtsp-media: Properly return first rtptime
4302 Instead we where returning first GstBuffer timestamp. This would result
4303 in clock skew and unwanted behaviour in RTSP playback.
4304 https://bugzilla.gnome.org/show_bug.cgi?id=746479
4306 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4308 * gst/rtsp-server/rtsp-stream.c:
4309 rtsp-stream: Don't leave buffer mapped
4310 If the seq is NULL, the RTP buffer was left mapped. We should always
4313 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
4318 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
4320 * gst/rtsp-server/rtsp-media-factory.c:
4321 * tests/check/gst/client.c:
4322 Fix double semicolons
4324 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
4326 * gst/rtsp-server/rtsp-stream.c:
4327 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
4328 This gives more accurate values than asking the payloader. There might be
4329 queueing happening between the payloader and the sink.
4330 https://bugzilla.gnome.org/show_bug.cgi?id=745704
4332 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
4334 * gst/rtsp-server/rtsp-media.c:
4335 rtsp-media: Don't seek for PLAY if the position will not change
4336 https://bugzilla.gnome.org/show_bug.cgi?id=745704
4338 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
4340 * gst/rtsp-server/rtsp-media.c:
4341 rtsp-media: Don't include payload type in the caps for framesize
4342 When the sdp media attribute framesize are converted to caps
4343 the <payload> should not be included.
4344 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
4345 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
4347 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
4349 * gst/rtsp-server/rtsp-sdp.c:
4350 rtsp-sdp: add payload type to the sdp framesize attribute
4351 The sdp framesize attribute is desribed in RFC6064. It is specified
4352 for payloading of H263 and has the following form
4353 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
4354 should be added to the caps in a payloader and the <payload type> should
4355 be added by the rtsp-server.
4356 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
4358 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
4360 * examples/test-uri.c:
4361 examples: test-uri: fix tainted variable
4362 Insignificant but this keeps Coverity happy.
4365 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
4367 * examples/.gitignore:
4368 * examples/Makefile.am:
4369 * examples/test-netclock-client.c:
4370 * examples/test-netclock.c:
4371 examples: Add a simple example of network synch for live streams.
4372 An example server and client that works for synchronising live streams
4373 only - as it can't support pause/play.
4375 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
4377 * gst/rtsp-server/rtsp-media-factory.c:
4378 * gst/rtsp-server/rtsp-media-factory.h:
4379 rtsp-media-factory: Add functions to set/get the media gtype
4380 Allow specifying the GType of a GstRtspMedia subclass to create
4381 as a simpler way to get the factory to create a custom
4382 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
4384 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
4386 * gst/rtsp-server/rtsp-media.c:
4387 rtsp-media: fix double unlock in _get_buffer_size()
4388 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
4389 because of double g_mutex_unlock () usage.
4390 https://bugzilla.gnome.org/show_bug.cgi?id=745434
4392 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
4394 * gst/rtsp-server/rtsp-session-pool.c:
4395 * gst/rtsp-server/rtsp-session.c:
4396 * gst/rtsp-server/rtsp-session.h:
4397 rtsp-session: Use monotonic time for RTSP session timeout
4398 Changed RTSP session timeout handling to monotonic time
4399 and deprecating the API for current system time.
4400 This fixes timeouts when the system time changes.
4401 https://bugzilla.gnome.org/show_bug.cgi?id=743346
4403 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
4405 * gst/rtsp-server/rtsp-client.c:
4406 * gst/rtsp-server/rtsp-media.c:
4407 rtsp-client: Only error out in PLAY if seeking actually failed
4408 If the media was just not seekable, we continue from whatever position we are
4409 and let the client decide if that is what is wanted or not.
4410 Only if the actual seek failed, we can't really recover and should error out.
4412 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
4414 * gst/rtsp-server/rtsp-stream.c:
4415 rtsp-stream: Add necessary queues between tee and multiudpsink
4416 https://bugzilla.gnome.org/show_bug.cgi?id=744379
4418 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
4420 * gst/rtsp-server/rtsp-client.c:
4421 * gst/rtsp-server/rtsp-media.c:
4422 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
4423 Instead error out properly the same way as if the SEEKING query already
4426 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
4428 * gst/rtsp-server/rtsp-stream.h:
4429 rtsp-stream: minor code formatting fix
4431 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
4433 * gst/rtsp-server/rtsp-media.c:
4434 rtsp-media: fix logic for collect_streams
4435 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
4436 all streams it knows if it got any, and can check if the transport mode is OK.
4439 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
4441 * gst/rtsp-server/rtsp-media.c:
4442 rtsp-media: Don't set the transport mode based on what elements we find
4443 Just print a warning if the one that was set before disagrees with what
4444 elements we found. It must already be set to something before as this
4445 function is called after we received the SDP from ANNOUNCE in RECORD mode,
4446 and we would reject ANNOUNCE if the RECORD flag was not set.
4448 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
4450 * tests/check/gst/rtspserver.c:
4451 tests: rtspserver: rename shadowed variable
4452 We have two different 'sink' variables here,
4453 rename one of them for clarity.
4455 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
4457 * gst/rtsp-server/rtsp-client.c:
4458 rtsp-client: fix awkward if clause
4460 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
4462 * examples/test-uri.c:
4463 examples: test-uri: improve uri argument handling and accept file names
4464 Print an error if the argument passed is not a URI and can't
4465 be converted into one, or no arguments have been provided.
4467 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
4469 * examples/test-uri.c:
4470 examples: test-uri: don't remove mount point after 10 seconds
4471 It's very irritating when trying to test stuff repeatedly
4472 and serves no real purpose other than showing that it can
4475 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
4477 * examples/.gitignore:
4478 examples: add new test-record to .gitignore
4480 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
4482 * examples/test-record.c:
4483 * gst/rtsp-server/rtsp-client.c:
4484 * gst/rtsp-server/rtsp-media-factory.c:
4485 * gst/rtsp-server/rtsp-media-factory.h:
4486 * gst/rtsp-server/rtsp-media.c:
4487 * gst/rtsp-server/rtsp-media.h:
4488 * tests/check/gst/rtspserver.c:
4489 rtsp-media: Use flags to distinguish between PLAY and RECORD media
4491 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
4493 * examples/test-record.c:
4494 test-record: Set latency for playback-style example to 2s instead of 200ms
4496 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
4498 * tests/check/gst/rtspserver.c:
4499 tests: add some unit tests for ANNOUNCE and RECORD
4500 https://bugzilla.gnome.org/show_bug.cgi?id=743175
4502 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
4504 * gst/rtsp-server/rtsp-client.c:
4505 rtsp-client: fix a couple of leaks in handle_announce
4507 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
4509 * gst/rtsp-server/rtsp-media-factory.c:
4510 * gst/rtsp-server/rtsp-media-factory.h:
4511 * gst/rtsp-server/rtsp-media.c:
4512 * gst/rtsp-server/rtsp-media.h:
4513 rtsp-media: Expose latency setting for setting the rtpbin latency
4515 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
4517 * examples/test-record.c:
4518 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
4520 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
4522 * gst/rtsp-server/rtsp-stream.c:
4523 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
4525 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
4527 * examples/Makefile.am:
4528 * examples/test-record.c:
4529 * gst/rtsp-server/rtsp-client.c:
4530 * gst/rtsp-server/rtsp-client.h:
4531 * gst/rtsp-server/rtsp-media-factory.c:
4532 * gst/rtsp-server/rtsp-media-factory.h:
4533 * gst/rtsp-server/rtsp-media.c:
4534 * gst/rtsp-server/rtsp-media.h:
4535 * gst/rtsp-server/rtsp-session-media.c:
4536 * gst/rtsp-server/rtsp-stream.c:
4537 * gst/rtsp-server/rtsp-stream.h:
4538 Add initial support for RECORD
4539 We currently only support media that is RECORD or PLAY only, not both at once.
4540 https://bugzilla.gnome.org/show_bug.cgi?id=743175
4542 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
4544 * gst/rtsp-server/rtsp-stream.c:
4545 rtsp-stream: RTCP and RTP transport cache cookies seperated
4546 RTCP packets were not sent because the same tr_cache_cookie was used for
4547 both RTP and RTCP. So only one of the tr_cache lists were populated
4548 depending on which one was sent first. If the tr_cache list is not
4549 populated then no packets can be sent. Most often this happened to be
4550 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
4551 resulted in both the tr_cache_lists to be populated regardless of which
4553 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
4555 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
4557 * gst/rtsp-server/rtsp-stream.c:
4558 rtsp-stream: fix false compiler warning
4559 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
4561 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
4563 * gst/rtsp-server/rtsp-client.c:
4564 rtsp-client: log interleaved data received
4566 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
4568 * gst/rtsp-server/rtsp-client.c:
4569 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
4571 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
4573 * gst/rtsp-server/rtsp-client.c:
4574 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
4576 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
4578 * gst/rtsp-server/rtsp-client.c:
4579 rtsp-client: Use a random session ID in the SDP
4580 RFC4566 Section 5.2 says that it should make the username, session id,
4581 nettype, addrtype and unicast address tuple globally unique. Always using
4582 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
4583 Instead let's create a 64 bit random number, which at least brings us
4584 closer to the goal of global uniqueness.
4585 https://tools.ietf.org/html/rfc4566#section-5.2
4587 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
4589 * examples/test-launch.c:
4590 * examples/test-mp4.c:
4591 * examples/test-ogg.c:
4592 * examples/test-uri.c:
4593 examples: Don't call gst_init() and gst_get_option_group()
4594 The latter calls the former at the appropriate time.
4596 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
4598 * gst/rtsp-server/rtsp-client.c:
4599 rtsp-client: Drop trailing \0 of RTSP DATA messages
4600 We add a trailing \0 in GstRTSPConnection to make parsing of
4601 string message bodies easier (e.g. the SDP from DESCRIBE) but
4602 for actual data this means we have to drop it or otherwise
4603 create invalid data.
4605 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
4607 * gst/rtsp-server/rtsp-stream.c:
4608 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
4609 Fixes crash when two threads access handle_new_sample() at the same
4610 time, one for RTP, one for RTCP.
4611 Otherwise, when iterating over the transports cache, it might be modified by
4612 another thread at the same time if the transports cookie has changed.
4613 https://bugzilla.gnome.org/show_bug.cgi?id=742954
4615 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
4617 * gst/rtsp-server/rtsp-stream.c:
4618 rtsp-stream: Set format=TIME on our app sources for TCP
4620 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
4622 * gst/rtsp-server/rtsp-session-pool.c:
4623 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
4624 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
4625 RFC 2326 states that session IDs may consist of alphanumeric as well as
4626 the safe characters $-_.+ -- N.B. the percent character is not allowed.
4627 Previously the session ID was URI-escaped, this meant that any character
4628 which was not alphanumeric or any of the characters +-._~ would be
4629 percent encoded. While the RFC (surprisingly) mentions that linear white
4630 space in session IDs should be URI-escaped, it does not say anything
4631 about other characters. Moreover no white space is allowed in the
4632 session ID. Finally the percent character which is the result of
4633 URI-escaping is not allowed in a session ID.
4634 So there is no reason to do any URI-escaping, and now it is removed.
4635 https://bugzilla.gnome.org/show_bug.cgi?id=742869
4637 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
4640 Automatic update of common submodule
4641 From f2c6b95 to bc76a8b
4643 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
4646 Fix 'make check' from top-level directory
4648 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
4650 * examples/test-launch.c:
4651 * examples/test-mp4.c:
4652 * examples/test-ogg.c:
4653 * examples/test-uri.c:
4654 examples: Add command-line parsing and take a 'port' argument
4655 This allows users to run multiple servers on different ports for testing.
4656 Only done for examples that actually take arguments and hence are capable of
4657 outputting different streams for each instance on each port.
4658 https://bugzilla.gnome.org/show_bug.cgi?id=742115
4660 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
4662 * gst/rtsp-server/rtsp-client.c:
4663 * gst/rtsp-server/rtsp-client.h:
4664 rtsp-client: Add a send_message default signal handler
4665 This allows subclasses to easily hook into the response sending
4666 mechanism without doing everything from a signal, which seems
4667 awkward from subclasses.
4669 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
4672 Automatic update of common submodule
4673 From ef1ffdc to f2c6b95
4675 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4679 configure: add --disable-examples switch
4680 https://bugzilla.gnome.org/show_bug.cgi?id=741678
4682 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
4684 * examples/.gitignore:
4685 * examples/Makefile.am:
4686 * examples/test-video-rtx.c:
4687 examples: add a retransmisison example implementing RFC4588
4688 Currently only SSRC-multiplexed rtx streams are supported
4690 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
4692 * gst/rtsp-server/rtsp-stream.c:
4693 rtsp-stream: Fix some minor memory leaks
4695 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
4697 * gst/rtsp-server/rtsp-media.c:
4698 rtsp-media: Some minor cleanup
4700 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
4702 * gst/rtsp-server/rtsp-stream.c:
4703 rtsp-stream: Fix compiler warnings
4704 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
4705 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
4707 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
4708 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
4711 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
4713 * docs/libs/gst-rtsp-server-sections.txt:
4714 * gst/rtsp-server/rtsp-media-factory.c:
4715 * gst/rtsp-server/rtsp-media-factory.h:
4716 * gst/rtsp-server/rtsp-media.c:
4717 * gst/rtsp-server/rtsp-media.h:
4718 * gst/rtsp-server/rtsp-sdp.c:
4719 * gst/rtsp-server/rtsp-stream.c:
4720 * gst/rtsp-server/rtsp-stream.h:
4721 media: implement ssrc-multiplexed retransmission support
4722 based off RFC 4588 and the server-rtpaux example in -good
4724 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
4726 * gst/rtsp-server/rtsp-client.c:
4727 * gst/rtsp-server/rtsp-stream-transport.c:
4728 * gst/rtsp-server/rtsp-stream.c:
4729 rtsp: Ref transports in hash table.
4730 Also ref streams for transports.
4731 This solves a crash when reciving a rtcp after teardown but before
4733 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
4735 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
4738 Automatic update of common submodule
4739 From 7bb2bce to ef1ffdc
4741 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
4743 * gst/rtsp-server/rtsp-client.c:
4744 client: refactor cleanup of cached media
4746 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
4748 * tests/check/gst/client.c:
4750 The session leak is now fixed, lets remove those FIXME comments.
4752 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
4754 * tests/check/gst/rtspserver.c:
4755 tests: Test to setup two sessions on one connection
4756 https://bugzilla.gnome.org/show_bug.cgi?id=739112
4758 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
4760 * tests/check/gst/rtspserver.c:
4761 tests: Test setup with tcp transport
4762 https://bugzilla.gnome.org/show_bug.cgi?id=739112
4764 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
4766 * gst/rtsp-server/rtsp-client.c:
4767 client: Configure transport after creating session media
4768 The default implementation of configure_client_transport() in
4769 rtsp-client uses the session media when it chooses channels for
4770 interleaved traffic.
4771 https://bugzilla.gnome.org/show_bug.cgi?id=739112
4773 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
4775 * gst/rtsp-server/rtsp-client.c:
4776 * gst/rtsp-server/rtsp-session-media.c:
4777 client: Stop caching media in client when doing setup
4778 If the media has been managed by a session media, it should not be
4779 cached in the client any longer. The GstRTSPSessionMedia object is now
4780 responsible for unpreparing the GstRTSPMedia object using
4781 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
4783 https://bugzilla.gnome.org/show_bug.cgi?id=739112
4785 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4787 * gst/rtsp-server/rtsp-stream.c:
4788 rtsp-stream: unref srtp decoder when leaving bin
4789 https://bugzilla.gnome.org/show_bug.cgi?id=739481
4791 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4793 * gst/rtsp-server/rtsp-client.c:
4794 rtsp-client: mikey memory leaks
4795 https://bugzilla.gnome.org/show_bug.cgi?id=739383
4797 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
4800 Automatic update of common submodule
4801 From 84d06cd to 7bb2bce
4803 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
4806 Parallelise 'make check-valgrind'
4808 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
4811 Automatic update of common submodule
4812 From a8c8939 to 84d06cd
4814 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
4817 Automatic update of common submodule
4818 From 36388a1 to a8c8939
4820 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
4822 * gst/rtsp-server/rtsp-media.c:
4823 rtsp-media: deactivate media when shutting down from paused
4824 This was only done when going directly from playing.
4825 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
4827 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4829 * gst/rtsp-server/rtsp-client.c:
4830 * gst/rtsp-server/rtsp-context.h:
4831 rtsp-client: add stream transport to context
4832 We add the stream transport to the context so we can get the configured
4833 client stream transport in the setup request signal.
4834 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
4836 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4838 * gst/rtsp-server/rtsp-stream.c:
4839 stream: release lock even not all transports have been removed
4840 We don't want to keep the lock even we return FALSE because not all the
4841 transports have been removed. This could lead into a deadlock.
4842 https://bugzilla.gnome.org/show_bug.cgi?id=737797
4844 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
4846 * gst/rtsp-server/rtsp-sdp.c:
4847 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
4848 These were renamed in GstRTPBasePayload in 1.0
4850 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4852 * gst/rtsp-server/rtsp-client.c:
4853 client: set session media to NULL without the lock
4854 We need to set session medias to NULL without the client lock otherwise
4855 we can end up in a deadlock if another thread is waiting for the lock
4856 and media unprepare is also waiting for that thread to end.
4857 https://bugzilla.gnome.org/show_bug.cgi?id=737690
4859 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
4861 * gst/rtsp-server/rtsp-media.c:
4862 rtsp-media: Set state to UNPREPARING in all cases
4864 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
4866 * gst/rtsp-server/rtsp-media.c:
4867 media: set state to unpreparing when unprepare is initiated
4868 https://bugzilla.gnome.org/show_bug.cgi?id=737675
4870 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
4872 * gst/rtsp-server/rtsp-client.c:
4873 rtsp-client: Remove backlog limit while processings requests
4874 If the backlog limit is kept two cases of deadlocks may be
4875 encountered when streaming over TCP. Without the backlog
4876 limit this deadlocks can not happen, at the expence of
4878 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
4880 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
4882 * gst/rtsp-server/rtsp-client.c:
4883 rtsp-client: do not free main context before rtsp watch
4884 https://bugzilla.gnome.org/show_bug.cgi?id=737110
4886 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
4888 * tests/check/gst/rtspserver.c:
4889 tests: Extend unit test timeout to accomodate for valgrind
4890 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
4892 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
4894 * gst/rtsp-server/rtsp-client.c:
4895 * gst/rtsp-server/rtsp-session.c:
4896 * gst/rtsp-server/rtsp-stream-transport.c:
4897 rtsp-*: Treat sending packets to clients as keepalive
4898 As long as gst-rtsp-server can successfully send RTP/RTCP data to
4899 clients then the client must be reading. This change makes the server
4900 timeout the connection if the client stops reading.
4901 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
4903 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
4905 * gst/rtsp-server/rtsp-client.c:
4906 rtsp-client: Allow backlog to grow while expiring session
4907 Allow the send backlog in the RTSP watch to grow to unlimited size while
4908 attempting to bring the media pipeline to NULL due to a session
4909 expiring. Without this change the appsink element cannot change state
4910 because it is blocked while rendering data in the new_sample callback.
4911 This callback will block until it has successfully put the data into the
4912 send backlog. There is a chance that the send backlog is full at this
4913 point which means that the callback may block for a long time, possibly
4914 forever. Therefore the media pipeline may also be prevented from
4915 changing state for a long time.
4916 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
4918 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
4920 * gst/rtsp-server/rtsp-client.c:
4921 rtsp-client: Make old compilers happy
4922 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
4923 Just in case that guint8 doesn't fit in a pointer. Just in case ...
4925 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
4927 * gst/rtsp-server/rtsp-client.c:
4928 client: raise the backlog limits before pausing
4929 We need to raise the backlog limits before pausing the pipeline or else
4930 the appsink might be blocking in the render method in wait_backlog() and
4931 we would deadlock waiting for paused.
4932 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
4934 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
4936 * gst/rtsp-server/rtsp-client.c:
4937 client: make define for the WATCH_BACKLOG
4938 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
4940 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
4942 * gst/rtsp-server/rtsp-client.c:
4943 client: simplify session transport handling
4944 link/unlink of the transport in a session was done to keep track of all
4945 TCP transports and to send RTP/RTCP data to the streams. We can simplify
4946 that by putting all the TCP transports in a hashtable indexed with the
4948 We also don't need to link/unlink the transports when we pause/resume
4949 the streams. The same effect is already achieved when we pause/play the
4950 media. Indeed, when we pause the media, the transport is removed from
4951 the media and the callbacks will not be called anymore.
4952 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
4954 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
4956 * gst/rtsp-server/rtsp-stream-transport.c:
4957 * gst/rtsp-server/rtsp-stream-transport.h:
4958 stream-transport: make method to handle received data
4959 Make a method to handle the data received on a channel. It sends the
4960 data to the stream of the transport on the RTP or RTCP pads based on
4963 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
4965 * examples/test-mp4.c:
4966 test: add example of dumping RTCP reports
4968 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
4970 * gst/rtsp-server/rtsp-media.c:
4971 * gst/rtsp-server/rtsp-stream.c:
4972 * gst/rtsp-server/rtsp-stream.h:
4973 rtsp-media: Make sure that sequence numbers are monotonic after pause
4974 The sequence number is not monotonic for RTP packets after pause. The
4975 reason is basepayloader generates a randon sequence number when the
4976 pipeline goes from ready to pause. With this fix generation of sequence
4977 number will be monotonic when going from pause to play request.
4978 https://bugzilla.gnome.org/show_bug.cgi?id=736017
4980 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
4982 * gst/rtsp-server/rtsp-client.c:
4983 rtsp-client: Protect saved clients watch with a mutex
4984 Fixes a crash when close() is called while merging clients
4985 in handle_tunnel(). In that case close() would destroy the
4986 watch while it is still being used in handle_tunnel().
4987 https://bugzilla.gnome.org/show_bug.cgi?id=735570
4989 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
4991 * gst/rtsp-server/rtsp-stream.c:
4992 rtsp-stream: Remove the multicast group udp sources when removing from the bin
4994 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
4996 * gst/rtsp-server/rtsp-media.c:
4997 * gst/rtsp-server/rtsp-stream.c:
4998 * gst/rtsp-server/rtsp-stream.h:
4999 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
5000 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
5001 seeking and will always continue counting the time. This leads to
5002 the NPT after a backwards seek to be something completely different
5003 to the actual seek position.
5004 https://bugzilla.gnome.org/show_bug.cgi?id=732644
5006 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
5008 * examples/test-appsrc.c:
5009 examples: fix another reference leak
5010 gst_rtsp_media_get_element() returns a new ref.
5012 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
5014 * examples/test-appsrc.c:
5015 examples: unref element after usage
5016 gst_bin_get_by_name_recurse_up() returns an element
5017 reference that must be unreffed after usage.
5018 https://bugzilla.gnome.org/show_bug.cgi?id=734546
5020 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
5022 * gst/rtsp-server/rtsp-media.c:
5023 signals: Fix copy-pasto in target-state signal offset
5025 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
5029 Makefile: Add usage of build-checks step
5030 Allows building checks without running them
5032 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
5034 * gst/rtsp-server/rtsp-stream.c:
5035 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
5036 When a UDP multicast transport is used it is expected that the server listens
5037 for RTP and RTCP packets on the multicast group with the corresponding port.
5038 Without this we will never get RTCP packets from clients in multicast mode.
5039 https://bugzilla.gnome.org/show_bug.cgi?id=732238
5041 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
5046 === release 1.4.0 ===
5048 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
5054 * gst-rtsp-server.doap:
5057 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
5059 * gst/rtsp-server/rtsp-media.h:
5060 media: correct misspelled words in description
5061 https://bugzilla.gnome.org/show_bug.cgi?id=733244
5063 === release 1.3.91 ===
5065 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
5071 * gst-rtsp-server.doap:
5074 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
5076 * docs/libs/gst-rtsp-server-sections.txt:
5079 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
5081 * gst/rtsp-server/rtsp-server.c:
5082 server: implement client REMOVE filter
5084 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
5086 * gst/rtsp-server/rtsp-client.c:
5087 * gst/rtsp-server/rtsp-client.h:
5088 client: expose _close() method
5089 Expose a previously internal close method to close the client
5092 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
5094 * gst/rtsp-server/rtsp-session-pool.c:
5095 session-pool: signal session-removed outside of the lock
5096 Release the lock before emiting the session-removed signal.
5098 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
5100 * gst/rtsp-server/rtsp-client.c:
5101 * gst/rtsp-server/rtsp-server.c:
5102 * gst/rtsp-server/rtsp-session-pool.c:
5103 * gst/rtsp-server/rtsp-session.c:
5104 * gst/rtsp-server/rtsp-stream.c:
5105 filter: Release lock in filter functions
5106 Release the object lock before calling the filter functions. We need to
5107 keep a cookie to detect when the list changed during the filter
5108 callback. We also keep a hashtable to make sure we only call the filter
5109 function once for each object in case of concurrent modification.
5110 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
5112 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
5114 * gst/rtsp-server/rtsp-client.c:
5115 client: check if watch is set in handle_teardown()
5116 The unit tests run without a watch
5118 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
5120 * tests/check/gst/client.c:
5121 client tests: send teardown to cleanup session
5123 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
5125 * tests/check/gst/rtspserver.c:
5126 server tests: send teardown to cleanup session
5128 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
5130 * gst/rtsp-server/rtsp-client.c:
5131 client: keep ref to client for the session removed handler
5132 This extra ref will be dropped when all client sessions have been
5133 removed. A session is removed when a client sends teardown, closes its
5134 endpoint of the TCP connection or the sessions expires.
5135 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
5137 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
5139 * gst/rtsp-server/rtsp-client.c:
5140 * gst/rtsp-server/rtsp-session.c:
5141 * tests/check/gst/client.c:
5142 client: manage media in session as a last step
5143 Once we manage a media in a session, we can't unmanage it anymore
5144 without destroying it. Therefore, first check everything before we
5145 manage the media, otherwise if something is wrong we have no way to
5147 If we created a new session and something went wrong, remove the session
5148 again. Fixes a leak in the unit test.
5150 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
5152 * examples/test-mp4.c:
5153 * examples/test-ogg.c:
5154 examples: print 'stream ready at url' for mp4 and ogg example
5156 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
5158 * gst/rtsp-server/rtsp-client.c:
5159 * gst/rtsp-server/rtsp-sdp.c:
5160 rtsp: fix for MIKEY api change
5162 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
5164 * gst/rtsp-server/rtsp-client.c:
5165 client: free watch context only once
5166 The watch context is freed when the source is destroyed. Avoids
5167 a CRITICAL when we try to unref the context twice.
5169 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
5171 * gst/rtsp-server/rtsp-client.c:
5174 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
5176 * gst/rtsp-server/rtsp-client.c:
5177 client: protect sessions with lock
5178 Protect the list of sessions with the lock.
5179 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
5181 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
5183 * gst/rtsp-server/rtsp-client.c:
5184 Client: keep a ref to the session
5185 Don't just keep a weak ref to the session objects but use a hard ref. We
5186 will be notified when a session is removed from the pool (expired) with
5187 the new session-removed signal.
5188 Don't automatically close the RTSP connection when all the sessions of
5189 a client are removed, a client can continue to operate and it can create
5190 a new session if it wants. If you want to remove the client from the
5191 server, you have to use gst_rtsp_server_client_filter() now.
5192 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
5193 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
5195 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
5197 * gst/rtsp-server/rtsp-session-pool.c:
5198 * gst/rtsp-server/rtsp-session-pool.h:
5199 session-pool: add session-removed signal
5200 Add a signal to be notified when a session is removed from the pool.
5202 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
5204 * gst/rtsp-server/Makefile.am:
5205 * gst/rtsp-server/rtsp-server.h:
5206 Make rtsp-server.h a single-include header, use it for G-I
5207 https://bugzilla.gnome.org/show_bug.cgi?id=732411
5209 === release 1.3.90 ===
5211 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
5217 * gst-rtsp-server.doap:
5220 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
5222 * gst/rtsp-server/rtsp-stream.c:
5223 stream: crypto can be NULL
5225 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
5227 * gst/rtsp-server/rtsp-client.c:
5228 * gst/rtsp-server/rtsp-media.c:
5229 * gst/rtsp-server/rtsp-mount-points.c:
5230 introspection: add missing allow-none annotations
5231 https://bugzilla.gnome.org/show_bug.cgi?id=730952
5233 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
5235 * gst/rtsp-server/rtsp-address-pool.c:
5236 * gst/rtsp-server/rtsp-media.c:
5237 * gst/rtsp-server/rtsp-session-media.c:
5238 * gst/rtsp-server/rtsp-session-pool.c:
5239 * gst/rtsp-server/rtsp-stream-transport.c:
5240 * gst/rtsp-server/rtsp-stream.c:
5241 * gst/rtsp-server/rtsp-token.c:
5242 introspection: add (nullable) annotations to return values
5243 https://bugzilla.gnome.org/show_bug.cgi?id=730952
5245 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
5247 * gst/rtsp-server/rtsp-client.c:
5248 * gst/rtsp-server/rtsp-stream.c:
5249 gi: improve annotations
5250 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
5252 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
5254 * gst/rtsp-server/rtsp-client.c:
5255 * gst/rtsp-server/rtsp-media-factory.c:
5256 * gst/rtsp-server/rtsp-media.c:
5257 * gst/rtsp-server/rtsp-server.c:
5258 signals: use generic marshal function
5259 Use the generic C marshal function.
5260 Use more explicit type instead of G_TYPE_POINTER
5262 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
5264 * gst/rtsp-server/rtsp-context.h:
5265 context: add type macro
5267 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
5269 * gst/rtsp-server/rtsp-client.c:
5270 * gst/rtsp-server/rtsp-sdp.c:
5271 * gst/rtsp-server/rtsp-sdp.h:
5272 sdp: hide key length defines
5273 They don't have a namespace.
5275 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
5280 === release 1.3.3 ===
5282 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
5288 * gst-rtsp-server.doap:
5291 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
5293 * gst/rtsp-server/rtsp-client.c:
5294 * gst/rtsp-server/rtsp-sdp.c:
5295 * gst/rtsp-server/rtsp-sdp.h:
5296 mikey: add different key length parameters
5297 Add encryption and authentication key length parameters to MIKEY. For
5298 the encoders, the key lengths are obtained from the cipher and auth
5299 algorithms set in the caps. For the decoders, they are obtained while
5300 parsing the key management from the client.
5301 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
5303 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
5305 * tests/check/gst/stream.c:
5306 stream tests: Make sure we get right multicast address from stream
5307 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
5309 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
5311 * gst/rtsp-server/rtsp-client.c:
5312 client: ref the context until rtsp watch is alive
5313 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
5315 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
5317 * gst/rtsp-server/rtsp-client.c:
5318 client: Destroy the rtsp watch after connection close
5320 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
5322 * gst/rtsp-server/rtsp-media.c:
5323 media: fix confusing comment
5325 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
5327 * gst/rtsp-server/rtsp-session.c:
5328 rtsp-session: Timeout in header.
5329 Adding the possbilty to always have timout in header.
5330 This is configurabe with setting "timeout-always-visible".
5331 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
5333 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
5338 === release 1.3.2 ===
5340 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
5347 * gst-rtsp-server.doap:
5350 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
5353 Automatic update of common submodule
5354 From 211fa5f to 1f5d3c3
5356 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
5358 * gst/rtsp-server/rtsp-client.c:
5359 client: store TCP ports in transport
5360 Store the TCP ports in the transport when we are doing RTSP over TCP.
5361 This way, we can easily get to the ports from the transport.
5362 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
5364 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
5366 * gst/rtsp-server/rtsp-stream.c:
5367 stream: add signals for new RTP/RTCP encoders
5368 New signals to allow the user to configure the dynamically created
5370 https://bugzilla.gnome.org/show_bug.cgi?id=730228
5372 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
5374 * gst/rtsp-server/rtsp-media.c:
5375 * gst/rtsp-server/rtsp-media.h:
5376 media: Make suspend()/unsuspend() virtual
5377 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
5379 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
5381 * gst/rtsp-server/rtsp-client.c:
5382 client: fix send-message signal marshaller
5383 Use generic marshalling for the send-message signal. It has
5384 two POINTER arguments, not just one.
5385 https://bugzilla.gnome.org/show_bug.cgi?id=729900
5387 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
5389 * tests/check/gst/media.c:
5390 tests: add and remove pads only once
5391 In this test we simulate a dynamic pad by watching the caps event.
5392 Because of renegotiation in the base payloader now, this caps is sent
5393 multiple times but we can only deal with 1 invocation, use a variable to
5394 only 'add and remove' the pad once.
5396 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
5398 * tests/check/gst/rtspserver.c:
5399 tests: add unit test for correct handling of Require headers
5400 https://bugzilla.gnome.org/show_bug.cgi?id=729426
5402 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
5404 * gst/rtsp-server/rtsp-client.c:
5405 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
5406 Servers must handle Require headers and must report a failure
5407 if they don't handle any of the Required options, see RFC 2326,
5408 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
5409 https://bugzilla.gnome.org/show_bug.cgi?id=729426
5411 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
5416 === release 1.3.1 ===
5418 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
5424 * gst-rtsp-server.doap:
5427 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
5430 Automatic update of common submodule
5431 From bcb1518 to 211fa5f
5433 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
5438 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
5440 * tests/check/gst/sessionmedia.c:
5441 tests: fix memory leak in sessionmedia unit test
5443 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
5445 * gst/rtsp-server/rtsp-client.c:
5446 client: emit a signal before sending a message
5447 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
5449 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
5451 * gst/rtsp-server/rtsp-client.c:
5452 client: pass context to send_message
5453 Pass the current context to send_message, we will need it later.
5455 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
5457 * gst/rtsp-server/rtsp-client.c:
5458 client: fix typo in comment
5460 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
5462 * gst/rtsp-server/rtsp-media.c:
5463 media: Do not stop thread twice if default_prepare() fails
5465 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
5467 * gst/rtsp-server/rtsp-client.c:
5468 client: set the watch to flushing before going to NULL
5469 First set the watch to flushing so that we unblock any current and
5470 future attempt to send data on the watch, Then set the pipeline to
5472 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
5474 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
5476 * gst/rtsp-server/rtsp-session-pool.c:
5477 * tests/check/gst/sessionpool.c:
5478 rtsp-session-pool: Fixes annotation
5479 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
5480 in the sessionpool test.
5481 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
5483 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
5485 * gst/rtsp-server/rtsp-media.c:
5486 * gst/rtsp-server/rtsp-media.h:
5487 media: make media_prepare virtual
5488 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
5490 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
5492 * gst/rtsp-server/rtsp-media.c:
5493 * tests/check/gst/media.c:
5494 media: stop the thread in more error cases
5496 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
5498 * gst/rtsp-server/rtsp-media.c:
5499 * tests/check/gst/media.c:
5500 media: allow NULL as the thread
5501 Use the default context whan passing a NULL thread.
5503 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
5505 * gst/rtsp-server/rtsp-client.c:
5506 rtsp-client: indent cleanup
5507 Coverity was moaning about unreachable code, and I think it was just
5508 confused by { being before the label. We'll see if it pops up again.
5511 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
5513 * gst/rtsp-server/rtsp-client.c:
5514 * gst/rtsp-server/rtsp-media.c:
5515 client: Add drop-backlog property
5516 When we have too many messages queued for a client (currently hardcoded
5517 to 100) we overflow and drop the messages. Add a drop-backlog property
5518 to control this behaviour. Setting this property to FALSE will retry
5519 to send the messages to the client by waiting for more room in the
5521 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
5523 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
5525 * gst/rtsp-server/rtsp-client.c:
5526 client: support for POST before GET when setting up a tunnel
5528 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
5530 * gst/rtsp-server/rtsp-client.c:
5531 client: remove watch of the second client after http tunnel setup
5532 The second client will be freed after the HTTP tunnel has been set up.
5533 Make sure it's RTSP watch is never dispatched again.
5534 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
5536 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
5538 * gst/rtsp-server/rtsp-media.c:
5539 * tests/check/gst/media.c:
5540 media: Make media_prepare() fail if port allocation fails
5541 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
5543 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
5545 * tests/check/gst/media.c:
5546 media test: cleanup the thread pool in tests
5548 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
5550 * gst/rtsp-server/rtsp-media.c:
5551 * tests/check/gst/media.c:
5552 rtsp-media: Unblock blocked streams in unprepare
5553 The streams will be blocked when a live media is prepared.
5554 The streams should be unblocked in gst_rtsp_media_unprepare.
5555 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
5557 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
5559 * gst/rtsp-server/rtsp-media.c:
5560 media: release the state lock when going to NULL
5561 Set our state to UNPREPARING and release the state-lock before
5562 setting the pipeline to the NULL state. This way, any pad-added
5563 callback will be able to take the state-lock and check that we are now
5564 unpreparing instead of deadlocking.
5565 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
5567 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
5569 * gst/rtsp-server/rtsp-media.c:
5570 media: protect status with lock
5571 Make sure we only update the status with the lock.
5573 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
5575 * gst/rtsp-server/rtsp-client.c:
5576 * gst/rtsp-server/rtsp-sdp.c:
5577 rtsp: update for MIKEY API changes
5579 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
5581 * gst/rtsp-server/rtsp-client.c:
5582 client: parse the mikey response from the client
5583 Parse the mikey response from the client and update the policy for
5586 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
5588 * gst/rtsp-server/rtsp-stream.c:
5589 * gst/rtsp-server/rtsp-stream.h:
5590 stream: add method to set crypto info
5591 Make a method to configure the crypto information of a stream.
5592 Set udpsrc in READY instead of PAUSED so that we can configure caps
5595 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
5597 * gst/rtsp-server/rtsp-client.c:
5598 client: cleanup error paths
5600 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
5602 * gst/rtsp-server/rtsp-media.c:
5605 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
5607 * examples/test-video.c:
5608 test: enable SRTP only on RTSPS
5609 We only want to enable SRTP when doing rtsp over TLS so that we can
5610 exchange the keys in a secure way.
5612 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
5614 * examples/test-video.c:
5615 test: print an error on failure
5617 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
5620 * examples/test-video.c:
5621 * gst/rtsp-server/rtsp-sdp.c:
5622 * gst/rtsp-server/rtsp-stream.c:
5623 * tests/check/Makefile.am:
5624 stream: add SRTP support
5625 Install srtp encoder and decoder elements in rtpbin
5628 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5630 * tests/check/Makefile.am:
5631 * tests/check/gst/sessionpool.c:
5632 tests: Add unit tests for sessionpool
5633 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
5635 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5637 * tests/check/gst/threadpool.c:
5638 tests: Improve code coverage of rtsp-threadpool tests
5639 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
5641 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5643 * tests/check/gst/sessionmedia.c:
5644 tests: Improve code coverage for rtsp-session-media
5645 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
5647 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5649 gobject-introspection: Add annotations to support language bindings
5650 In addition a few cosmetic changes:
5651 * Adjust the order of arguments
5652 * Fix typo: occured -> occurred
5653 * Fix indentation after Return:-clauses
5654 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
5656 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5658 * gst/rtsp-server/rtsp-stream.c:
5659 rtsp-stream: Don't mix IPv4 and IPv6 addresses
5660 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
5662 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
5664 * gst/rtsp-server/rtsp-stream.c:
5665 stream: take caps after the session manager
5666 Take the caps for the SDP after they leave the rtpbin so that we can
5667 also get the properties added by rtpbin elements.
5669 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
5671 * gst/rtsp-server/rtsp-stream.c:
5672 stream: release lock while pushing out packets
5673 Keep a cache of the transports and use this to iterate the transport
5674 while pushing packets. This allows us to release the lock early.
5675 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
5677 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
5679 * gst/rtsp-server/rtsp-client.c:
5680 * gst/rtsp-server/rtsp-client.h:
5681 rtsp-client: vmethod for modifying tunnel GET response
5682 Add a vmethod tunnel_http_response where the response to the HTTP GET
5683 for tunneled connections can be modified.
5684 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
5686 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
5688 * gst/rtsp-server/rtsp-sdp.c:
5689 sdp: make 1 media line per profile
5690 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
5691 line in the SDP for each profile. The client is then supposed to pick
5692 one of the profiles in the SETUP request. Because the m= lines have the
5693 same pt, the client also knows that only 1 option is possible.
5695 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
5697 * gst/rtsp-server/rtsp-media-factory.c:
5698 * gst/rtsp-server/rtsp-media-factory.h:
5699 * gst/rtsp-server/rtsp-media.c:
5700 factory: add profile property and pass to media and streams
5702 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
5704 * examples/test-multicast.c:
5705 * gst/rtsp-server/rtsp-sdp.c:
5706 sdp: pass multicast connection for multicast-only stream
5707 Pass the multicast address of the stream in the connection info in the
5708 SDP so that clients try a multicast connection first.
5709 Only allow multicast connections in the test-multicast example. Also
5710 increase the TTL a little.
5712 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5715 .gitignore: Ignore gcov intermediate files
5716 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
5718 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
5720 * gst/rtsp-server/rtsp-stream.c:
5721 stream: release some locks in error cases
5723 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5725 docs: Enable and fix gtk-doc warnings
5726 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
5727 * addresspool/mediafactory: Add missing annotation colon
5728 * stream: Annotate return value
5729 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
5731 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
5734 Automatic update of common submodule
5735 From fe1672e to bcb1518
5737 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
5740 Automatic update of common submodule
5741 From 1a07da9 to fe1672e
5743 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
5745 * examples/Makefile.am:
5746 examples: use LDADD for libs instead of LDFLAGS
5748 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
5751 configure: make sure releases are in .doap file
5753 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
5755 * examples/test-cgroups.c:
5756 examples: test-cgroups: don't put code with side effects into g_assert()
5757 The g_assert() might get compiled out with the right
5758 compiler/preprocessor flags.
5760 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
5762 * examples/.gitignore:
5763 examples: add cgroup test binary to .gitignore
5765 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
5767 * examples/test-cgroups.c:
5768 examples: fix cgroup test build
5769 Fixes build failure caused by compiler warning:
5770 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
5772 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
5775 .gitignore: ignore temp files created in the course of 'make check'
5777 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
5779 * gst/rtsp-server/rtsp-media.c:
5780 rtsp-media: don't loose frames handling new PLAY request
5781 If client supplied a range check if the range specifies the start point.
5782 If not, then do an accurate seek to the current position. If a start
5783 point was specified do do a key unit seek to make sure the streaming
5784 starts with decodeable frames.
5785 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
5787 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
5789 * gst/rtsp-server/rtsp-media.c:
5790 Revert "media: only flush when setting a new start position"
5791 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
5792 We need to do the flush in all cases, demuxer block currently for
5795 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
5797 * gst/rtsp-server/rtsp-media.c:
5798 media: only flush when setting a new start position
5799 Only flush the pipeline when we change the start position with
5801 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
5803 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
5805 * gst/rtsp-server/rtsp-stream.c:
5806 stream: set ttl-mc before adding the socket
5807 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
5808 never be set on socket.
5809 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
5811 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
5813 * gst/rtsp-server/rtsp-media.c:
5814 media: stop thread if media is already prepared
5815 in gst_rtsp_media_prepare() the thread is not used if media is already
5816 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
5818 https://bugzilla.gnome.org/show_bug.cgi?id=724182
5820 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
5823 build: Ship gst-rtsp-server.doap file
5825 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
5827 * tests/check/gst/rtspserver.c:
5828 tests: Fix another compiler warning with gcc
5830 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
5832 * gst/rtsp-server/rtsp-client.c:
5833 * gst/rtsp-server/rtsp-mount-points.c:
5834 * gst/rtsp-server/rtsp-stream.c:
5835 * tests/check/gst/client.c:
5836 rtsp-server: Fix lots of compiler warnings with clang
5838 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
5841 * gst-rtsp-server.doap:
5842 * tests/Makefile.am:
5843 configure: Synchronise with the configure scripts of the other modules
5845 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
5848 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
5850 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
5852 * gst/rtsp-server/rtsp-media.c:
5853 * gst/rtsp-server/rtsp-stream.c:
5854 Revert "rtsp-server: support build against last stable release"
5855 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
5856 Let us require 1.2.3 now, which is going to be released in a few
5859 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
5861 * gst/rtsp-server/rtsp-session-media.c:
5862 * gst/rtsp-server/rtsp-stream-transport.c:
5863 session: improve RTP-Info
5864 Ignore streams that can't generate RTP-Info instead of failing.
5865 Don't return the empty string when all streams are unconfigured but
5866 return NULL so that we don't generate and empty RTP-Info header.
5867 Improve docs a little.
5869 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
5871 * gst/rtsp-server/rtsp-session-media.c:
5872 Don't free rtpinfo GString when it is NULL
5873 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
5875 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
5877 * gst/rtsp-server/rtsp-media.c:
5878 media: only set keyframe flag when modifying start
5879 Only set the keyframe flag when we modify the start position. The
5880 keyframe flag should probably be ignored when no change is requested but
5881 until we can claim this is all documented properly and all demuxer
5882 implement this, avoid setting the flag.
5883 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
5885 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
5887 * gst/rtsp-server/rtsp-thread-pool.c:
5888 thread-pool: Unref source after mainloop has quit to avoid races in GLib
5889 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
5891 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
5893 * gst/rtsp-server/rtsp-stream.c:
5894 stream: handle NULL seqnum and rtptime arguments
5896 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
5898 * gst/rtsp-server/rtsp-thread-pool.c:
5899 * tests/check/gst/threadpool.c:
5900 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
5901 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
5903 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
5905 * gst/rtsp-server/rtsp-stream.c:
5906 stream: add fallback for missing stats property
5907 Use a fallback when the payloader does not have a stats property
5908 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
5910 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
5913 Automatic update of common submodule
5914 From f7bc1c3 to 1a07da9
5916 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
5918 * gst/rtsp-server/rtsp-stream.c:
5919 stream: don't leak stats structure
5920 Don't leak the stats structure and deal with NULL stats.
5922 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
5924 * gst/rtsp-server/rtsp-stream.c:
5925 stream: Get rtpinfo properties atomically from payloader
5926 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
5928 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
5930 * gst/rtsp-server/rtsp-media.c:
5931 media: refactor state change functions and signals
5932 Make functions to set the target state and the pipeline state and emit
5933 the signals from those functions.
5935 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
5937 * gst/rtsp-server/rtsp-media.c:
5938 * gst/rtsp-server/rtsp-media.h:
5939 media: add signal to notify of pending state changes
5941 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
5943 * gst/rtsp-server/rtsp-media.c:
5944 * gst/rtsp-server/rtsp-stream.c:
5945 rtsp-server: support build against last stable release
5946 Until 1.2.3 is out with the new get_type function and we
5949 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
5951 * gst/rtsp-server/rtsp-stream.c:
5952 stream: fix compilation
5954 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
5956 * gst/rtsp-server/rtsp-media.c:
5957 * gst/rtsp-server/rtsp-media.h:
5958 * gst/rtsp-server/rtsp-stream.c:
5959 * gst/rtsp-server/rtsp-stream.h:
5960 stream: add property to configure profiles
5962 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
5964 * gst/rtsp-server/rtsp-client.c:
5965 client: let stream check supported transport
5966 Delegate the check if a transport is allowed to the stream.
5967 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
5969 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
5971 * gst/rtsp-server/rtsp-stream.c:
5972 * gst/rtsp-server/rtsp-stream.h:
5973 stream: add method to check supported transport
5974 Add a method to check if a transport is supported
5976 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
5979 configure.ac: Only check for gstreamer-check, not check
5980 We include check in gstreamer-check since quite some time now.
5982 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
5984 * gst/rtsp-server/rtsp-session-media.c:
5985 * gst/rtsp-server/rtsp-stream-transport.c:
5986 * gst/rtsp-server/rtsp-stream.c:
5987 * gst/rtsp-server/rtsp-stream.h:
5988 stream: return clock-rate from get_rtpinfo
5989 And use it to correct the rtptime to the requested start-time.
5990 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
5992 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
5994 * gst/rtsp-server/rtsp-session-media.c:
5995 * gst/rtsp-server/rtsp-stream-transport.c:
5996 * gst/rtsp-server/rtsp-stream-transport.h:
5997 session-media: calculate start-time
5999 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
6001 * gst/rtsp-server/rtsp-stream-transport.c:
6002 * gst/rtsp-server/rtsp-stream.c:
6003 * gst/rtsp-server/rtsp-stream.h:
6004 stream: also return the running-time
6005 Return the running-time in the rtpinfo as well.
6007 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
6009 * gst/rtsp-server/rtsp-client.c:
6010 * gst/rtsp-server/rtsp-session-media.c:
6011 * gst/rtsp-server/rtsp-session-media.h:
6012 * gst/rtsp-server/rtsp-stream-transport.c:
6013 * gst/rtsp-server/rtsp-stream-transport.h:
6014 session-media: let the session-media make the RTPInfo
6015 Add method to create the RTPInfo for a stream-transport.
6016 Add method to create the RTPInfo for all stream-transports in a
6018 Use the session-media RTPInfo code in client. This allows us to refactor
6019 another method to link the TCP callbacks.
6021 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
6023 mount-points: sort sequence before g_sequence_lookup
6024 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
6025 sort sequence if dirty, otherwise lookup will fail.
6026 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
6028 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
6031 configure: rename package from gst-rtsp to gst-rtsp-server
6032 To match git module name and avoid confusion with the
6033 rtsp lib in gst-plugins-base and rtsp plugin in -good.
6035 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
6038 configure: bump core/base/good requirement to 1.2.0
6039 Bump to released stable version and make implicit
6040 requirements explicit.
6042 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
6047 Fix broken gettext setup which is not used anyway
6049 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
6052 Automatic update of common submodule
6053 From dbedaa0 to d48bed3
6055 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
6057 * gst/rtsp-server/rtsp-client.c:
6058 * gst/rtsp-server/rtsp-media.c:
6059 * gst/rtsp-server/rtsp-media.h:
6060 media: add setup_sdp vmethod
6061 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
6062 gst_rtsp_media_setup_sdp.
6063 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
6065 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
6067 * gst/rtsp-server/rtsp-stream.c:
6068 rtsp-stream: Check return value of sscanf
6069 streamid is only valid if sscanf matched something.
6071 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
6073 * gst/rtsp-server/rtsp-client.c:
6074 rtsp-client: Fix iteration
6075 Wouldn't even enter the code block otherwise (i++ was used as the check
6076 and not the postfix).
6078 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
6080 * gst/rtsp-server/rtsp-client.c:
6081 * gst/rtsp-server/rtsp-client.h:
6082 client: add vmethod to configure media and streams
6083 Implement a vmethod that can be used to configure the media and the
6084 streams based on the current context. Handle the blocksize handling in
6085 the default handler.
6086 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
6088 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
6091 Make git ignore more unit test binaries
6093 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
6095 * gst/rtsp-server/rtsp-address-pool.h:
6096 * gst/rtsp-server/rtsp-auth.h:
6097 * gst/rtsp-server/rtsp-client.h:
6098 * gst/rtsp-server/rtsp-context.h:
6099 * gst/rtsp-server/rtsp-media-factory-uri.h:
6100 * gst/rtsp-server/rtsp-media-factory.h:
6101 * gst/rtsp-server/rtsp-media.h:
6102 * gst/rtsp-server/rtsp-mount-points.h:
6103 * gst/rtsp-server/rtsp-server.h:
6104 * gst/rtsp-server/rtsp-session-media.h:
6105 * gst/rtsp-server/rtsp-session-pool.h:
6106 * gst/rtsp-server/rtsp-session.h:
6107 * gst/rtsp-server/rtsp-stream-transport.h:
6108 * gst/rtsp-server/rtsp-stream.h:
6109 * gst/rtsp-server/rtsp-thread-pool.h:
6110 * gst/rtsp-server/rtsp-token.h:
6111 rtsp-server: add padding to many public structures
6112 Not mini objects though, since they are not subclassable
6113 anyway, nor kept on the stack or inlined in a structure.
6115 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
6117 media: add new create_rtpbin vmethod
6118 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
6119 https://bugzilla.gnome.org/show_bug.cgi?id=719734
6121 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
6123 * tests/check/gst/media.c:
6124 tests: fix memory leak, free test's thread pool
6125 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
6127 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
6129 * gst/rtsp-server/rtsp-stream-transport.c:
6130 stream-transport: free url in finalize
6132 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
6134 * gst/rtsp-server/rtsp-media.c:
6135 media: also do state change in suspended state
6137 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
6139 * gst/rtsp-server/rtsp-client.c:
6140 * gst/rtsp-server/rtsp-media.c:
6141 media: also handle prepare and range in suspended state
6142 When we are suspended, we are already prepared.
6143 We can get the range in the suspended state.
6145 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
6147 * tests/check/Makefile.am:
6148 * tests/check/gst/sessionmedia.c:
6149 check: add test for uri in setup
6150 Added unit tests for the new functionality in GstRTSPStreamTransport.
6151 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
6153 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
6155 * gst/rtsp-server/rtsp-client.c:
6156 client: store setup uri and use in PLAY response
6157 Store the uri used when doing the setup and use that in the PLAY
6159 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
6161 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
6163 * gst/rtsp-server/rtsp-stream-transport.c:
6164 * gst/rtsp-server/rtsp-stream-transport.h:
6165 stream-transport: add method to get/set url
6167 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
6169 * gst/rtsp-server/rtsp-client.c:
6170 client: suspend after SDP and unsuspend before PLAYING
6171 Based on patches by Ognyan Tonchev <ognyan@axis.com>
6172 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
6174 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
6176 * gst/rtsp-server/rtsp-media-factory.c:
6177 * gst/rtsp-server/rtsp-media-factory.h:
6178 * gst/rtsp-server/rtsp-media.c:
6179 * gst/rtsp-server/rtsp-media.h:
6180 * gst/rtsp-server/rtsp-session-media.c:
6181 * gst/rtsp-server/rtsp-session.c:
6182 * tests/check/gst/media.c:
6183 * tests/check/gst/mediafactory.c:
6184 media: add suspend modes
6185 Add support for different suspend modes. The stream is suspended right after
6186 producing the SDP and after PAUSE. Different suspend modes are available that
6187 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
6188 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
6189 state and RESET will bring the pipeline to the NULL state.
6190 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
6191 this means that the pipeline needs to be prerolled again.
6192 Base on patches by Ognyan Tonchev <ognyan@axis.com>
6193 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
6195 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
6197 * gst/rtsp-server/rtsp-media.c:
6198 media: start live streams in blocked state
6199 Start live streams in the blocked state and make them preroll using the
6200 messages. This ensure that no data is played by the sink until we explicitly
6201 unblock the stream right before going to PLAYING.
6202 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
6204 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
6206 * gst/rtsp-server/rtsp-media.c:
6207 media: refactor starting and waiting for preroll
6208 Based on patches from Ognyan Tonchev <ognyan@axis.com>
6209 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
6211 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
6213 * gst/rtsp-server/rtsp-stream.c:
6214 * gst/rtsp-server/rtsp-stream.h:
6215 stream: add API to block streams
6216 Add an API to block on the streams and make it post a message.
6217 Based on patch by Ognyan Tonchev <ognyan@axis.com>
6218 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
6220 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
6222 * docs/libs/Makefile.am:
6223 docs: Specify the override file
6224 Even if it's empty (for now) it avoids make distcheck complaining
6226 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
6228 * gst/rtsp-server/rtsp-media.c:
6229 media: move default implementations to where they are used
6231 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
6233 * gst/rtsp-server/rtsp-media.c:
6234 media: take the right lock in gst_rtsp_media_set_pipeline_state()
6235 We need to take the state_lock when calling this method.
6237 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
6239 * gst/rtsp-server/rtsp-media.c:
6240 media: handle add-added on non-bins too
6241 Handle dynamic payloaders that are not bins, as used in the unit-test.
6243 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
6245 * gst/rtsp-server/rtsp-media-factory.c:
6246 * gst/rtsp-server/rtsp-media-factory.h:
6247 * gst/rtsp-server/rtsp-media.c:
6248 rtsp-media/-factory: Fix request pad name comments
6249 These must be escaped for gtk-doc to parse the comments without warnings.
6251 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
6253 rtsp-media: remove transports if media is in error status
6254 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
6255 trying to change to GST_STATE_NULL and media is in error status, we
6256 remove all transports.
6257 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
6259 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
6261 * gst/rtsp-server/rtsp-media.c:
6262 rtsp-media: use element metadata to find payloader
6263 Use the element metadata to find the payloader instead of checking
6265 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
6267 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
6269 rtsp-stream: add getter for payload type
6270 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
6271 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
6272 element and create the stream with this one instead of the dynpay%d
6274 https://bugzilla.gnome.org/show_bug.cgi?id=712396
6276 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
6278 * gst/rtsp-server/rtsp-client.c:
6279 * gst/rtsp-server/rtsp-context.h:
6280 * gst/rtsp-server/rtsp-media.c:
6281 * gst/rtsp-server/rtsp-mount-points.c:
6282 * gst/rtsp-server/rtsp-server.c:
6283 * gst/rtsp-server/rtsp-token.c:
6284 rtsp-*: Refer to NULL as a constant in comments
6286 https://bugzilla.gnome.org/show_bug.cgi?id=714988
6288 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
6290 rtsp-*: Fix type name typos in comments
6291 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
6292 * rtsp-auth: Refer to part of constant name as text
6293 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
6294 * rtsp-session-media: Fix GstRTSPSessionMedia typo
6295 * rtsp-stream: Fix typo when refering to GstBin
6296 https://bugzilla.gnome.org/show_bug.cgi?id=714988
6298 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
6301 * docs/libs/gst-rtsp-server-docs.sgml:
6302 * docs/libs/gst-rtsp-server-sections.txt:
6303 docs: Improve documentation
6304 * Include annotation-glossary to quiet gtk-doc
6305 * Rename remaining ClientState -> Context
6306 * Rename object hierarchy file
6307 * Remove stale chapter references
6308 * Add missing function and object references
6309 * Include missing GstRTSPAddressPoolResult
6310 https://bugzilla.gnome.org/show_bug.cgi?id=714988
6312 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
6314 * gst/rtsp-server/rtsp-client.c:
6315 * gst/rtsp-server/rtsp-server.c:
6316 * gst/rtsp-server/rtsp-session-pool.c:
6317 * gst/rtsp-server/rtsp-session.c:
6318 * gst/rtsp-server/rtsp-stream.c:
6319 rtsp-server: sprinkle some allow-none annotations for g-i
6321 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
6323 * gst/rtsp-server/rtsp-stream.c:
6324 * gst/rtsp-server/rtsp-stream.h:
6325 stream: add method to filter transports
6326 Add a method to safely iterate and collect the stream transports
6327 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
6329 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
6331 * gst/rtsp-server/rtsp-client.c:
6332 * gst/rtsp-server/rtsp-server.c:
6333 * gst/rtsp-server/rtsp-session-pool.c:
6334 * gst/rtsp-server/rtsp-session.c:
6335 rtsp: allow NULL func in filters
6336 Passing a null function make the filters return a list of
6339 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
6341 * gst/rtsp-server/rtsp-address-pool.c:
6342 * tests/check/gst/addresspool.c:
6343 address-pool: fix address increment
6344 Use a guint instead of guint8 to increment the address. It's still not
6345 completely correct because a guint might not be able to hold the complete
6346 address range, but that's an enhacement for later.
6347 Add unit test to test improved behaviour.
6348 https://bugzilla.gnome.org/show_bug.cgi?id=708237
6350 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
6352 * gst/rtsp-server/rtsp-client.c:
6353 * tests/check/gst/client.c:
6354 client: allow absolute path in requests
6355 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
6357 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
6359 * gst/rtsp-server/rtsp-client.c:
6360 * gst/rtsp-server/rtsp-client.h:
6361 client: make make_path_from_uri a vmethod
6363 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
6365 * docs/libs/gst-rtsp-server-sections.txt:
6366 * gst/rtsp-server/rtsp-stream.c:
6367 * gst/rtsp-server/rtsp-stream.h:
6368 * tests/check/Makefile.am:
6369 * tests/check/gst/stream.c:
6370 stream: Add functions to get rtp and rtcp sockets
6371 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
6373 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
6375 * gst/rtsp-server/rtsp-context.c:
6376 * gst/rtsp-server/rtsp-context.h:
6377 context: defing a GType for the context
6378 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
6380 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6382 * gst/rtsp-server/Makefile.am:
6383 * gst/rtsp-server/rtsp-auth.c:
6384 * gst/rtsp-server/rtsp-context.c:
6385 * gst/rtsp-server/rtsp-media.c:
6386 * gst/rtsp-server/rtsp-mount-points.c:
6387 * gst/rtsp-server/rtsp-server.h:
6388 * gst/rtsp-server/rtsp-session-media.c:
6389 * gst/rtsp-server/rtsp-session.c:
6390 * gst/rtsp-server/rtsp-stream.c:
6391 Fixed several GIR warnings
6393 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
6395 * gst/rtsp-server/rtsp-auth.c:
6398 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6400 * tests/check/Makefile.am:
6401 * tests/check/gst/token.c:
6402 tests: Add unit tests for token
6403 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
6405 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6407 * gst/rtsp-server/rtsp-token.c:
6408 token: Validate args for gst_rtsp_token_is_allowed
6409 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
6411 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6413 * gst/rtsp-server/rtsp-token.c:
6414 token: Fix bug when creating empty token
6415 We always want to have a valid GstStructure in the token.
6416 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
6418 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
6420 * gst/rtsp-server/rtsp-thread-pool.c:
6421 thread-pool: avoid race in shutdown
6422 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
6423 don't actually stop the mainloop ever. Solve this race by adding an idle source
6424 to the mainloop that calls the _quit. This way we immediately exit the mainloop
6425 if quit was called before we started it.
6427 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6429 * tests/check/Makefile.am:
6430 * tests/check/gst/permissions.c:
6431 tests: Add unit tests for permissions
6432 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
6434 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6436 * tests/check/gst/mediafactory.c:
6437 tests: Test mediafactory permissions
6438 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
6440 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6442 * gst/rtsp-server/rtsp-permissions.c:
6443 permissions: Fix refcounting when adding/removing roles
6444 Previously a role that was removed was unreffed twice, and when
6445 replacing an existing role the replaced role was freed while still being
6446 referenced. Both bugs are now fixed.
6447 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
6449 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6451 * tests/check/gst/media.c:
6452 * tests/check/gst/mediafactory.c:
6453 * tests/check/gst/rtspserver.c:
6454 tests: Check gst_rtsp_url_parse return value
6455 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
6457 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
6460 Automatic update of common submodule
6461 From 865aa20 to dbedaa0
6463 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
6465 * gst/rtsp-server/rtsp-server.c:
6466 rtsp-server: Fix socket leak
6467 https://bugzilla.gnome.org/show_bug.cgi?id=710088
6469 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
6471 * gst/rtsp-server/rtsp-session-pool.c:
6472 rtsp-session-pool: Make sure session IDs are properly URI-escaped
6473 https://bugzilla.gnome.org/show_bug.cgi?id=643812
6475 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
6477 * examples/.gitignore:
6478 * examples/test-video.c:
6479 examples: fix compilation when WITH_AUTH is defined
6480 https://bugzilla.gnome.org/show_bug.cgi?id=710228
6482 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
6485 gitignore: Add new test binary
6487 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
6489 * tests/check/Makefile.am:
6490 * tests/check/gst/threadpool.c:
6491 thread-pool: Add unit test for the thread pools
6492 https://bugzilla.gnome.org/show_bug.cgi?id=710228
6494 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
6496 * gst/rtsp-server/rtsp-thread-pool.c:
6497 thread-pool: Fix thread leak when reusing threads
6498 https://bugzilla.gnome.org/show_bug.cgi?id=709730
6500 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
6502 * gst/rtsp-server/rtsp-server.c:
6503 * tests/check/gst/rtspserver.c:
6504 tests: fixed racy behavior in rtspserver tests
6505 https://bugzilla.gnome.org/show_bug.cgi?id=710078
6507 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6509 * tests/check/gst/addresspool.c:
6510 tests: Improve address pool unit tests
6511 Add a range with mixed IPV4 and IPV6 addresses to pool.
6512 Get an IPV4 address from an IPV6-only pool.
6513 Get an IPV6 address from an IPV4-only pool.
6514 Reserve a IPV6 address from an IPV4-only pool.
6515 Check for unicast addresses in multicast-only pool.
6516 Check for unicast addresses in uni-/multicast-mixed pool.
6517 https://bugzilla.gnome.org/show_bug.cgi?id=710128
6519 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6521 * gst/rtsp-server/rtsp-client.c:
6522 client: append query string in PAUSE/PLAY/TEARDOWN as well
6524 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
6526 * gst/rtsp-server/rtsp-client.c:
6527 client: Add query to control path
6528 If the SETUP url contains a query it must be appended to the control
6529 path so that it matches any already created stream in the media. The
6530 query will also be appended to the session media path.
6532 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6534 * gst/rtsp-server/rtsp-media.c:
6535 rtsp-media: remove old line
6537 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
6539 * gst/rtsp-server/rtsp-stream.c:
6540 stream: Correct control comparison
6541 https://bugzilla.gnome.org/show_bug.cgi?id=709176
6543 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6545 * gst/rtsp-server/rtsp-media.c:
6546 media: Check dynamically if the pipeline supports seeking
6547 We should not depend on whether or not the pipeline state change
6548 returned NO_PREROLL or not. A media could dynamically change its
6549 element and switch from seekable to non seekable so it's best to test
6550 the seekable nature of the pipeline dynamically when we try to do a seek.
6552 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6554 * gst/rtsp-server/rtsp-media.c:
6555 media: Return FALSE if seeking is not supported
6557 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6559 * gst/rtsp-server/rtsp-media.c:
6560 rtsp-media: don't seek accurate by default
6561 Accurate seeking is perhaps a little overkill in the most common situation and
6562 causes some formats (mp3) over slow media to seek extremely slowly.
6564 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
6566 * tests/check/gst/rtspserver.c:
6567 tests: fix unit test
6568 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
6570 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
6572 * gst/rtsp-server/rtsp-client.c:
6573 client: Reply 400 if media cannot be constructed
6574 Reply 400 Bad Request instead of 503 Service Unavailable if media
6575 cannot be constructed in SETUP.
6576 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
6578 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
6580 * gst/rtsp-server/rtsp-client.c:
6581 client: Send setup reply once only
6582 If find_media() failed in handle_setup_request() two replies was sent.
6583 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
6585 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
6588 Automatic update of common submodule
6589 From 6b03ba7 to 865aa20
6591 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
6593 * gst/rtsp-server/rtsp-server.c:
6594 server: Emit client-connected signal earlier
6595 Emit client-connected before the client ref is given to a GSource,
6596 otherwise client-connected can be emitted after the client object has
6599 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
6601 * gst/rtsp-server/rtsp-address-pool.c:
6602 * gst/rtsp-server/rtsp-address-pool.h:
6603 * gst/rtsp-server/rtsp-stream.c:
6604 * tests/check/gst/addresspool.c:
6605 addresspool: return reason of failure
6606 Let gst_rtsp_address_pool_reserve_address() return the reason why
6607 the address could not be reserved.
6608 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
6610 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
6613 autogen.sh: Sync behaviour with other GStreamer modules
6614 Allows building from outside of tree amongst other things
6616 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
6619 Automatic update of common submodule
6620 From b613661 to 6b03ba7
6622 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
6625 Automatic update of common submodule
6626 From 74a6857 to b613661
6628 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
6631 Automatic update of common submodule
6632 From 01a7a46 to 74a6857
6634 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
6636 * gst/rtsp-server/rtsp-client.c:
6637 client: Do not read beyond end of path string
6638 If the setup was done without a control url, make sure we don't try to read the
6639 non-existing control string and crash.
6641 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6643 * gst/rtsp-server/rtsp-client.c:
6644 client: Fix RTPInfo header
6645 Refactor the method to make the content_base.
6646 Use the content-base and the control url to construct the RTPInfo
6649 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6651 * gst/rtsp-server/rtsp-client.c:
6652 client: map url to path only in describe
6653 Only map the request url to a path in the DESCRIBE method. The SDP then
6654 contains the base and control urls that should be used to SETUP/PAUSE/
6655 PLAY/TEARDOWN the media.
6657 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6659 * gst/rtsp-server/rtsp-client.c:
6660 Revert "client: map URL to path in requests"
6661 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
6662 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
6663 contains the base and control urls which are used in the SETUP, PLAY,
6664 PAUSE and TEARDOWN requests.
6666 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6668 * gst/rtsp-server/rtsp-client.c:
6669 client: map URL to path in requests
6671 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6673 * gst/rtsp-server/rtsp-client.c:
6674 * gst/rtsp-server/rtsp-mount-points.c:
6675 * gst/rtsp-server/rtsp-mount-points.h:
6676 mount-points: make vmethod to make path from uri
6677 Make a vmethod to transform an url into a path. The path is then used to lookup
6678 the factory. This makes it possible to also use other bits of the url, such as
6679 the query parameters, to locate the factory.
6681 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
6683 * gst/rtsp-server/rtsp-thread-pool.c:
6684 * gst/rtsp-server/rtsp-thread-pool.h:
6685 thread-pool: Add cleanup to wait for the threadpool to finish
6686 Also fix race condition if two threads are asking for the first
6687 thread from the thread pool at once. This would case two internal
6688 GThreadPools to be created.
6689 https://bugzilla.gnome.org/show_bug.cgi?id=707753
6691 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
6693 * gst/rtsp-server/rtsp-client.c:
6694 * tests/check/gst/client.c:
6695 client: free threadpool
6696 https://bugzilla.gnome.org/show_bug.cgi?id=707638
6698 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
6700 * tests/check/gst/mountpoints.c:
6701 mountpoints tests: unref matched factories
6702 https://bugzilla.gnome.org/show_bug.cgi?id=707638
6704 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
6706 * tests/check/gst/media.c:
6707 media tests: unref thread pool and caps
6708 https://bugzilla.gnome.org/show_bug.cgi?id=707638
6710 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
6712 * gst/rtsp-server/rtsp-auth.c:
6713 * gst/rtsp-server/rtsp-media-factory.c:
6714 * gst/rtsp-server/rtsp-media.c:
6715 auth, media, media-factory: unref permissions
6716 https://bugzilla.gnome.org/show_bug.cgi?id=707638
6718 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6720 * examples/Makefile.am:
6721 Makefile: add rule for appsrc example
6723 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6725 * examples/test-appsrc.c:
6726 tests: add appsrc example
6727 Add an example on how to use appsrc to feed the server pipeline with data.
6729 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
6731 * gst/rtsp-server/rtsp-client.c:
6732 rtsp-client: remove query part from content-base string
6733 Make sure that after the control url has been resolved, it's
6734 not a part of the query-string.
6735 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
6737 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6739 * gst/rtsp-server/rtsp-client.c:
6740 client: don't check url in response
6741 There is no url or method in the response to check
6743 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6745 * gst/rtsp-server/rtsp-client.c:
6746 * gst/rtsp-server/rtsp-client.h:
6747 Add handle-response signal for when we receive a GET_PARAMETER response
6749 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6751 * gst/rtsp-server/rtsp-server.c:
6752 Fix gst_rtsp_server_client_filter, using wrong variable type
6754 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
6756 * gst/rtsp-server/rtsp-media-factory-uri.c:
6757 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
6758 For AAC we need to check for framed=true instead of parsed=true.
6759 https://bugzilla.gnome.org/show_bug.cgi?id=701384
6761 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6763 * gst/rtsp-server/rtsp-stream.c:
6764 stream: optimize pipeline for protocols
6765 When TCP is not an allowed protocol for the stream, avoid creating the
6766 appsrc/appsink/queue and tee elements.
6768 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6770 * gst/rtsp-server/rtsp-media.c:
6771 media: set protocols on streams
6773 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6775 * gst/rtsp-server/rtsp-client.c:
6776 client: use protocols supported by stream
6778 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6780 * gst/rtsp-server/rtsp-media-factory.c:
6781 * gst/rtsp-server/rtsp-media.c:
6782 * gst/rtsp-server/rtsp-stream.c:
6783 media-factory: allow all protocols
6785 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6787 * gst/rtsp-server/rtsp-media.c:
6788 media: configure protocols in new streams
6790 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6792 * gst/rtsp-server/rtsp-stream.c:
6793 * gst/rtsp-server/rtsp-stream.h:
6794 stream: add protocols property
6796 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6798 * gst/rtsp-server/rtsp-media.c:
6799 rtsp-media: send state in "new-state" signal
6800 https://bugzilla.gnome.org/show_bug.cgi?id=705110
6802 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
6805 build: add subdir-objects to AM_INIT_AUTOMAKE
6806 Fixes warnings with automake 1.14
6807 https://bugzilla.gnome.org/show_bug.cgi?id=705350
6809 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6811 * docs/libs/gst-rtsp-server-sections.txt:
6812 * gst/rtsp-server/rtsp-client.c:
6813 * gst/rtsp-server/rtsp-server.c:
6814 * gst/rtsp-server/rtsp-server.h:
6815 server: add method to iterate clients of server
6817 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6819 * gst/rtsp-server/rtsp-media.c:
6820 * gst/rtsp-server/rtsp-media.h:
6821 Add vmethod for rtsp-media subclass to access rtpbin
6823 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6825 * gst/rtsp-server/rtsp-client.h:
6826 small documentation fix
6828 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6830 * gst/rtsp-server/rtsp-client.c:
6831 Do not take range header if range is invalid
6833 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6835 * docs/libs/gst-rtsp-server-sections.txt:
6836 * gst/rtsp-server/rtsp-media.c:
6837 media: add docs for new method
6839 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6841 * gst/rtsp-server/rtsp-media.c:
6842 * gst/rtsp-server/rtsp-media.h:
6843 Add API to rtsp-media set the pipeline's state
6845 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6847 * gst/rtsp-server/rtsp-media.c:
6848 Update current position/duration when gst_rtsp_media_get_range_string is called
6850 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6852 * examples/test-cgroups.c:
6853 tests: add some more docs
6855 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6857 * examples/test-cgroups.c:
6858 * gst/rtsp-server/Makefile.am:
6859 * gst/rtsp-server/rtsp-auth.c:
6860 * gst/rtsp-server/rtsp-auth.h:
6861 * gst/rtsp-server/rtsp-client.c:
6862 * gst/rtsp-server/rtsp-client.h:
6863 * gst/rtsp-server/rtsp-context.c:
6864 * gst/rtsp-server/rtsp-context.h:
6865 * gst/rtsp-server/rtsp-params.c:
6866 * gst/rtsp-server/rtsp-params.h:
6867 * gst/rtsp-server/rtsp-server.c:
6868 * gst/rtsp-server/rtsp-thread-pool.c:
6869 * gst/rtsp-server/rtsp-thread-pool.h:
6870 * tests/check/gst/client.c:
6871 ClientState -> Context
6872 Rename the clientstate to context and put the code in a separate file.
6874 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6876 * examples/test-auth.c:
6877 * gst/rtsp-server/rtsp-auth.c:
6878 * gst/rtsp-server/rtsp-auth.h:
6879 auth: add support for default token
6880 The default token is used when the user is not authenticated and can be used to
6881 give minimal permissions.
6883 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6885 * examples/test-auth.c:
6886 * gst/rtsp-server/rtsp-auth.c:
6887 auth: use defines when possible
6889 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6891 * gst/rtsp-server/rtsp-address-pool.c:
6892 address-pool: improve docs
6894 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6896 * gst/rtsp-server/rtsp-permissions.c:
6897 permissions: add the role to the copy
6899 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
6901 * gst/rtsp-server/rtsp-permissions.c:
6902 permissions: Also copy the roles
6904 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
6906 * gst/rtsp-server/rtsp-permissions.c:
6907 permissions: Make it build
6909 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6911 * gst/rtsp-server/rtsp-address-pool.h:
6914 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6916 * docs/libs/gst-rtsp-server-sections.txt:
6917 * gst/rtsp-server/rtsp-auth.c:
6918 * gst/rtsp-server/rtsp-auth.h:
6919 * gst/rtsp-server/rtsp-media.c:
6920 * gst/rtsp-server/rtsp-session-media.c:
6921 * gst/rtsp-server/rtsp-stream-transport.c:
6922 * gst/rtsp-server/rtsp-stream-transport.h:
6923 * gst/rtsp-server/rtsp-stream.c:
6924 * tests/check/gst/client.c:
6927 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6929 * docs/libs/gst-rtsp-server-sections.txt:
6930 * gst/rtsp-server/rtsp-address-pool.c:
6931 * gst/rtsp-server/rtsp-address-pool.h:
6932 * tests/check/gst/addresspool.c:
6933 * tests/check/gst/rtspserver.c:
6934 address-pool: cleanups
6935 Remove redundant method, improve docs.
6937 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6939 * docs/libs/gst-rtsp-server-sections.txt:
6940 * gst/rtsp-server/rtsp-auth.h:
6941 * gst/rtsp-server/rtsp-permissions.c:
6942 * gst/rtsp-server/rtsp-permissions.h:
6943 * gst/rtsp-server/rtsp-token.c:
6946 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6948 * gst/rtsp-server/rtsp-permissions.c:
6949 permissions: implement _remove_role
6951 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6953 * gst/rtsp-server/rtsp-permissions.c:
6954 permissions: update docs
6956 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6958 * tests/check/gst/client.c:
6959 tests: simplify tests
6960 Client settings are now disabled by default so we don't need an auth
6961 module to disable them.
6963 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6965 * gst/rtsp-server/rtsp-auth.c:
6966 auth: add default authorizations
6967 When no auth module is specified, use our table of defaults to look up the
6968 default value of the check instead of always allowing everything. This was
6969 we can disallow client settings by default.
6971 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6974 README: update readme
6976 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6978 * gst/rtsp-server/rtsp-thread-pool.c:
6979 * gst/rtsp-server/rtsp-thread-pool.h:
6980 thread-pool: add more docs
6982 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6984 * gst/rtsp-server/rtsp-thread-pool.c:
6985 * gst/rtsp-server/rtsp-thread-pool.h:
6986 thread-pool: fix race in thread reuse
6987 If we try to reuse a thread right after we made it stop, we end up using a
6988 stopped thread. Catch this case and only reuse threads that are not stopping.
6990 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6992 * gst/rtsp-server/rtsp-server.c:
6993 server: add small debug
6995 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6997 * tests/check/gst/client.c:
6999 Add some permissions to media so we can use the auth and enable
7002 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7004 * gst/rtsp-server/rtsp-client.c:
7005 client: support pushed context in handle_request
7006 If we already have a pushed state, reuse it and add our own things. This makes
7007 it easier to write tests.
7009 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7011 * gst/rtsp-server/rtsp-auth.c:
7012 auth: don't auth on methods
7013 Don't authorize on methods anymore but on the resources that we
7014 try to access, this is more flexible.
7015 Move the authorization checks to where they are needed and let the
7016 check return the response on error.
7018 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7020 * gst/rtsp-server/rtsp-mount-points.c:
7021 mount-points: add some debug
7023 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7025 * tests/check/gst/client.c:
7026 tests: almost fix test
7028 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7030 * gst/rtsp-server/rtsp-auth.c:
7031 * gst/rtsp-server/rtsp-auth.h:
7032 * gst/rtsp-server/rtsp-client.c:
7033 * gst/rtsp-server/rtsp-client.h:
7034 * gst/rtsp-server/rtsp-server.c:
7035 * gst/rtsp-server/rtsp-server.h:
7036 auth: let the auth module check client_settings
7037 Let the auth module decide if client settings are allowed for the
7040 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7042 * gst/rtsp-server/rtsp-token.c:
7043 * gst/rtsp-server/rtsp-token.h:
7044 token: add method to check boolean permission
7046 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7048 * examples/test-auth.c:
7049 * examples/test-cgroups.c:
7050 * gst/rtsp-server/rtsp-token.c:
7051 * gst/rtsp-server/rtsp-token.h:
7052 token: simplify token constructor
7053 Use variable arguments to make easier API.
7055 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7057 * examples/test-auth.c:
7058 * examples/test-cgroups.c:
7059 * gst/rtsp-server/rtsp-media-factory.c:
7060 * gst/rtsp-server/rtsp-media-factory.h:
7061 media-factory: add convenience API for factory
7063 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7065 * examples/test-auth.c:
7066 * examples/test-cgroups.c:
7067 * gst/rtsp-server/rtsp-permissions.c:
7068 * gst/rtsp-server/rtsp-permissions.h:
7069 permissions: simplify API a little
7070 Avoid passing GstStructure in the add_role method, use varargs instead
7071 to construct the structure behind the scenes. We can then also use the
7072 structure name as the role and simplify some more logic.
7074 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7076 * gst/rtsp-server/rtsp-auth.c:
7079 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7081 * gst/rtsp-server/rtsp-auth.c:
7082 * gst/rtsp-server/rtsp-auth.h:
7083 * gst/rtsp-server/rtsp-client.c:
7084 auth: handle unauthorized response
7085 Move handling of the unauthorized response to the auth module, it can add
7086 the appropriate headers to request authorization for the required method
7087 much better than the client.
7089 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7091 * gst/rtsp-server/rtsp-client.c:
7092 * gst/rtsp-server/rtsp-client.h:
7093 client: allow for sending any message, not only requests
7094 Change the _send_request() method to _send_message() so that we
7095 can both send requests and replies.
7097 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7099 * docs/libs/gst-rtsp-server-sections.txt:
7100 * gst/rtsp-server/rtsp-server.h:
7103 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7105 * examples/test-video.c:
7106 * gst/rtsp-server/rtsp-auth.c:
7107 * gst/rtsp-server/rtsp-auth.h:
7108 * gst/rtsp-server/rtsp-server.c:
7109 * gst/rtsp-server/rtsp-server.h:
7110 auth: move TLS handling to auth module
7111 Remove the TLS settings on the server and move it to the auth module because
7112 that is where security related bits go.
7114 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7116 * gst/rtsp-server/rtsp-client.c:
7117 * gst/rtsp-server/rtsp-client.h:
7118 client: add state push/pop
7120 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7122 * gst/rtsp-server/rtsp-client.c:
7123 * gst/rtsp-server/rtsp-client.h:
7124 client: add connection to state
7126 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7128 * gst/rtsp-server/rtsp-mount-points.c:
7129 mount-points: fix debug
7131 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7133 * tests/check/gst/media.c:
7134 tests: fix media test
7136 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7138 * gst/rtsp-server/rtsp-thread-pool.c:
7139 thread-pool: we don't require a state
7141 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7143 * gst/rtsp-server/rtsp-server.c:
7144 server: let context ref the server
7145 So that we don't risk losing the server object early anc crash.
7147 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7149 * tests/check/gst/client.c:
7150 tests: fix client test
7152 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7155 * docs/libs/gst-rtsp-server-docs.sgml:
7156 * docs/libs/gst-rtsp-server-sections.txt:
7157 * gst/rtsp-server/rtsp-address-pool.c:
7158 * gst/rtsp-server/rtsp-auth.c:
7159 * gst/rtsp-server/rtsp-client.c:
7160 * gst/rtsp-server/rtsp-client.h:
7161 * gst/rtsp-server/rtsp-media-factory-uri.c:
7162 * gst/rtsp-server/rtsp-media-factory.c:
7163 * gst/rtsp-server/rtsp-media-factory.h:
7164 * gst/rtsp-server/rtsp-media.c:
7165 * gst/rtsp-server/rtsp-mount-points.c:
7166 * gst/rtsp-server/rtsp-params.c:
7167 * gst/rtsp-server/rtsp-permissions.c:
7168 * gst/rtsp-server/rtsp-sdp.c:
7169 * gst/rtsp-server/rtsp-server.c:
7170 * gst/rtsp-server/rtsp-server.h:
7171 * gst/rtsp-server/rtsp-session-media.c:
7172 * gst/rtsp-server/rtsp-session-pool.c:
7173 * gst/rtsp-server/rtsp-session.c:
7174 * gst/rtsp-server/rtsp-stream-transport.c:
7175 * gst/rtsp-server/rtsp-stream.c:
7176 * gst/rtsp-server/rtsp-thread-pool.c:
7177 * gst/rtsp-server/rtsp-token.c:
7180 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7182 * gst/rtsp-server/rtsp-session-pool.c:
7183 * gst/rtsp-server/rtsp-session-pool.h:
7184 session-pool: make vmethod to create a session
7185 Make a vmethod to create a sessions so that subclasses can create
7186 custom session objects
7188 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7190 * gst/rtsp-server/rtsp-auth.c:
7191 * gst/rtsp-server/rtsp-media-factory.h:
7192 * gst/rtsp-server/rtsp-media.h:
7193 * gst/rtsp-server/rtsp-mount-points.h:
7194 * gst/rtsp-server/rtsp-session-pool.h:
7195 * gst/rtsp-server/rtsp-stream.h:
7198 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7200 * docs/libs/gst-rtsp-server-docs.sgml:
7201 * docs/libs/gst-rtsp-server-sections.txt:
7202 * gst/rtsp-server/rtsp-address-pool.c:
7203 * gst/rtsp-server/rtsp-address-pool.h:
7204 * gst/rtsp-server/rtsp-auth.c:
7205 * gst/rtsp-server/rtsp-client.h:
7206 * gst/rtsp-server/rtsp-media-factory.h:
7207 * gst/rtsp-server/rtsp-media.c:
7208 * gst/rtsp-server/rtsp-media.h:
7209 * gst/rtsp-server/rtsp-permissions.c:
7210 * gst/rtsp-server/rtsp-permissions.h:
7211 * gst/rtsp-server/rtsp-server.h:
7212 * gst/rtsp-server/rtsp-session-media.c:
7213 * gst/rtsp-server/rtsp-session-media.h:
7214 * gst/rtsp-server/rtsp-session-pool.h:
7215 * gst/rtsp-server/rtsp-session.h:
7216 * gst/rtsp-server/rtsp-stream-transport.h:
7217 * gst/rtsp-server/rtsp-stream.c:
7218 * gst/rtsp-server/rtsp-thread-pool.h:
7221 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7224 * examples/Makefile.am:
7225 configure: compile cgroup example conditionally
7226 Only compile the cgroup example when we have libcgroup
7228 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7231 * examples/Makefile.am:
7232 * examples/test-cgroups.c:
7233 examples: add cgroups example
7235 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7237 * tests/check/gst/rtspserver.c:
7238 tests: fix compilation
7240 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7242 * gst/rtsp-server/rtsp-thread-pool.c:
7243 thread-pool: fix vmethod invocation
7245 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7247 * gst/rtsp-server/rtsp-thread-pool.c:
7248 * gst/rtsp-server/rtsp-thread-pool.h:
7249 thread-pool: store thread type in thread
7251 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7253 * gst/rtsp-server/rtsp-client.c:
7254 client: pass thread from pool to media _prepare
7255 Get a thread from the configured threadpool and pass it to the prepare method of
7258 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7260 * gst/rtsp-server/rtsp-media.c:
7261 * gst/rtsp-server/rtsp-media.h:
7262 media: Accept a thread in _prepare
7263 Remove out own threadpool handling and use the provided thread and
7264 maincontext for the bus messages and the state changes.
7266 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7268 * gst/rtsp-server/rtsp-server.c:
7269 server: configure client thread pool
7271 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7273 * gst/rtsp-server/rtsp-client.c:
7274 * gst/rtsp-server/rtsp-client.h:
7275 client: add method to configure thread pool
7277 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7279 * gst/rtsp-server/rtsp-client.h:
7280 * gst/rtsp-server/rtsp-server.c:
7281 * gst/rtsp-server/rtsp-server.h:
7282 server: use thread pool
7283 Use the thread pool instead of doing our own thing.
7285 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7287 * gst/rtsp-server/Makefile.am:
7288 * gst/rtsp-server/rtsp-thread-pool.c:
7289 * gst/rtsp-server/rtsp-thread-pool.h:
7290 thread-pool: add object to manage threads
7291 Add an object to manage the client and media threads.
7293 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7295 * gst/rtsp-server/rtsp-auth.c:
7296 auth: debug authorization check
7298 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7300 * gst/rtsp-server/rtsp-media.c:
7301 media: start media pipeline in context
7302 Start the media pipeline in the provided context (or our default one
7303 when NULL). This makes sure that we run the bus thread in this context and that
7304 all media threads are children of this context.
7306 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7308 * gst/rtsp-server/rtsp-media-factory.c:
7309 factory: pass permissions to media by default
7311 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7313 * examples/test-auth.c:
7314 test: add permissions to auth test
7315 Ass some permissions to the media factory in the test.
7317 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7319 * gst/rtsp-server/rtsp-auth.c:
7320 * gst/rtsp-server/rtsp-auth.h:
7321 * gst/rtsp-server/rtsp-client.c:
7322 auth: simplify auth checks
7323 Remove client from methods, it's now in the state
7324 Perform the check specified by the string, use the information from the
7325 thread local context.
7327 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7329 * gst/rtsp-server/rtsp-client.c:
7330 * gst/rtsp-server/rtsp-client.h:
7331 client: add state to current thread
7332 Add the client to the ClientState object.
7333 Place the ClientState on the current thread.
7335 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7337 * gst/rtsp-server/rtsp-media-factory.c:
7338 * gst/rtsp-server/rtsp-media-factory.h:
7339 * gst/rtsp-server/rtsp-media.c:
7340 * gst/rtsp-server/rtsp-media.h:
7341 media: make it possible to set permissions
7342 Make it possible to set permissions on media and media factory objects
7344 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7346 * gst/rtsp-server/Makefile.am:
7347 * gst/rtsp-server/rtsp-permissions.c:
7348 * gst/rtsp-server/rtsp-permissions.h:
7349 permissions: add permissions object
7350 Add a mini object to store permissions based on a role.
7352 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7354 * examples/test-auth.c:
7355 * gst/rtsp-server/rtsp-auth.c:
7356 * gst/rtsp-server/rtsp-auth.h:
7357 * gst/rtsp-server/rtsp-client.c:
7358 auth: add auth checks
7359 Add an enum with auth checks and implement the checks in the auth object.
7360 Perform the checks from the client.
7362 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7364 * examples/test-auth.c:
7365 * gst/rtsp-server/rtsp-auth.c:
7366 * gst/rtsp-server/rtsp-auth.h:
7367 * gst/rtsp-server/rtsp-client.h:
7368 auth: use the token after authentication
7369 After we authenticated a user, keep the Token around in the state.
7371 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7373 * gst/rtsp-server/rtsp-client.c:
7374 * gst/rtsp-server/rtsp-media.c:
7375 * gst/rtsp-server/rtsp-media.h:
7376 * tests/check/gst/media.c:
7377 media: add optional context for bus messages
7378 Add an optional mainloop to _prepare that will handle the bus messages instead
7379 of always using the shared mainloop.
7381 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7383 * gst/rtsp-server/Makefile.am:
7384 * gst/rtsp-server/rtsp-token.c:
7385 * gst/rtsp-server/rtsp-token.h:
7386 token: add authorization token
7387 Add a simply miniobject that contains the authorizations. The object contains a
7388 GstStructure that hold all authorization fields. When a user is authenticated,
7389 the auth module will create a Token for the user. The token is then used to
7390 check what operations the user is allowed to do and various other configuration
7393 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7395 * examples/test-auth.c:
7396 * gst/rtsp-server/rtsp-auth.c:
7397 * gst/rtsp-server/rtsp-auth.h:
7398 * gst/rtsp-server/rtsp-client.c:
7399 * gst/rtsp-server/rtsp-client.h:
7400 * gst/rtsp-server/rtsp-media-factory.c:
7401 * gst/rtsp-server/rtsp-media-factory.h:
7402 * gst/rtsp-server/rtsp-media.c:
7403 * gst/rtsp-server/rtsp-media.h:
7404 auth: remove auth from media and factory
7405 Remove the auth object from media and factory. We want to have the RTSPClient
7406 authenticate and authorize resources, there is no need to place another auth
7407 manager on the media/factory.
7409 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7411 * examples/test-auth.c:
7412 * gst/rtsp-server/rtsp-auth.c:
7413 * gst/rtsp-server/rtsp-auth.h:
7414 * gst/rtsp-server/rtsp-client.h:
7415 auth: add support for multiple basic auth tokens
7416 Make it possible to add multiple basic authorisation tokens to one authorization
7417 object. Associate with each token an authorization group that will define what
7418 capabilities are allowed.
7420 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7422 * gst/rtsp-server/rtsp-client.c:
7423 client: error out on non-aggregate control
7424 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
7426 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7428 * gst/rtsp-server/rtsp-client.c:
7429 client: rework setup request a little
7430 Cache the media in DESCRIBE based on the longest matching path with the uri
7431 that we can find in the mount points.
7432 Rework the setup request a little to get the media from the session or from
7433 the longest matching path, this way we can derive the control string as
7434 everything after the path instead of hardcoding it.
7435 Find the stream based on the control string and only open a session when all
7438 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7440 * gst/rtsp-server/rtsp-media.c:
7441 * gst/rtsp-server/rtsp-media.h:
7442 media: add method to find a stream by control url
7444 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7446 * gst/rtsp-server/rtsp-stream.c:
7447 * gst/rtsp-server/rtsp-stream.h:
7448 stream: add method to check control url of stream
7450 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7452 * gst/rtsp-server/rtsp-client.c:
7453 * gst/rtsp-server/rtsp-session-media.c:
7454 * gst/rtsp-server/rtsp-session-media.h:
7455 * gst/rtsp-server/rtsp-session.c:
7456 * gst/rtsp-server/rtsp-session.h:
7457 session: use path matching for session media
7458 Use a path string instead of a uri to lookup session media in the sessions. Also
7459 use path matching to find the largest possible path that matches.
7461 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7463 * gst/rtsp-server/rtsp-client.c:
7464 * gst/rtsp-server/rtsp-mount-points.c:
7465 * gst/rtsp-server/rtsp-mount-points.h:
7466 * tests/check/gst/mountpoints.c:
7467 mount-points: remove useless vmethod
7468 Making lookups in the mount points should not be done with a URL, if there is a
7469 mapping to be done from URL to mount points, we'll need to do it somewhere
7472 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7474 * gst/rtsp-server/rtsp-mount-points.c:
7475 * gst/rtsp-server/rtsp-mount-points.h:
7476 * tests/check/gst/mountpoints.c:
7477 mount-points: improve mount point searching
7478 Use a GSequence to keep track of the mount points.
7479 Match a URL to the longest matching registered mount point. This should be the
7480 URL to perform aggreagate control and the remainder is the stream specific
7482 Add some unit tests for this.
7484 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
7486 * gst/rtsp-server/Makefile.am:
7487 rtsp-server: Allow building of static library
7489 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7491 * tests/check/gst/mediafactory.c:
7492 tests: fix compilation
7494 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7496 * gst/rtsp-server/rtsp-sdp.c:
7497 sdp: get control string from stream
7498 Use the control string as configured in the stream.
7500 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7502 * gst/rtsp-server/rtsp-stream.c:
7503 * gst/rtsp-server/rtsp-stream.h:
7504 stream: add methods and property to set control string
7506 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7508 * gst/rtsp-server/rtsp-client.c:
7510 Rename variables for clarity
7511 Keep media in state when we can
7513 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7515 * gst/rtsp-server/rtsp-client.c:
7516 * gst/rtsp-server/rtsp-stream.c:
7517 * gst/rtsp-server/rtsp-stream.h:
7518 stream: add more support for IPv6
7519 Rename _get_address to _get_multicast_address in GstRTSPStream to
7520 make it clear that this function only deals with multicast.
7521 Make it possible to have both an IPv4 and IPv6 multicast address on
7522 a stream. Give the client an IPv4 or IPv6 address depending on the
7523 address it used to connect to the server.
7524 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
7526 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7528 * gst/rtsp-server/rtsp-client.c:
7531 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7533 * gst/rtsp-server/rtsp-stream.c:
7534 stream: handle failed port allocation
7535 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
7536 can't allocate any family at all. Also keep track of what port families we
7538 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
7540 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7542 * gst/rtsp-server/rtsp-stream.c:
7543 stream: improve docs
7545 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7547 * gst/rtsp-server/rtsp-stream-transport.c:
7548 stream-transport: remove old if 0 block
7550 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
7552 * tests/check/gst/client.c:
7554 gst_rtsp_client_get_uri() has been removed
7555 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
7557 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7559 * gst/rtsp-server/rtsp-client.c:
7560 * gst/rtsp-server/rtsp-client.h:
7561 client: add method to filter managed sessions
7562 Add a method to filter the sessions managed by this client connection.
7563 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
7565 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7567 * gst/rtsp-server/rtsp-client.c:
7568 * gst/rtsp-server/rtsp-client.h:
7569 client: remove _get_uri() method
7570 Remove the get_uri() method on the client. A client has no uri, the uri
7571 property is an internal property to manage the last cached media for
7574 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7576 * gst/rtsp-server/rtsp-media-factory.h:
7577 media-factory: fix typo
7579 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
7581 * gst/rtsp-server/rtsp-media.c:
7582 rtsp-media: Do not leak the query in default_query_stop
7583 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
7585 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7587 * gst/rtsp-server/rtsp-media.c:
7588 media: don't unlock when conversion fails
7589 Don't unlock the state lock when conversion fails because it was not locked.
7591 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
7593 * gst/rtsp-server/rtsp-media.c:
7594 * gst/rtsp-server/rtsp-media.h:
7595 Add query_position and query_stop vmethods to rtsp-media
7597 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
7599 * gst/rtsp-server/rtsp-media.c:
7600 Fix typo in property install for rtsp-media's time-provider
7602 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7604 * gst/rtsp-server/rtsp-client.c:
7605 * gst/rtsp-server/rtsp-client.h:
7606 client: clean some variables
7607 Clean some variables and add some guards to _send_request()
7609 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
7611 * gst/rtsp-server/rtsp-client.c:
7612 * gst/rtsp-server/rtsp-client.h:
7613 Add gst_rtsp_client_send_request API
7614 This makes it possible to send arbitrary messages to a client, such as
7615 SET_PARAMETER or GET_PARAMETER
7617 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7619 * gst/rtsp-server/rtsp-media.c:
7620 * gst/rtsp-server/rtsp-media.h:
7621 media: add _get_element() method
7622 Add method to get the element used when creating the media.
7623 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
7625 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7627 * gst/rtsp-server/rtsp-media.c:
7630 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
7632 * gst/rtsp-server/rtsp-stream.c:
7633 * gst/rtsp-server/rtsp-stream.h:
7634 stream: allow access to the rtp session
7635 https://bugzilla.gnome.org/show_bug.cgi?id=703004
7637 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
7639 * gst/rtsp-server/rtsp-stream.c:
7640 * gst/rtsp-server/rtsp-stream.h:
7641 dscp qos support in gst-rtsp-stream
7642 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
7644 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7646 * tests/check/gst/rtspserver.c:
7648 Actually do what the comment says. Also keep the old code around, not sure what
7649 should happen when you get a 454 from a TEARDOWN, does it close the connection?
7650 it currently doesn't.
7652 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7654 * gst/rtsp-server/rtsp-client.c:
7655 client: also watch newly created session
7656 When we newly created a session, start watching it immediately instead of
7657 on the next request.
7659 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
7661 * tests/check/gst/client.c:
7662 tests: add unit test for new-session
7663 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
7665 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7667 * gst/rtsp-server/rtsp-client.c:
7668 client: emit new-session when new session is created
7669 Only emit new-session when we created a new session for a client, not when a
7670 client picked up a previous session.
7671 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
7673 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
7675 * gst/rtsp-server/rtsp-client.c:
7676 client: handle asterisk as path in requests
7677 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
7679 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7681 * gst/rtsp-server/rtsp-media.c:
7682 media: handle segment query format mismatch
7683 It's possible that the segment query returns with a different format than what
7684 we asked for, handle this case also.
7686 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
7688 * gst/rtsp-server/rtsp-media.c:
7689 media: use segment stop in collect_media_stats
7690 Use segment stop instead of duration as range end point.
7691 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
7693 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
7695 * gst/rtsp-server/rtsp-media.c:
7696 * tests/check/gst/media.c:
7697 rtsp-media: Do not leak the element in take_pipeline
7698 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
7700 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
7702 * gst/rtsp-server/rtsp-client.c:
7703 * gst/rtsp-server/rtsp-client.h:
7704 rtsp-client: Make configure_client_transport virtual
7705 This patch makes configure_client_transport virtual. The functionality is
7706 needed to handle some weird clients sending multicast transport settings as url
7708 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
7710 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
7712 * gst/rtsp-server/rtsp-client.c:
7713 * gst/rtsp-server/rtsp-client.h:
7714 rtsp-client: Make param_set and param_get virtual
7715 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
7717 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
7719 * gst/rtsp-server/rtsp-client.c:
7720 * gst/rtsp-server/rtsp-media.c:
7721 * gst/rtsp-server/rtsp-media.h:
7722 media: convert_range replaces get_range_times
7723 get_range_times worked for handling UTC ranges for seeks, but we also
7724 need to convert back from NPT to the requested unit in
7725 get_range_string. convert_range is now used for both.
7726 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
7728 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7730 * gst/rtsp-server/rtsp-client.c:
7731 * gst/rtsp-server/rtsp-sdp.c:
7732 * gst/rtsp-server/rtsp-sdp.h:
7733 sdp: cleanup sdp info
7734 We don't need to pass the proto, we can more easily check a boolean.
7735 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
7737 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
7739 * gst/rtsp-server/rtsp-sdp.c:
7740 use 0.0.0.0 or :: for c= line instead of server address
7742 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
7744 * gst/rtsp-server/rtsp-client.c:
7745 use local address, not remote, in SDP
7746 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
7748 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7751 Automatic update of common submodule
7752 From 098c0d7 to 01a7a46
7754 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
7756 * gst/rtsp-server/rtsp-media.c:
7757 * gst/rtsp-server/rtsp-media.h:
7758 media: possibility to override range time conversion
7759 Make it possible to override the conversion from GstRTSPTimeRange to
7760 GstClockTimes, that is done before seeking on the media
7761 pipeline. Overriding can be useful for UTC ranges, where the default
7762 conversion gives nanoseconds since 1900.
7763 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
7765 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
7767 * gst/rtsp-server/rtsp-server.c:
7768 * gst/rtsp-server/rtsp-server.h:
7769 rtsp-server: Expose the use_client_settings API
7770 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
7772 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
7774 * gst/rtsp-server/rtsp-client.c:
7775 * gst/rtsp-server/rtsp-stream.c:
7776 * gst/rtsp-server/rtsp-stream.h:
7777 rtspstream: handle both ipv4 and ipv6 clients
7778 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
7780 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7782 * gst/rtsp-server/rtsp-sdp.c:
7783 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
7784 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
7785 We already have a way to place extra attributes in the SDP by using a string
7786 property with prefix x- or a- in the caps.
7788 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7790 * gst/rtsp-server/rtsp-sdp.c:
7791 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
7792 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
7793 We already have a way to place extra attributes in the SDP, just make a string
7794 property in the payloader with a- or x- prefix.
7796 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7798 * gst/rtsp-server/rtsp-sdp.c:
7799 rtsp: place a- and x- properties as attributes
7800 application/x-rtp has properties with a- and x- prefixes that should be
7801 placed as attributes in the SDP for the media instead of being added to the
7804 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7806 * examples/Makefile.am:
7807 * examples/test-video.c:
7808 example: add TLS example
7810 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7812 * gst/rtsp-server/rtsp-server.c:
7813 * gst/rtsp-server/rtsp-server.h:
7814 server: add support for TLS
7815 Add methods to set and get a TLS certificate.
7816 Add vmethod to configure a new connection. By default, configure the TLS
7817 certificate in a new connection if needed.
7819 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7821 * gst/rtsp-server/rtsp-server.c:
7822 * gst/rtsp-server/rtsp-server.h:
7823 server: remove accept_client vmethod
7824 This vmethod is not very useful so remove it.
7826 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7828 * gst/rtsp-server/rtsp-server.c:
7829 server: don't crash on NULL GError
7831 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
7833 * gst/rtsp-server/rtsp-session-pool.c:
7834 rtsp-session-pool: corrected session timeout detection
7835 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
7837 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7839 * gst/rtsp-server/rtsp-client.c:
7840 client: improve debug
7842 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7844 * gst/rtsp-server/rtsp-client.c:
7845 * gst/rtsp-server/rtsp-client.h:
7846 * gst/rtsp-server/rtsp-server.c:
7847 server: refactor connection setup
7848 Let the server accept the socket connection and construct a GstRTSPConnection
7849 from it. Remove the code from the client and let the client only deal with
7850 a fully configure GstRTSPConnection object.
7851 We will need this later when the server will configure the connection for
7854 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7856 * gst/rtsp-server/rtsp-stream.c:
7857 stream: keep the transport object alive
7858 Keep the transport object alive while we have it as qdata on the
7861 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
7863 * gst/rtsp-server/rtsp-client.c:
7864 * gst/rtsp-server/rtsp-server.c:
7865 rtsp-server: Do not crash on nmapping of server
7866 * generate error when gst_rtsp_connection_accept fails
7867 * do not stop accepting incoming connections because
7868 accepting a client fails
7869 https://bugzilla.gnome.org/show_bug.cgi?id=701072
7871 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
7873 * gst/rtsp-server/rtsp-client.c:
7874 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
7875 https://bugzilla.gnome.org/show_bug.cgi?id=700953
7877 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
7879 * gst/rtsp-server/rtsp-sdp.c:
7880 rtsp-sdp: Parse framerate caps field and set SDP attribute
7881 The SDP attribute and its format is described in RFC4566.
7882 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
7884 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
7886 * gst/rtsp-server/rtsp-sdp.c:
7887 rtsp-sdp: Parse width/height from caps and set SDP attribute
7888 The SDP attribute and its format is described in RFC6064.
7889 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
7891 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
7893 * gst/rtsp-server/rtsp-sdp.c:
7894 * tests/check/gst/client.c:
7895 rtsp-sdp: add bandwidth line
7896 https://bugzilla.gnome.org/show_bug.cgi?id=699220
7898 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7901 Automatic update of common submodule
7902 From 5edcd85 to 098c0d7
7904 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
7906 * tests/check/gst/media.c:
7907 tests: add dynamic payloader prepare/unprepare check
7909 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7911 * gst/rtsp-server/rtsp-media.c:
7912 media: release lock when removing fakesink
7914 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7916 * gst/rtsp-server/rtsp-stream.c:
7917 stream: set elements to NULL before removing
7918 When removing a stream, set the elements to NULL first. This avoids
7919 element-is-not-in-NULL-state errors when we dispose the elements.
7921 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
7924 Automatic update of common submodule
7925 From 3cb3d3c to 5edcd85
7927 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7929 * gst/rtsp-server/rtsp-media.c:
7930 * gst/rtsp-server/rtsp-media.h:
7931 media: listen to pad-removed signals
7932 Listen to the pad-removed signal and remove the stream associated with the
7934 Add signal to be notified of the removed pad.
7935 Remove the fakesink in unprepare()
7936 Fix signatures of the signal methods
7938 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7940 * examples/test-sdp.c:
7941 tests: add example of reusable pipelines
7943 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
7945 * gst/rtsp-server/rtsp-stream.c:
7946 * gst/rtsp-server/rtsp-stream.h:
7947 stream: add method to get the srcpad
7949 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
7951 * tests/check/gst/media.c:
7952 check: add media prepare/unprepare test
7953 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
7955 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
7957 * gst/rtsp-server/rtsp-media.c:
7958 media: disconnect from signal handlers in unprepare()
7959 We connected to the pad-added and no-more-pads signals in prepare() so
7960 we need to disconnect from them in unprepare().
7961 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
7963 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
7965 * gst/rtsp-server/rtsp-media.c:
7966 media: don't free streams array
7967 Don't free the streams array in the unprepare() method, they were not
7969 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
7971 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
7973 * gst/rtsp-server/rtsp-media.c:
7974 media: don't unref the pipeline in unprepare
7975 Unprepare() should undo what prepare() does. Because the pipeline is
7976 not created in prepare(), we should not unref it in unprepare()
7978 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
7980 * gst/rtsp-server/rtsp-stream.c:
7981 stream: clear session and caps for reuse
7982 Set the session and caps to NULL after unref otherwise we might unref
7984 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
7986 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
7988 * gst/rtsp-server/rtsp-client.c:
7989 client: send out teardown signal before tearing down
7990 The advantage is that in the signal handler you get direct access to
7991 information about what streams are about to get torn down (in the
7992 GstRTSPClientState).
7993 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
7995 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
7997 * gst/rtsp-server/rtsp-client.c:
7998 * gst/rtsp-server/rtsp-client.h:
7999 client: expose connection
8000 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
8002 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
8005 Automatic update of common submodule
8006 From aed87ae to 3cb3d3c
8008 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8010 * gst/rtsp-server/rtsp-media.c:
8011 * gst/rtsp-server/rtsp-media.h:
8012 * gst/rtsp-server/rtsp-session-media.c:
8013 * gst/rtsp-server/rtsp-session-media.h:
8014 media: add method to get the base_time of the pipeline
8015 Together with a shared clock, this base-time could eventually be sent to
8016 the client so that it can reconstruct the exact running-time of the clock
8019 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8021 * gst/rtsp-server/Makefile.am:
8022 * gst/rtsp-server/rtsp-media.c:
8023 * gst/rtsp-server/rtsp-media.h:
8024 * gst/rtsp-server/rtsp-sdp.c:
8025 media: add GstNetTimeProvider support
8026 Add a property to let the media provide a GstNetTimeProvider for its clock.
8027 Make methods to get the clock and nettimeprovider
8028 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
8029 provider and also the current time of the clock. This should make it possible
8030 for (GStreamer) clients to slave their clock to the server clock.
8032 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
8035 Automatic update of common submodule
8036 From 04c7a1e to aed87ae
8038 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8040 * gst/rtsp-server/rtsp-media.c:
8041 media: wait for buffering to complete
8042 Wait for buffering to complete before changing the state to the target state.
8044 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8046 * gst/rtsp-server/rtsp-media.c:
8047 media: small cleanup
8049 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
8051 * tests/check/gst/rtspserver.c:
8052 tests: remove extra unref in test_setup_non_existing_stream
8053 The unref is not needed anymore, teardown runs without it.
8054 https://bugzilla.gnome.org/show_bug.cgi?id=696542
8056 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
8058 * tests/check/gst/rtspserver.c:
8059 tests: GSocketService cleanup in test_bind_already_in_use
8060 Use g_socket_service_stop so the rtspserver test stops listening for
8061 incoming connections in test_bind_already_in_use.
8062 https://bugzilla.gnome.org/show_bug.cgi?id=696541
8064 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
8066 * gst/rtsp-server/rtsp-media-factory.c:
8067 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
8068 Instead use a GWeakRef which is safe to use
8069 This is a known GLib bug, see:
8070 https://bugzilla.gnome.org/show_bug.cgi?id=667145
8072 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
8074 * gst/rtsp-server/rtsp-client.c:
8075 * gst/rtsp-server/rtsp-media.c:
8076 * gst/rtsp-server/rtsp-media.h:
8077 * gst/rtsp-server/rtsp-sdp.c:
8078 * tests/check/gst/media.c:
8079 * tests/check/gst/rtspserver.c:
8080 rtsp-media/client: Reply to PLAY request with same type of Range
8081 Remember the type of Range from the PLAY request and use the same type for
8084 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
8086 * gst/rtsp-server/rtsp-client.c:
8087 * gst/rtsp-server/rtsp-client.h:
8088 * tests/check/gst/client.c:
8089 rtsp-client: expose uri
8091 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
8093 * tests/check/gst/mediafactory.c:
8094 tests: Hold ref while creating second media
8095 To test if the media aren't shared, make sure we keep the first one while creating a second
8096 otherwise the same memory address may be reused.
8098 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
8101 configure: remove out-of-date comment
8103 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
8106 .gitignore: ignore more build files
8108 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
8110 * tests/check/Makefile.am:
8111 tests: use right _LIBS variable for gst-plugins-base libs
8113 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8115 * tests/check/Makefile.am:
8116 check: add librtp to libs
8118 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
8120 * tests/check/gst/rtspserver.c:
8121 tests: Add test to check selecting a port the server will send from
8123 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
8125 * tests/check/gst/rtspserver.c:
8126 tests: Make sure packets are actually received
8128 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
8130 * gst/rtsp-server/rtsp-stream.c:
8131 stream: Select unicast address from pool if appropriate
8133 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
8135 * gst/rtsp-server/rtsp-stream.c:
8136 stream: Properties are always there in Gst 1.0
8138 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
8140 * tests/check/gst/addresspool.c:
8141 tests: Add tests for unicast addresses in pool
8143 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
8145 * gst/rtsp-server/rtsp-address-pool.c:
8146 * tests/check/gst/addresspool.c:
8147 address-pool: Verify that multicast addresses are used for multicast and vice-versa
8149 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
8151 * docs/libs/gst-rtsp-server-sections.txt:
8152 * gst/rtsp-server/rtsp-address-pool.c:
8153 * gst/rtsp-server/rtsp-address-pool.h:
8154 * gst/rtsp-server/rtsp-stream.c:
8155 * tests/check/gst/addresspool.c:
8156 address-pool: Add unicast addresses
8158 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
8161 * gst/rtsp-server/rtsp-server.c:
8162 * tests/check/gst/rtspserver.c:
8163 rtsp-server: Limit the number of threads per server instance
8164 If we exceed the maximum, just round robin the clients over the existing
8167 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
8169 * gst/rtsp-server/rtsp-server.c:
8170 rtsp-server: No need to store the GMainContext in the client context
8172 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
8174 * tests/check/gst/rtspserver.c:
8175 tests: Add test for client disconnection
8177 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
8179 * tests/check/gst/rtspserver.c:
8180 tests: Test client and session timeouts with multiple threads
8182 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
8184 * gst/rtsp-server/rtsp-address-pool.c:
8185 * gst/rtsp-server/rtsp-auth.c:
8186 * gst/rtsp-server/rtsp-client.c:
8187 * gst/rtsp-server/rtsp-media-factory-uri.c:
8188 * gst/rtsp-server/rtsp-media-factory.c:
8189 * gst/rtsp-server/rtsp-media.c:
8190 * gst/rtsp-server/rtsp-mount-points.c:
8191 * gst/rtsp-server/rtsp-server.c:
8192 * gst/rtsp-server/rtsp-session-media.c:
8193 * gst/rtsp-server/rtsp-session-pool.c:
8194 * gst/rtsp-server/rtsp-session.c:
8195 Document locking and its order
8197 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
8199 * tests/check/gst/rtspserver.c:
8200 tests: Test that slow DESCRIBE don't block other clients
8202 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
8204 * tests/check/gst/client.c:
8205 tests: Add tests for client-requested multicast address
8207 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
8209 * docs/libs/gst-rtsp-server-sections.txt:
8210 docs: Put the various functions in the right sections
8212 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
8214 * docs/libs/gst-rtsp-server-docs.sgml:
8215 * docs/libs/gst-rtsp-server-sections.txt:
8216 * gst/rtsp-server/rtsp-address-pool.c:
8217 * gst/rtsp-server/rtsp-address-pool.h:
8218 docs: Generate docs for GstRTSPAddressPool
8220 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
8222 * gst/rtsp-server/rtsp-client.c:
8223 * gst/rtsp-server/rtsp-stream.c:
8224 * gst/rtsp-server/rtsp-stream.h:
8225 client: Check client provided addresses against the address pool
8227 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
8229 * gst/rtsp-server/rtsp-address-pool.c:
8230 * gst/rtsp-server/rtsp-address-pool.h:
8231 * tests/check/gst/addresspool.c:
8232 address-pool: Add API to request a specific address from the pool
8233 Also add relevant unit tests.
8235 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
8237 * tests/check/gst/mediafactory.c:
8238 tests: Check the passing around of a RTSPAddressPool
8239 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
8240 way down to the stream.
8242 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
8244 * tests/check/gst/addresspool.c:
8245 tests: Add more tests for the address pool
8247 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
8249 * gst/rtsp-server/rtsp-address-pool.c:
8250 address-pool: Fix off by one error
8251 When splitting a port range, the port after a skip is not part of range.
8253 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
8256 Automatic update of common submodule
8257 From 2de221c to 04c7a1e
8259 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
8262 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
8263 AM_CONFIG_HEADER was removed in automake 1.13
8264 https://bugzilla.gnome.org/show_bug.cgi?id=693368
8266 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
8269 Automatic update of common submodule
8270 From a942293 to 2de221c
8272 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8274 * gst/rtsp-server/rtsp-client.c:
8275 client: make sure the watch exists while sending data
8276 Protect the send_func with a lock. This allows us to wait for sending
8277 to complete before changing the send_func and user_data. We add an
8278 extra ref to the watch to make sure that it remains valid during
8280 When closing the connection, set the send_func to NULL
8281 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
8283 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8285 * tests/check/Makefile.am:
8286 tests: use GST_*_1_0 environment variables everywhere
8287 The _1_0 suffixed environment variables override the
8288 non-suffixed ones, so if we're in an environment that
8289 sets the _1_0 suffixed ones, such as jhbuild, we need
8290 to set those to make sure ours actually always get
8293 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8296 Automatic update of common submodule
8297 From acb04d9 to a942293
8299 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8301 * gst/rtsp-server/rtsp-client.c:
8302 rtsp-client: set the client backlog
8303 Set the client backlog to a reasonable default
8305 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
8307 * gst/rtsp-server/rtsp-media.c:
8308 rtsp-media: Make the element a constructor parameter
8309 https://bugzilla.gnome.org/show_bug.cgi?id=689594
8311 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8313 * docs/libs/Makefile.am:
8314 docs: Link with gcov library when gcov is enabled
8315 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
8317 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8319 * gst/rtsp-server/rtsp-media.c:
8320 media: match prepare with unprepare
8321 Really unprepare when there were an equal amount of prepare calls.
8323 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8325 * gst/rtsp-server/rtsp-media.c:
8326 media: media has to be unprepared in finalize
8327 Because unprepare takes away the last ref on the media.
8329 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8331 * gst/rtsp-server/rtsp-client.c:
8332 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
8333 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
8334 We can't use the refcount to trigger unprepare because it is the unprepare call
8335 that removes the last refcount after all messages are consumed. What we should
8336 probably do is make a prepared refcount and only unprepare when the refcount
8339 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8341 * gst/rtsp-server/rtsp-media.c:
8342 media: let the source unref the last media ref
8343 the last ref to the media is held by the source so we don't need to add more ref
8344 and unrefs, we simply destroy the media when the source is gone.
8346 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8348 * gst/rtsp-server/rtsp-media.c:
8349 media: improve debug
8351 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8353 * gst/rtsp-server/rtsp-media.c:
8355 Make sure we are in the right state when collecting the position and duration.
8356 Only make ourselves PREPARED when we were previously PREPARING.
8358 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8360 * gst/rtsp-server/rtsp-media.c:
8361 media: use g_object_ref/unref for GObjects
8363 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
8365 * gst/rtsp-server/rtsp-client.c:
8366 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
8367 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
8368 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
8369 isn't being used anymore.
8371 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
8373 * gst/rtsp-server/rtsp-media.c:
8374 Fix compiler warning
8376 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
8378 * gst/rtsp-server/rtsp-media-factory-uri.c:
8379 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
8381 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8383 * gst/rtsp-server/rtsp-session-media.h:
8386 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8388 * gst/rtsp-server/rtsp-media.c:
8389 * tests/check/gst/media.c:
8390 media: avoid element leak
8392 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8394 * gst/rtsp-server/rtsp-media.c:
8395 media: require an element in media constructor
8397 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8399 * gst/rtsp-server/rtsp-client.c:
8400 Revert "client: TEARDOWN brings that state to Init again"
8401 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
8402 The object is already disposed, there is no point in setting the state.
8404 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8406 * gst/rtsp-server/rtsp-client.c:
8407 client: TEARDOWN brings that state to Init again
8409 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8411 * docs/libs/gst-rtsp-server-sections.txt:
8412 * examples/test-auth.c:
8413 * gst/rtsp-server/rtsp-auth.c:
8414 * gst/rtsp-server/rtsp-auth.h:
8415 * gst/rtsp-server/rtsp-client.c:
8416 * gst/rtsp-server/rtsp-client.h:
8417 * gst/rtsp-server/rtsp-media-factory-uri.c:
8418 * gst/rtsp-server/rtsp-media-factory-uri.h:
8419 * gst/rtsp-server/rtsp-media-factory.c:
8420 * gst/rtsp-server/rtsp-media-factory.h:
8421 * gst/rtsp-server/rtsp-media.c:
8422 * gst/rtsp-server/rtsp-media.h:
8423 * gst/rtsp-server/rtsp-mount-points.c:
8424 * gst/rtsp-server/rtsp-mount-points.h:
8425 * gst/rtsp-server/rtsp-sdp.c:
8426 * gst/rtsp-server/rtsp-server.c:
8427 * gst/rtsp-server/rtsp-server.h:
8428 * gst/rtsp-server/rtsp-session-media.c:
8429 * gst/rtsp-server/rtsp-session-media.h:
8430 * gst/rtsp-server/rtsp-session-pool.c:
8431 * gst/rtsp-server/rtsp-session-pool.h:
8432 * gst/rtsp-server/rtsp-session.c:
8433 * gst/rtsp-server/rtsp-session.h:
8434 * gst/rtsp-server/rtsp-stream-transport.c:
8435 * gst/rtsp-server/rtsp-stream-transport.h:
8436 * gst/rtsp-server/rtsp-stream.c:
8437 * gst/rtsp-server/rtsp-stream.h:
8438 * tests/check/gst/media.c:
8439 rtsp: make object details private
8440 Make all object details private
8441 Add methods to access private bits
8443 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8445 * tests/check/Makefile.am:
8446 * tests/check/gst/media.c:
8447 tests: add media tests
8449 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8451 * gst/rtsp-server/rtsp-media.c:
8452 media: check if prepared for some methods
8453 Check that the media object is prepared before doing seek and getting the
8454 current position etc.
8455 Add some g_return checks.
8457 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8459 * tests/check/Makefile.am:
8460 * tests/check/gst/mediafactory.c:
8461 tests: add mediafactory test
8463 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8465 * gst/rtsp-server/rtsp-stream.c:
8466 stream: improve debug
8468 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8470 * gst/rtsp-server/rtsp-media.c:
8471 * gst/rtsp-server/rtsp-media.h:
8472 media: unref pipeline in finalize to avoid leaking it
8474 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8476 * gst/rtsp-server/rtsp-media-factory-uri.c:
8477 * gst/rtsp-server/rtsp-media.c:
8478 rtsp: use gst_object_unref on GstObjects
8480 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8482 * gst/rtsp-server/rtsp-media-factory.c:
8483 media-factory: require an url
8485 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8487 * examples/test-uri.c:
8488 examples: fix include
8490 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8492 * gst/rtsp-server/rtsp-server.h:
8493 server: remove unused include
8495 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8497 * tests/check/Makefile.am:
8498 * tests/check/gst/mountpoints.c:
8499 tests: add test for mountpoints
8501 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8503 * gst/rtsp-server/rtsp-client.c:
8504 client: fix factory leak
8505 Keep the factory in the state object only for authorization checks and make
8506 sure we unref it on failure. Also don't keep invalid objects in the state
8509 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8511 * gst/rtsp-server/rtsp-mount-points.c:
8512 mounts: add g_return_if guards
8514 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8516 * tests/check/gst/client.c:
8517 tests: add more tests
8519 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8521 * gst/rtsp-server/rtsp-client.c:
8522 client: improve debug
8524 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8526 * gst/rtsp-server/rtsp-client.c:
8527 client: improve debug and fix leaks
8528 Cleanup the uri and session when there is a bad request.
8530 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8535 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8537 * tests/check/gst/client.c:
8538 test: add test for session in options request
8540 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8542 * gst/rtsp-server/rtsp-client.c:
8543 client: use 454 when session can't be found
8544 We should use 454 when a session can't be found because there was no session
8545 pool configured in the server. This is not a server configuration problem
8546 because the server on which the request is done might not be the same one that
8547 will keep the sessions for us and so it does not need to support sessions.
8549 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8551 * gst/rtsp-server/rtsp-client.c:
8552 client: only free connection when there is one
8553 It's possible that the client doesn't have a connection when we try to free it.
8555 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8557 * tests/check/Makefile.am:
8558 * tests/check/gst/client.c:
8559 tests: add unit test for the client object
8561 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8563 * gst/rtsp-server/rtsp-client.c:
8564 client: small cleanup
8566 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8568 * gst/rtsp-server/rtsp-client.h:
8569 client: remove unused include
8571 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8573 * gst/rtsp-server/rtsp-client.c:
8574 client: fix compilation
8576 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8578 * gst/rtsp-server/rtsp-client.c:
8579 client: call destroy without the lock
8581 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8583 * gst/rtsp-server/rtsp-client.c:
8584 * gst/rtsp-server/rtsp-client.h:
8585 client: make the client usable without a socket
8586 Make a method to let the client handle a message and a callback when the client
8587 wants us to send a response message back. This makes it possible to also use the
8588 client object without the sockets, which should make it easier to test.
8590 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8592 * gst/rtsp-server/rtsp-client.c:
8593 * gst/rtsp-server/rtsp-client.h:
8594 client: small cleanup
8596 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8598 * docs/libs/gst-rtsp-server-sections.txt:
8599 * gst/rtsp-server/rtsp-client.c:
8600 * gst/rtsp-server/rtsp-client.h:
8601 * gst/rtsp-server/rtsp-server.c:
8602 client: remove reference to server
8603 We don't need to keep a ref to the server
8605 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8607 * gst/rtsp-server/rtsp-client.c:
8608 * gst/rtsp-server/rtsp-client.h:
8610 Also add some g_return_if()
8612 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8614 * gst/rtsp-server/rtsp-client.c:
8615 client: log more errors
8617 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8619 * gst/rtsp-server/rtsp-client.c:
8620 client: fix compilation
8622 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8624 * gst/rtsp-server/rtsp-client.c:
8625 * gst/rtsp-server/rtsp-client.h:
8626 client: add generic close-after-send support
8627 Add a property to send_response() to close the connection after the response has
8628 been sent to the client.
8630 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8633 * docs/libs/gst-rtsp-server-docs.sgml:
8634 * docs/libs/gst-rtsp-server-sections.txt:
8635 * docs/libs/gst-rtsp-server.types:
8636 * examples/test-auth.c:
8637 * examples/test-launch.c:
8638 * examples/test-mp4.c:
8639 * examples/test-multicast.c:
8640 * examples/test-multicast2.c:
8641 * examples/test-ogg.c:
8642 * examples/test-readme.c:
8643 * examples/test-sdp.c:
8644 * examples/test-uri.c:
8645 * examples/test-video.c:
8646 * gst/rtsp-server/Makefile.am:
8647 * gst/rtsp-server/rtsp-auth.h:
8648 * gst/rtsp-server/rtsp-client.c:
8649 * gst/rtsp-server/rtsp-client.h:
8650 * gst/rtsp-server/rtsp-media-mapping.c:
8651 * gst/rtsp-server/rtsp-media-mapping.h:
8652 * gst/rtsp-server/rtsp-mount-points.c:
8653 * gst/rtsp-server/rtsp-mount-points.h:
8654 * gst/rtsp-server/rtsp-server.c:
8655 * gst/rtsp-server/rtsp-server.h:
8656 * gst/rtsp-server/rtsp-session-media.c:
8657 * gst/rtsp-server/rtsp-session-pool.c:
8658 * gst/rtsp-server/rtsp-session-pool.h:
8659 * tests/check/gst/rtspserver.c:
8660 MediaMapping -> MountPoints
8661 Describes better what the object manages.
8663 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8666 configure: bump required version of -base
8668 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8670 * gst/rtsp-server/rtsp-media.c:
8673 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8675 * gst/rtsp-server/rtsp-media.c:
8676 * gst/rtsp-server/rtsp-media.h:
8677 media: support more Range formats
8678 Use the new -base methods to convert the Range string into a seek start and stop
8681 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8683 * examples/test-launch.c:
8684 examples: fix whitespace
8686 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8688 * examples/test-auth.c:
8689 test-auth: add example of how to remove sessions
8690 Add an example of the session filter api.
8692 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8694 * examples/test-uri.c:
8695 test-uri: remove mapping example
8697 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8699 * examples/test-uri.c:
8700 test-uri: fix callback signature
8702 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8704 * gst/rtsp-server/rtsp-media-factory.c:
8705 factory: keep ref to factory while media active
8706 While the media from a factory is alive, keep a ref to the factory.
8707 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
8709 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8711 * gst/rtsp-server/rtsp-media-factory-uri.c:
8712 factory-uri: add some debug
8714 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8716 * gst/rtsp-server/rtsp-stream.c:
8717 stream: set udp sources to PLAYING
8718 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
8719 so that it doesn't cause our pipeline to produce ASYNC-DONE.
8721 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8723 * gst/rtsp-server/rtsp-media-factory-uri.c:
8724 factory-uri: take ref to factory
8725 Take a ref to the factory that we place in our list.
8727 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8729 * tests/Makefile.am:
8730 * tests/test-reuse.c:
8731 test: add test for server reuse
8732 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
8734 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
8736 * gst/rtsp-server/rtsp-server.c:
8737 server: start and stop multiple times
8738 Stop listening on the RTSP port when the GSource is removed, so clients
8739 can't connect and the server can be started again.
8740 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
8742 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8744 * gst/rtsp-server/rtsp-server.c:
8745 server: fix small leak
8747 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8749 * gst/rtsp-server/rtsp-media.c:
8750 media: unref source in finish_unprepare
8751 The source is created in prepare, unref it in finish_unprepare.
8752 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
8754 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
8756 * gst/rtsp-server/rtsp-client.c:
8757 * gst/rtsp-server/rtsp-media.c:
8758 rtsp-media: remove bus watch before finalizing
8759 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
8760 * An extra media ref is added for the bus watch. This extra ref is unreffed by
8761 the GDestroyNotify function.
8762 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
8763 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
8764 gst_rtsp_media_unprepare before unreffing the media.
8765 This way, the bus watch will be removed before the media is finalized.
8766 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
8768 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
8770 * gst/rtsp-server/rtsp-client.c:
8771 * gst/rtsp-server/rtsp-client.h:
8772 client: wait until the TEARDOWN response is sent to close the connection
8773 Responses can be sent async so we need to wait until the TEARDOWN response has
8774 been written before we close the connection to the client. This avoids the risk
8775 of writing/polling closed sockets.
8776 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
8778 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
8780 * gst/rtsp-server/rtsp-stream.c:
8781 rtsp-stream: plug socket leak
8782 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
8784 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
8787 Automatic update of common submodule
8788 From 6bb6951 to a72faea
8790 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
8792 * gst/rtsp-server/rtsp-media-factory-uri.c:
8793 rtsp-server: don't use deprecated API
8795 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
8797 * gst/rtsp-server/rtsp-client.c:
8798 rtsp-client: fix unused-but-set-variable compiler warning
8799 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
8801 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8804 * docs/libs/gst-rtsp-server-sections.txt:
8805 * gst/rtsp-server/rtsp-client.c:
8808 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8810 * examples/Makefile.am:
8811 * examples/test-multicast2.c:
8812 examples: add another multicast example
8813 Add an example for how to configure separate multicast ranges for each media
8816 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8818 * examples/test-multicast.c:
8821 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8823 * gst/rtsp-server/rtsp-client.c:
8824 * gst/rtsp-server/rtsp-media.c:
8825 * gst/rtsp-server/rtsp-session-media.c:
8826 * gst/rtsp-server/rtsp-session-media.h:
8827 * gst/rtsp-server/rtsp-stream-transport.c:
8828 * gst/rtsp-server/rtsp-stream-transport.h:
8829 stream: use the address managed by the stream
8830 Use the address managed by the stream for multicast. This allows us to have 1
8831 multicast address for each stream.
8832 Because the address is now managed by the stream we don't have to pass it around
8834 Set the address pool on the streams.
8836 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8838 * gst/rtsp-server/rtsp-client.c:
8839 * gst/rtsp-server/rtsp-media.c:
8840 * gst/rtsp-server/rtsp-stream.c:
8843 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8845 * gst/rtsp-server/rtsp-media.c:
8846 * gst/rtsp-server/rtsp-media.h:
8847 media: add signal for new streams
8848 This allows applications to listen for new streams and configure properties on
8849 them, like the address pool.
8851 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8853 * gst/rtsp-server/rtsp-media.c:
8854 media: configure address pool in new streams
8856 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8858 * gst/rtsp-server/rtsp-stream.c:
8859 * gst/rtsp-server/rtsp-stream.h:
8860 stream: add methods to deal with address pool
8861 Add methods to get and set the address pool for the stream
8862 Add method to allocate and get the multicast addresses for this stream.
8864 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8866 * docs/libs/gst-rtsp-server-sections.txt:
8867 * gst/rtsp-server/rtsp-media.c:
8868 * gst/rtsp-server/rtsp-media.h:
8869 media: remove MTU property
8870 It is a stream property
8872 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8874 * gst/rtsp-server/rtsp-client.c:
8875 client: set blocksize only on stream
8876 Set the blocksize only on the current stream.
8878 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8880 * gst/rtsp-server/rtsp-stream.c:
8881 stream: share src and sink sockets
8882 the allocated socket is in the used-socket property, not socket.
8884 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8886 * gst/rtsp-server/rtsp-address-pool.c:
8887 * gst/rtsp-server/rtsp-address-pool.h:
8888 * gst/rtsp-server/rtsp-client.c:
8889 * gst/rtsp-server/rtsp-session-media.c:
8890 * gst/rtsp-server/rtsp-session-media.h:
8891 * gst/rtsp-server/rtsp-stream-transport.c:
8892 * gst/rtsp-server/rtsp-stream-transport.h:
8893 * tests/check/gst/addresspool.c:
8894 rtsp: make address-pool return an address object
8895 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
8896 store more info in the structure and allows us to more easily return the address
8897 to the right pool when no longer needed.
8898 Pass the address to the StreamTransport so that we can return it to the pool
8899 when the stream transport is freed or changed.
8901 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8903 * examples/Makefile.am:
8904 * examples/test-multicast.c:
8905 examples: add multicast example
8906 Show how to set up the multicast address pool so that media can be
8907 server with multicast.
8909 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8911 * gst/rtsp-server/rtsp-client.c:
8912 * gst/rtsp-server/rtsp-media-factory.c:
8913 * gst/rtsp-server/rtsp-media-factory.h:
8914 * gst/rtsp-server/rtsp-media.c:
8915 * gst/rtsp-server/rtsp-media.h:
8916 rtsp: use AddressPool
8917 Remove the multicast_group property.
8918 Use the configured addresspool to allocate multicast addresses.
8920 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8922 * gst/rtsp-server/rtsp-address-pool.c:
8923 * gst/rtsp-server/rtsp-address-pool.h:
8924 address-pool: add clear method
8926 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8928 * gst/rtsp-server/rtsp-address-pool.c:
8929 address-pool: small cleanups
8931 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8933 * tests/check/Makefile.am:
8934 * tests/check/gst/addresspool.c:
8935 tests: add addresspool unit test
8937 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8939 * gst/rtsp-server/Makefile.am:
8940 * gst/rtsp-server/rtsp-address-pool.c:
8941 * gst/rtsp-server/rtsp-address-pool.h:
8942 address-pool: add object to manage multicast addresses
8943 Make an object that can manage a rage of multicast addresses and ports.
8945 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8947 * gst/rtsp-server/rtsp-server.c:
8948 server: set default max-threads property
8950 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8952 * gst/rtsp-server/rtsp-media.c:
8953 media: wait for concurrent _prepare
8954 If a prepare is busy, wait for the result.
8956 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8958 * gst/rtsp-server/rtsp-media.c:
8959 media: add lock around message handler
8960 We don't want to dispatch messages while we are still processing the result of
8963 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8965 * gst/rtsp-server/rtsp-media.c:
8966 * gst/rtsp-server/rtsp-media.h:
8967 media: add lock to protect state changes
8969 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8971 * gst/rtsp-server/rtsp-stream.c:
8972 * gst/rtsp-server/rtsp-stream.h:
8975 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8977 * gst/rtsp-server/rtsp-stream-transport.c:
8978 * gst/rtsp-server/rtsp-stream-transport.h:
8979 * gst/rtsp-server/rtsp-stream.c:
8980 stream-transport: add keep-alive method
8982 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8984 * gst/rtsp-server/rtsp-stream-transport.c:
8985 * gst/rtsp-server/rtsp-stream-transport.h:
8986 * gst/rtsp-server/rtsp-stream.c:
8987 stream-transport: add method to handle RTP/RTCP
8988 Call new methods instead of poking into the structures directly.
8990 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8992 * gst/rtsp-server/rtsp-session-media.c:
8993 * gst/rtsp-server/rtsp-session-media.h:
8994 session-media: add locking
8996 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8998 * gst/rtsp-server/rtsp-session.c:
8999 * gst/rtsp-server/rtsp-session.h:
9000 session: add locking
9002 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9004 * gst/rtsp-server/rtsp-server.c:
9005 server: free old socket
9007 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9009 * gst/rtsp-server/rtsp-media-mapping.c:
9010 * gst/rtsp-server/rtsp-media-mapping.h:
9011 mapping: add locking
9013 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9015 * gst/rtsp-server/rtsp-media-factory.c:
9016 media-factory: add locking
9018 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9020 * gst/rtsp-server/rtsp-auth.c:
9021 * gst/rtsp-server/rtsp-auth.h:
9024 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9026 * gst/rtsp-server/rtsp-server.c:
9027 * gst/rtsp-server/rtsp-server.h:
9028 server: add max-thread property
9030 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9032 * gst/rtsp-server/rtsp-server.c:
9033 * gst/rtsp-server/rtsp-server.h:
9034 server: use a threadpool for the mainloops
9036 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9038 * gst/rtsp-server/rtsp-client.c:
9039 * gst/rtsp-server/rtsp-client.h:
9040 client: rename method
9041 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
9042 don't really create the client from the socket, we use the socket for the
9045 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9047 * gst/rtsp-server/rtsp-client.c:
9048 * gst/rtsp-server/rtsp-client.h:
9049 * gst/rtsp-server/rtsp-server.c:
9050 server: rework maincontext handling in clients
9051 Make a separate method to attach a client to a MainContext.
9052 Let the server decide in what GMainContext the client will operate and give this
9053 context to the client in attach. Then the server can later decide to use a
9054 separate thread for each client or just use the mainthread.
9056 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9058 * gst/rtsp-server/rtsp-client.c:
9059 * gst/rtsp-server/rtsp-session.c:
9060 * gst/rtsp-server/rtsp-session.h:
9061 session: move session header code in session object
9063 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
9067 * examples/test-auth.c:
9068 * examples/test-launch.c:
9069 * examples/test-mp4.c:
9070 * examples/test-ogg.c:
9071 * examples/test-readme.c:
9072 * examples/test-sdp.c:
9073 * examples/test-uri.c:
9074 * examples/test-video.c:
9075 * gst/rtsp-server/rtsp-auth.c:
9076 * gst/rtsp-server/rtsp-auth.h:
9077 * gst/rtsp-server/rtsp-client.c:
9078 * gst/rtsp-server/rtsp-client.h:
9079 * gst/rtsp-server/rtsp-media-factory-uri.c:
9080 * gst/rtsp-server/rtsp-media-factory-uri.h:
9081 * gst/rtsp-server/rtsp-media-factory.c:
9082 * gst/rtsp-server/rtsp-media-factory.h:
9083 * gst/rtsp-server/rtsp-media-mapping.c:
9084 * gst/rtsp-server/rtsp-media-mapping.h:
9085 * gst/rtsp-server/rtsp-media.c:
9086 * gst/rtsp-server/rtsp-media.h:
9087 * gst/rtsp-server/rtsp-params.c:
9088 * gst/rtsp-server/rtsp-params.h:
9089 * gst/rtsp-server/rtsp-sdp.c:
9090 * gst/rtsp-server/rtsp-sdp.h:
9091 * gst/rtsp-server/rtsp-server.c:
9092 * gst/rtsp-server/rtsp-server.h:
9093 * gst/rtsp-server/rtsp-session-media.c:
9094 * gst/rtsp-server/rtsp-session-media.h:
9095 * gst/rtsp-server/rtsp-session-pool.c:
9096 * gst/rtsp-server/rtsp-session-pool.h:
9097 * gst/rtsp-server/rtsp-session.c:
9098 * gst/rtsp-server/rtsp-session.h:
9099 * gst/rtsp-server/rtsp-stream-transport.c:
9100 * gst/rtsp-server/rtsp-stream-transport.h:
9101 * gst/rtsp-server/rtsp-stream.c:
9102 * gst/rtsp-server/rtsp-stream.h:
9103 * tests/check/gst/rtspserver.c:
9104 * tests/test-cleanup.c:
9107 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9109 * gst/rtsp-server/rtsp-media.c:
9110 * gst/rtsp-server/rtsp-session-media.c:
9111 * gst/rtsp-server/rtsp-session.c:
9112 rtsp-server: added annotations to indicate type of ownership transfer of return values
9113 https://bugzilla.gnome.org/show_bug.cgi?id=680777
9115 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
9118 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
9120 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
9123 * bindings/Makefile.am:
9124 * bindings/vala/Makefile.am:
9125 * bindings/vala/gst-rtsp-server-0.10.deps:
9126 * bindings/vala/gst-rtsp-server-0.10.vapi:
9127 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
9128 * bindings/vala/packages/gst-rtsp-server-0.10.files:
9129 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9130 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9131 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
9133 bindings: remove vala bindings
9134 They'll be reunited with the other GStreamer bindings
9135 https://bugzilla.gnome.org/show_bug.cgi?id=680777
9137 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9139 * gst/rtsp-server/rtsp-client.c:
9140 * gst/rtsp-server/rtsp-session-media.c:
9141 * gst/rtsp-server/rtsp-session-media.h:
9142 * gst/rtsp-server/rtsp-stream-transport.c:
9143 * gst/rtsp-server/rtsp-stream-transport.h:
9144 rtsp: only create transport when needed
9145 Only create the StreamTransport when configured.
9147 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9149 * gst/rtsp-server/rtsp-client.c:
9150 client: small cleanup
9152 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9154 * gst/rtsp-server/rtsp-client.c:
9155 * gst/rtsp-server/rtsp-client.h:
9156 * gst/rtsp-server/rtsp-stream-transport.c:
9157 * gst/rtsp-server/rtsp-stream-transport.h:
9158 rtsp: refactor configuration of transport
9159 Move the configuration of the transport to a place where it makes
9162 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9164 * gst/rtsp-server/rtsp-client.c:
9165 client: refactor transport parsing
9167 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9169 * gst/rtsp-server/rtsp-client.c:
9170 client: refuse to change the MTU on shared media
9171 If we change the MTU of chared media, it changes for all clients.
9172 We don't want to set the MTU to something large for clients that
9175 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9177 * examples/test-mp4.c:
9178 * gst/rtsp-server/rtsp-media.c:
9179 small fixes to docs and debug
9181 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9183 * gst/rtsp-server/rtsp-stream.c:
9184 stream: transports must already have been removed
9186 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9188 * gst/rtsp-server/rtsp-media.c:
9189 * gst/rtsp-server/rtsp-stream.c:
9190 * gst/rtsp-server/rtsp-stream.h:
9191 stream: improve join and leave of the pipeline
9193 Do the cleanup properly
9196 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9198 * gst/rtsp-server/rtsp-media.c:
9199 media: move unprepare below default implementation
9200 Makes it easier to find the default implementation
9202 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9204 * gst/rtsp-server/rtsp-media.c:
9205 media: signal unprepared when we actually finish
9207 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9209 * gst/rtsp-server/rtsp-media.c:
9210 media: no need to unlock, unprepare does that when needed
9212 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9214 * docs/libs/gst-rtsp-server-sections.txt:
9215 * gst/rtsp-server/rtsp-media-factory.h:
9216 * gst/rtsp-server/rtsp-media-mapping.c:
9217 * gst/rtsp-server/rtsp-media.h:
9218 * gst/rtsp-server/rtsp-params.c:
9219 * gst/rtsp-server/rtsp-server.c:
9220 * gst/rtsp-server/rtsp-session-pool.h:
9221 * gst/rtsp-server/rtsp-session.c:
9222 * gst/rtsp-server/rtsp-session.h:
9223 * gst/rtsp-server/rtsp-stream-transport.h:
9224 * gst/rtsp-server/rtsp-stream.h:
9227 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9229 * gst/rtsp-server/rtsp-client.c:
9230 * gst/rtsp-server/rtsp-media-mapping.h:
9231 * gst/rtsp-server/rtsp-media.c:
9232 * gst/rtsp-server/rtsp-media.h:
9233 * gst/rtsp-server/rtsp-server.h:
9234 * gst/rtsp-server/rtsp-stream.c:
9235 * gst/rtsp-server/rtsp-stream.h:
9236 rtsp: fix MTU setting
9237 Fix setting of the MTU. There is no need for a vmethod.
9239 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9244 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9247 configure: bump version number after refactoring
9249 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9251 * gst/rtsp-server/Makefile.am:
9252 * gst/rtsp-server/rtsp-client.c:
9253 * gst/rtsp-server/rtsp-client.h:
9254 * gst/rtsp-server/rtsp-media-factory-uri.c:
9255 * gst/rtsp-server/rtsp-media-factory.c:
9256 * gst/rtsp-server/rtsp-media-factory.h:
9257 * gst/rtsp-server/rtsp-media.c:
9258 * gst/rtsp-server/rtsp-media.h:
9259 * gst/rtsp-server/rtsp-sdp.c:
9260 * gst/rtsp-server/rtsp-session-media.c:
9261 * gst/rtsp-server/rtsp-session-media.h:
9262 * gst/rtsp-server/rtsp-session.c:
9263 * gst/rtsp-server/rtsp-session.h:
9264 * gst/rtsp-server/rtsp-stream-transport.c:
9265 * gst/rtsp-server/rtsp-stream-transport.h:
9266 * gst/rtsp-server/rtsp-stream.c:
9267 * gst/rtsp-server/rtsp-stream.h:
9268 rtsp: massive refactoring
9269 Make GObjects from the remaining simple structures.
9270 Remove GstRTSPSessionStream, it's not needed.
9271 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
9272 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
9273 a GstRTSPStream should be transported to a client.
9274 Rename GstRTSPMediaFactory::get_element -> create_element because that
9275 more accurately describes what it does.
9276 Make nice methods instead of poking in the structures.
9277 Move some methods inside the relevant object source code.
9278 Use GPtrArray to store objects instead of plain arrays, it is more
9279 natural and allows us to more easily clean up.
9280 Move the allocation of udp ports to the Stream object. The Stream object
9281 contains the elements needed to stream the media to a client.
9282 Improve the prepare and unprepare methods. Unprepare should now undo
9283 everything prepare did. Improve also async unprepare when doing EOS on
9284 shutdown. Make sure we always unprepare correctly.
9286 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
9288 * gst/rtsp-server/rtsp-client.c:
9289 rtsp-client: Unref server address clients connected to
9290 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
9292 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
9294 * gst/rtsp-server/rtsp-server.c:
9295 rtsp-server: don't ref server socket if it is NULL
9296 Fixes test_bind_already_in_use unit test again after commit 6a497440.
9297 https://bugzilla.gnome.org/show_bug.cgi?id=686644
9299 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
9301 * tests/check/Makefile.am:
9302 tests: Add libgio link dependency
9303 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
9305 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9307 * gst/rtsp-server/rtsp-media-mapping.c:
9308 * gst/rtsp-server/rtsp-media-mapping.h:
9309 rtsp-media-mapping: rename find_media vfunc to find_factory
9310 The virtual method and class method should have the same name
9311 so it is correctly represented in GIR file
9312 https://bugzilla.gnome.org/show_bug.cgi?id=680777
9314 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9316 * gst/rtsp-server/rtsp-auth.c:
9317 * gst/rtsp-server/rtsp-client.c:
9318 * gst/rtsp-server/rtsp-media-factory-uri.c:
9319 * gst/rtsp-server/rtsp-media-factory.c:
9320 * gst/rtsp-server/rtsp-media-mapping.c:
9321 * gst/rtsp-server/rtsp-media.c:
9322 * gst/rtsp-server/rtsp-server.c:
9323 * gst/rtsp-server/rtsp-session-pool.c:
9324 * gst/rtsp-server/rtsp-session.c:
9325 rtsp-server: fixed comments and GIR annotations
9326 https://bugzilla.gnome.org/show_bug.cgi?id=680777
9328 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
9330 * gst/rtsp-server/rtsp-media-mapping.c:
9331 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
9333 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
9335 * gst/rtsp-server/rtsp-server.c:
9336 rtsp-server: allow binding on port 0 (binds on a random port)
9338 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
9340 * gst/rtsp-server/rtsp-server.c:
9341 * gst/rtsp-server/rtsp-server.h:
9342 rtsp-server: add bound-port property
9343 bound-port can be used to retrieve the port number when the server is bound on
9344 port 0, which binds on a random port.
9346 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
9348 * gst/rtsp-server/rtsp-media-factory.c:
9349 * gst/rtsp-server/rtsp-media-factory.h:
9350 rtsp-media-factory: make ::get_element overridable by GI bindings
9351 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
9352 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
9353 as the invoker for ::get_element(), making it overridable by GI generated
9356 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
9358 * gst/rtsp-server/rtsp-media-factory-uri.c:
9359 rtsp-media-factory-uri: don't autoplug parsers in a loop
9360 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
9363 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
9365 * gst/rtsp-server/Makefile.am:
9366 Explicitly link against gio. Fix link error on mac.
9368 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
9370 * gst/rtsp-server/rtsp-session.c:
9371 session: add ttl to the transport header in SETUP
9372 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
9374 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
9376 * gst/rtsp-server/rtsp-client.c:
9377 * gst/rtsp-server/rtsp-client.h:
9378 * gst/rtsp-server/rtsp-media.c:
9379 client: Use client transport settings for multicast if allowed.
9380 This patch makes it possible for the client to send transport settings for
9381 multicast (destination && ttl). Client settings must be explicitly allowed or
9382 the server will use its own settings.
9383 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
9385 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
9388 Automatic update of common submodule
9389 From 6c0b52c to 6bb6951
9391 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
9393 * gst/rtsp-server/rtsp-client.c:
9394 rtsp-client: do not destroy the rtsp watch
9395 Don't destroy the client watch while dispatching. The rtsp watch is
9396 automatically destroyed after the rtsp watch function closed() has
9398 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
9400 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
9403 Automatic update of common submodule
9404 From 4f962f7 to 6c0b52c
9406 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
9408 * gst/rtsp-server/rtsp-media.c:
9409 media: fix check for seekability
9411 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9413 * gst/rtsp-server/rtsp-client.c:
9414 client: use more GIO
9415 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
9417 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9419 * gst/rtsp-server/rtsp-server.c:
9420 server: remove obsolete includes
9422 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
9424 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
9425 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
9426 be available in "on_new_ssrc". The transports are added in
9427 gst_rtsp_media_set_state when going to PLAYING state. However,
9428 "on_new_ssrc" might be called before this happens.
9429 https://bugzilla.gnome.org/show_bug.cgi?id=683304
9431 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
9433 * gst/rtsp-server/rtsp-client.c:
9434 * gst/rtsp-server/rtsp-client.h:
9435 rtsp-client: add signals for rtsp requests (fixes #683287)
9437 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
9439 * gst/rtsp-server/rtsp-client.c:
9440 * gst/rtsp-server/rtsp-client.h:
9441 add new-session signal to rtsp-client (fixes #683058)
9443 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
9446 Automatic update of common submodule
9447 From 668acee to 4f962f7
9449 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
9451 * gst/rtsp-server/rtsp-server.c:
9452 * tests/check/gst/rtspserver.c:
9453 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
9454 Do not assume that *error is set in g_socket_address_enumerator_next.
9455 Added test_bind_already_in_use unit-test.
9456 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
9458 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
9461 Automatic update of common submodule
9462 From 94ccf4c to 668acee
9464 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
9466 * gst/rtsp-server/rtsp-client.c:
9467 * gst/rtsp-server/rtsp-client.h:
9468 rtsp-client: make create_sdp virtual method
9469 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
9471 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9474 Automatic update of common submodule
9475 From 98e386f to 94ccf4c
9477 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9479 * gst/rtsp-server/rtsp-client.c:
9482 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
9484 * gst/rtsp-server/rtsp-client.c:
9485 * gst/rtsp-server/rtsp-client.h:
9486 * gst/rtsp-server/rtsp-server.c:
9487 * gst/rtsp-server/rtsp-server.h:
9488 rtsp-server: use an existing socket to establish HTTP tunnel
9489 Make it possible to transfer a socket from an HTTP server to be used as
9490 an RTSP over HTTP tunnel.
9492 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
9494 * gst/rtsp-server/rtsp-client.c:
9495 * gst/rtsp-server/rtsp-media.c:
9496 * gst/rtsp-server/rtsp-media.h:
9497 rtsp: Handle the blocksize parameter
9498 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
9500 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
9502 * tests/check/Makefile.am:
9503 * tests/check/gst/rtspserver.c:
9504 Have unit test get header from source dir, not installed dir
9505 This makes compilation of unit tests work in a build directory other
9506 than the source directory.
9507 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
9509 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
9511 * gst/rtsp-server/rtsp-media.c:
9512 rtsp-media: update for gst_element_make_from_uri() changes
9514 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
9517 * tests/Makefile.am:
9518 * tests/check/Makefile.am:
9519 * tests/check/gst/rtspserver.c:
9521 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
9523 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
9525 * gst/rtsp-server/rtsp-media.c:
9526 rtsp-media: don't collect media stats when going to NULL
9527 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
9529 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9531 * gst/rtsp-server/rtsp-client.c:
9532 client: don't leak transports
9534 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
9536 * gst/rtsp-server/rtsp-client.c:
9537 rtsp-client: free transport on no_stream in SETUP handler
9539 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
9541 * gst/rtsp-server/rtsp-client.c:
9542 rtsp-client: changed session media iteration
9543 In client_unlink_session: now don't iterate in session->medias
9544 list where items are removed by gst_rtsp_session_release_media.
9545 Instead, repeatedly remove the first item.
9547 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
9549 * gst/rtsp-server/rtsp-client.c:
9550 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
9551 GstRTSPSessionMedia is not a GObject type. When the
9552 GstRTSPSession is freed, it will free the media.
9554 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
9556 * gst/rtsp-server/rtsp-media-factory.c:
9557 factory: plug pad leak in collect_streams
9558 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
9559 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
9560 will take one reference, and the other reference will otherwise
9563 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
9566 configure: suppress some warnings when debug is disabled
9567 Warnings about unused variables should be suppressed if core has the
9568 debug system disabled.
9569 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
9571 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9573 * docs/libs/Makefile.am:
9574 docs: fix build in uninstalled setup
9575 Include gst-plugins-base libs properly.
9577 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
9579 * docs/libs/gst-rtsp-server.types:
9580 docs: include headers defining rtsp-server object types
9581 Fixes compiler warnings during docs build.
9582 https://bugzilla.gnome.org/show_bug.cgi?id=676824
9584 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
9587 configure: Add warning flags for compiler when configuring
9588 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
9590 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
9593 Automatic update of common submodule
9594 From 03a0e57 to 98e386f
9596 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
9599 Automatic update of common submodule
9600 From 1fab359 to 03a0e57
9602 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
9604 * gst/rtsp-server/rtsp-client.c:
9605 client: fix GSocketAddress leak in gst_rtsp_client_accept
9606 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
9608 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
9611 Automatic update of common submodule
9612 From f1b5a96 to 1fab359
9614 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9617 Automatic update of common submodule
9618 From 92b7266 to f1b5a96
9620 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9623 Automatic update of common submodule
9624 From ec1c4a8 to 92b7266
9626 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9629 Automatic update of common submodule
9630 From 3429ba6 to ec1c4a8
9632 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
9634 * gst/rtsp-server/rtsp-auth.c:
9635 * gst/rtsp-server/rtsp-client.c:
9636 * gst/rtsp-server/rtsp-media-factory-uri.c:
9637 * gst/rtsp-server/rtsp-server.c:
9638 rtsp: fix compiler warnings
9639 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
9641 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9644 Automatic update of common submodule
9645 From dc70203 to 3429ba6
9647 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9649 * gst/rtsp-server/rtsp-client.c:
9650 * gst/rtsp-server/rtsp-media-factory.c:
9651 * gst/rtsp-server/rtsp-media-factory.h:
9652 * gst/rtsp-server/rtsp-media.c:
9653 * gst/rtsp-server/rtsp-media.h:
9654 * gst/rtsp-server/rtsp-server.c:
9655 * gst/rtsp-server/rtsp-server.h:
9656 * gst/rtsp-server/rtsp-session-pool.c:
9657 * gst/rtsp-server/rtsp-session-pool.h:
9658 rtsp-server: port to new thread API
9660 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9663 Automatic update of common submodule
9664 From 6db25be to dc70203
9666 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9668 * gst/rtsp-server/rtsp-auth.c:
9669 * gst/rtsp-server/rtsp-auth.h:
9670 * gst/rtsp-server/rtsp-client.c:
9671 rtsp-server: Fix compilation and compiler warnings
9673 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9677 * gst/rtsp-server/Makefile.am:
9678 configure: Modernize autotools setup a bit
9679 Also we now only create tar.bz2 and tar.xz tarballs.
9681 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9684 Automatic update of common submodule
9685 From 464fe15 to 6db25be
9687 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9690 Automatic update of common submodule
9691 From 7fda524 to 464fe15
9693 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9696 * docs/libs/Makefile.am:
9697 * docs/version.entities.in:
9699 * gst/rtsp-server/Makefile.am:
9700 * pkgconfig/Makefile.am:
9701 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
9702 * pkgconfig/gstreamer-rtsp-server.pc.in:
9703 * tests/Makefile.am:
9704 rtsp-server: Update versioning
9706 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9708 Merge remote-tracking branch 'origin/0.10'
9710 gst/rtsp-server/rtsp-session-pool.c
9712 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9714 * gst/rtsp-server/rtsp-session-pool.c:
9715 rtsp-server: Don't use deprecated GLib API
9717 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9719 Replace master with 0.11
9721 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9723 Merge branch 'master' into 0.11
9725 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9727 Merge branch 'master' into 0.11
9729 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
9732 A couple minor typo fixes
9734 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9736 * gst/rtsp-server/rtsp-media.c:
9737 media: fix state of the appqueue
9739 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9741 * gst/rtsp-server/rtsp-media-factory-uri.c:
9742 factory: use videoconvert
9744 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9746 * gst/rtsp-server/rtsp-media-factory-uri.c:
9747 factory: change to new style caps
9749 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9751 * gst/rtsp-server/rtsp-client.c:
9752 * gst/rtsp-server/rtsp-client.h:
9753 * gst/rtsp-server/rtsp-media-factory-uri.c:
9754 * gst/rtsp-server/rtsp-media.c:
9755 * gst/rtsp-server/rtsp-server.c:
9756 * gst/rtsp-server/rtsp-server.h:
9757 * gst/rtsp-server/rtsp-session-pool.c:
9758 rtsp-server: port to GIO
9761 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9764 configure: fix build
9766 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9769 docs: fix for gst_rtsp_server_set_port() -> _set_service()
9770 https://bugzilla.gnome.org/show_bug.cgi?id=666548
9772 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9775 * examples/Makefile.am:
9776 First rule of gst-rtsp-server club: don't talk about gst-phonon
9778 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9781 * pkgconfig/Makefile.am:
9782 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
9783 * pkgconfig/gstreamer-rtsp-server.pc.in:
9784 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
9785 For consistency with all other modules.
9787 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9789 * gst/rtsp-server/rtsp-client.c:
9790 rtsp-client: update for new map API
9792 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9795 * bindings/Makefile.am:
9796 * bindings/python/Makefile.am:
9797 * bindings/python/arg-types.py:
9798 * bindings/python/codegen/Makefile.am:
9799 * bindings/python/codegen/__init__.py:
9800 * bindings/python/codegen/argtypes.py:
9801 * bindings/python/codegen/code-coverage.py:
9802 * bindings/python/codegen/codegen.py:
9803 * bindings/python/codegen/definitions.py:
9804 * bindings/python/codegen/defsparser.py:
9805 * bindings/python/codegen/docextract.py:
9806 * bindings/python/codegen/docgen.py:
9807 * bindings/python/codegen/fileprefix.override:
9808 * bindings/python/codegen/fileprefixmodule.c:
9809 * bindings/python/codegen/h2def.py:
9810 * bindings/python/codegen/mergedefs.py:
9811 * bindings/python/codegen/mkskel.py:
9812 * bindings/python/codegen/override.py:
9813 * bindings/python/codegen/reversewrapper.py:
9814 * bindings/python/codegen/scmexpr.py:
9815 * bindings/python/rtspserver-types.defs:
9816 * bindings/python/rtspserver.defs:
9817 * bindings/python/rtspserver.override:
9818 * bindings/python/rtspservermodule.c:
9819 * bindings/python/test.py:
9821 python: remove pygst-based python bindings
9822 pygi is the future, apparently.
9824 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
9827 Automatic update of common submodule
9828 From c463bc0 to 7fda524
9830 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9833 Automatic update of common submodule
9834 From 2a59016 to c463bc0
9836 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9839 Automatic update of common submodule
9840 From 0807187 to 2a59016
9842 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9845 Automatic update of common submodule
9846 From 11f0cd5 to 0807187
9848 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9850 * examples/test-auth.c:
9851 example: update for new caps
9853 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9855 * examples/test-video.c:
9856 * gst/rtsp-server/rtsp-client.c:
9857 * gst/rtsp-server/rtsp-media-factory-uri.c:
9858 * gst/rtsp-server/rtsp-media.c:
9859 * gst/rtsp-server/rtsp-media.h:
9860 * gst/rtsp-server/rtsp-session.c:
9861 * gst/rtsp-server/rtsp-session.h:
9862 rtsp-server: port some more to 0.11
9864 Remove bufferlist stuff
9866 Add queue before appsink now that preroll-queue-len is gone.
9867 Update for request pad changes.
9869 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9871 Merge branch 'master' into 0.11
9873 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
9875 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9876 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
9877 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
9879 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
9881 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9882 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
9883 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
9885 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9887 Merge branch 'master' into 0.11
9889 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9891 * gst/rtsp-server/rtsp-media.c:
9892 * gst/rtsp-server/rtsp-media.h:
9893 media: add a seekable boolean
9894 Maintain the seekable state with a new variable instead of reusing the
9897 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
9899 * gst/rtsp-server/rtsp-media.c:
9900 Disallow seek in live media
9902 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9904 Merge branch 'master' into 0.11
9906 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
9908 * gst/rtsp-server/rtsp-server.c:
9909 #ifdef statements for windows socket creation were missing
9911 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
9914 Automatic update of common submodule
9915 From a39eb83 to 11f0cd5
9917 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
9920 Automatic update of common submodule
9921 From 605cd9a to a39eb83
9923 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9925 Merge branch 'master' into 0.11
9927 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9929 * gst/rtsp-server/rtsp-client.c:
9930 client: use method to access property
9932 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9934 * gst/rtsp-server/rtsp-media-factory.c:
9935 * gst/rtsp-server/rtsp-media-factory.h:
9936 media-factory: add protocols property
9937 Add a property to configure the allowed protocols in the media created from the
9940 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9942 * gst/rtsp-server/rtsp-media-factory.c:
9943 * gst/rtsp-server/rtsp-media-factory.h:
9944 media-factory: add media-configure signal
9945 Add signal to allow the application to configure the media after it was created
9948 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9950 * gst/rtsp-server/rtsp-client.c:
9951 client: use method to access property
9953 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9955 * gst/rtsp-server/rtsp-media-factory.c:
9956 * gst/rtsp-server/rtsp-media-factory.h:
9957 media-factory: add protocols property
9958 Add a property to configure the allowed protocols in the media created from the
9961 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9963 * gst/rtsp-server/rtsp-media-factory.c:
9964 * gst/rtsp-server/rtsp-media-factory.h:
9965 media-factory: add media-configure signal
9966 Add signal to allow the application to configure the media after it was created
9969 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9971 Merge branch 'master' into 0.11
9973 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9975 * gst/rtsp-server/rtsp-client.c:
9976 client: use media multicast group
9978 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9980 * gst/rtsp-server/rtsp-media-factory.h:
9981 * gst/rtsp-server/rtsp-server.h:
9982 * gst/rtsp-server/rtsp-session-pool.h:
9983 * gst/rtsp-server/rtsp-session.h:
9986 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
9988 * gst/rtsp-server/rtsp-client.c:
9989 * gst/rtsp-server/rtsp-sdp.h:
9990 sdp: copy and free the server ip address
9991 Copy and free the server ip address to make memory management easier later.
9993 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9995 * gst/rtsp-server/rtsp-media-factory.c:
9996 media-factory: configure multicast in media
9998 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10000 * gst/rtsp-server/rtsp-media.c:
10001 * gst/rtsp-server/rtsp-media.h:
10002 media: add property for multicast group
10003 Add a property to configure the multicast group in the media.
10004 Based on patches from Marc Leeman and Robert Krakora.
10006 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10008 * gst/rtsp-server/rtsp-media-factory.c:
10009 * gst/rtsp-server/rtsp-media-factory.h:
10010 media-factory: add property for multicast group
10011 Add a property to configure the multicast group in the media factory.
10012 Based on patches from Marc Leeman and Robert Krakora.
10014 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10016 * gst/rtsp-server/rtsp-client.c:
10017 client: do configuration of transport in one place
10018 Move the configuration of the transport destination address to where we also
10019 configure the other bits.
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 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10071 Merge branch 'master' into 0.11
10073 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
10075 * gst/rtsp-server/rtsp-client.c:
10076 client: destroy pipeline on client disconnect with no prior TEARDOWN.
10077 The problem occurs when the client abruptly closes the connection without
10078 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
10079 server is where the pipeline gets torn down. Since this handler is not called,
10080 the pipeline remains and is up and running. Subsequent clients get their own
10081 pipelines and if the do not issue TEARDOWNs then those pipelines will also
10082 remain up and running. This is a resource leak.
10084 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10086 Merge branch 'master' into 0.11
10088 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
10090 * gst/rtsp-server/rtsp-media-factory.c:
10091 * gst/rtsp-server/rtsp-media-factory.h:
10092 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
10093 For example, it can be used to retrieve source elements like appsrc, in a more
10094 convenient way than subclassing get_element.
10096 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10098 Merge branch 'master' into 0.11
10100 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
10102 * gst/rtsp-server/rtsp-server.c:
10103 rtsp-server: hold on to reference while using object
10105 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10107 * gst/rtsp-server/rtsp-media.c:
10110 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10113 configure: use unstable api
10115 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
10117 * gst/rtsp-server/rtsp-client.c:
10118 client: fix reference counting
10120 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
10122 * gst/rtsp-server/rtsp-client.c:
10123 * gst/rtsp-server/rtsp-media.c:
10124 fix compiler warnings about unused variables
10126 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
10128 * examples/test-launch.c:
10129 * examples/test-readme.c:
10130 * examples/test-uri.c:
10131 * examples/test-video.c:
10132 examples: tell rtsp uri when ready
10134 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
10137 Automatic update of common submodule
10138 From 69b981f to 605cd9a
10140 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10142 * gst/rtsp-server/rtsp-client.c:
10143 client: update for buffer API change
10145 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
10147 * gst/rtsp-server/Makefile.am:
10148 Makefile.am: 0.10 => @GST_MAJORMINOR@
10150 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
10152 * gst/rtsp-server/rtsp-media-factory-uri.c:
10153 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
10155 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
10157 * gst/rtsp-server/.gitignore:
10158 .gitignore: 0.10 => 0.11
10160 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
10162 * gst/rtsp-server/Makefile.am:
10163 Makefile.am: 0.10 => @GST_MAJORMINOR@
10165 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10167 Merge branch 'master' into 0.11
10169 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
10172 Automatic update of common submodule
10173 From 9e5bbd5 to 69b981f
10175 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
10178 Automatic update of common submodule
10179 From fd35073 to 9e5bbd5
10181 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
10184 Automatic update of common submodule
10185 From 46dfcea to fd35073
10187 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10189 * gst/rtsp-server/rtsp-media-factory-uri.c:
10190 * gst/rtsp-server/rtsp-media.c:
10191 media: port to new caps API
10193 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10195 Merge branch 'master' into 0.11
10197 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
10199 * bindings/vala/gst-rtsp-server-0.10.vapi:
10200 Updated Vala bindings.
10201 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
10203 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
10205 * gst/rtsp-server/rtsp-server.c:
10206 * gst/rtsp-server/rtsp-server.h:
10207 Add a signal for newly connected clients.
10208 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
10210 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
10212 * bindings/python/rtspserver.override:
10213 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
10215 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10217 * gst/rtsp-server/Makefile.am:
10218 * gst/rtsp-server/rtsp-client.c:
10219 * gst/rtsp-server/rtsp-funnel.c:
10220 * gst/rtsp-server/rtsp-funnel.h:
10221 * gst/rtsp-server/rtsp-media.c:
10222 rtsp-server: port to 0.11
10224 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10229 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10231 Merge branch 'master' into 0.11
10236 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10239 Automatic update of common submodule
10240 From c3cafe1 to 46dfcea
10242 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
10244 * bindings/python/Makefile.am:
10245 * bindings/python/rtspserver.defs:
10246 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
10248 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
10250 * bindings/python/arg-types.py:
10251 python bindings: add GstRTSPUrlParam
10252 Needed to implement MediaFactory virtual proxies
10254 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
10256 * bindings/python/arg-types.py:
10257 python bindings: fix returning GstRTSPUrl types
10259 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
10261 * bindings/python/arg-types.py:
10262 python bindings: add arg type for GstRTSPUrl
10264 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
10266 * bindings/python/rtspserver.defs:
10267 python bindings: fix the definition of MediaFactory.collect_stream
10269 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
10272 Automatic update of common submodule
10273 From 1ccbe09 to c3cafe1
10275 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10278 Automatic update of common submodule
10279 From 193b717 to 1ccbe09
10281 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
10284 Automatic update of common submodule
10285 From b77e2bf to 193b717
10287 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10290 build: Include lcov.mak to allow test coverage report generation
10292 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10295 Automatic update of common submodule
10296 From d8814b6 to b77e2bf
10298 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10301 Automatic update of common submodule
10302 From 6aaa286 to d8814b6
10304 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
10307 Automatic update of common submodule
10308 From 6aec6b9 to 6aaa286
10310 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
10313 autogen: wingo signed comment
10315 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
10317 * gst/rtsp-server/rtsp-session-pool.c:
10318 session: use full charset for RTSP session ID
10319 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
10320 session ID more difficult.
10321 https://bugzilla.gnome.org/show_bug.cgi?id=643812
10323 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10325 * gst/rtsp-server/Makefile.am:
10326 rtsp-server: Don't install the funnel header
10328 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
10331 Automatic update of common submodule
10332 From 1de7f6a to 6aec6b9
10334 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10337 configure: require core/base 0.10.31
10338 Needed at least for gst_plugin_feature_rank_compare_func().
10340 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
10343 Automatic update of common submodule
10344 From f94d739 to 1de7f6a
10346 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10348 * gst/rtsp-server/rtsp-media.c:
10349 media: remove more unused code
10351 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10353 * gst/rtsp-server/rtsp-media.c:
10354 * gst/rtsp-server/rtsp-media.h:
10355 media: remove duplicate filtering
10356 Remove the duplicate filtering code now that we have a released -good version.
10357 Give a warning instead.
10359 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10361 * gst/rtsp-server/rtsp-media-factory.c:
10362 * gst/rtsp-server/rtsp-media.c:
10363 media: fix default buffer size
10365 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10367 * gst/rtsp-server/rtsp-media-factory.c:
10368 * gst/rtsp-server/rtsp-media-factory.h:
10369 media-factory: add property to configure the buffer-size
10370 Add a property to configure the kernel UDP buffer size.
10372 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10374 * gst/rtsp-server/rtsp-media.c:
10375 * gst/rtsp-server/rtsp-media.h:
10376 media: add property to configure kernel buffer sizes
10377 Add a property to configure the kernel UDP buffer size.
10379 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10382 configure: set PYGOBJECT_REQ before using it
10383 https://bugzilla.gnome.org/show_bug.cgi?id=640641
10385 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10387 * docs/Makefile.am:
10388 docs: recursive into sub-directories on 'make upload'
10390 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10392 * docs/libs/gst-rtsp-server-docs.sgml:
10393 * docs/version.entities.in:
10394 docs: mention full version these docs are for, not just major-minor
10396 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10399 back to development
10401 === release 0.10.8 ===
10403 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10408 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10410 * gst/rtsp-server/rtsp-server.c:
10411 rtsp-server: clarify docs a little
10413 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10415 * gst/rtsp-server/rtsp-media.c:
10416 media: init debug category before starting thread
10418 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10420 * gst/rtsp-server/rtsp-auth.c:
10421 auth: add realm to make it more spec compliant
10423 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10425 * gst/rtsp-server/rtsp-server.c:
10426 * gst/rtsp-server/rtsp-server.h:
10427 server: add locking
10429 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10431 * examples/test-video.c:
10432 example: improve example docs a little
10434 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10436 * gst/rtsp-server/rtsp-server.c:
10437 server: ensure the watch has a ref to the server
10439 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10441 * gst/rtsp-server/rtsp-server.c:
10442 server: simpify channel function
10444 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10446 * gst/rtsp-server/rtsp-server.c:
10447 * gst/rtsp-server/rtsp-server.h:
10448 server: simplify management of channel and source
10449 We don't need to keep around the channel and source objects. Let the mainloop
10450 and the source manage the source and channel respectively.
10452 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10458 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10460 * tests/.gitignore:
10461 * tests/Makefile.am:
10462 * tests/test-cleanup.c:
10463 tests: add tests directory and cleanup test
10465 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10467 * gst/rtsp-server/rtsp-media-factory-uri.c:
10468 * gst/rtsp-server/rtsp-media-factory.c:
10469 * gst/rtsp-server/rtsp-media-mapping.c:
10470 * gst/rtsp-server/rtsp-media.c:
10471 * gst/rtsp-server/rtsp-session-pool.c:
10472 * gst/rtsp-server/rtsp-session.c:
10473 server: improve debugging in various objects
10475 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10477 * gst/rtsp-server/rtsp-server.c:
10478 server: chain up to the parent finalize
10480 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
10482 * bindings/python/rtspserver-types.defs:
10483 * bindings/python/rtspserver.defs:
10484 * bindings/python/rtspserver.override:
10485 * bindings/python/test.py:
10486 gst-rtsp-server: update python bindings
10488 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10490 * gst/rtsp-server/rtsp-client.c:
10491 client: use the response from the clientstate
10492 Create the response object only once and store in the client state.
10493 Make all methods use the state response,
10495 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10497 * gst/rtsp-server/rtsp-server.c:
10498 server: use signal to keep track of clients
10499 Keep track of all the clients that the server creates and remove them when they
10500 fire the 'closed' signal.
10502 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10504 * gst/rtsp-server/rtsp-client.c:
10505 * gst/rtsp-server/rtsp-client.h:
10506 client: emit signal when closing
10508 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10510 * examples/.gitignore:
10511 * examples/Makefile.am:
10512 * examples/test-auth.c:
10513 * examples/test-video.c:
10514 * gst/rtsp-server/rtsp-auth.c:
10515 * gst/rtsp-server/rtsp-auth.h:
10516 * gst/rtsp-server/rtsp-client.c:
10517 * gst/rtsp-server/rtsp-media-factory.c:
10518 * gst/rtsp-server/rtsp-media.c:
10519 * gst/rtsp-server/rtsp-media.h:
10520 * gst/rtsp-server/rtsp-session-pool.h:
10521 * gst/rtsp-server/rtsp-session.h:
10522 media: enable per factory authorisations
10523 Allow for adding a GstRTSPAuth on the factory and media level and check
10524 permissions when accessing the factory.
10525 Add hints to the auth methods for future more fine grained authorisation.
10526 Add example application for per factory authentication.
10528 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10530 * gst/rtsp-server/rtsp-auth.c:
10531 * gst/rtsp-server/rtsp-auth.h:
10532 * gst/rtsp-server/rtsp-client.c:
10533 * gst/rtsp-server/rtsp-client.h:
10534 * gst/rtsp-server/rtsp-params.c:
10535 * gst/rtsp-server/rtsp-params.h:
10536 rtsp-server: Pass ClientState structure arround
10537 Pass the collected information for the ongoing request in a GstRTSPClientState
10538 structure that we can then pass around to simplify the method arguments. This
10539 will also be handy when we implement logging functionality.
10541 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10543 * gst/rtsp-server/rtsp-media-factory.c:
10544 * gst/rtsp-server/rtsp-media-factory.h:
10545 media-factory: add methods to configure authorisation
10547 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10549 * gst/rtsp-server/rtsp-client.c:
10550 client: unref auth in finalize
10552 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10554 * gst/rtsp-server/rtsp-server.c:
10555 server: unref auth in finalize
10557 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10559 * docs/libs/gst-rtsp-server-docs.sgml:
10560 * docs/libs/gst-rtsp-server-sections.txt:
10561 * docs/libs/gst-rtsp-server.types:
10562 docs: add more docs
10564 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10566 * gst/rtsp-server/rtsp-server.c:
10567 * gst/rtsp-server/rtsp-server.h:
10568 server: separate create and accept
10569 Create separate create and accept methods so that subclasses can create custom
10571 Configure the server in the client object and prepare for keeping track of
10574 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10576 * gst/rtsp-server/rtsp-client.c:
10577 * gst/rtsp-server/rtsp-client.h:
10578 client: add support for setting the server.
10579 Add support for keeping a ref to the server that started this client
10582 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10584 * gst/rtsp-server/rtsp-auth.c:
10585 auth: fix memleak and add some docs
10586 Fix a memleak of the basic auth token.
10587 Add docs for the helper function
10589 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10591 * gst/rtsp-server/rtsp-auth.c:
10592 * gst/rtsp-server/rtsp-auth.h:
10593 * gst/rtsp-server/rtsp-client.c:
10594 client: delegate setup of auth to the manager
10595 Delegate the configuration of the authentication tokens to the manager object
10598 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10600 * examples/test-video.c:
10601 * gst/rtsp-server/Makefile.am:
10602 * gst/rtsp-server/rtsp-auth.c:
10603 * gst/rtsp-server/rtsp-auth.h:
10604 * gst/rtsp-server/rtsp-client.c:
10605 * gst/rtsp-server/rtsp-client.h:
10606 * gst/rtsp-server/rtsp-server.c:
10607 * gst/rtsp-server/rtsp-server.h:
10608 auth: add authentication object
10609 Add an object that can check the authorization of requests.
10610 Implement basic authentication.
10611 Add example authentication to test-video
10613 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10615 * gst/rtsp-server/rtsp-server.c:
10616 * gst/rtsp-server/rtsp-server.h:
10617 server: move includes back
10618 the includes are needed for sockaddr_in.
10620 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10622 * gst/rtsp-server/rtsp-client.c:
10623 * gst/rtsp-server/rtsp-client.h:
10624 * gst/rtsp-server/rtsp-server.c:
10625 * gst/rtsp-server/rtsp-server.h:
10626 rtsp: move network includes where they are needed
10628 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
10630 * gst/rtsp-server/rtsp-media.h:
10631 rtsp-media.h: Minor corrections in comments.
10634 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
10637 Automatic update of common submodule
10638 From e572c87 to f94d739
10640 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
10644 * docs/libs/.gitignore:
10645 * examples/.gitignore:
10646 * gst/rtsp-server/.gitignore:
10649 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
10651 * docs/libs/Makefile.am:
10652 docs: We don't build ps/pdf for API reference docs
10654 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10657 Automatic update of common submodule
10658 From ccbaa85 to e572c87
10660 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10663 Automatic update of common submodule
10664 From 46445ad to ccbaa85
10666 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10668 * gst/rtsp-server/Makefile.am:
10669 * gst/rtsp-server/rtsp-funnel.c:
10670 * gst/rtsp-server/rtsp-funnel.h:
10671 * gst/rtsp-server/rtsp-media.c:
10672 funnel: rename fsfunnel to rtspfunnel
10673 Rename the funnel to avoid conflicts with the farsight one.
10675 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10677 * gst/rtsp-server/Makefile.am:
10678 * gst/rtsp-server/fs-funnel.c:
10679 * gst/rtsp-server/fs-funnel.h:
10680 * gst/rtsp-server/rtsp-media.c:
10681 rtsp-media: add and use fsfunnel
10682 Add a copy of fsfunnel to the build because input-selector removed the (broken)
10683 select-all property that we need.
10685 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10687 * gst/rtsp-server/Makefile.am:
10688 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
10689 Use PKG_CONFIG_PATH specified at configure time (if any) as well
10690 for the g-ir-compiler, rather than just assuming the env var has
10693 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10700 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
10702 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10705 * gst/rtsp-server/Makefile.am:
10706 gobject-introspection: fix g-i build for uninstalled setup
10707 Requires gst-plugins-base git (> 0.10.31.2).
10709 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10711 * examples/test-uri.c:
10712 examples: add some more options and comments
10714 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10716 * gst/rtsp-server/rtsp-media-factory-uri.c:
10717 factory-uri: use right property type
10719 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10721 * gst/rtsp-server/rtsp-media-factory-uri.c:
10722 factory-uri: attempt to configure buffer-lists
10723 Attempt to configure buffer lists in the payloader for improved performance.
10725 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10727 * gst/rtsp-server/rtsp-media.c:
10728 media: attempt to configure bigger UDP buffers
10729 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
10730 send buffers with high bitrate streams.
10732 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
10734 * gst/rtsp-server/rtsp-client.c:
10735 client: use the socket length from getsockname
10736 Use the length returned by getsockname to perform the getnameinfo call because
10737 the size can depend on the socket type and platform.
10740 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10742 * docs/libs/gst-rtsp-server-docs.sgml:
10743 * docs/libs/gst-rtsp-server-sections.txt:
10744 docs: add uri factory to the docs
10746 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10748 * gst/rtsp-server/rtsp-client.c:
10749 * gst/rtsp-server/rtsp-media.h:
10752 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10754 * gst/rtsp-server/rtsp-client.c:
10755 * gst/rtsp-server/rtsp-media.c:
10756 * gst/rtsp-server/rtsp-media.h:
10757 * gst/rtsp-server/rtsp-session.c:
10758 * gst/rtsp-server/rtsp-session.h:
10759 rtsp-server: add support for buffer lists
10760 Add support for sending bufferlists received from appsink.
10763 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10765 * gst/rtsp-server/rtsp-client.c:
10766 * gst/rtsp-server/rtsp-media.c:
10767 * gst/rtsp-server/rtsp-media.h:
10768 * gst/rtsp-server/rtsp-sdp.c:
10769 media: make method to retrieve the play range
10770 Make a method to retrieve the playback range so that we can conditionally create
10771 a different range for the SDP and the PLAY requests.
10773 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10775 * gst/rtsp-server/rtsp-media.c:
10776 * gst/rtsp-server/rtsp-media.h:
10777 media: add signal to notify of state changes
10779 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10781 * gst/rtsp-server/rtsp-client.h:
10782 client: cleanup headers
10784 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10786 * gst/rtsp-server/rtsp-client.c:
10789 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10791 * gst/rtsp-server/rtsp-media-factory-uri.c:
10792 * gst/rtsp-server/rtsp-media-factory-uri.h:
10793 factory-uri: add support for gstpay
10794 Add an option to prefer gstpay over decoder + raw payloader.
10796 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10798 * gst/rtsp-server/rtsp-media-factory-uri.c:
10799 * gst/rtsp-server/rtsp-media-factory-uri.h:
10800 factory-uri: rework the autoplugger.
10801 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
10804 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10806 * gst/rtsp-server/rtsp-media-factory-uri.c:
10807 factory-uri: use better factory filter
10808 Make better payloader filter based on autoplug rank and RTP use case.
10810 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
10813 Automatic update of common submodule
10814 From 169462a to 46445ad
10816 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10818 * gst/rtsp-server/rtsp-server.c:
10819 server: set SO_REUSEADDR before bind
10820 Set the SO_REUSEADDR _before_ bind() to make it actually work.
10822 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10824 * gst/rtsp-server/rtsp-media.c:
10825 * gst/rtsp-server/rtsp-media.h:
10826 media: emit prepared signal when prepared
10827 Make a 'prepared' signal and emit it when we successfully prepared the element.
10828 This signal can be used to configure the media object after it has been prepared
10831 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
10834 Automatic update of common submodule
10835 From 011bcc8 to 169462a
10837 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
10839 python an optional dependency
10840 * configure.ac: Move up valgrind and g-i checks. Make the python
10841 dependency optional, as it was before.
10843 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10845 Merge branch 'master' into 0.11
10850 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10852 * gst/rtsp-server/rtsp-media.c:
10853 media: update range when active clients changed
10854 When we changed the number of active clients, update the current range
10855 information because we want the second client connecting to a shared resource
10856 continue from where the stream currently.
10858 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10860 * gst/rtsp-server/rtsp-media-factory-uri.c:
10861 * gst/rtsp-server/rtsp-media-factory-uri.h:
10862 factory-uri: add colorspace and fix pt
10863 Rework the way we pass data to the autoplugger.
10864 When we have raw caps, plug a converter element to make pluggin to raw
10865 payloaders more successful.
10866 Make sure all dynamically plugged payloaders have a unique payload types.
10868 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10870 * examples/Makefile.am:
10871 * examples/test-uri.c:
10872 example: add example of the uri factory
10874 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10876 * gst/rtsp-server/Makefile.am:
10877 * gst/rtsp-server/rtsp-media-factory-uri.c:
10878 * gst/rtsp-server/rtsp-media-factory-uri.h:
10879 * gst/rtsp-server/rtsp-server.h:
10880 factory-uri: add a factory to stream any URI
10881 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
10884 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10886 * gst/rtsp-server/rtsp-media.c:
10887 * gst/rtsp-server/rtsp-media.h:
10888 media: ignore spurious ASYNC_DONE messages
10889 When we are dynamically adding pads, the addition of the udpsrc elements will
10890 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
10891 the real ASYNC_DONE when everything is prerolled.
10893 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10895 * gst/rtsp-server/rtsp-media-factory.c:
10896 * gst/rtsp-server/rtsp-media-factory.h:
10897 media-factory: make lock macro
10899 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
10901 * gst/rtsp-server/rtsp-client.c:
10902 rtsp-server: Remove unused variable and dead assignment
10904 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
10906 * examples/test-launch.c:
10907 * examples/test-mp4.c:
10908 * examples/test-ogg.c:
10909 * examples/test-readme.c:
10910 * examples/test-sdp.c:
10911 * examples/test-video.c:
10912 examples: Run gst-indent
10914 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
10916 * gst/rtsp-server/rtsp-client.c:
10917 * gst/rtsp-server/rtsp-media-factory.c:
10918 * gst/rtsp-server/rtsp-media-mapping.c:
10919 * gst/rtsp-server/rtsp-media.c:
10920 * gst/rtsp-server/rtsp-params.c:
10921 * gst/rtsp-server/rtsp-sdp.c:
10922 * gst/rtsp-server/rtsp-server.c:
10923 * gst/rtsp-server/rtsp-session-pool.c:
10924 * gst/rtsp-server/rtsp-session.c:
10925 rtsp-server: Run gst-indent
10926 Since it wasn't using the upstream common previously, there was no
10927 indentation check before commiting.
10929 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
10931 * gst/rtsp-server/rtsp-media-mapping.h:
10932 * gst/rtsp-server/rtsp-media.c:
10933 * gst/rtsp-server/rtsp-media.h:
10934 * gst/rtsp-server/rtsp-sdp.c:
10935 * gst/rtsp-server/rtsp-session-pool.h:
10936 * gst/rtsp-server/rtsp-session.c:
10937 * gst/rtsp-server/rtsp-session.h:
10938 rtsp-server: Some more doc fixups
10940 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
10943 Makefile: Add cruft-cleaning support
10945 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
10949 * docs/Makefile.am:
10950 * docs/libs/Makefile.am:
10951 * docs/libs/gst-rtsp-server-docs.sgml:
10952 * docs/libs/gst-rtsp-server-sections.txt:
10953 * docs/libs/gst-rtsp-server.types:
10954 * docs/version.entities.in:
10955 docs: Add gtk-doc build system
10957 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
10959 * gst/rtsp-server/Makefile.am:
10960 Makefile.am: Use standard GIR make behaviour
10962 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
10966 autogen/configure: Bring more in sync to standard gst module behaviour
10968 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10970 * gst/rtsp-server/rtsp-media.c:
10971 media: warn and fail when gstrtpbin is not found
10973 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10976 configure: open 0.11 branch
10978 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
10982 Add common submodule
10984 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
10986 * common/ChangeLog:
10987 * common/Makefile.am:
10988 * common/c-to-xml.py:
10989 * common/check.mak:
10990 * common/coverage/coverage-report-entry.pl:
10991 * common/coverage/coverage-report.pl:
10992 * common/coverage/coverage-report.xsl:
10993 * common/coverage/lcov.mak:
10994 * common/gettext.patch:
10995 * common/glib-gen.mak:
10996 * common/gst-autogen.sh:
10997 * common/gst-xmlinspect.py:
10999 * common/gstdoc-scangobj:
11000 * common/gtk-doc-plugins.mak:
11001 * common/gtk-doc.mak:
11002 * common/m4/.gitignore:
11003 * common/m4/Makefile.am:
11004 * common/m4/README:
11005 * common/m4/as-ac-expand.m4:
11006 * common/m4/as-auto-alt.m4:
11007 * common/m4/as-compiler-flag.m4:
11008 * common/m4/as-compiler.m4:
11009 * common/m4/as-docbook.m4:
11010 * common/m4/as-libtool-tags.m4:
11011 * common/m4/as-libtool.m4:
11012 * common/m4/as-python.m4:
11013 * common/m4/as-scrub-include.m4:
11014 * common/m4/as-version.m4:
11015 * common/m4/ax_create_stdint_h.m4:
11016 * common/m4/check.m4:
11017 * common/m4/glib-gettext.m4:
11018 * common/m4/gst-arch.m4:
11019 * common/m4/gst-args.m4:
11020 * common/m4/gst-check.m4:
11021 * common/m4/gst-debuginfo.m4:
11022 * common/m4/gst-default.m4:
11023 * common/m4/gst-doc.m4:
11024 * common/m4/gst-error.m4:
11025 * common/m4/gst-feature.m4:
11026 * common/m4/gst-function.m4:
11027 * common/m4/gst-gettext.m4:
11028 * common/m4/gst-glib2.m4:
11029 * common/m4/gst-libxml2.m4:
11030 * common/m4/gst-plugindir.m4:
11031 * common/m4/gst-valgrind.m4:
11032 * common/m4/gtk-doc.m4:
11033 * common/m4/introspection.m4:
11034 * common/m4/pkg.m4:
11035 * common/mangle-tmpl.py:
11036 * common/plugins.xsl:
11038 * common/release.mak:
11039 * common/scangobj-merge.py:
11040 * common/upload.mak:
11041 common: Remove static version
11043 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
11045 * common/m4/introspection.m4:
11046 Update introspection.m4 to match usage
11048 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11052 Remove old stuff from the README
11054 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11057 back to development
11059 === release 0.10.7 ===
11061 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11066 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11068 * examples/test-ogg.c:
11069 test-ogg: remove parsers
11070 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
11071 buffers with timestamps. Using the parsers also seems to break things.
11073 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11075 * bindings/vala/gst-rtsp-server-0.10.vapi:
11076 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11077 Updated Vala bindings
11079 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11081 * common/m4/introspection.m4:
11083 * gst/rtsp-server/Makefile.am:
11084 Added initial gobject-introspection support
11086 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11088 * gst/rtsp-server/rtsp-media-factory.c:
11089 media-factory: don't use host for shared hash key
11090 When we generate the key to share made between connections, don't include the
11091 host used to connect so that we can share media even if between clients that
11092 connected with localhost and ones with the ip address.
11094 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11096 * bindings/vala/Makefile.am:
11097 build: fix distcheck
11099 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11101 * bindings/vala/gst-rtsp-server-0.10.vapi:
11102 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11103 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11104 Update Vala bindings
11106 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11108 * bindings/vala/Makefile.am:
11110 Fix configure checks and installation location for Vala bindings
11113 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11116 back to development
11118 === release 0.10.6 ===
11120 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11123 configure: release 0.10.6
11125 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11127 * gst/rtsp-server/rtsp-media.c:
11128 media: help the compiler a little
11130 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11132 * gst/rtsp-server/rtsp-media.c:
11133 * gst/rtsp-server/rtsp-media.h:
11134 * gst/rtsp-server/rtsp-session.c:
11135 media: cleanup media transport before freeing
11136 Cleanup the media transport data before freeing. In particular, remove the qdata
11137 from the rtpsource object.
11139 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11141 * gst/rtsp-server/rtsp-media-factory.c:
11142 * gst/rtsp-server/rtsp-media-factory.h:
11143 * gst/rtsp-server/rtsp-media.c:
11144 * gst/rtsp-server/rtsp-media.h:
11145 media-factory: add eos-shutdown property
11146 Add an eos-shutdown property that will send an EOS to the pipeline before
11147 shutting it down. This allows for nice cleanup in case of a muxer.
11150 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11152 * gst/rtsp-server/rtsp-media.c:
11153 * gst/rtsp-server/rtsp-media.h:
11154 media: use multiudpsink send-duplicates when we can
11155 If we have a new enough multiudpsink with the send-duplicates property, use this
11156 instead of doing our own filtering. Our custom filtering code should eventually
11157 be removed when we can depend on a released -good.
11159 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11161 * gst/rtsp-server/rtsp-media.c:
11162 media: don't leak destinations
11163 Refactor and cleanup the destinations array when the stream is destroyed.
11165 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11167 * gst/rtsp-server/rtsp-media.c:
11168 * gst/rtsp-server/rtsp-media.h:
11169 media: don't add udp addresses multiple times
11170 Keep track of the udp addresses we added to udpsink and never add the same udp
11171 destination twice. This avoids duplicate packets when using multicast.
11173 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11175 * gst/rtsp-server/rtsp-server.c:
11176 server: disable use of SO_LINGER
11177 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
11178 server close()s the connection.
11180 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11182 * gst/rtsp-server/rtsp-server.c:
11183 server: use 5 second linger period in SO_LINGER
11184 Wait 5 seconds before clearing the send buffers and reseting the connection with
11185 the client when we do a close. This should be enough time to get the message to
11189 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
11191 * gst/rtsp-server/rtsp-server.c:
11192 server: use SO_LINGER
11193 SO_LINGER on the socket will make sure that any pending data on the socket is
11194 flushed ASAP and that the socket connection is reset. This makes sure that the
11195 socket can be reused immediately.
11198 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11201 README: add blurb about shared media factories
11203 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
11205 * gst/rtsp-server/rtsp-media.c:
11206 Add stdlib.h for atoi()
11208 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11210 * bindings/python/Makefile.am:
11211 * bindings/vala/Makefile.am:
11212 build: distcheck fixes
11213 Fix 'make distcheck', somewhat (it still fails because it tries to
11214 install files into /usr/share/vala/vapi/ irrespective of the
11215 configured prefix).
11217 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11220 configure: bump core/base requirements to released version
11221 Makes things less confusing for people.
11223 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11226 configure: fail if GStreamer core/base requirements are not met
11228 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11230 * gst/rtsp-server/rtsp-client.c:
11231 client: improve client cleanups
11232 Make sure the session does not timeout when using TCP. We need to do this
11233 because quicktime player does not send RTCP for some reason in tunneled
11235 Refactor some cleanup code.
11238 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11240 * gst/rtsp-server/rtsp-session.c:
11241 * gst/rtsp-server/rtsp-session.h:
11242 session: add support for prevent session timeouts
11243 Add an atomix counter to prevent session timeouts when we are, for example,
11244 streaming over TCP.
11246 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11248 * gst/rtsp-server/rtsp-client.c:
11249 client: fix unlink on session timeouts
11250 When our session times out, make sure we unlink all streams in this
11252 Remove the tunnelid when closing the connection.
11254 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11256 * gst/rtsp-server/rtsp-session.c:
11257 session: small cleanups
11259 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11261 * gst/rtsp-server/rtsp-client.c:
11262 client: handle lost_tunnel callbacks
11263 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
11264 hashtable so that we can reuse it for when the client reopens the POST
11266 Close the connection after a TEARDOWN.
11267 Make sure or watchid is cleared when the watch is removed.
11270 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11272 * gst/rtsp-server/rtsp-client.c:
11273 * gst/rtsp-server/rtsp-media.c:
11274 * gst/rtsp-server/rtsp-sdp.c:
11275 rtsp-server: add more support for multicast
11277 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11280 * gst/rtsp-server/rtsp-media.c:
11281 * gst/rtsp-server/rtsp-media.h:
11282 media: allow configuration of allowed lower transport
11284 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11286 * gst/rtsp-server/rtsp-client.h:
11287 * gst/rtsp-server/rtsp-media.c:
11288 * gst/rtsp-server/rtsp-media.h:
11289 * gst/rtsp-server/rtsp-sdp.c:
11290 * gst/rtsp-server/rtsp-sdp.h:
11291 * gst/rtsp-server/rtsp-server.c:
11292 rtsp: keep track of server ip and ipv6
11293 Keep track of how the client connected to the server and setup the udp ports
11294 with the same protocol.
11295 Copy the server ip address in the SDP so that clients can send RTCP back to
11298 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11300 * gst/rtsp-server/rtsp-session.c:
11303 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11305 * gst/rtsp-server/rtsp-client.c:
11306 client: use right size for malloc
11308 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11310 * gst/rtsp-server/rtsp-server.c:
11311 server: comment ipv6 server listening address
11313 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11315 * gst/rtsp-server/rtsp-media.c:
11316 media: allow for ipv6 sockets
11318 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11320 * gst/rtsp-server/rtsp-server.c:
11321 * gst/rtsp-server/rtsp-server.h:
11322 server: rework server part
11323 Allow setting a bind address, make sure we can deal with ipv6.
11324 Remove the port property and change with the service property.
11326 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11328 * gst/rtsp-server/rtsp-media.h:
11329 media: update comments a little
11331 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11333 * gst/rtsp-server/rtsp-client.c:
11334 client: make content-base better
11335 Use the URI formatting functions to make a content-base. Also make sure that
11336 there is a trailing / at the end.
11338 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11340 * gst/rtsp-server/rtsp-client.c:
11341 client: guard against invalid paths
11343 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11345 * examples/test-video.c:
11346 test: catch server bind errors
11348 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
11350 * gst/rtsp-server/rtsp-media.c:
11351 rtspmedia: emit "unprepared" if _prepare fails.
11352 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
11353 media object is removed from its factory's cache.
11355 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11357 * gst/rtsp-server/rtsp-media.c:
11358 media: collect media position when seek completes
11360 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
11362 * gst/rtsp-server/rtsp-client.c:
11363 client: call unlink_streams in client finalize
11366 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11368 * gst/rtsp-server/rtsp-media.c:
11369 media: limit the time to wait to something huge
11370 Avoid waiting forever but limit the timeout to 20 seconds.
11372 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11374 * gst/rtsp-server/rtsp-sdp.c:
11375 sdp: reindent and check for prepared status
11377 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11379 * gst/rtsp-server/rtsp-media.c:
11380 * gst/rtsp-server/rtsp-media.h:
11381 * gst/rtsp-server/rtsp-session.c:
11382 media: avoid doing _get_state() for state changes
11383 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
11384 until the media is prerolled or in error. This avoids doing a blocking call of
11385 gst_element_get_state() that can cause lockups when there is an error.
11388 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11390 * gst/rtsp-server/rtsp-media.c:
11393 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11395 * gst/rtsp-server/rtsp-media-factory.c:
11396 media-factory: better error handling
11397 Improve the error handling a bit.
11399 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11401 * gst/rtsp-server/rtsp-client.c:
11402 client: rework transport parsing
11403 Rework the transport parsing code so that we can ignore transports we don't
11404 support instead of just picking the first one we can parse.
11405 Configure a (for now hardcoded) destination for multicast transports.
11407 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11409 * gst/rtsp-server/rtsp-media.c:
11410 media: set multicast sink parameters
11411 Disable loop and automatic multicast join on the udpsink elements.
11412 Add some more debug info.
11413 Reset some state variables in the right place.
11414 Use the right port numbers for multicast.
11416 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11418 * gst/rtsp-server/rtsp-session.c:
11419 session: handle transport setup correctly
11420 Handle UDP, MCAST and TCP transport negotiation more correctly.
11421 Store the server session SSRC in the transport.
11423 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11425 * gst/rtsp-server/rtsp-client.c:
11426 rtsp-client: implement error_full
11427 Implement error_full to avoid some segfaults when the rtspconnection calls it.
11430 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11433 * gst/rtsp-server/rtsp-client.c:
11434 * gst/rtsp-server/rtsp-server.c:
11435 docs: update docs and comments
11437 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
11439 * gst/rtsp-server/rtsp-sdp.c:
11440 sdp: make server work better when behind a proxy
11442 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11444 * gst/rtsp-server/rtsp-client.c:
11445 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
11447 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11449 * gst/rtsp-server/rtsp-client.c:
11450 * gst/rtsp-server/rtsp-media-factory.c:
11451 * gst/rtsp-server/rtsp-media-mapping.c:
11452 * gst/rtsp-server/rtsp-media.c:
11453 * gst/rtsp-server/rtsp-server.c:
11454 * gst/rtsp-server/rtsp-session-pool.c:
11455 * gst/rtsp-server/rtsp-session.c:
11456 Use GStreamer's debugging subsystem
11458 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11460 * gst/rtsp-server/rtsp-media-factory.c:
11461 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
11463 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11466 back to development
11468 === release 0.10.5 ===
11470 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11475 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11478 configure: bump required versions
11480 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
11482 * gst/rtsp-server/rtsp-client.c:
11483 client: call weak-unref on client->sessions from finalize
11486 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11488 * gst/rtsp-server/rtsp-media.c:
11489 media: Fixed crasher where caps got unref'ed too often
11491 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11494 * pkgconfig/.gitignore:
11495 * pkgconfig/Makefile.am:
11496 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
11497 Added pkg-config file to use gst-rtsp-server uninstalled
11499 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11501 * gst/rtsp-server/rtsp-media.c:
11502 media: add some docs
11504 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
11506 * gst/rtsp-server/rtsp-client.c:
11507 rtsp: Use gst_rtsp_watch_send_message().
11508 Use gst_rtsp_watch_send_message() since the old API which used
11509 gst_rtsp_watch_queue_message() has been deprecated.
11511 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11514 back to development
11516 === release 0.10.4 ===
11518 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11523 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11525 * gst/rtsp-server/rtsp-client.c:
11526 * gst/rtsp-server/rtsp-session.c:
11527 * gst/rtsp-server/rtsp-session.h:
11528 rtsp: allocate channels in TCP mode
11529 When the client does not provide us with channels in TCP mode, allocate channels
11532 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11534 * gst/rtsp-server/rtsp-client.c:
11535 client: don't crash when tunnelid is missing
11536 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
11537 don't crash but return an error response to the client.
11540 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11542 * bindings/vala/gst-rtsp-server-0.10.vapi:
11543 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11544 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11545 bindings: update vala bindings with new method
11547 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11549 * gst/rtsp-server/rtsp-session-pool.c:
11550 * gst/rtsp-server/rtsp-session-pool.h:
11551 sessionpool: add function to filter sessions
11552 Add generic function to retrieve/remove sessions.
11554 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11557 configure: bump core/base requirements to release
11559 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11561 * gst/rtsp-server/rtsp-media.c:
11562 media: fix indentation
11564 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11566 * gst/rtsp-server/rtsp-media.c:
11567 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
11569 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11571 * gst/rtsp-server/rtsp-media.c:
11572 set state and remove elements of media in for loop
11574 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
11576 * bindings/vala/gst-rtsp-server-0.10.vapi:
11577 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11578 Added gst_rtsp_media_remove_elements function to Vala bindings
11580 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
11582 * gst/rtsp-server/rtsp-media.c:
11583 * gst/rtsp-server/rtsp-media.h:
11584 Added gst_rtsp_media_remove_elements function
11586 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
11588 * gst/rtsp-server/rtsp-media.c:
11589 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
11591 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11593 * bindings/vala/gst-rtsp-server-0.10.vapi:
11594 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11595 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11596 Updated Vala bindings
11598 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11600 * gst/rtsp-server/rtsp-media.c:
11601 * gst/rtsp-server/rtsp-media.h:
11602 Added vmethod unprepare to GstRTSPMedia
11603 The default implementation sets the state of the pipeline to GST_STATE_NULL
11605 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11607 * gst/rtsp-server/rtsp-media-factory.c:
11608 * gst/rtsp-server/rtsp-media-factory.h:
11609 Made collect_streams function public
11611 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11613 * gst/rtsp-server/rtsp-media-factory.c:
11614 * gst/rtsp-server/rtsp-media-factory.h:
11615 * gst/rtsp-server/rtsp-media.c:
11616 Added vmethod create_pipeline to GstRTSPMediaFactory
11617 The pipeline is created in this method and the GstRTSPMedia's element is added to it
11619 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11621 * gst/rtsp-server/rtsp-client.c:
11622 client: use g_source_destroy()
11623 We need to use g_source_destroy() because we might have added the source to a
11624 different main context than the default one.
11626 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11628 * gst/rtsp-server/Makefile.am:
11629 * gst/rtsp-server/rtsp-client.c:
11630 * gst/rtsp-server/rtsp-params.c:
11631 * gst/rtsp-server/rtsp-params.h:
11632 rtsp: prepare for handling GET/SET_PARAMETER
11633 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
11635 Fix return codes of handlers.
11637 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11639 * gst/rtsp-server/rtsp-media.c:
11640 media: don't leak session pads
11642 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11644 * gst/rtsp-server/rtsp-media.c:
11645 media: clean up the messages a bit
11647 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11649 * gst/rtsp-server/rtsp-sdp.c:
11650 sdp: warn and skip streams without media
11652 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11654 * bindings/vala/gst-rtsp-server-0.10.vapi:
11655 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11656 vala: Fixed typo in header file of RTSPMediaStream
11658 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11660 * gst/rtsp-server/rtsp-media.c:
11662 Fix a debug message
11663 Make dumping RTCP stats configurable
11665 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11667 * gst/rtsp-server/rtsp-media.c:
11668 media: be less verbose and leak less
11670 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11672 * gst/rtsp-server/rtsp-media.c:
11673 media: don't leak the destination address
11675 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11677 * gst/rtsp-server/rtsp-client.c:
11678 * gst/rtsp-server/rtsp-media.c:
11679 * gst/rtsp-server/rtsp-media.h:
11680 * gst/rtsp-server/rtsp-session.c:
11681 * gst/rtsp-server/rtsp-session.h:
11682 rtsp: use RTCP to keep the session alive
11683 Use the RTCP rtcp-from stats field to find the associated session and use this
11684 to keep the session alive.
11686 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11688 * gst/rtsp-server/rtsp-session.c:
11689 session: add 5sec to the real session timeout
11690 Allow the session to live 5sec longer before really timing out. This should give
11691 clients some extra time to keep the session active.
11693 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11695 * gst/rtsp-server/rtsp-client.c:
11696 client: replay OK to GET/SET_PARAMETER
11697 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
11698 so that we return OK for those requests.
11700 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11702 * gst/rtsp-server/rtsp-media.c:
11703 * gst/rtsp-server/rtsp-media.h:
11704 media: keep track of active transports
11705 Keep track of which transport is active to avoid closing the connection too
11707 Remove the destination transport also when going to NULL.
11708 Print some stats about the SDES and other RTCP messages we receive from the
11711 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11713 * examples/.gitignore:
11714 * examples/Makefile.am:
11715 * examples/test-sdp.c:
11716 example: add SDP relay example
11718 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11720 * gst/rtsp-server/rtsp-media.c:
11721 media: also count active TCP connections
11723 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11725 * gst/rtsp-server/rtsp-media-factory.c:
11726 * gst/rtsp-server/rtsp-media.c:
11727 * gst/rtsp-server/rtsp-media.h:
11728 rtsp: add support for dynamic elements
11729 Add support for dynamic elements.
11730 Don't set live pipelines back to paused.
11732 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11734 * gst/rtsp-server/rtsp-sdp.c:
11735 sdp: don't add encoding name when absent in caps
11737 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11739 * gst/rtsp-server/rtsp-client.c:
11740 client: warn when we can't do RTP-Info
11742 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11744 * gst/rtsp-server/rtsp-media-factory.c:
11745 factory: factor out the stream construction
11747 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11749 * gst/rtsp-server/rtsp-client.c:
11750 client: only add RTP-Info when we have the info
11751 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
11754 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11757 back to development
11759 === release 0.10.3 ===
11761 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11765 - Fixes a bug where it put the wrong verion in pkgconfig
11766 - Link RTP and RTCP sources
11768 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11770 * gst/rtsp-server/rtsp-media.c:
11771 * gst/rtsp-server/rtsp-media.h:
11772 media: link the RTP udpsrc to the session manager
11773 Link the RTP udpsrc and the appsrc to the session manager so that they don't
11774 shut down when the client sends a packet to open firewalls.
11776 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11778 * pkgconfig/gst-rtsp-server.pc.in:
11779 Don't use hard-coded version number in pkg-config file
11781 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11784 back to development
11786 === release 0.10.2 ===
11788 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11793 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11796 * common/m4/.gitignore:
11797 * examples/.gitignore:
11798 * pkgconfig/.gitignore:
11799 add some .gitignore files
11801 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11803 * gst/rtsp-server/rtsp-media.c:
11804 media: seek to key frames
11806 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11808 * gst/rtsp-server/rtsp-media.c:
11809 media: emit the unprepared signal by id
11810 Emit the unprepared signal by id instead of name and set the media as
11813 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11815 * gst/rtsp-server/rtsp-media.c:
11816 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
11818 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11820 * gst/rtsp-server/rtsp-server.c:
11821 Added finalize function to GstRTPSPServer to unref session pool and media mapping
11823 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11825 * bindings/vala/gst-rtsp-server-0.10.vapi:
11826 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11827 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11828 Updated vala bindings
11830 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11832 * gst/rtsp-server/Makefile.am:
11833 * gst/rtsp-server/rtsp-client.c:
11834 * gst/rtsp-server/rtsp-media.c:
11835 server: use appsink and appsrc with the API
11836 Use the appsink/appsrc API instead of the signals for higher
11839 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11841 * examples/test-ogg.c:
11842 tests: set the payload type correctly
11844 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11846 * gst/rtsp-server/rtsp-media-factory.c:
11847 factory: connect to the unprepare signal
11848 Connect to the unprepare signal for non-reusable media so that we can remove
11849 them from the cache.
11851 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11853 * gst/rtsp-server/rtsp-media.c:
11854 * gst/rtsp-server/rtsp-media.h:
11855 media: add signal to notify of unprepare
11857 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11859 * gst/rtsp-server/rtsp-media.c:
11860 * gst/rtsp-server/rtsp-media.h:
11861 media: more work on making the media shared
11862 Add a reusable flag to medias, indicating that they can be reused after a state
11866 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11868 * examples/test-readme.c:
11869 examples: mark the example as shared for testing
11871 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11873 * gst/rtsp-server/rtsp-media.c:
11874 * gst/rtsp-server/rtsp-media.h:
11875 client: support shared media
11876 Always perform the state actions even if the target state of the pipeline is
11877 already correct, we still want to add/remove the transports when we are dealing
11879 Keep a counter of the number of active transports for a media so that we can use
11880 this to perform a state change when needed.
11881 Perform a state change of the pipeline only when the first transport was added
11882 or when there are no active transports.
11884 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11886 * gst/rtsp-server/rtsp-client.c:
11887 client: fix refcounting crasher
11888 Don't need to remove the weak refs in the finalize methods, they are already
11889 removed in the dispose.
11890 Don't register the callback with a DestroyNofity.
11892 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11894 * gst/rtsp-server/rtsp-client.c:
11895 Fix rtsp client refcount management in TCP mode.
11896 Don't unref a client ref we never had. Fixes an unref
11897 of an already-free client object after a client
11898 teardown request for me.
11900 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11902 * gst/rtsp-server/rtsp-session.c:
11903 docs: fix typo in API docs
11905 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11907 * gst/rtsp-server/rtsp-media.c:
11908 More seeking fixes.
11909 Keep the udp sources in playing even if we go to paused. unlock the sources when
11911 Add some more debug info.
11912 Only seek when we need to.
11913 Keep track of the position when we go to paused.
11915 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11917 * gst/rtsp-server/rtsp-client.c:
11918 * gst/rtsp-server/rtsp-media.c:
11919 * gst/rtsp-server/rtsp-media.h:
11920 Add beginnings of seeking.
11921 Parse the Range header and perform a seek on the pipeline for the requested
11922 position. It's disabled currently until I figure out what's going wrong.
11924 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11926 * gst/rtsp-server/rtsp-client.c:
11927 allow pause requests for now.
11930 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11932 * gst/rtsp-server/rtsp-client.c:
11933 Remove weak ref on the session in teardown
11934 We need to remove our weakref from the session when we do a teardown because
11935 else we close the TCP connection prematurely.
11937 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11939 * gst/rtsp-server/rtsp-client.c:
11940 * gst/rtsp-server/rtsp-client.h:
11941 * gst/rtsp-server/rtsp-session-pool.c:
11942 Do some more session cleanup
11943 Make session timeout kill the TCP connection that currently watches the
11945 Remove the client timeout property.
11947 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11949 * gst/rtsp-server/rtsp-client.c:
11950 * gst/rtsp-server/rtsp-client.h:
11951 * gst/rtsp-server/rtsp-media.c:
11952 * gst/rtsp-server/rtsp-media.h:
11953 * gst/rtsp-server/rtsp-server.c:
11954 * gst/rtsp-server/rtsp-session.c:
11955 * gst/rtsp-server/rtsp-session.h:
11957 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
11960 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11962 * examples/Makefile.am:
11963 * examples/test-launch.c:
11964 Add example server that takes launch lines
11965 Add an example server that streams any -launch line.
11967 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11969 * examples/test-readme.c:
11970 * gst/rtsp-server/rtsp-client.c:
11971 * gst/rtsp-server/rtsp-media.c:
11972 * gst/rtsp-server/rtsp-media.h:
11973 Add support for live streams
11974 Add support for live streams and ranges
11975 Start on handling TCP data transfer.
11977 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11979 * gst/rtsp-server/rtsp-media.c:
11980 Free the pipeline before other things
11983 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11985 * gst/rtsp-server/rtsp-client.c:
11986 Only free the pending tunnel if there is one
11989 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11991 * gst/rtsp-server/rtsp-client.c:
11992 * gst/rtsp-server/rtsp-client.h:
11993 * gst/rtsp-server/rtsp-media.c:
11994 rtsp-server: Add support for tunneling
11995 Add support for tunneling over HTTP.
11996 Use new connection methods to retrieve the url.
11997 Dispatch messages based on the message type instead of blindly
11998 assuming it's always a request.
11999 Keep track of the watch id so that we can remove it later.
12000 Set the media pipeline to NULL before unreffing the pipeline.
12002 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12004 * gst/rtsp-server/rtsp-client.c:
12005 * gst/rtsp-server/rtsp-client.h:
12006 Fix for channel -> watch rename in gstreamer
12007 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
12009 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12011 * gst/rtsp-server/rtsp-client.c:
12012 * gst/rtsp-server/rtsp-client.h:
12014 Use the async RTSP channels instead of spawning a new thread for each client.
12015 If a sessionid is specified in a request, fail if we don't have the session.
12017 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12019 * gst/rtsp-server/rtsp-media.c:
12020 Add better debug info
12021 Add some better debug info.
12023 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12025 * examples/test-video.c:
12027 Add support for session timeouts in the example.
12029 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12031 * gst/rtsp-server/rtsp-session-pool.c:
12032 * gst/rtsp-server/rtsp-session-pool.h:
12033 Pass GTimeVal around for performance reasons
12034 Get the current time only once and pass it around so that sessions don't have to
12035 get the current time anymore.
12036 Add experimental support for a GSource that dispatches when the session needs to
12039 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12041 * gst/rtsp-server/rtsp-session.c:
12042 * gst/rtsp-server/rtsp-session.h:
12043 Add better support for session timeouts
12044 Add a method to request the number of milliseconds when a session will timeout.
12046 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12048 * gst/rtsp-server/rtsp-media.c:
12049 * gst/rtsp-server/rtsp-media.h:
12050 Add suport for RTP manager monitoring
12051 Add the first stage in monitoring the rtp manager.
12052 Make sure we don't update the state to something we don't want.
12054 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12056 * gst/rtsp-server/rtsp-client.c:
12057 Add support for session keepalive
12058 Get and update the session timeout for all requests. get the session as early as
12061 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12063 * gst/rtsp-server/rtsp-media-factory.h:
12064 * gst/rtsp-server/rtsp-media.c:
12065 * gst/rtsp-server/rtsp-media.h:
12066 Handle media bus messages
12067 Handle media bus messages in a custom mainloop and dispatch them to the
12068 RTSPMedia objects. Let the default implementation handle some common messages.
12070 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12072 * gst/rtsp-server/rtsp-client.c:
12073 * gst/rtsp-server/rtsp-session-pool.c:
12074 * gst/rtsp-server/rtsp-session.c:
12075 Some more session timeout handling
12076 Move the session header setting code to a central place so that we always add
12077 the timeout parameter too.
12078 Handle timeouts by running the session cleanup code.
12079 Stop media before cleaning up.
12081 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12083 * gst/rtsp-server/rtsp-client.c:
12084 * gst/rtsp-server/rtsp-client.h:
12085 Add timeout property
12086 Add a timeout property ot the client and make the other properties into GObject
12089 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12091 * gst/rtsp-server/rtsp-session-pool.c:
12092 Use getters and setters in property code
12093 Use the getters and setters for the timeout property instead of locking
12096 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12098 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
12100 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12102 * gst/rtsp-server/rtsp-session-pool.c:
12103 * gst/rtsp-server/rtsp-session-pool.h:
12104 * gst/rtsp-server/rtsp-session.c:
12105 * gst/rtsp-server/rtsp-session.h:
12106 Add more timeout stuff
12107 Add method to check if a session is expired.
12108 Add method to perform cleanup on a session pool.
12110 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12112 * gst/rtsp-server/rtsp-client.c:
12113 * gst/rtsp-server/rtsp-session-pool.c:
12114 * gst/rtsp-server/rtsp-session-pool.h:
12115 * gst/rtsp-server/rtsp-session.c:
12116 * gst/rtsp-server/rtsp-session.h:
12117 Add beginnings of session timeouts and limits
12118 Add the timeout value to the Session header for unusual timeout values.
12119 Allow us to configure a limit to the amount of active sessions in a pool. Set a
12120 limit on the amount of retry we do after a sessionid collision.
12121 Add properties to the sessionid and the timeout of a session. Keep track of
12122 creation time and last access time for sessions.
12124 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12126 * gst/rtsp-server/rtsp-client.c:
12127 * gst/rtsp-server/rtsp-media.c:
12128 * gst/rtsp-server/rtsp-media.h:
12129 * gst/rtsp-server/rtsp-sdp.c:
12130 * gst/rtsp-server/rtsp-session-pool.c:
12131 * gst/rtsp-server/rtsp-session.c:
12132 * gst/rtsp-server/rtsp-session.h:
12133 Cleanup of sessions and more
12134 Fix the refcounting of media and sessions in the client. Properly clean up the
12135 session data when the client performs a teardown.
12136 Add Server header to responses.
12137 Allow for multiple uri setups in one session.
12138 Add Range header to the PLAY response and add the range attribute to the SDP
12140 Fix the session pool remove method, it used the wrong key in the hashtable. Also
12141 give the ownership of the sessionid to the session object.
12143 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12145 * gst/rtsp-server/rtsp-server.c:
12146 * gst/rtsp-server/rtsp-server.h:
12148 Rename the 'server_port' variable to simply 'port'.
12150 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12153 * gst/rtsp-server/rtsp-client.c:
12154 * gst/rtsp-server/rtsp-media.c:
12155 * gst/rtsp-server/rtsp-media.h:
12156 * gst/rtsp-server/rtsp-session.c:
12157 * gst/rtsp-server/rtsp-session.h:
12158 Rework the way we handle transports for streams
12159 Make the media accept an array of transports for the streams that we have
12160 configured for the play/pause requests.
12161 Implement server states for a client and its media.
12162 Require 0.10.22.1 (git HEAD) of gstreamer.
12164 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12166 * gst/rtsp-server/rtsp-client.c:
12167 * gst/rtsp-server/rtsp-media-factory.c:
12168 Drop const from functions dealing with urls
12169 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
12170 have the right const in them.
12172 2009-01-30 17:06:26 +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-sdp.c:
12180 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12182 * gst/rtsp-server/rtsp-client.c:
12183 * gst/rtsp-server/rtsp-media-factory.c:
12184 * gst/rtsp-server/rtsp-media.c:
12185 * gst/rtsp-server/rtsp-media.h:
12187 Don't keep a reference to the GstRTSPMedia in the stream.
12188 Free more things when freeing the GstRTSPMedia.
12190 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12193 * gst/rtsp-server/rtsp-media-factory.c:
12194 * gst/rtsp-server/rtsp-media-factory.h:
12195 * gst/rtsp-server/rtsp-media.c:
12196 * gst/rtsp-server/rtsp-media.h:
12197 * gst/rtsp-server/rtsp-server.c:
12198 * gst/rtsp-server/rtsp-server.h:
12199 More docs and small cleanups
12200 Add some more docs and update the README
12201 Cleanup some method names.
12202 Remove an unneeded idx field in the GstRTSPMediaStream
12204 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12207 * examples/Makefile.am:
12208 * examples/test-readme.c:
12209 Add a README and more example code
12210 Add a README file that contains a small introduction on how to use the server
12211 along with the example code explained in the readme.
12213 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12215 * gst/rtsp-server/rtsp-media.c:
12216 * gst/rtsp-server/rtsp-server.c:
12217 Fix some leaks and change default port
12218 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
12219 we finished the initial preroll. If we keep them locked, setting the pipeline to
12220 NULL will not stop and clean up the sources correctly.
12221 Change the default RTSP port to 8554 aka the official alternative RTSP port.
12223 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12225 * gst/rtsp-server/rtsp-session.c:
12226 * gst/rtsp-server/rtsp-session.h:
12227 Cleanups to the session object
12228 Remove some unneeded variables in the session state of a stream such as the
12229 owner media and the server transport.
12230 Get the configuration of a media stream in a session based on the media_stream
12231 in the original object instead of our cached index.
12232 Free more data in the finalize method.
12234 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12236 * gst/rtsp-server/rtsp-client.c:
12237 * gst/rtsp-server/rtsp-client.h:
12238 Cleanups and reuse media from DESCRIBE
12239 Handle thread create errors.
12240 Rename some internal methods to better match what they actually do.
12241 Handle misconfiguration of session_pool and media_mapping gracefully.
12242 Cache the DESCRIBE media and uri in the client connection and reuse them when
12243 we receive a SETUP request in the same connection for the same uri.
12244 Cleanup the client connection object.
12246 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12248 * gst/rtsp-server/rtsp-media-factory.c:
12249 * gst/rtsp-server/rtsp-media-factory.h:
12250 * gst/rtsp-server/rtsp-media.c:
12251 * gst/rtsp-server/rtsp-media.h:
12252 Add shared properties to media and factory
12253 Add the shared property to media.
12254 Implement some simple caching in the factory depending on if the media is shared
12257 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12259 * gst/rtsp-server/rtsp-client.c:
12260 Add a little comment
12261 Add some comment about the content-base header.
12263 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12265 * examples/Makefile.am:
12266 * examples/test-mp4.c:
12267 * examples/test-ogg.c:
12268 * examples/test-video.c:
12269 * gst/rtsp-server/Makefile.am:
12270 * gst/rtsp-server/rtsp-client.c:
12271 * gst/rtsp-server/rtsp-client.h:
12272 * gst/rtsp-server/rtsp-media-factory.c:
12273 * gst/rtsp-server/rtsp-media-factory.h:
12274 * gst/rtsp-server/rtsp-media.c:
12275 * gst/rtsp-server/rtsp-media.h:
12276 * gst/rtsp-server/rtsp-sdp.c:
12277 * gst/rtsp-server/rtsp-sdp.h:
12278 * gst/rtsp-server/rtsp-server.c:
12279 * gst/rtsp-server/rtsp-server.h:
12280 * gst/rtsp-server/rtsp-session.c:
12281 * gst/rtsp-server/rtsp-session.h:
12282 Reorganize things, prepare for media sharing
12283 Added various other test server examples
12284 Move the SDP message generation to a separate helper.
12285 Refactor common code for finding the session.
12286 Add content-base for realplayer compatibility
12287 Clean up request uris before processing for better vlc compatibility.
12288 Move prerolling and pipeline construction to the RTSPMedia object.
12289 Use multiudpsink for future pipeline reuse.
12291 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12294 Back to development
12297 === release 0.10.1 ===
12299 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12302 Make 0.10.1 release
12305 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12307 * bindings/vala/Makefile.am:
12309 Add more directories and files to the dist.
12311 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
12313 * bindings/python/Makefile.am:
12314 * bindings/python/rtspserver.override:
12315 Fixed compile error of python bindings
12317 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
12319 * bindings/vala/gst-rtsp-server-0.10.vapi:
12320 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
12321 Marked values as nullable accordingly
12323 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
12325 * bindings/vala/gst-rtsp-server-0.10.vapi:
12326 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
12327 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
12328 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
12329 Updated Vala bindings
12331 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12333 * gst/rtsp-server/rtsp-client.c:
12334 * gst/rtsp-server/rtsp-media-mapping.c:
12335 * gst/rtsp-server/rtsp-media-mapping.h:
12336 * gst/rtsp-server/rtsp-media.h:
12337 * gst/rtsp-server/rtsp-session-pool.h:
12338 Cleanups and doc updates
12339 Add some more documentation and do some minor cleanups here and there.
12341 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12343 * gst/rtsp-server/rtsp-client.c:
12344 * gst/rtsp-server/rtsp-media-factory.c:
12345 * gst/rtsp-server/rtsp-media-factory.h:
12346 * gst/rtsp-server/rtsp-media.c:
12347 * gst/rtsp-server/rtsp-media.h:
12348 * gst/rtsp-server/rtsp-session.c:
12349 * gst/rtsp-server/rtsp-session.h:
12351 Rename GstRTSPMediaBin to GstRTSPMedia
12352 Parse the request url into a GstRTSPUri object and pass this object to the
12353 various handlers and methods that require the uri.
12355 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12359 Add some more docs and remove some old code from the example.
12361 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12363 * gst/rtsp-server/rtsp-client.c:
12364 Handle state change failures better
12365 Handle state change failures better when changing the state of the pipeline to
12368 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12370 * gst/rtsp-server/rtsp-media-factory.c:
12371 * gst/rtsp-server/rtsp-media-factory.h:
12372 Make element creation more extendible
12373 Add get_element vmethod to the default MediaFactory so that subclasses can just
12374 override that method and still use the default logic for making a MediaBin from
12377 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12380 * gst/rtsp-server/Makefile.am:
12381 * gst/rtsp-server/rtsp-client.c:
12382 * gst/rtsp-server/rtsp-client.h:
12383 * gst/rtsp-server/rtsp-media-factory.c:
12384 * gst/rtsp-server/rtsp-media-factory.h:
12385 * gst/rtsp-server/rtsp-media-mapping.c:
12386 * gst/rtsp-server/rtsp-media-mapping.h:
12387 * gst/rtsp-server/rtsp-media.c:
12388 * gst/rtsp-server/rtsp-media.h:
12389 * gst/rtsp-server/rtsp-server.c:
12390 * gst/rtsp-server/rtsp-server.h:
12391 * gst/rtsp-server/rtsp-session.c:
12392 * gst/rtsp-server/rtsp-session.h:
12393 Make the server handle arbitrary pipelines
12394 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
12395 The GstMediaBin object has a handle to a bin with elements and to a list of
12396 GstMediaStream objects that this bin produces.
12397 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
12398 with methods to register and remove those mappings.
12399 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
12400 used by the server instance.
12401 Modify the example application so that it shows how to create custom pipelines
12402 attached to a specific mount point.
12403 Various misc cleanps.
12405 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12407 * gst/rtsp-server/rtsp-server.c:
12408 * gst/rtsp-server/rtsp-server.h:
12409 Allow setting a custom media factory for a server
12411 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12413 * gst/rtsp-server/rtsp-client.c:
12414 * gst/rtsp-server/rtsp-client.h:
12415 Allow setting a custom media factory for a client.
12417 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12419 * gst/rtsp-server/Makefile.am:
12420 Add Makefile entry for the media factory
12422 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12424 * gst/rtsp-server/rtsp-media-factory.c:
12425 * gst/rtsp-server/rtsp-media-factory.h:
12426 Add media factory to map urls to media pipeline objects.
12428 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12430 * gst/rtsp-server/rtsp-media.c:
12431 * gst/rtsp-server/rtsp-media.h:
12432 Add comments. Remove unused field
12434 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12436 * gst/rtsp-server/rtsp-session-pool.c:
12437 * gst/rtsp-server/rtsp-session-pool.h:
12438 Allow custom session pools to override the session id allocation algorithms Add some comments.
12440 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12442 * gst/rtsp-server/rtsp-session.h:
12445 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12447 * gst/rtsp-server/rtsp-client.c:
12448 * gst/rtsp-server/rtsp-client.h:
12449 Move the connection code in one place Add some comments
12451 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12453 * gst/rtsp-server/rtsp-server.c:
12454 * gst/rtsp-server/rtsp-server.h:
12455 Make vmethod to create and accept new clients. Add some docs.
12457 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12459 * gst/rtsp-server/rtsp-server.c:
12460 * gst/rtsp-server/rtsp-server.h:
12461 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
12463 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12465 * gst/rtsp-server/rtsp-client.c:
12466 * gst/rtsp-server/rtsp-client.h:
12467 Name the parameters more appropriately.
12469 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12471 * gst/rtsp-server/rtsp-session-pool.c:
12472 Do some more cleanup of the session pool.
12474 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12476 * gst/rtsp-server/Makefile.am:
12477 * gst/rtsp-server/rtsp-client.c:
12478 Check if return value of gst_rtsp_session_get_media is not NULL
12480 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12482 * gst/rtsp-server/Makefile.am:
12483 Install rtsp-session and rtsp-session-pool headers
12485 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12490 * bindings/python/Makefile.am:
12491 * bindings/python/arg-types.py:
12492 * bindings/python/codegen/Makefile.am:
12493 * bindings/python/codegen/__init__.py:
12494 * bindings/python/codegen/argtypes.py:
12495 * bindings/python/codegen/code-coverage.py:
12496 * bindings/python/codegen/codegen.py:
12497 * bindings/python/codegen/definitions.py:
12498 * bindings/python/codegen/defsparser.py:
12499 * bindings/python/codegen/docextract.py:
12500 * bindings/python/codegen/docgen.py:
12501 * bindings/python/codegen/fileprefix.override:
12502 * bindings/python/codegen/fileprefixmodule.c:
12503 * bindings/python/codegen/h2def.py:
12504 * bindings/python/codegen/mergedefs.py:
12505 * bindings/python/codegen/mkskel.py:
12506 * bindings/python/codegen/override.py:
12507 * bindings/python/codegen/reversewrapper.py:
12508 * bindings/python/codegen/scmexpr.py:
12509 * bindings/python/rtspserver-types.defs:
12510 * bindings/python/rtspserver.defs:
12511 * bindings/python/rtspserver.override:
12512 * bindings/python/rtspservermodule.c:
12514 Add python bindings.
12516 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12518 * bindings/Makefile.am:
12520 Don't go into python dir when requirements for python bindings are missing
12522 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12524 * bindings/Makefile.am:
12525 * bindings/vala/Makefile.am:
12527 Install Vala bindings if vala is available
12529 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
12531 * bindings/vala/gst-rtsp-server-0.10.deps:
12532 * bindings/vala/gst-rtsp-server-0.10.vapi:
12533 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
12534 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
12535 * bindings/vala/packages/gst-rtsp-server-0.10.files:
12536 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
12537 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
12538 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
12539 Regenerated Vala bindings
12541 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
12543 * bindings/vala/gst-rtsp-server.vapi:
12544 * bindings/vala/packages/gst-rtsp-server.metadata:
12545 Fixed typo in included headers for vala bindings
12547 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12551 * pkgconfig/Makefile.am:
12552 * pkgconfig/gst-rtsp-server.pc.in:
12553 Added pkgconfig file
12555 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
12557 * bindings/vala/gst-rtsp-server.vapi:
12558 * bindings/vala/packages/gst-rtsp-server.excludes:
12559 * bindings/vala/packages/gst-rtsp-server.gi:
12560 * bindings/vala/packages/gst-rtsp-server.metadata:
12561 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
12563 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
12565 * bindings/vala/gst-rtsp-server.vapi:
12566 * bindings/vala/packages/gst-rtsp-server.deps:
12567 * bindings/vala/packages/gst-rtsp-server.files:
12568 * bindings/vala/packages/gst-rtsp-server.gi:
12569 * bindings/vala/packages/gst-rtsp-server.metadata:
12570 * bindings/vala/packages/gst-rtsp-server.namespace:
12571 Added Vala bindings
12573 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
12575 * gst/rtsp-server/rtsp-session.c:
12576 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
12578 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
12580 * examples/Makefile.am:
12581 * gst/rtsp-server/Makefile.am:
12582 Put GStreamer version in library name
12584 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12586 * examples/Makefile.am:
12587 * gst/rtsp-server/Makefile.am:
12588 Fix some issues to pass distcheck
12590 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12592 * gst/rtsp-server/rtsp-server.c:
12593 Added port property to GstRTSPServer class.
12595 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12600 * examples/Makefile.am:
12603 * gst/rtsp-server/Makefile.am:
12604 * gst/rtsp-server/rtsp-client.c:
12605 * gst/rtsp-server/rtsp-client.h:
12606 * gst/rtsp-server/rtsp-media.c:
12607 * gst/rtsp-server/rtsp-media.h:
12608 * gst/rtsp-server/rtsp-server.c:
12609 * gst/rtsp-server/rtsp-server.h:
12610 * gst/rtsp-server/rtsp-session-pool.c:
12611 * gst/rtsp-server/rtsp-session-pool.h:
12612 * gst/rtsp-server/rtsp-session.c:
12613 * gst/rtsp-server/rtsp-session.h:
12615 Split in library and example program
12617 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
12619 * src/rtsp-client.h:
12620 Removed obsolete variable
12622 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
12624 * src/rtsp-client.c:
12625 * src/rtsp-client.h:
12626 Removed pipeline variable GstRTSPClient, because it's only used in one function
12628 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12630 * src/rtsp-media.c:
12631 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
12633 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
12635 * src/rtsp-session.c:
12636 Initialize some more vars.
12638 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
12640 * src/rtsp-session.c:
12641 Initialize variable to avoid compiler warning.
12643 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
12646 Add a reasonable generic .gitignore