3 2019-04-19 00:34:54 +0100 Tim-Philipp Müller <tim@centricular.com>
9 * gst-rtsp-server.doap:
13 2019-04-15 20:33:01 +0300 Sebastian Dröge <sebastian@centricular.com>
15 * gst/rtsp-sink/gstrtspclientsink.c:
16 rtspclientsink: Notify the stream transport about each written message
17 Otherwise it will never try to send us the next one: it tries to keep
18 exactly one message in-flight all the time.
19 In gst-rtsp-server this is done asynchronously via the GstRTSPWatch but
20 in the client sink we always write data out synchronously.
22 2019-04-02 08:05:03 +0200 Göran Jönsson <goranjn@axis.com>
24 * gst/rtsp-server/rtsp-stream.c:
25 rtsp_server: Free thread pool before clean transport cache
26 If not waiting for free thread pool before clean transport caches, there
27 can be a crash if a thread is executing in transport list loop in
28 function send_tcp_message.
29 Also add a check if priv->send_pool in on_message_sent to avoid that a
30 new thread is pushed during wait of free thread pool. This is possible
31 since when waiting for free thread pool mutex have to be unlocked.
33 === release 1.15.90 ===
35 2019-04-11 00:35:55 +0100 Tim-Philipp Müller <tim@centricular.com>
41 * gst-rtsp-server.doap:
45 2019-04-10 10:32:53 +0200 Ulf Olsson <ulfo@axis.com>
47 * gst/rtsp-server/rtsp-stream.c:
48 rtsp-stream: Add support for GCM (RFC 7714)
51 2019-03-28 00:27:37 +0100 Erlend Eriksen <erlend_ne@hotmail.com>
53 * gst/rtsp-server/rtsp-session-pool.c:
54 session pool: fix missing klass-> in klass->create_session
56 2019-03-23 19:16:17 +0000 Tim-Philipp Müller <tim@centricular.com>
59 g-i: pass --quiet to g-ir-scanner
60 This suppresses the annoying 'g-ir-scanner: link: cc ..' output
61 that we get even if everything works just fine.
62 We still get g-ir-scanner warnings and compiler warnings if
65 2019-03-23 19:15:48 +0000 Tim-Philipp Müller <tim@centricular.com>
68 g-i: silence 'nested extern' compiler warnings when building scanner binary
69 We need a nested extern in our init section for the scanner binary
70 so we can call gst_init to make sure GStreamer types are initialised
71 (they are not all lazy init via get_type functions, but some are in
72 exported variables). There doesn't seem to be any other mechanism to
73 achieve this, so just remove that warning, it's not important at all.
75 2019-03-21 11:49:10 +0000 Tim-Philipp Müller <tim@centricular.com>
78 meson: pass -Wno-unused to compiler if gstreamer debug system is disabled
80 2019-03-14 07:37:26 +0100 Göran Jönsson <goranjn@axis.com>
82 * gst/rtsp-server/rtsp-media.c:
83 * tests/check/gst/media.c:
84 rtsp-media: Handle set state when preparing.
85 Handle the situation when a call to gst_rtsp_media_set_state is done
86 when media status is preparing.
87 Also add unit test for this scenario.
88 The unit test simulate on a media level when two clients share a (live)
90 Both clients have done SETUP and got responses. Now client 1 is doing
91 play and client 2 is just closing the connection.
92 Then without patch there are a problem when
93 client1 is calling gst_rtsp_media_unsuspend in handle_play_request.
94 And client2 is doing closing connection we can end up in a call
95 to gst_rtsp_media_set_state when
96 priv->status == GST_RTSP_MEDIA_STATUS_PREPARING and all the logic for
97 shut down media is jumped over .
98 With this patch and this scenario we wait until
99 priv->status == GST_RTSP_MEDIA_STATUS_PREPARED and then continue to
100 execute after that and now we will execute the logic for
103 2019-03-04 09:13:30 +0000 Tim-Philipp Müller <tim@centricular.com>
111 === release 1.15.2 ===
113 2019-02-26 11:58:53 +0000 Tim-Philipp Müller <tim@centricular.com>
119 * gst-rtsp-server.doap:
123 2019-02-19 09:45:08 +0100 Göran Jönsson <goranjn@axis.com>
125 * gst/rtsp-server/rtsp-media.c:
126 * tests/check/gst/client.c:
127 rtsp-media: Fix multicast use case with common media
136 2019-01-16 12:59:11 +0100 Göran Jönsson <goranjn@axis.com>
138 * gst/rtsp-server/rtsp-client.c:
139 * gst/rtsp-server/rtsp-stream.c:
140 * gst/rtsp-server/rtsp-stream.h:
141 rtsp-server: remove recursive behavior
142 Introduce a threadpool to send rtp and rtcp to avoid recursive behavior.
144 2019-01-25 14:22:42 +0200 Sebastian Dröge <sebastian@centricular.com>
146 * gst/rtsp-server/rtsp-client.c:
147 rtsp-client: Only allow to set either a send_func or send_messages_func but not both
148 And route all messages through the send_func if no send_messages_func
150 We otherwise break backwards compatibility.
152 2018-09-17 22:18:46 +0300 Sebastian Dröge <sebastian@centricular.com>
154 * docs/libs/gst-rtsp-server-sections.txt:
155 * gst/rtsp-server/rtsp-client.c:
156 * gst/rtsp-server/rtsp-client.h:
157 * gst/rtsp-server/rtsp-stream.c:
158 rtsp-client: Add support for sending buffer lists directly
159 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
161 2018-06-27 12:17:07 +0200 Sebastian Dröge <sebastian@centricular.com>
163 * docs/libs/gst-rtsp-server-sections.txt:
164 * gst/rtsp-server/rtsp-client.c:
165 * gst/rtsp-server/rtsp-media.c:
166 * gst/rtsp-server/rtsp-stream-transport.c:
167 * gst/rtsp-server/rtsp-stream-transport.h:
168 * gst/rtsp-server/rtsp-stream.c:
169 * gst/rtsp-sink/gstrtspclientsink.c:
170 rtsp-server: Add support for buffer lists
171 This adds new functions for passing buffer lists through the different
172 layers without breaking API/ABI, and enables the appsink to actually
173 provide buffer lists.
174 This should already reduce CPU usage and potentially context switches a
175 bit by passing a whole buffer list from the appsink instead of
176 individual buffers. As a next step it would be necessary to
177 a) Add support for a vector of data for the GstRTSPMessage body
178 b) Add support for sending multiple messages at once to the
179 GstRTSPWatch and let it be handled internally
180 c) Adding API to GOutputStream that works like writev()
181 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
183 2018-12-04 14:12:04 +0100 Benjamin Berg <bberg@redhat.com>
185 * gst/rtsp-server/rtsp-client.c:
186 client: Fix crash in close handler
187 The close handler could trigger a crash because it invalidated the
188 watch_context while still leaving a source attached to it which would be
189 cleaned up at a later point.
191 2019-01-29 14:42:35 +0100 Edward Hervey <edward@centricular.com>
193 * gst/rtsp-server/rtsp-stream.c:
194 rtsp-stream: Use cached address when allocating sockets
195 If an address/port was previously decided upon (ex: multicast in the
196 SDP), then use that instead of re-creating another one
197 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/57
199 2018-12-27 11:28:17 +0100 Lars Wiréen <larswi@axis.com>
201 * gst/rtsp-server/rtsp-media.c:
202 rtsp-media: Fix race codition in finish_unprepare
203 The previous fix for race condition around finish_unprepare where the
204 function could be called twice assumed that the status wouldn't change
205 during execution of the function. This assumption is incorrect as the
206 state may change, for example if an error message arrives from the
208 Instead a flag keeping track on whether the finish_unprepare function
209 is currently executing is introduced and checked.
210 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/59
212 === release 1.15.1 ===
214 2019-01-17 02:26:48 +0000 Tim-Philipp Müller <tim@centricular.com>
220 * gst-rtsp-server.doap:
224 2018-12-05 15:07:25 +0100 Patricia Muscalu <patricia@axis.com>
226 * gst/rtsp-server/rtsp-stream.c:
227 Add source elements to the pipeline before activation
228 In plug_src we changed the element state before adding it to
229 the owner container. This prevented the pipeline from intercepting
230 a GST_STREAM_STATUS_TYPE_CREATE message from the pad in order
231 to assign a custom task pool.
232 Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/53
234 2018-12-05 17:24:59 -0300 Thibault Saunier <tsaunier@igalia.com>
237 Automatic update of common submodule
238 From ed78bee to 59cb678
240 2018-11-20 19:12:09 +0100 Ingo Randolf <ingo.randolf@servus.at>
242 * examples/test-appsrc.c:
243 examples: test-appsrc: fix coding style error
245 2018-11-20 11:07:48 +0100 Ingo Randolf <ingo.randolf@servus.at>
247 * examples/test-appsrc.c:
248 examples: test-appsrc: fix buffer leak
250 2018-11-17 19:19:54 +0100 Patricia Muscalu <patricia@axis.com>
252 * gst/rtsp-server/rtsp-media.c:
253 rtsp-media: Update priv->blocked when linked streams are unblocked.
254 Media is considered to be blocked when all streams that belong to
255 that media are blocked.
256 This patch solves the problem of inconsistent updates of
257 priv->blocked that are not synchronized with the media state.
259 2018-11-17 18:18:27 +0100 Patricia Muscalu <patricia@axis.com>
261 * gst/rtsp-server/rtsp-media.c:
262 rtsp-media: Don't block streams before seeking
263 Before the seek operation is performed on media, it's required that
264 its pipeline is prepared <=> the pipeline is in the PAUSED state.
265 At this stage, all transport parts (transport sinks) have been successfully
266 added to the pipeline and there is no need for blocking the streams.
268 2018-11-17 16:11:53 +0100 Patricia Muscalu <patricia@axis.com>
270 * tests/check/gst/rtspserver.c:
271 tests: rtspserver: Add shared media test case for TCP
273 2018-11-06 18:21:54 +0100 Linus Svensson <linussn@axis.com>
275 * gst/rtsp-server/rtsp-stream.c:
276 rtsp-stream: Use seqnum-offset for rtpinfo
277 The sequence number in the rtpinfo is supposed to be the first RTP
278 sequence number. The "seqnum" property on a payloader is supposed to be
279 the number from the last processed RTP packet. The sequence number for
280 payloaders that inherit gstrtpbasepayload will not be correct in case of
281 buffer lists. In order to fix the seqnum property on the payloaders
282 gst-rtsp-server must get the sequence number for rtpinfo elsewhere and
283 "seqnum-offset" from the "stats" property contains the value of the
284 very first RTP packet in a stream. The server will, however, try to look
285 at the last simple in the sink element and only use properties on the
286 payloader in case there no sink elements yet, and by looking at the last
287 sample of the sink gives the server full control of which RTP packet it
288 looks at. If the payloader does not have the "stats" property, "seqnum"
289 is still used since "seqnum-offset" is only present in as part of
290 "stats" and this is still an issue not solved with this patch.
291 Needed for gst-plugins-base!17
293 2018-11-06 18:10:56 +0100 Linus Svensson <linussn@axis.com>
295 * gst/rtsp-server/rtsp-stream.c:
296 rtsp-stream: Plug memory leak
297 Attaching a GSource to a context will increase the refcount. The idle
298 source will never be free'd since the initial reference is never
301 2018-11-12 16:06:39 +0200 Jordan Petridis <jordan@centricular.com>
304 Add Gitlab CI configuration
305 This commit adds a .gitlab-ci.yml file, which uses a feature
306 to fetch the config from a centralized repository. The intent is
307 to have all the gstreamer modules use the same configuration.
308 The configuration is currently hosted at the gst-ci repository
309 under the gitlab/ci_template.yml path.
310 Part of https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/29
312 2018-11-05 05:56:35 +0000 Matthew Waters <matthew@centricular.com>
315 * gst-rtsp-server.doap:
316 Update git locations to gitlab
318 2018-11-01 14:20:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
320 * gst/rtsp-server/meson.build:
321 meson: add new onvif types
323 2018-11-01 12:49:51 +0200 Sebastian Dröge <sebastian@centricular.com>
325 * gst/rtsp-server/meson.build:
326 Add ONVIF subclass headers to the installed headers in meson.build too
328 2018-11-01 11:29:01 +0200 Sebastian Dröge <sebastian@centricular.com>
330 * gst/rtsp-server/rtsp-server-object.h:
331 * gst/rtsp-server/rtsp-server.h:
332 rtsp-server: Declare GstRTSPServer struct before anything else
333 It's needed by all kinds of other headers, including the ones that are
334 required for defining the GstRTSPServer struct itself and its API.
336 2018-11-01 10:23:02 +0200 Sebastian Dröge <sebastian@centricular.com>
338 * gst/rtsp-server/rtsp-onvif-client.h:
339 * gst/rtsp-server/rtsp-onvif-media-factory.h:
340 * gst/rtsp-server/rtsp-onvif-media.h:
341 * gst/rtsp-server/rtsp-onvif-server.h:
342 Mark all ONVIF-specific subclasses as Since 1.14
344 2018-11-01 10:18:22 +0200 Sebastian Dröge <sebastian@centricular.com>
346 * gst/rtsp-server/Makefile.am:
347 * gst/rtsp-server/meson.build:
348 * gst/rtsp-server/rtsp-context.h:
349 * gst/rtsp-server/rtsp-onvif-server.c:
350 * gst/rtsp-server/rtsp-onvif-server.h:
351 * gst/rtsp-server/rtsp-server-object.h:
352 * gst/rtsp-server/rtsp-server-prelude.h:
353 * gst/rtsp-server/rtsp-server.c:
354 * gst/rtsp-server/rtsp-server.h:
355 * gst/rtsp-server/rtsp-session.h:
356 Include ONVIF types from single-include rtsp-server.h
357 ... by actually making it a single-include header and moving everything
358 related to the GstRTSPServer type to rtsp-server-object.h instead.
359 Otherwise there are too many circular includes.
360 https://bugzilla.gnome.org/show_bug.cgi?id=797361
362 2018-10-18 07:25:05 +0200 Göran Jönsson <goranjn@axis.com>
364 * gst/rtsp-server/rtsp-client.c:
365 * gst/rtsp-server/rtsp-latency-bin.c:
366 * gst/rtsp-server/rtsp-stream.c:
367 * gst/rtsp-server/rtsp-stream.h:
368 rtsp-stream: use idle source in on_message_sent
369 When the underlying layers are running on_message_sent, this sometimes
370 causes the underlying layer to send more data, which will cause the
371 underlying layer to run callback on_message_sent again. This can go on
373 To break this chain, we introduce an idle source that takes care of
374 sending data if there are more to send when running callback
375 https://bugzilla.gnome.org/show_bug.cgi?id=797289
377 2018-10-20 16:14:53 +0200 Edward Hervey <edward@centricular.com>
379 * gst/rtsp-server/rtsp-client.c:
380 rtsp-client: Remove timeout GSource on cleanup
381 Avoids ending up with races where a timeout would still be around
382 *after* a client was gone. This could happen rather easily in
383 RTSP-over-HTTP mode on a local connection, where each RTSP message
384 would be sent as a different HTTP connection with the same tunnelid.
385 If not properly removed, that timeout would then try to free again
386 a client (and its contents).
388 2018-10-04 14:31:59 +0100 Tim-Philipp Müller <tim@centricular.com>
390 * gst/rtsp-server/Makefile.am:
391 autotools: fix distcheck
393 2018-09-12 11:55:15 +0200 Ognyan Tonchev <ognyan@axis.com>
395 * gst/rtsp-server/Makefile.am:
396 * gst/rtsp-server/meson.build:
397 * gst/rtsp-server/rtsp-latency-bin.c:
398 * gst/rtsp-server/rtsp-latency-bin.h:
399 * gst/rtsp-server/rtsp-onvif-media.c:
400 onvif: encapsulate onvif part into a bin
401 ...and thus do not let onvif affect pipelines latency
402 https://bugzilla.gnome.org/show_bug.cgi?id=797174
404 2018-09-27 19:57:13 +0200 Patricia Muscalu <patricia@dovakhiin.com>
406 * tests/check/gst/client.c:
407 tests: client: Avoid bind() failures in tests
408 https://bugzilla.gnome.org/show_bug.cgi?id=797059
410 2018-09-06 16:17:33 +0200 Patricia Muscalu <patricia@axis.com>
412 * gst/rtsp-server/rtsp-media-factory.c:
413 * gst/rtsp-server/rtsp-media-factory.h:
414 * gst/rtsp-server/rtsp-media.c:
415 * gst/rtsp-server/rtsp-media.h:
416 * gst/rtsp-server/rtsp-stream.c:
417 * gst/rtsp-server/rtsp-stream.h:
418 * tests/check/gst/client.c:
419 * tests/check/gst/mediafactory.c:
420 New property for socket binding to mcast addresses
421 By default the multicast sockets are bound to INADDR_ANY,
422 as it's not allowed to bind sockets to multicast addresses
423 in Windows. This default behaviour can be changed by setting
424 bind-mcast-address property on the media-factory object.
425 https://bugzilla.gnome.org/show_bug.cgi?id=797059
427 2018-09-24 09:36:21 +0100 Tim-Philipp Müller <tim@centricular.com>
430 * gst/rtsp-server/Makefile.am:
431 * gst/rtsp-server/meson.build:
432 * gst/rtsp-server/rtsp-address-pool.c:
433 * gst/rtsp-server/rtsp-auth.c:
434 * gst/rtsp-server/rtsp-client.c:
435 * gst/rtsp-server/rtsp-context.c:
436 * gst/rtsp-server/rtsp-media-factory-uri.c:
437 * gst/rtsp-server/rtsp-media-factory.c:
438 * gst/rtsp-server/rtsp-media.c:
439 * gst/rtsp-server/rtsp-mount-points.c:
440 * gst/rtsp-server/rtsp-params.c:
441 * gst/rtsp-server/rtsp-permissions.c:
442 * gst/rtsp-server/rtsp-sdp.c:
443 * gst/rtsp-server/rtsp-server-prelude.h:
444 * gst/rtsp-server/rtsp-server.c:
445 * gst/rtsp-server/rtsp-session-media.c:
446 * gst/rtsp-server/rtsp-session-pool.c:
447 * gst/rtsp-server/rtsp-session.c:
448 * gst/rtsp-server/rtsp-stream-transport.c:
449 * gst/rtsp-server/rtsp-stream.c:
450 * gst/rtsp-server/rtsp-thread-pool.c:
451 * gst/rtsp-server/rtsp-token.c:
453 libs: fix API export/import and 'inconsistent linkage' on MSVC
454 Export rtsp-server library API in headers when we're building the
455 library itself, otherwise import the API from the headers.
456 This fixes linker warnings on Windows when building with MSVC.
457 Fix up some missing config.h includes when building the lib which
458 is needed to get the export api define from config.h
459 https://bugzilla.gnome.org/show_bug.cgi?id=797185
461 2018-09-19 14:31:56 +0200 Edward Hervey <edward@centricular.com>
463 * gst/rtsp-server/rtsp-media-factory.c:
464 rtsp-media-factory: Add missing break statements
465 This resulted in warnings/assertions whenever one accessed the
466 max-mcast-ttl property.
470 2018-09-19 12:21:30 +0100 Tim-Philipp Müller <tim@centricular.com>
474 meson: add gobject-cast-checks, glib-asserts, glib-checks options
476 2018-09-19 12:17:57 +0100 Tim-Philipp Müller <tim@centricular.com>
480 * tests/check/meson.build:
481 meson: add option to disable build of rtspclientsink plugin
483 2018-09-19 12:10:14 +0100 Tim-Philipp Müller <tim@centricular.com>
486 meson: re-arrange options
488 2018-09-01 11:21:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
492 * tests/check/meson.build:
494 meson: Use feature option for tests option
495 This was somehow missed the last time around.
497 2018-08-31 14:42:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
499 * gst/rtsp-server/meson.build:
501 meson: Maintain macOS ABI through dylib versioning
502 Requires Meson 0.48, but the feature will be ignored on older versions
503 so it's safe to add it without bumping the requirement.
505 https://github.com/mesonbuild/meson/blob/master/docs/markdown/Reference-manual.md#shared_library
507 2018-08-31 17:20:47 +1000 Matthew Waters <matthew@centricular.com>
509 * gst/rtsp-sink/meson.build:
511 meson: add pkg-config file for the rtspclientsink plugin
513 2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
515 * gst/rtsp-server/rtsp-client.c:
516 * tests/check/gst/client.c:
517 rtsp-client: Avoid reuse of channel numbers for interleaved
518 If a (strange) client would reuse interleaved channel numbers in
519 multiple SETUP requests, we should not accept them. The channel
520 numbers are used for looking up stream transports in the
521 priv->transports hash table, and transports disappear from the table
522 if channel numbers are reused.
523 RFC 7826 (RTSP 2.0), Section 18.54, clarifies that it is OK for the
524 server to change the channel numbers suggested by the client.
525 https://bugzilla.gnome.org/show_bug.cgi?id=796988
527 2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
529 * tests/check/gst/client.c:
530 rtsp-client: Add unit test of SETUP for RTSP/RTP/TCP
531 Allow regex for matching transport header against expected pattern.
532 https://bugzilla.gnome.org/show_bug.cgi?id=796988
534 2018-08-15 18:57:27 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
536 * tests/check/meson.build:
537 meson: There is no gstreamer-plugins-good-1.0.pc
538 There is no installed version of that, only an uninstalled version.
540 2018-08-14 14:31:55 +0300 Sebastian Dröge <sebastian@centricular.com>
542 * gst/rtsp-server/rtsp-client.c:
543 * tests/check/gst/stream.c:
544 Fix indentation again
546 2018-07-26 12:01:16 +0200 Patricia Muscalu <patricia@axis.com>
548 * gst/rtsp-server/rtsp-client.c:
549 * gst/rtsp-server/rtsp-stream.c:
550 * gst/rtsp-server/rtsp-stream.h:
551 * tests/check/gst/client.c:
552 * tests/check/gst/stream.c:
553 stream: Added a list of multicast client addresses
554 When media is shared, the same media stream can be sent
555 to multiple multicast groups. Currently, there is no API
556 to retrieve multicast addresses from the stream.
557 When calling gst_rtsp_stream_get_multicast_address() function,
558 only the first multicast address is returned.
559 With this patch, each multicast destination requested in SETUP
560 will be stored in an internal list (call to
561 gst_rtsp_stream_add_multicast_client_address()).
562 The list of multicast groups requested by the clients can be
563 retrieved by calling gst_rtsp_stream_get_multicast_client_addresses().
564 There still exist some problems with the current implementation
565 in the multicast case:
566 1) The receiving part is currently only configured with
567 regard to the first multicast client (see
568 https://bugzilla.gnome.org/show_bug.cgi?id=796917).
569 2) Secondly, of security reasons, some constraints should be
570 put on the requested multicast destinations (see
571 https://bugzilla.gnome.org/show_bug.cgi?id=796916).
572 Change-Id: I6b060746e472a0734cc2fd828ffe4ea2956733ea
573 https://bugzilla.gnome.org/show_bug.cgi?id=793441
575 2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
577 * gst/rtsp-server/rtsp-client.c:
578 * gst/rtsp-server/rtsp-stream.c:
579 * gst/rtsp-server/rtsp-stream.h:
580 * tests/check/gst/client.c:
581 stream: Choose the maximum ttl value provided by multicast clients
582 The maximum ttl value provided so far by the multicast clients
583 will be chosen and reported in the response to the current
585 Change-Id: I5408646e3b5a0a224d907ae215bdea60c4f1905f
586 https://bugzilla.gnome.org/show_bug.cgi?id=793441
588 2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
590 * gst/rtsp-server/rtsp-stream.c:
591 * tests/check/gst/client.c:
592 rtsp-stream: Don't require address pool in the transport specific case
593 If "transport.client-settings" parameter is set to true, the client is
594 allowed to specify destination, ports and ttl.
595 There is no need for pre-configured address pool.
596 Change-Id: I6ae578fb5164d78e8ec1e2ee82dc4eaacd0912d1
597 https://bugzilla.gnome.org/show_bug.cgi?id=793441
599 2018-07-24 14:02:40 +0200 Patricia Muscalu <patricia@axis.com>
601 * gst/rtsp-server/rtsp-client.c:
602 * tests/check/gst/client.c:
603 client: Don't reserve multicast address in the client setting case
604 When two multicast clients request specific transport
605 configurations, and "transport.client-settings" parameter is
606 set to true, it's wrong to actually require that these two
607 clients request the same multicast group.
608 Removed test_client_multicast_invalid_transport_specific test
609 cases as they wrongly require that the requested destination
610 address is supposed to be present in the address pool, also in
611 the case when "transport.client-settings" parameter is set to true.
612 Change-Id: I4580182ef35996caf644686d6139f72ec599c9fa
613 https://bugzilla.gnome.org/show_bug.cgi?id=793441
615 2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
617 * gst/rtsp-server/rtsp-media-factory.c:
618 * gst/rtsp-server/rtsp-media-factory.h:
619 * gst/rtsp-server/rtsp-media.c:
620 * gst/rtsp-server/rtsp-media.h:
621 * gst/rtsp-server/rtsp-stream.c:
622 * gst/rtsp-server/rtsp-stream.h:
623 * tests/check/gst/mediafactory.c:
624 Add new API for setting/getting maximum multicast ttl value
625 Change-Id: I5ef4758188c14785e17fb8fbf42a3dc0cb054233
626 https://bugzilla.gnome.org/show_bug.cgi?id=793441
628 2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
630 * gst/rtsp-server/rtsp-stream.c:
631 rtsp-stream: avoid duplicating the first multicast client
632 In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
633 clients were dynamically added and removed to the multicast
634 udp sinks, as such we should no longer add a first client in
635 set_multicast_socket_for_udpsink
636 https://bugzilla.gnome.org/show_bug.cgi?id=793441
638 2018-08-14 14:25:53 +0300 Sebastian Dröge <sebastian@centricular.com>
640 * gst/rtsp-server/rtsp-stream.c:
641 Revert "rtsp-stream: avoid duplicating the first multicast client"
642 This reverts commit 33570944401747f44d8ebfec535350651413fb92.
643 Commits where accidentially squashed together
645 2018-08-14 14:25:42 +0300 Sebastian Dröge <sebastian@centricular.com>
647 * gst/rtsp-server/rtsp-client.c:
648 * gst/rtsp-server/rtsp-media-factory.c:
649 * gst/rtsp-server/rtsp-media-factory.h:
650 * gst/rtsp-server/rtsp-media.c:
651 * gst/rtsp-server/rtsp-media.h:
652 * gst/rtsp-server/rtsp-stream.c:
653 * gst/rtsp-server/rtsp-stream.h:
654 * tests/check/gst/client.c:
655 * tests/check/gst/mediafactory.c:
656 Revert "Add new API for setting/getting maximum multicast ttl value"
657 This reverts commit 7f0ae77e400fb8a0462a76a5dd2e63e12c4a2e52.
658 Commits where accidentially squashed together
660 2018-08-14 14:25:37 +0300 Sebastian Dröge <sebastian@centricular.com>
662 * gst/rtsp-server/rtsp-stream.c:
663 * tests/check/gst/client.c:
664 Revert "rtsp-stream: Don't require address pool in the transport specific case"
665 This reverts commit a9db3e7f092cfeb5475e9aa24b1e91906c141d52.
666 Commits where accidentially squashed together
668 2018-08-14 14:25:14 +0300 Sebastian Dröge <sebastian@centricular.com>
670 * gst/rtsp-server/rtsp-client.c:
671 * gst/rtsp-server/rtsp-stream.c:
672 * gst/rtsp-server/rtsp-stream.h:
673 * tests/check/gst/client.c:
674 * tests/check/gst/stream.c:
675 Revert "stream: Choose the maximum ttl value provided by multicast clients"
676 This reverts commit 499e437e501215849d24cdaa157e0edf4de097d0.
677 Commits where accidentially squashed together
679 2018-08-14 14:10:56 +0300 Sebastian Dröge <sebastian@centricular.com>
681 * examples/test-auth-digest.c:
682 examples: Fix indentation
684 2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
686 * gst/rtsp-server/rtsp-client.c:
687 * gst/rtsp-server/rtsp-stream.c:
688 * gst/rtsp-server/rtsp-stream.h:
689 * tests/check/gst/client.c:
690 * tests/check/gst/stream.c:
691 stream: Choose the maximum ttl value provided by multicast clients
692 The maximum ttl value provided so far by the multicast clients
693 will be chosen and reported in the response to the current
695 https://bugzilla.gnome.org/show_bug.cgi?id=793441
697 2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
699 * gst/rtsp-server/rtsp-stream.c:
700 * tests/check/gst/client.c:
701 rtsp-stream: Don't require address pool in the transport specific case
702 If "transport.client-settings" parameter is set to true, the client is
703 allowed to specify destination, ports and ttl.
704 There is no need for pre-configured address pool.
705 https://bugzilla.gnome.org/show_bug.cgi?id=793441
707 2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
709 * gst/rtsp-server/rtsp-client.c:
710 * gst/rtsp-server/rtsp-media-factory.c:
711 * gst/rtsp-server/rtsp-media-factory.h:
712 * gst/rtsp-server/rtsp-media.c:
713 * gst/rtsp-server/rtsp-media.h:
714 * gst/rtsp-server/rtsp-stream.c:
715 * gst/rtsp-server/rtsp-stream.h:
716 * tests/check/gst/client.c:
717 * tests/check/gst/mediafactory.c:
718 Add new API for setting/getting maximum multicast ttl value
719 https://bugzilla.gnome.org/show_bug.cgi?id=793441
721 2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
723 * gst/rtsp-server/rtsp-stream.c:
724 rtsp-stream: avoid duplicating the first multicast client
725 In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
726 clients were dynamically added and removed to the multicast
727 udp sinks, as such we should no longer add a first client in
728 set_multicast_socket_for_udpsink
729 https://bugzilla.gnome.org/show_bug.cgi?id=793441
731 2018-08-06 15:33:04 -0400 Thibault Saunier <tsaunier@igalia.com>
733 * gst/rtsp-server/Makefile.am:
734 rtsp-server: Add gstreamer-base gir dir in autotools
736 2018-07-25 19:54:55 +0200 Mathieu Duponchelle <mathieu@centricular.com>
738 * gst/rtsp-server/rtsp-client.c:
739 * gst/rtsp-server/rtsp-stream.c:
740 rtsp-client: always allocate both IPV4 and IPV6 sockets
741 multiudpsink does not support setting the socket* properties
742 after it has started, which meant that rtsp-server could no
743 longer serve on both IPV4 and IPV6 sockets since the patches
744 from https://bugzilla.gnome.org/show_bug.cgi?id=757488 were
746 When first connecting an IPV6 client then an IPV4 client,
747 multiudpsink fell back to using the IPV6 socket.
748 When first connecting an IPV4 client, then an IPV6 client,
749 multiudpsink errored out, released the IPV4 socket, then
750 crashed when trying to send a message on NULL nevertheless,
751 that is however a separate issue.
752 This could probably be fixed by handling the setting of
753 sockets in multiudpsink after it has started, that will
754 however be a much more significant effort.
755 For now, this commit simply partially reverts the behaviour
756 of rtsp-stream: it will continue to only create the udpsinks
757 when needed, as was the case since the patches were merged,
758 it will however when creating them, always allocate both
759 sockets and set them on the sink before it starts, as was
760 the case prior to the patches.
761 Transport configuration will only error out if the allocation
762 of UDP sockets fails for the actual client's family, this
763 also downgrades the GST_ERRORs in alloc_ports_one_family
764 to GST_WARNINGs, as failing to allocate is no longer
766 https://bugzilla.gnome.org/show_bug.cgi?id=796875
768 2018-07-25 17:22:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
772 meson: Convert common options to feature options
773 These are necessary for gst-build to set options correctly. The
774 remaining automagic option is cgroup support in examples.
775 https://bugzilla.gnome.org/show_bug.cgi?id=795107
777 2018-07-23 18:03:51 +0300 Sebastian Dröge <sebastian@centricular.com>
779 * gst/rtsp-server/rtsp-stream.c:
780 rtsp-stream: Slightly simplify locking
782 2018-06-28 11:22:21 +0200 David Svensson Fors <davidsf@axis.com>
784 * gst/rtsp-server/rtsp-client.c:
785 * gst/rtsp-server/rtsp-stream-transport.c:
786 * gst/rtsp-server/rtsp-stream-transport.h:
787 * gst/rtsp-server/rtsp-stream.c:
788 Limit queued TCP data messages to one per stream
789 Before, the watch backlog size in GstRTSPClient was changed
790 dynamically between unlimited and a fixed size, trying to avoid both
791 unlimited memory usage and deadlocks while waiting for place in the
792 queue. (Some of the deadlocks were described in a long comment in
794 In the previous commit, we changed to a fixed backlog size of 100.
795 This is possible, because we now handle RTP/RTCP data messages differently
796 from RTSP request/response messages.
797 The data messages are messages tunneled over TCP. We allow at most one
798 queued data message per stream in GstRTSPClient at a time, and
799 successfully sent data messages are acked by sending a "message-sent"
800 callback from the GstStreamTransport. Until that ack comes, the
801 GstRTSPStream does not call pull_sample() on its appsink, and
802 therefore the streaming thread in the pipeline will not be blocked
803 inside GstRTSPClient, waiting for a place in the queue.
804 pull_sample() is called when we have both an ack and a "new-sample"
805 signal from the appsink. Then, we know there is a buffer to write.
806 RTSP request/response messages are not acked in the same way as data
807 messages. The rest of the 100 places in the queue are used for
808 them. If the queue becomes full of request/response messages, we
809 return an error and close the connection to the client.
810 Change-Id: I275310bc90a219ceb2473c098261acc78be84c97
812 2018-06-28 11:22:13 +0200 David Svensson Fors <davidsf@axis.com>
814 * gst/rtsp-server/rtsp-client.c:
815 rtsp-client: Use fixed backlog size
816 Change to using a fixed backlog size WATCH_BACKLOG_SIZE.
817 Preparation for the next commit, which changes to a different way of
818 avoiding both deadlocks and unlimited memory usage with the watch
821 2018-07-16 21:57:08 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
823 * gst/rtsp-server/rtsp-media.c:
824 rtsp-media: unref clock (if set) when finalizing
825 https://bugzilla.gnome.org/show_bug.cgi?id=796814
827 2018-07-16 21:56:44 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
829 * docs/libs/gst-rtsp-server-sections.txt:
830 rtsp-media: add gst_rtsp_media_*_set_clock to docs
831 https://bugzilla.gnome.org/show_bug.cgi?id=796814
833 2018-07-12 19:01:54 +0100 Tim-Philipp Müller <tim@centricular.com>
835 * gst/rtsp-server/rtsp-media-factory.c:
836 media-factory: unref old clock when setting new clock
837 https://bugzilla.gnome.org/show_bug.cgi?id=796724
839 2018-06-29 15:20:57 -0700 Brendan Shanks <brendan.shanks@teradek.com>
841 * gst/rtsp-server/rtsp-media-factory.c:
842 media-factory: unref clock in finalize
843 https://bugzilla.gnome.org/show_bug.cgi?id=796724
845 2018-07-12 18:57:21 +0100 Tim-Philipp Müller <tim@centricular.com>
847 * gst/rtsp-server/rtsp-onvif-media.c:
848 rtsp-onvif-media: fix g-ir-scanner warnings
850 2018-07-10 23:56:23 +0100 Tim-Philipp Müller <tim@centricular.com>
853 .gitignore: add another example binary
855 2018-07-10 23:55:20 +0100 Tim-Philipp Müller <tim@centricular.com>
857 * examples/meson.build:
858 meson: add new test-appsrc2 example to meson build
860 2018-07-10 23:53:41 +0100 Tim-Philipp Müller <tim@centricular.com>
862 * examples/Makefile.am:
863 examples: fix build of new test-appsrc2 example
864 Need to link against libgstapp-1.0.
866 2018-07-11 01:25:51 +1000 Jan Schmidt <jan@centricular.com>
868 * examples/.gitignore:
869 * examples/Makefile.am:
870 * examples/test-appsrc2.c:
871 examples: Add test-appsrc2
872 Add an example of feeding both audio and video into an RTSP
875 2016-01-08 18:12:14 -0500 Louis-Francis Ratté-Boulianne <lfrb@collabora.com>
877 * gst/rtsp-server/rtsp-client.c:
878 client: Strip transport parts as whitespaces could be around commas
879 https://bugzilla.gnome.org/show_bug.cgi?id=758428
881 2018-06-27 08:30:42 +0200 Göran Jönsson <goranjn@axis.com>
883 * gst/rtsp-server/rtsp-stream.c:
884 rtsp-stream: avoid pushing data on unlinked udpsrc pad during setup
885 Fix race when setting up source elements.
886 Since we set the source element(s) to PLAYING state before hooking
887 them up to the downstream funnel, it's possible for the source element
888 to receive packets before we actually get to linking it to the funnel,
889 in which case buffers would be pushed out on an unlinked pad, causing
890 it to error out and stop receiving more data.
891 We fix this by blocking the source's srcpad until we have linked it.
892 https://bugzilla.gnome.org/show_bug.cgi?id=796160
894 2018-03-21 10:56:51 +0100 Ognyan Tonchev <ognyan@axis.com>
896 * gst/rtsp-server/rtsp-stream.c:
897 rtsp-stream: Fix mismatch between allowed and configured protocols
898 https://bugzilla.gnome.org/show_bug.cgi?id=796679
900 2017-02-01 09:44:50 +0100 Ulf Olsson <ulfo@axis.com>
902 * gst/rtsp-server/rtsp-stream.c:
903 rtsp-stream: Emit a signal when the SRTP decoder is created
904 https://bugzilla.gnome.org/show_bug.cgi?id=778080
906 2018-03-13 11:10:35 +0100 Patricia Muscalu <patricia@axis.com>
908 * gst/rtsp-server/rtsp-stream.c:
909 rtsp-stream: Don't require presence of sinks in _get_*_socket()
910 Transport specific sink elements are added to the pipeline
911 in PLAY request and sockets are already created in SETUP so
912 it's actually wrong to require the presence of sinks in
913 _get_*_socket() functions.
914 https://bugzilla.gnome.org/show_bug.cgi?id=793441
916 2018-02-14 10:41:02 +0100 Patricia Muscalu <patricia@axis.com>
918 * gst/rtsp-server/rtsp-stream.c:
919 rtsp-stream: Update transport for multicast clients as well
920 If a multicast client requests different transport settings
921 than the existing one make sure that this new transport
922 configuruation is propagated to the multicast udp sink.
923 https://bugzilla.gnome.org/show_bug.cgi?id=793441
925 2018-02-13 11:04:36 +0100 Patricia Muscalu <patricia@axis.com>
927 * gst/rtsp-server/rtsp-stream.c:
928 rtsp-stream: Set the multicast TTL parameter on multicast udp sinks
929 And not on unicast udp sinks
930 https://bugzilla.gnome.org/show_bug.cgi?id=793441
932 2018-06-24 12:44:26 +0200 Tim-Philipp Müller <tim@centricular.com>
934 * gst/rtsp-server/rtsp-address-pool.c:
935 * gst/rtsp-server/rtsp-auth.c:
936 * gst/rtsp-server/rtsp-client.c:
937 * gst/rtsp-server/rtsp-media-factory-uri.c:
938 * gst/rtsp-server/rtsp-media-factory.c:
939 * gst/rtsp-server/rtsp-media.c:
940 * gst/rtsp-server/rtsp-mount-points.c:
941 * gst/rtsp-server/rtsp-server.c:
942 * gst/rtsp-server/rtsp-session-media.c:
943 * gst/rtsp-server/rtsp-session-pool.c:
944 * gst/rtsp-server/rtsp-session.c:
945 * gst/rtsp-server/rtsp-stream-transport.c:
946 * gst/rtsp-server/rtsp-stream.c:
947 * gst/rtsp-server/rtsp-thread-pool.c:
948 Update for g_type_class_add_private() deprecation in recent GLib
950 2018-06-24 12:45:49 +0200 Tim-Philipp Müller <tim@centricular.com>
952 * gst/rtsp-server/rtsp-auth.c:
953 * gst/rtsp-server/rtsp-media.c:
954 * gst/rtsp-server/rtsp-sdp.c:
955 * gst/rtsp-server/rtsp-stream.c:
958 2018-06-22 23:17:08 +1000 Jan Schmidt <jan@centricular.com>
960 * examples/Makefile.am:
961 * examples/test-video-disconnect.c:
962 examples: Add test-video-disconnect example
963 Simple example which cuts off all clients 10 seconds
964 after the first one connects.
966 2018-06-20 04:37:11 +0200 Mathieu Duponchelle <mathieu@centricular.com>
968 * docs/libs/gst-rtsp-server-sections.txt:
969 * examples/test-auth-digest.c:
970 * gst/rtsp-server/rtsp-auth.c:
971 * gst/rtsp-server/rtsp-auth.h:
972 rtsp-auth: Add support for parsing .htdigest files
973 Passwords are usually not stored in clear text, but instead
974 stored already hashed in a .htdigest file.
975 Add support for parsing such files, add API to allow setting
976 a custom realm in RTSPAuth, and update the digest example.
977 https://bugzilla.gnome.org/show_bug.cgi?id=796637
979 2018-06-19 14:53:02 +1000 Matthew Waters <matthew@centricular.com>
981 * gst/rtsp-sink/gstrtspclientsink.c:
982 * gst/rtsp-sink/gstrtspclientsink.h:
983 rtspclientsink: fix waiting for multiple streams
984 We were previously only ever waiting for a single stream to notify it's
985 blocked status through GstRTSPStreamBlocking. Actually count streams to
987 Fixes rtspclientsink sending SDP's without out some of the input
989 https://bugzilla.gnome.org/show_bug.cgi?id=796624
991 2018-06-20 04:30:04 +0200 Mathieu Duponchelle <mathieu@centricular.com>
993 * docs/libs/gst-rtsp-server-sections.txt:
994 docs: add missing auth methods
996 2018-06-20 00:10:18 +0200 Mathieu Duponchelle <mathieu@centricular.com>
998 * gst/rtsp-server/rtsp-stream.c:
999 rtsp-stream: only create funnel if it didn't exist already.
1000 This precented using multiple protocols for the same stream.
1001 https://bugzilla.gnome.org/show_bug.cgi?id=796634
1003 2018-06-20 01:35:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1005 * examples/meson.build:
1006 meson: build auth-digest example
1008 2018-06-05 08:44:44 +0200 Patricia Muscalu <patricia@axis.com>
1010 * gst/rtsp-server/rtsp-client.c:
1011 * gst/rtsp-server/rtsp-media.c:
1012 * gst/rtsp-server/rtsp-sdp.c:
1013 * gst/rtsp-server/rtsp-session-media.c:
1014 * gst/rtsp-server/rtsp-stream-transport.c:
1015 Get payloader stats only for the sending streams
1016 Get/set payloader properties only for streams that actually
1017 contain a payloader element.
1018 https://bugzilla.gnome.org/show_bug.cgi?id=796523
1020 2018-05-18 14:53:49 +0200 Edward Hervey <edward@centricular.com>
1022 * gst/rtsp-server/Makefile.am:
1023 Makefile: Don't hardcode libtool for g-i build
1024 Similar to the other commits in core/base/bad
1026 2018-05-08 14:13:31 +0200 Johan Bjäreholt <johanbj@axis.com>
1028 * gst/rtsp-server/rtsp-onvif-media-factory.h:
1029 rtsp-onvif-media-factory: export gst_rtsp_onvif_media_factory_requires_backchannel
1030 https://bugzilla.gnome.org/show_bug.cgi?id=796229
1032 2018-05-09 04:09:02 +1000 Jan Schmidt <jan@centricular.com>
1034 * gst/rtsp-sink/gstrtspclientsink.c:
1035 rtspclientsink: Don't deadlock in preroll on early close
1036 If the connection is closed very early, the flushing
1037 marker might not get set and rtspclientsink can get
1038 deadlocked waiting for preroll forever.
1039 https://bugzilla.gnome.org/show_bug.cgi?id=786961
1041 2018-05-05 19:51:52 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
1044 * meson_options.txt:
1045 meson: Update option names to omit disable_ and with- prefixes
1046 Also yield common options to the outer project (gst-build in our case)
1047 so that they don't have to be set manually.
1049 2018-04-25 11:00:32 +0100 Tim-Philipp Müller <tim@centricular.com>
1052 meson: use -Wl,-Bsymbolic-functions where supported
1053 Just like the autotools build.
1055 2018-04-22 20:09:01 +0100 Tim-Philipp Müller <tim@centricular.com>
1058 * tests/check/Makefile.am:
1059 configure: check for -good and -bad plugins only in uninstalled setup
1060 Avoids confusing configure messages looking or a -good .pc file
1062 Also use plugindir variables that common macros set while at it.
1063 https://bugzilla.gnome.org/show_bug.cgi?id=795466
1065 2018-04-17 11:03:11 +0200 Joakim Johansson <joakimj@axis.com>
1067 * gst/rtsp-server/rtsp-client.c:
1068 rtsp-client: Fix session timeout
1069 When streaming data over TCP then is not the keep-alive
1070 functionality working.
1071 The reason is that the function do_send_data have changed
1072 to boolean but the code is still checking the received result
1073 from send_func with GST_RTSP_OK.
1074 The result is that a successful send_func will always lead to
1075 that do_send_data is returning false and the keep-alive will
1077 https://bugzilla.gnome.org/show_bug.cgi?id=795321
1079 2018-04-02 22:49:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1081 * docs/libs/gst-rtsp-server-sections.txt:
1082 * gst/rtsp-server/rtsp-media.c:
1083 * gst/rtsp-server/rtsp-sdp.c:
1084 * gst/rtsp-server/rtsp-stream.c:
1085 * gst/rtsp-server/rtsp-stream.h:
1086 * gst/rtsp-sink/gstrtspclientsink.c:
1087 * gst/rtsp-sink/gstrtspclientsink.h:
1088 Implement support for ULP Forward Error Correction
1089 In this initial commit, interface is only exposed for RECORD,
1090 further work will be needed in rtspsrc to support this for
1092 https://bugzilla.gnome.org/show_bug.cgi?id=794911
1094 2018-04-17 17:47:30 +0300 Sebastian Dröge <sebastian@centricular.com>
1096 * gst/rtsp-server/rtsp-onvif-media.c:
1097 Revert "rtsp-server: Switch around sendonly/recvonly attributes"
1098 This reverts commit 3d275b1345b76151418e3f56ed014d9089ac1a57.
1099 While RFC 3264 (SDP) says that sendonly/recvonly are from the point of view of
1100 the requester, the actual RTSP RFCs (RFC 2326 / 7826) disagree and say
1101 the opposite, just like the ONVIF standard.
1102 Let's follow those RFCs as we're doing RTSP here, and add a property at
1103 a later time if needed to switch to the SDP RFC behaviour.
1104 https://bugzilla.gnome.org/show_bug.cgi?id=793964
1106 2018-04-16 10:53:52 +0100 Tim-Philipp Müller <tim@centricular.com>
1109 Automatic update of common submodule
1110 From 3fa2c9e to ed78bee
1112 2018-04-04 10:06:06 +0300 Sebastian Dröge <sebastian@centricular.com>
1114 * gst/rtsp-server/rtsp-client.c:
1115 * gst/rtsp-server/rtsp-media-factory.c:
1116 * gst/rtsp-server/rtsp-media.c:
1117 * gst/rtsp-server/rtsp-stream.c:
1118 * tests/check/gst/rtspclientsink.c:
1119 gst: Run everything through gst-indent again
1121 2018-04-03 08:57:47 +0200 Branko Subasic <branko@axis.com>
1123 * gst/rtsp-server/rtsp-media.c:
1124 * tests/check/gst/media.c:
1125 rtsp-media: query the position on active streams if media is complete
1126 If the media is complete, i.e. one or more streams have been configured
1127 with sinks, then we want to query the position on those streams only.
1128 A query on an incomplete stream may return a position that originates from
1130 https://bugzilla.gnome.org/show_bug.cgi?id=794964
1132 2018-04-02 12:35:04 +0100 Tim-Philipp Müller <tim@centricular.com>
1134 * gst/rtsp-sink/gstrtspclientsink.c:
1135 rtspclientsink: make sure not to use freed string
1136 Set transport string to NULL after freeing it, so that
1137 at worst we get a NULL pointer if constructing a new
1138 transport string fails (which shouldn't really fail here).
1139 Also check return value of that, just in case.
1142 2018-03-30 23:34:01 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1144 * gst/rtsp-server/rtsp-client.c:
1145 rtsp-client: do not free string passed to take_header
1147 2018-03-30 23:10:10 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1149 * gst/rtsp-server/rtsp-stream.c:
1150 rtsp-stream: do not take lock in request_aux_receiver
1151 Added it right before pushing the previous commit, it is
1152 incorrect and deadlocks because this function gets called
1153 from the join_bin thread, which already holds the lock,
1154 that's the reason why request_aux_sender didn't take the
1157 2018-03-29 22:49:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1159 * docs/libs/gst-rtsp-server-sections.txt:
1160 * gst/rtsp-server/rtsp-media-factory.c:
1161 * gst/rtsp-server/rtsp-media-factory.h:
1162 * gst/rtsp-server/rtsp-media.c:
1163 * gst/rtsp-server/rtsp-media.h:
1164 * gst/rtsp-server/rtsp-stream.c:
1165 * gst/rtsp-server/rtsp-stream.h:
1166 rtsp-server: add API to enable retransmission requests
1167 "do-retransmission" was previously set when rtx-time != 0,
1168 which made no sense as do-retransmission is used to enable
1169 the sending of retransmission requests, where as rtx-time
1170 is used by the peer to enable storing of buffers in order
1171 to respond to retransmission requests.
1172 rtsp-media now also provides a callback for the
1173 request-aux-receiver signal.
1174 https://bugzilla.gnome.org/show_bug.cgi?id=794822
1176 2018-03-29 16:18:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1178 * gst/rtsp-sink/gstrtspclientsink.c:
1179 rtspclientsink: add rtx ssrc to mikey's crypto sessions
1180 https://bugzilla.gnome.org/show_bug.cgi?id=794813
1182 2018-03-29 16:15:45 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1184 * gst/rtsp-sink/gstrtspclientsink.c:
1185 rtspclientsink: Handle the KeyMgmt header in ANNOUNCE response
1186 This in order to be able to decrypt the RTCP backchannel
1187 https://bugzilla.gnome.org/show_bug.cgi?id=794813
1189 2018-03-29 16:12:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1191 * gst/rtsp-server/rtsp-client.c:
1192 rtsp-client: Send KeyMgmt header in ANNOUNCE response
1193 When sending back an encrypted RTCP back channel, it is useful
1194 for the client to know the encryption key.
1195 https://bugzilla.gnome.org/show_bug.cgi?id=794813
1197 2018-03-29 16:06:31 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1199 * gst/rtsp-server/rtsp-client.c:
1200 * gst/rtsp-server/rtsp-stream.c:
1201 * gst/rtsp-server/rtsp-stream.h:
1202 rtsp-stream: extract handle_keymgmt from rtsp-client
1203 rtspclientsink will also need to parse KeyMgmt headers
1204 sent by the server to decrypt the RTCP backchannel stream
1205 https://bugzilla.gnome.org/show_bug.cgi?id=794813
1207 2018-03-29 02:51:02 +0200 Mathieu Duponchelle <mathieu@centricular.com>
1209 * gst/rtsp-sink/gstrtspclientsink.c:
1210 * tests/check/gst/rtspclientsink.c:
1211 rtspclientsink: Fix client ports for the RTCP backchannel
1212 This was broken since the work for delayed transport creation
1213 was merged: the creation of the transports string depends on
1214 calling stream_get_server_port, which only starts returning
1215 something meaningful after a call to stream_allocate_udp_sockets
1216 has been made, this function expects a transport that we parse
1217 from the transport string ...
1218 Significant refactoring is in order, but does not look entirely
1219 trivial, for now we put a band aid on and create a second transport
1220 string after the stream has been completed, to pass it in
1221 the request headers instead of the previous, incomplete one.
1222 https://bugzilla.gnome.org/show_bug.cgi?id=794789
1224 2018-02-15 13:26:16 +0100 Göran Jönsson <goranjn@axis.com>
1226 * gst/rtsp-server/rtsp-client.c:
1227 rtsp-client:Error handling when equal http session cookie
1228 There are some clients that are sending same session cookie on random
1230 https://bugzilla.gnome.org/show_bug.cgi?id=753616
1232 2018-03-20 16:21:37 +0200 Sebastian Dröge <sebastian@centricular.com>
1234 * gst/rtsp-server/rtsp-media-factory-uri.c:
1235 rtsp-media-factory-uri: Fix compilation with latest GLib
1236 rtsp-media-factory-uri.c: In function ‘rtsp_media_factory_uri_create_element’:
1237 rtsp-media-factory-uri.c:621:17: error: assignment from incompatible pointer type [-Werror=incompatible-pointer-types]
1238 data->factory = g_object_ref (factory);
1241 2018-03-20 10:21:36 +0000 Tim-Philipp Müller <tim@centricular.com>
1249 === release 1.14.0 ===
1251 2018-03-19 20:27:04 +0000 Tim-Philipp Müller <tim@centricular.com>
1257 * gst-rtsp-server.doap:
1261 === release 1.13.91 ===
1263 2018-03-13 19:28:33 +0000 Tim-Philipp Müller <tim@centricular.com>
1269 * gst-rtsp-server.doap:
1273 2018-03-13 13:30:41 +0000 Tim-Philipp Müller <tim@centricular.com>
1275 * gst/rtsp-server/Makefile.am:
1276 * gst/rtsp-server/meson.build:
1277 * gst/rtsp-server/rtsp-address-pool.h:
1278 * gst/rtsp-server/rtsp-auth.h:
1279 * gst/rtsp-server/rtsp-client.h:
1280 * gst/rtsp-server/rtsp-context.h:
1281 * gst/rtsp-server/rtsp-media-factory-uri.h:
1282 * gst/rtsp-server/rtsp-media-factory.h:
1283 * gst/rtsp-server/rtsp-media.h:
1284 * gst/rtsp-server/rtsp-mount-points.h:
1285 * gst/rtsp-server/rtsp-onvif-client.h:
1286 * gst/rtsp-server/rtsp-onvif-media-factory.h:
1287 * gst/rtsp-server/rtsp-onvif-media.h:
1288 * gst/rtsp-server/rtsp-onvif-server.h:
1289 * gst/rtsp-server/rtsp-params.h:
1290 * gst/rtsp-server/rtsp-permissions.h:
1291 * gst/rtsp-server/rtsp-sdp.h:
1292 * gst/rtsp-server/rtsp-server-prelude.h:
1293 * gst/rtsp-server/rtsp-server.h:
1294 * gst/rtsp-server/rtsp-session-media.h:
1295 * gst/rtsp-server/rtsp-session-pool.h:
1296 * gst/rtsp-server/rtsp-session.h:
1297 * gst/rtsp-server/rtsp-stream-transport.h:
1298 * gst/rtsp-server/rtsp-stream.h:
1299 * gst/rtsp-server/rtsp-thread-pool.h:
1300 * gst/rtsp-server/rtsp-token.h:
1301 rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API
1302 We need different export decorators for the different libs.
1303 For now no actual change though, just rename before the release,
1304 and add prelude headers to define the new decorator to GST_EXPORT.
1306 2018-03-07 12:20:05 +0200 Sebastian Dröge <sebastian@centricular.com>
1308 * gst/rtsp-server/rtsp-onvif-media-factory.c:
1309 rtsp-onvif-media-factory: Document that backchannel pipelines must end with async=false sinks
1310 https://bugzilla.gnome.org/show_bug.cgi?id=794143
1312 === release 1.13.90 ===
1314 2018-03-03 22:49:34 +0000 Tim-Philipp Müller <tim@centricular.com>
1320 * gst-rtsp-server.doap:
1324 2018-03-02 16:24:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1326 * gst/rtsp-server/rtsp-media-factory.c:
1327 * gst/rtsp-server/rtsp-permissions.c:
1328 permissions: add Since tags and example for new API
1330 2018-03-02 01:36:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1332 * docs/libs/gst-rtsp-server-sections.txt:
1333 * gst/rtsp-server/rtsp-media-factory.c:
1334 * gst/rtsp-server/rtsp-media-factory.h:
1335 * gst/rtsp-server/rtsp-permissions.c:
1336 * gst/rtsp-server/rtsp-permissions.h:
1337 * tests/check/gst/permissions.c:
1338 permissions: more bindings-friendly API
1339 https://bugzilla.gnome.org/show_bug.cgi?id=793975
1341 2018-03-01 19:28:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1344 meson: enable more warnings
1346 2018-02-28 21:12:43 +0200 Sebastian Dröge <sebastian@centricular.com>
1348 * gst/rtsp-server/rtsp-client.c:
1349 rtsp-client: Place netaddress meta on packets received via TCP
1350 This allows us to later map signals from rtpbin/rtpsource back to the
1351 corresponding stream transport, and allows to do keep-alive based on
1352 RTCP packets in case of TCP media transport.
1353 https://bugzilla.gnome.org/show_bug.cgi?id=789646
1355 2018-02-27 20:34:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1357 * gst/rtsp-sink/gstrtspclientsink.c:
1358 rtspclientsink: if OPEN failed, unqueue next command
1359 As READY_TO_PAUSED can no longer return async, the RECORD
1360 command will be queued before the OPEN command fails
1361 (for example in case the server could not be connected),
1362 and record then waits for ever.
1363 https://bugzilla.gnome.org/show_bug.cgi?id=793896
1365 2018-02-26 22:59:17 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1367 * gst/rtsp-sink/gstrtspclientsink.c:
1368 rtspclientsink: fix retrieval of custom payloader caps
1369 If a bin is passed as the custom payloader, the caps of
1370 its factory will be empty, the correct way to obtain the caps
1371 is to query its sinkpad.
1373 2018-02-26 22:59:00 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1375 * gst/rtsp-sink/gstrtspclientsink.c:
1376 rtspclientsink: fix extra unref of custom payloader
1378 2018-02-26 22:57:39 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1380 * gst/rtsp-sink/gstrtspclientsink.c:
1381 rspclientsink: fix recent code indentation
1383 2018-02-26 20:27:57 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1385 * gst/rtsp-sink/gstrtspclientsink.c:
1386 rtspclientsink: add missing get_type prototype
1388 2018-02-24 03:52:15 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1390 * gst/rtsp-sink/gstrtspclientsink.c:
1391 rtspclientsink: allow setting payloader as pad property
1392 This was a FIXME item, and can be quite useful, also
1393 allowing to specify payloader properties from the command
1394 line, which is always nice.
1395 https://bugzilla.gnome.org/show_bug.cgi?id=793776
1397 2018-02-26 14:16:54 +0100 Carlos Rafael Giani <dv@pseudoterminal.org>
1399 * gst/rtsp-server/rtsp-media.c:
1400 rtsp-media: Replace g_print() log line
1401 https://bugzilla.gnome.org/show_bug.cgi?id=793838
1403 2018-02-22 20:17:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1405 * gst/rtsp-server/rtsp-media.c:
1406 * tests/check/gst/rtspclientsink.c:
1407 rtsp-media: fix RECORD getting stuck
1408 The test_record case was working because async=false had
1409 been added in https://bugzilla.gnome.org/show_bug.cgi?id=757488
1410 but that was incorrect, as it should not be needed.
1411 Removing async=false made the test fail as expected, this is
1412 fixed by not trying to preroll when preparing the media for
1413 RECORD, as start_prepare is called upon receiving ANNOUNCE,
1414 and our peer will not start sending media until it has received
1415 a response to that request, and sent and received a response
1416 to RECORD as well, thus obviously preventing preroll.
1417 https://bugzilla.gnome.org/show_bug.cgi?id=793738
1419 2018-02-23 03:26:21 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1421 * gst/rtsp-server/rtsp-auth.c:
1422 rtsp-auth: fix set_tls_authentication_mode annotation
1424 2018-02-19 11:57:29 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
1426 * gst/rtsp-server/rtsp-onvif-media.c:
1427 rtp-server: remove redefined variable
1428 res is a boolean variable which is defined in the function scope and
1429 redefined, with no reason, in the loop scope. This patch removes the
1431 https://bugzilla.gnome.org/show_bug.cgi?id=793592
1433 2018-02-05 11:49:07 +0100 Ognyan Tonchev <ognyan@axis.com>
1435 * gst/rtsp-server/rtsp-media.c:
1436 * gst/rtsp-server/rtsp-stream.c:
1437 * gst/rtsp-server/rtsp-stream.h:
1438 stream: Add functions for checking if stream is receiver or sender
1439 ...and replace all checks for RECORD in GstRTSPMedia which are really
1440 for "sender-only". This way the code becomes more generic and introducing
1441 support for onvif-backchannel later on will require no changes in
1444 2017-10-21 14:06:30 +0200 Ognyan Tonchev <ognyan@axis.com>
1446 * gst/rtsp-server/rtsp-onvif-media-factory.c:
1447 * gst/rtsp-server/rtsp-onvif-media-factory.h:
1448 onvif: Make requires_backchannel() public
1449 ...in order to let subclasses building the onvif part of the pipeline
1450 check whether backchannel shall be included or not.
1452 2018-01-22 12:46:34 +0200 Sebastian Dröge <sebastian@centricular.com>
1454 * gst/rtsp-server/rtsp-onvif-media.c:
1455 rtsp-server: Switch around sendonly/recvonly attributes
1456 They are wrong in the ONVIF streaming spec. The backchannel should be
1457 recvonly and the normal media should be sendonly: direction is always
1458 from the point of view of the SDP offerer (the server) according to
1461 2017-09-25 19:41:05 +0300 Sebastian Dröge <sebastian@centricular.com>
1463 * docs/libs/gst-rtsp-server-docs.sgml:
1464 * docs/libs/gst-rtsp-server-sections.txt:
1465 * examples/.gitignore:
1466 * examples/Makefile.am:
1467 * examples/test-onvif-backchannel.c:
1468 * gst/rtsp-server/Makefile.am:
1469 * gst/rtsp-server/rtsp-media.h:
1470 * gst/rtsp-server/rtsp-onvif-client.c:
1471 * gst/rtsp-server/rtsp-onvif-client.h:
1472 * gst/rtsp-server/rtsp-onvif-media-factory.c:
1473 * gst/rtsp-server/rtsp-onvif-media-factory.h:
1474 * gst/rtsp-server/rtsp-onvif-media.c:
1475 * gst/rtsp-server/rtsp-onvif-media.h:
1476 * gst/rtsp-server/rtsp-onvif-server.c:
1477 * gst/rtsp-server/rtsp-onvif-server.h:
1478 * gst/rtsp-server/rtsp-sdp.c:
1479 * gst/rtsp-server/rtsp-sdp.h:
1480 rtsp: Add support for ONVIF backchannel
1481 This adds a new RTSP server, client, media-factory and media subclass
1482 for handling the specifics of the backchannel. Ideally this later can be
1483 extended with other ONVIF specific features.
1485 2017-10-12 21:00:16 +0300 Sebastian Dröge <sebastian@centricular.com>
1487 * gst/rtsp-server/rtsp-media.c:
1488 rtsp-media: Add support for sending+receiving medias
1489 We need to add an appsrc/appsink in that case because otherwise the
1490 media bin will be a sink and a source for rtpbin, causing a pipeline
1492 https://bugzilla.gnome.org/show_bug.cgi?id=788950
1494 2018-02-15 19:44:28 +0000 Tim-Philipp Müller <tim@centricular.com>
1500 === release 1.13.1 ===
1502 2018-02-15 17:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
1506 * gst-rtsp-server.doap:
1510 2018-02-14 17:11:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1512 * gst/rtsp-server/rtsp-session-pool.c:
1513 session-pool: remove nullable return annotation
1514 create_watch can only return NULL from the API guards, no
1517 2018-02-13 18:59:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1519 * gst/rtsp-server/rtsp-media-factory.c:
1520 * gst/rtsp-server/rtsp-media.c:
1521 set_clock functions: Add nullable annotations
1523 2018-02-10 00:07:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1525 * gst/rtsp-server/rtsp-auth.c:
1526 * gst/rtsp-server/rtsp-client.c:
1527 * gst/rtsp-server/rtsp-media-factory.c:
1528 * gst/rtsp-server/rtsp-media.c:
1529 * gst/rtsp-server/rtsp-mount-points.c:
1530 * gst/rtsp-server/rtsp-server.c:
1531 * gst/rtsp-server/rtsp-session-media.c:
1532 * gst/rtsp-server/rtsp-session-pool.c:
1533 * gst/rtsp-server/rtsp-session.c:
1534 * gst/rtsp-server/rtsp-stream-transport.c:
1535 * gst/rtsp-server/rtsp-stream.c:
1536 * gst/rtsp-server/rtsp-thread-pool.c:
1537 All around: add annotations and API guards
1539 2018-02-12 19:12:35 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1541 * tests/test-cleanup.c:
1542 test-cleanup: bind any port
1543 The meson test suite runs tests in parallel, trying to bind
1544 a single port made the test fail.
1546 2018-02-08 19:15:10 +0000 Tim-Philipp Müller <tim@centricular.com>
1549 meson: make version numbers ints and fix int/string comparison
1550 WARNING: Trying to compare values of different types (str, int).
1551 The result of this is undefined and will become a hard error
1552 in a future Meson release.
1554 2018-02-06 18:00:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1556 * gst/rtsp-server/rtsp-context.c:
1557 gst_rtsp_context_get_current: add (skip) annotation
1558 The return value type is defined with G_DEFINE_POINTER_TYPE,
1559 and gi emits the following warning:
1560 Invalid non-constant return of bare structure or union; register as
1561 boxed type or (skip)
1563 2018-02-06 17:58:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
1565 * gst/rtsp-server/rtsp-client.c:
1566 rtsp-client: add type annotations
1567 gi doesn't seem to be able to figure out the type of the
1568 signal parameters when defined with G_DEFINE_POINTER_TYPE
1570 2018-02-04 12:24:09 +0100 Tim-Philipp Müller <tim@centricular.com>
1573 autotools: use -fno-strict-aliasing where supported
1574 https://bugzilla.gnome.org/show_bug.cgi?id=769183
1576 2018-01-30 20:35:21 +0000 Tim-Philipp Müller <tim@centricular.com>
1579 meson: use -fno-strict-aliasing where supported
1580 https://bugzilla.gnome.org/show_bug.cgi?id=769183
1582 2018-01-25 12:09:03 +0000 Tim-Philipp Müller <tim@centricular.com>
1584 * gst/rtsp-server/rtsp-mount-points.c:
1585 mount-points: bail out of loop again when matching mount points
1586 Previous patch led to us iterating the entire sequence. Bail out
1587 of the loop again if we have a match but are moving away from it.
1588 https://bugzilla.gnome.org/show_bug.cgi?id=771555
1590 2018-01-25 12:06:57 +0000 Tim-Philipp Müller <tim@centricular.com>
1592 * tests/check/gst/mountpoints.c:
1593 tests: mountpoints: add more checks for mount point path matching
1594 https://bugzilla.gnome.org/show_bug.cgi?id=771555
1596 2016-09-16 20:41:19 +0000 Andrew Bott <andrew.bott@blackmoth.com>
1598 * gst/rtsp-server/rtsp-mount-points.c:
1599 mount-points: fix matching of paths where there's also an entry with a common prefix
1600 e.g. with the following mount points
1604 _match() would not match /raw/video and /raw/snapshot correctly.
1605 https://bugzilla.gnome.org/show_bug.cgi?id=771555
1607 2018-01-18 23:53:20 +0000 Tim-Philipp Müller <tim@centricular.com>
1609 * docs/libs/gst-rtsp-server-sections.txt:
1610 * gst/rtsp-server/rtsp-permissions.c:
1611 * gst/rtsp-server/rtsp-permissions.h:
1612 * tests/check/gst/permissions.c:
1613 permissions: add some new API to make this usable from bindings
1614 https://bugzilla.gnome.org/show_bug.cgi?id=787073
1616 2018-01-18 11:32:32 +0000 Tim-Philipp Müller <tim@centricular.com>
1618 * gst/rtsp-server/rtsp-token.c:
1619 rtsp-token: annotate constructors for bindings
1620 This maps _new_empty() to _new(), which also makes RTSPToken()
1621 work properly now. Since this API wasn't usable from bindings
1622 before, this should hopefully be fine.
1623 https://bugzilla.gnome.org/show_bug.cgi?id=787073
1625 2018-01-18 11:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
1627 * docs/libs/gst-rtsp-server-sections.txt:
1628 * gst/rtsp-server/rtsp-token.c:
1629 * gst/rtsp-server/rtsp-token.h:
1630 * tests/check/gst/token.c:
1631 rtsp-token: add some API to set fields from bindings
1632 The existing functions are all vararg-based and as such
1633 not usable from bindings.
1634 https://bugzilla.gnome.org/show_bug.cgi?id=787073
1636 2018-01-13 15:02:28 +0000 Tim-Philipp Müller <tim@centricular.com>
1638 * tests/check/gst/rtspclientsink.c:
1639 * tests/check/gst/rtspserver.c:
1640 * tests/check/gst/sessionpool.c:
1641 * tests/check/gst/stream.c:
1642 tests: fix indentation
1645 2018-01-13 14:58:55 +0000 Tim-Philipp Müller <tim@centricular.com>
1647 * tests/check/gst/rtspserver.c:
1648 tests: rtspserver: fix another ref leak
1649 Even if this didn't show up in valgrind.
1651 2018-01-13 14:58:00 +0000 Tim-Philipp Müller <tim@centricular.com>
1653 * tests/check/gst/rtspclientsink.c:
1654 tests: rtspclientsink: fix leak
1656 2018-01-02 14:19:31 +0100 Branko Subasic <branko@axis.com>
1658 * tests/check/gst/rtspserver.c:
1659 test: rtspserver: plug memory leak in test_no_session_timeout
1660 In test_no_session_timeout, unref the rtsp session object when the
1662 https://bugzilla.gnome.org/show_bug.cgi?id=792127
1664 2017-12-20 14:17:02 +0100 Edward Hervey <edward@centricular.com>
1666 * gst/rtsp-sink/gstrtspclientsink.c:
1667 rtpsclientsink: Initialize and clear newly added mutex and cond
1668 While it *did* work, glib would automatically create new mutex and cond
1669 ... which never got freed
1671 2017-12-19 11:34:37 +0200 Sebastian Dröge <sebastian@centricular.com>
1673 * gst/rtsp-server/rtsp-stream.c:
1674 rtsp-stream: Set multicast TTL on the multicast sockets
1675 And not if we do unicast UDP.
1676 https://bugzilla.gnome.org/show_bug.cgi?id=791743
1678 2017-12-19 11:14:48 +0200 Sebastian Dröge <sebastian@centricular.com>
1680 * gst/rtsp-server/rtsp-stream.c:
1681 rtsp-stream: Decide based on the sockets, not the addresses if we already allocated a socket
1682 In the multicast case (as in test-multicast, not test-multicast2), the
1683 address could be allocated/reserved (and thus set) already without
1684 allocating the actual socket. We need to allocate the socket here still
1685 instead of just claiming that it was already allocated.
1686 See https://bugzilla.gnome.org/show_bug.cgi?id=791743#c2
1688 2017-12-16 21:46:53 +0100 Patricia Muscalu <patricia@dovakhiin.com>
1690 * gst/rtsp-sink/gstrtspclientsink.c:
1691 * gst/rtsp-sink/gstrtspclientsink.h:
1692 rtspclientsink: Use the new rtsp-stream API
1693 https://bugzilla.gnome.org/show_bug.cgi?id=790412
1695 2017-12-16 21:01:43 +0100 Patricia Muscalu <patricia@dovakhiin.com>
1697 * gst/rtsp-sink/gstrtspclientsink.c:
1698 * gst/rtsp-sink/gstrtspclientsink.h:
1699 rtspclientsink: Wait until OPEN has been scheduled
1700 Make sure that the sink thread has started opening connection
1701 to the server before continuing.
1702 https://bugzilla.gnome.org/show_bug.cgi?id=790412
1704 2017-12-14 14:53:35 +1100 Matthew Waters <matthew@centricular.com>
1707 Automatic update of common submodule
1708 From e8c7a71 to 3fa2c9e
1710 2017-12-07 16:08:29 +0100 Edward Hervey <edward@centricular.com>
1712 * gst/rtsp-server/rtsp-media.c:
1713 * gst/rtsp-server/rtsp-session-media.c:
1714 * gst/rtsp-server/rtsp-stream.c:
1715 rtsp-server: Minor doc fixes
1718 2017-12-06 20:47:22 +0000 Tim-Philipp Müller <tim@centricular.com>
1721 * tests/Makefile.am:
1722 tests: disable all tests when --disable-tests is used
1723 Move conditional subdir include into top level.
1724 Based on patch by: Joel Holdsworth
1725 https://bugzilla.gnome.org/show_bug.cgi?id=757703
1727 2017-12-06 20:42:39 +0000 Tim-Philipp Müller <tim@centricular.com>
1730 * meson_options.txt:
1731 * tests/meson.build:
1732 meson: build more tests and add options to disable tests and examples
1734 2017-11-26 13:26:39 -0300 Thibault Saunier <tsaunier@gnome.org>
1736 * gst/rtsp-server/rtsp-session.c:
1737 Fix build when -Werror=deprecated-declarations is on
1738 As gst_rtsp_session_next_timeout is deprecated.
1740 ../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]
1741 res = (gst_rtsp_session_next_timeout (session, now) == 0);
1743 ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-session.c:685:1: note: declared here
1744 gst_rtsp_session_next_timeout (GstRTSPSession * session, GTimeVal * now)
1745 ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~
1748 2017-11-27 20:18:24 +1100 Matthew Waters <matthew@centricular.com>
1751 Automatic update of common submodule
1752 From 3f4aa96 to e8c7a71
1754 2017-11-25 20:34:16 +0100 Patricia Muscalu <patricia@dovakhiin.com>
1756 * tests/check/gst/media.c:
1757 check/media: Add seekability test case: not all streams are active
1758 Media contains two streams but only one is complete and prepared
1760 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1762 2017-11-25 20:32:02 +0100 Patricia Muscalu <patricia@dovakhiin.com>
1764 * gst/rtsp-server/rtsp-stream.c:
1765 rtsp-stream: Do not reset 'blocking' if stream is already blocked
1766 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1768 2017-11-25 20:45:44 +0100 Patricia Muscalu <patricia@dovakhiin.com>
1770 * gst/rtsp-server/rtsp-media.c:
1771 rtsp-media: Fix missing lock in gst_rtsp_media_seekable()
1772 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1774 2017-11-26 16:29:49 +0000 Tim-Philipp Müller <tim@centricular.com>
1777 meson: remove vs_module_defs_dir variable which is no longer needed
1779 2017-11-26 14:46:05 +0000 Tim-Philipp Müller <tim@centricular.com>
1781 * gst/rtsp-server/rtsp-session.h:
1784 2017-11-26 12:53:42 +0000 Tim-Philipp Müller <tim@centricular.com>
1787 * gst/rtsp-server/meson.build:
1789 * win32/common/libgstrtspserver.def:
1790 win32: remove .def file with exports
1791 They're no longer needed, symbol exporting is now explicit
1792 via GST_EXPORT in all cases (autotools, meson, incl. MSVC).
1794 2017-11-26 12:28:40 +0000 Tim-Philipp Müller <tim@centricular.com>
1797 autotools: stop controlling symbol visibility with -export-symbols-regex
1798 Instead, use -fvisibility=hidden and explicit exports via GST_EXPORT.
1799 This should result in consistent behaviour for the autotools and
1802 2017-11-26 12:47:08 +0000 Tim-Philipp Müller <tim@centricular.com>
1804 * gst/rtsp-server/rtsp-media.h:
1805 * gst/rtsp-server/rtsp-server.h:
1806 * gst/rtsp-server/rtsp-session.c:
1807 * gst/rtsp-server/rtsp-session.h:
1808 rtsp-server: add missing GST_EXPORT and export deprecated funcs
1810 2017-11-25 07:53:30 +0100 Edward Hervey <edward@centricular.com>
1812 * tests/check/gst/media.c:
1813 check: Add seekability testing on medias
1814 Make sure that once GstRTSPMedia are prepared they returned
1815 the expected seekability results
1816 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1818 2017-11-24 17:34:31 +0100 Edward Hervey <edward@centricular.com>
1820 * docs/libs/gst-rtsp-server-sections.txt:
1821 * gst/rtsp-server/rtsp-media.c:
1822 * gst/rtsp-server/rtsp-stream.c:
1823 * gst/rtsp-server/rtsp-stream.h:
1824 * win32/common/libgstrtspserver.def:
1825 rtsp-media: Enable seeking query before pipeline is complete
1826 SDP are now provided *before* the pipeline is fully complete. In order
1827 to know whether a media is seekable or not therefore requires asking
1828 the invididual streams.
1829 API: gst_rtsp_stream_seekable
1830 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1832 2017-11-23 20:34:03 +0100 Patricia Muscalu <patricia@axis.com>
1834 * gst/rtsp-server/rtsp-media.c:
1835 rtsp-media: Fix handling in default_unsuspend()
1836 Handle the case when streams are not blocked and media
1837 is suspended from PAUSED.
1838 Change-Id: I2f3d222ea7b9b20a0732ea5dc81a32d17ab75040
1839 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1841 2017-11-23 18:51:21 +0100 Patricia Muscalu <patricia@axis.com>
1843 * tests/check/gst/media.c:
1844 check/media: Fix thread pool leak.
1845 Change-Id: I0f92b1caca0ee518ae64a7dacfbd28a214c3eea1
1846 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1848 2017-11-23 18:39:44 +0100 Patricia Muscalu <patricia@axis.com>
1850 * gst/rtsp-server/rtsp-media.c:
1851 rtsp-media: Removed fakesink elements
1852 There is not need of adding fakesink elements to the media
1853 pipeline in the dynamic-payloader case.
1854 The media pipeline itself is dynamically updated with
1855 the receiver and sender parts that are based on the client
1856 transport information known after SETUP has been received.
1857 Change-Id: I4e88c9b500c04030669822f0d03b1842913f6cb9
1858 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1860 2017-11-23 09:10:54 +0100 Patricia Muscalu <patricia@axis.com>
1862 * gst/rtsp-server/rtsp-media.c:
1863 rtsp-media: Corrected ASYNC_DONE handling
1864 Media is complete when all the transport based parts are
1865 added to the media pipeline. At this point ASYNC_DONE is
1866 posted by the media pipeline and media is ready to enter
1868 Change-Id: I50fb8dfed88ebaf057d9a35fca2d7f0a70e9d1fa
1869 https://bugzilla.gnome.org/show_bug.cgi?id=790674
1871 2017-11-22 12:24:38 +0100 Edward Hervey <bilboed@bilboed.com>
1873 * tests/check/gst/media.c:
1874 check/media: Check that prepared media can provide a SDP
1875 Whenever a RTSPMedia is prepared, it should be able to provide a SDP
1877 2017-11-21 09:53:19 +0100 Edward Hervey <edward@centricular.com>
1879 * gst/rtsp-server/rtsp-client.c:
1880 rtsp-client: Don't leak addr
1883 2017-11-21 09:53:08 +0100 Edward Hervey <bilboed@bilboed.com>
1885 * gst/rtsp-server/rtsp-client.c:
1886 * gst/rtsp-server/rtsp-session-media.c:
1887 * gst/rtsp-server/rtsp-stream.c:
1890 2017-11-20 18:30:19 +0100 Edward Hervey <bilboed@bilboed.com>
1892 * gst/rtsp-server/rtsp-media.c:
1893 rtsp-media: Don't unblock with remaining dynamic payloaders
1894 If we still have some dynamic paylaoders which haven't posted
1895 no-more-pads yet, don't go to PREPARED if one of the streams
1897 The risk was that we would end up not exposing/using all specified
1899 The downside is that if you have _multiple_ _live_ _dynamic_ payloaders
1900 then it will take a bit more time to start. But only if those 3
1901 conditions are present.
1902 https://bugzilla.gnome.org/show_bug.cgi?id=769521
1904 2017-11-20 16:49:29 +0100 Edward Hervey <edward@centricular.com>
1906 * gst/rtsp-server/rtsp-media.c:
1909 2017-11-20 16:48:55 +0100 Edward Hervey <edward@centricular.com>
1911 * gst/rtsp-server/rtsp-media.c:
1912 rtsp-media: Don't set float on a gint64 variable
1913 Just use 0. Fixes 'undefined' behaviour from clang
1915 2017-11-20 18:29:02 +0100 Edward Hervey <edward@centricular.com>
1917 * gst/rtsp-server/rtsp-media.c:
1918 rtsp-media: Fix previous commit
1919 We only want to count dynamic payloaders
1921 2017-11-20 09:32:07 +0100 Edward Hervey <edward@centricular.com>
1923 * gst/rtsp-server/rtsp-media.c:
1924 * tests/check/gst/media.c:
1925 rtsp-media: Handle multiple dynamic elements
1926 If we have more than one dynamic payloader in the pipeline, we need
1927 to wait until the *last* one emits 'no-more-pads' before switching
1929 Failure to do so would result in a race where some of the streams
1930 wouldn't properly be prepared
1931 https://bugzilla.gnome.org/show_bug.cgi?id=769521
1933 2017-11-16 12:18:20 +0200 Sebastian Dröge <sebastian@centricular.com>
1935 * win32/common/libgstrtspserver.def:
1936 win32: Fix exported symbols list
1938 2017-11-15 19:52:29 +0200 Sebastian Dröge <sebastian@centricular.com>
1940 * gst/rtsp-server/rtsp-stream.c:
1941 rtsp-stream: Only update the RTP udpsink if it actually exists
1942 For send-only streams it does not exist, but the RTCP udpsink might.
1944 2017-11-15 18:15:53 +0200 Sebastian Dröge <sebastian@centricular.com>
1946 * win32/common/libgstrtspserver.def:
1947 win32: Update exports
1949 2017-10-23 09:49:09 +0200 Patricia Muscalu <patricia@axis.com>
1951 * gst/rtsp-server/rtsp-media.c:
1952 * gst/rtsp-server/rtsp-stream.c:
1953 * gst/rtsp-server/rtsp-stream.h:
1954 rtsp-media: seek on media pipelines that are complete
1955 Make sure that a seek is performed on pipelines that
1956 contain at least one sink element.
1957 Change-Id: Icf398e10add3191d104b1289de612412da326819
1958 https://bugzilla.gnome.org/show_bug.cgi?id=788340
1960 2017-10-17 10:44:33 +0200 Patricia Muscalu <patricia@axis.com>
1962 * gst/rtsp-server/rtsp-client.c:
1963 * gst/rtsp-server/rtsp-media.c:
1964 * gst/rtsp-server/rtsp-media.h:
1965 * gst/rtsp-server/rtsp-stream.c:
1966 * gst/rtsp-server/rtsp-stream.h:
1967 * tests/check/gst/client.c:
1968 * tests/check/gst/media.c:
1969 * tests/check/gst/rtspserver.c:
1970 * tests/check/gst/stream.c:
1971 Dynamically reconfigure pipeline in PLAY based on transports
1972 The initial pipeline does not contain specific transport
1973 elements. The receiver and the sender parts are added
1975 If the media is shared, the streams are dynamically
1976 reconfigured after each PLAY.
1977 https://bugzilla.gnome.org/show_bug.cgi?id=788340
1979 2017-10-16 12:40:57 +0200 Patricia Muscalu <patricia@axis.com>
1981 * gst/rtsp-server/rtsp-stream.c:
1982 rtsp-stream: obtain stream position from pad
1983 If no sinks have been added yet, obtain the current and
1984 the stop position of the stream from the send_src pad.
1985 Change-Id: Iacd4ab4bdc69f6b49370d06012880ce48a7d595a
1986 https://bugzilla.gnome.org/show_bug.cgi?id=788340
1988 2017-10-16 11:35:10 +0200 Patricia Muscalu <patricia@axis.com>
1990 * gst/rtsp-server/rtsp-session-media.c:
1991 * gst/rtsp-server/rtsp-session-media.h:
1992 rtsp-session-media: add function to get a list of transports
1993 Change-Id: I817e10624da0f3200f24d1b232cff481099278e3
1994 https://bugzilla.gnome.org/show_bug.cgi?id=788340
1996 2017-10-16 11:15:55 +0200 Patricia Muscalu <patricia@axis.com>
1998 * gst/rtsp-server/rtsp-stream.c:
1999 * gst/rtsp-server/rtsp-stream.h:
2000 rtsp-stream: add functions to get rtp and rtcp multicast sockets
2001 Change-Id: Iddfe6e0bd250cb0159096d5eba9e4202d22b56db
2002 https://bugzilla.gnome.org/show_bug.cgi?id=788340
2004 2017-10-20 12:21:48 +0200 Patricia Muscalu <patricia@axis.com>
2006 * gst/rtsp-server/rtsp-stream.c:
2007 stream: set async=sync=false only for RTCP appsink
2008 Change-Id: I929a218a9adf4759f61322b6f2063aacc5595f90
2009 https://bugzilla.gnome.org/show_bug.cgi?id=788340
2011 2017-10-16 10:10:17 +0200 Patricia Muscalu <patricia@axis.com>
2013 * gst/rtsp-server/rtsp-media.c:
2014 rtsp-media: return minimum value in query position case
2015 The minimum position should be returned as we are interested
2016 in the whole interval.
2017 Change-Id: I30e297fc040c995ae40c25dee8ff56321612fe2b
2018 https://bugzilla.gnome.org/show_bug.cgi?id=788340
2020 2017-08-09 11:52:38 +0200 Jonathan Karlsson <jonakn@axis.com>
2022 * gst/rtsp-server/rtsp-session.c:
2023 * tests/check/gst/rtspserver.c:
2024 rtsp-session: Handle the case when timeout=0
2025 According to the documentation, a timeout of value 0 means
2026 that the session never timeouts. This adds handling of that.
2027 If timeout=0 we just return with a -1 from
2028 gst_rtsp_session_next_timeout_usec ().
2029 https://bugzilla.gnome.org/show_bug.cgi?id=785058
2031 2017-07-17 17:15:22 +0300 Sebastian Dröge <sebastian@centricular.com>
2033 * gst/rtsp-sink/gstrtspclientsink.c:
2034 rtspclientsink: Add "accept-certificate" signal for manually checking a TLS certificate for validity
2035 https://bugzilla.gnome.org/show_bug.cgi?id=785024
2037 2017-10-26 14:43:19 +0200 Mathieu Duponchelle <mathieu@centricular.com>
2039 * docs/libs/gst-rtsp-server-sections.txt:
2040 * gst/rtsp-server/rtsp-media-factory.c:
2041 docs: add media factory transport mode accessors
2042 and fix the documentation for the return value of the getter
2044 2017-10-09 12:43:01 +0200 Branko Subasic <branko@axis.com>
2046 * gst/rtsp-server/rtsp-client.c:
2047 rtsp-client: unref 'pipelined_requests' in finalize
2048 The hash table priv->pipelined_requests is not unref:ed in the
2049 finalize funktion. Make sure it is.
2050 https://bugzilla.gnome.org/show_bug.cgi?id=788704
2052 2017-10-09 14:44:40 +0200 Thibault Saunier <thibault.saunier@osg.samsung.com>
2054 * gst/rtsp-server/rtsp-media.c:
2055 rtsp-media: Initialize scalar variable
2058 2017-10-06 10:27:34 +0200 Edward Hervey <edward@centricular.com>
2060 * win32/common/libgstrtspserver.def:
2061 win32: Update export file
2063 2017-04-22 09:26:07 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
2065 * gst/rtsp-server/rtsp-client.c:
2066 * gst/rtsp-server/rtsp-media.c:
2067 * gst/rtsp-server/rtsp-media.h:
2068 Start support for RTSP 2.0
2069 This adds basic support for new 2.0 features, though the protocol is
2070 subposdely backward incompatible, most semantics are the sames.
2073 * version negotiation
2074 * pipelined requests support
2075 * Media-Properties support
2076 * Accept-Ranges support
2078 * gst_rtsp_media_seekable
2079 The RTSP methods that have been removed when using 2.0 now return
2081 https://bugzilla.gnome.org/show_bug.cgi?id=781446
2083 2017-06-02 15:37:54 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
2085 * gst/rtsp-server/rtsp-stream.c:
2086 stream: Use stream duration as stream-stop if segment was not configured with a stop
2087 Allowing client to know stream duration when no seeking happened.
2088 https://bugzilla.gnome.org/show_bug.cgi?id=783435
2090 2017-09-25 19:40:17 +0300 Sebastian Dröge <sebastian@centricular.com>
2092 * gst/rtsp-server/rtsp-media-factory.c:
2093 rtsp-media-factory: Don't cache any media if NULL was returned as key
2094 The docs already mentioned this, but we actually stored it in the hash
2095 table with key==NULL and leaked its reference forever.
2097 2017-09-18 19:31:31 +0200 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
2099 * gst/rtsp-sink/gstrtspclientsink.c:
2100 * gst/rtsp-sink/gstrtspclientsink.h:
2101 rtspclientsink: Use a mutex for protecting against concurrent send/receives
2102 This is a simple port of:
2103 * a722f6e8329032c6eda4865d6a07f4ba5981d7ea
2104 * c438545dc9e2f14f657bc0ef261fff726449867b
2105 * cd17c71dcea5c9310d21f1347c7520983e5869ac
2106 in gst-plugins-good.
2108 2017-08-31 13:24:15 +0530 Satya Prakash Gupta <sp.gupta@samsung.com>
2110 * gst/rtsp-server/rtsp-sdp.c:
2111 sdp: fix Memory leak in error case
2112 https://bugzilla.gnome.org/show_bug.cgi?id=787059
2114 2017-08-18 17:37:01 +0100 Tim-Philipp Müller <tim@centricular.com>
2116 * pkgconfig/meson.build:
2117 meson: don't install -uninstalled.pc file
2118 https://bugzilla.gnome.org/show_bug.cgi?id=786457
2120 2017-08-17 12:26:17 +0100 Tim-Philipp Müller <tim@centricular.com>
2123 Automatic update of common submodule
2124 From 48a5d85 to 3f4aa96
2126 2017-08-14 21:04:23 +0300 Sebastian Dröge <sebastian@centricular.com>
2128 * gst/rtsp-server/rtsp-client.c:
2129 rtsp-client: Fix typo in debug message
2131 2017-08-11 14:14:32 +0100 Tim-Philipp Müller <tim@centricular.com>
2134 meson: hide symbols by default unless explicitly exported
2136 2017-08-10 14:20:12 +0100 Tim-Philipp Müller <tim@centricular.com>
2138 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
2139 pkgconfig: remove -I@srcdir@/.. which duplicates abs_top_srcdir
2140 Fixes meson warning about undefined @srcdir@.
2142 2017-07-21 13:36:00 +0100 Tim-Philipp Müller <tim@centricular.com>
2144 * tests/meson.build:
2145 meson: skip tests on windows for now
2146 As we do in the other modules. As libgstcheck is currently not
2147 built on windows. Fixes "Fallback variable 'gst_check_dep' in
2148 the subproject 'gstreamer' does not exist"" Meson error.
2150 2017-06-22 07:25:07 -0700 Julien Isorce <julien.isorce@gmail.com>
2152 * gst/rtsp-server/rtsp-stream.c:
2153 rtsp-stream: fix connection delay due to wrong assumption on last-sample
2154 Commit 852cc09f542af5cadd79ffd7fe79d6475cf57e14 assumed that
2155 multiudpsink's last-sample always comes from the payloader. Which
2156 is wrong if auxiliary streams are multiplexed in the same stream.
2157 So check the buffer's ssrc against the caps'ssrc before to use its
2158 seqnum. If not the same ssrc just use the payloader as done prior
2159 the commit above or when there is no last-sample yet.
2160 https://bugzilla.gnome.org/show_bug.cgi?id=784094
2162 2017-06-23 16:19:04 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
2165 meson: Allow using glib as a subproject
2167 2017-06-26 09:55:49 +0100 Tim-Philipp Müller <tim@centricular.com>
2170 meson: fix with-package-name option
2171 https://bugzilla.gnome.org/show_bug.cgi?id=784082
2173 2017-06-09 20:16:28 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
2176 Distribute meson_options.txt
2178 2017-06-09 20:11:47 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
2181 And config.h.meson is no longer dist either
2183 2017-06-09 21:27:09 +0100 Tim-Philipp Müller <tim@centricular.com>
2187 meson: config.h.meson is no longer needed
2189 2017-06-07 13:04:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
2191 * tests/check/meson.build:
2192 * tests/meson.build:
2193 meson: Fix building tests and activate them again
2195 2017-06-07 12:55:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
2197 * tests/check/meson.build:
2198 meson: Do not use path separator in test names
2199 Avoiding warnings like:
2200 WARNING: Target "elements/audioamplify" has a path separator in its name.
2202 2017-05-20 15:07:31 +0100 Tim-Philipp Müller <tim@centricular.com>
2205 * meson_options.txt:
2206 meson: add options to set package name and origin
2207 https://bugzilla.gnome.org/show_bug.cgi?id=782172
2209 2017-05-18 10:35:18 +0100 Tim-Philipp Müller <tim@centricular.com>
2211 * gst/rtsp-server/rtsp-address-pool.h:
2212 * gst/rtsp-server/rtsp-auth.h:
2213 * gst/rtsp-server/rtsp-client.h:
2214 * gst/rtsp-server/rtsp-context.h:
2215 * gst/rtsp-server/rtsp-media-factory-uri.h:
2216 * gst/rtsp-server/rtsp-media-factory.h:
2217 * gst/rtsp-server/rtsp-media.h:
2218 * gst/rtsp-server/rtsp-mount-points.h:
2219 * gst/rtsp-server/rtsp-params.h:
2220 * gst/rtsp-server/rtsp-permissions.h:
2221 * gst/rtsp-server/rtsp-sdp.h:
2222 * gst/rtsp-server/rtsp-server.h:
2223 * gst/rtsp-server/rtsp-session-media.h:
2224 * gst/rtsp-server/rtsp-session-pool.h:
2225 * gst/rtsp-server/rtsp-session.h:
2226 * gst/rtsp-server/rtsp-stream-transport.h:
2227 * gst/rtsp-server/rtsp-stream.h:
2228 * gst/rtsp-server/rtsp-thread-pool.h:
2229 * gst/rtsp-server/rtsp-token.h:
2230 Mark symbols explicitly for export with GST_EXPORT
2232 2017-05-16 14:44:43 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
2235 * gst/rtsp-sink/Makefile.am:
2236 Remove plugin specific static build option
2237 Static and dynamic plugins now have the same interface. The standard
2238 --enable-static/--enable-shared toggle are sufficient.
2240 2017-05-04 18:59:14 +0300 Sebastian Dröge <sebastian@centricular.com>
2246 === release 1.12.0 ===
2248 2017-05-04 15:40:46 +0300 Sebastian Dröge <sebastian@centricular.com>
2254 * gst-rtsp-server.doap:
2258 === release 1.11.91 ===
2260 2017-04-27 17:42:02 +0300 Sebastian Dröge <sebastian@centricular.com>
2266 * gst-rtsp-server.doap:
2270 2017-04-24 20:30:37 +0100 Tim-Philipp Müller <tim@centricular.com>
2273 Automatic update of common submodule
2274 From 60aeef6 to 48a5d85
2276 2017-04-13 14:20:10 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
2278 * gst/rtsp-server/rtsp-media-factory.c:
2279 * gst/rtsp-server/rtsp-media.c:
2280 * gst/rtsp-server/rtsp-session.c:
2281 * gst/rtsp-server/rtsp-stream.c:
2282 gi: Fix some annotations and docstrings
2284 2017-04-13 13:52:26 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
2286 * gst/rtsp-server/meson.build:
2288 * meson_options.txt:
2291 2017-04-10 23:51:12 +0100 Tim-Philipp Müller <tim@centricular.com>
2295 Automatic update of common submodule
2296 From 39ac2f5 to 60aeef6
2298 === release 1.11.90 ===
2300 2017-04-07 16:35:03 +0300 Sebastian Dröge <sebastian@centricular.com>
2306 * gst-rtsp-server.doap:
2310 2017-03-27 18:19:33 +0100 Tim-Philipp Müller <tim@centricular.com>
2312 * examples/test-launch.c:
2313 examples: make test-launch pipeline shared by default as well
2315 2017-02-27 19:10:44 +0200 Sebastian Dröge <sebastian@centricular.com>
2317 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
2318 gstreamer-rtsp-server: Add both srcdir and builddir to the include path
2319 Just the build dir is not going to work for srcdir!=builddir.
2321 2017-02-24 15:59:54 +0200 Sebastian Dröge <sebastian@centricular.com>
2324 meson: Update version
2326 2017-02-24 15:37:49 +0200 Sebastian Dröge <sebastian@centricular.com>
2331 === release 1.11.2 ===
2333 2017-02-24 15:10:07 +0200 Sebastian Dröge <sebastian@centricular.com>
2339 * gst-rtsp-server.doap:
2342 2017-02-14 20:40:26 +0000 Tim-Philipp Müller <tim@centricular.com>
2345 meson: dist meson build files
2346 Ship meson build files in tarballs, so people who use tarballs
2347 in their builds can start playing with meson already.
2349 2017-02-07 23:39:37 +1100 Jan Schmidt <jan@centricular.com>
2351 * examples/test-record.c:
2352 examples/test-record: Add extra line to initial printout
2353 Add an example line of how to deliver a stream to the
2356 2017-01-19 14:57:19 +0200 Sebastian Dröge <sebastian@centricular.com>
2358 * gst/rtsp-server/rtsp-client.c:
2359 rtsp-client: Also handle the (S|G)ET_PARAMETER case of size==0 || !data as keep-alive
2360 If there is no Content-Length header, no body would be allocated and the
2361 '\0' would also not be appended to the body.
2363 2017-01-19 14:24:07 +0200 Sebastian Dröge <sebastian@centricular.com>
2365 * gst/rtsp-server/rtsp-client.c:
2366 rtsp-client: Fix handling of keep-alive GET_PARAMETER/SET_PARAMETER
2367 While they logically have 0 bytes length, GstRTSPConnection is appending
2368 a '\0' to everything making the size be 1 instead.
2370 2017-01-13 12:39:36 +0000 Tim-Philipp Müller <tim@centricular.com>
2375 2017-01-12 19:04:23 +0200 Sebastian Dröge <sebastian@centricular.com>
2377 * gst/rtsp-server/rtsp-session.c:
2378 rtsp-session: Only remove deprecated API if requested to do so, not just when disabling
2379 gst_rtsp_session_is_expired() and gst_rtsp_session_next_timeout() were
2382 2017-01-12 16:32:59 +0200 Sebastian Dröge <sebastian@centricular.com>
2387 === release 1.11.1 ===
2389 2017-01-12 16:14:46 +0200 Sebastian Dröge <sebastian@centricular.com>
2395 * gst-rtsp-server.doap:
2396 * win32/common/libgstrtspserver.def:
2399 2017-01-10 08:34:50 +0100 Patricia Muscalu <patricia@axis.com>
2401 * gst/rtsp-server/rtsp-stream.c:
2402 rtsp-stream: corrected if-statement in _get_server_port()
2403 This bug was accidentally introduced while fixing a segfault
2404 in _get_server_port() function.
2405 https://bugzilla.gnome.org/show_bug.cgi?id=776345
2407 2017-01-09 14:12:05 +0100 Patricia Muscalu <patricia@axis.com>
2409 * gst/rtsp-server/rtsp-stream.c:
2410 * tests/check/gst/stream.c:
2411 rtsp-stream: fixed segmenation fault in _get_server_port()
2412 Calling function gst_rtsp_stream_get_server_port() results in
2413 segmenation fault in the RTP/RTSP/TCP case.
2414 Port that the server will use to receive RTCP makes only
2415 sense in the UDP case, however the function should handle
2416 the TCP case in a nicer way.
2417 https://bugzilla.gnome.org/show_bug.cgi?id=776345
2419 2017-01-09 12:22:40 +0300 Aleksandr Slobodeniuk <alenuke@yandex.ru>
2421 * gst/rtsp-server/rtsp-media-factory.c:
2422 dosc: Fix a little typo
2423 https://bugzilla.gnome.org/show_bug.cgi?id=777037
2425 2017-01-04 16:20:54 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
2427 * pkgconfig/Makefile.am:
2428 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
2429 * pkgconfig/meson.build:
2430 meson: generate pkg-config -uninstalled pc files
2431 Generating those files is useful for users building the GStreamer stack
2432 using meson and having to link it to another project which is still
2433 using the autotools.
2434 https://bugzilla.gnome.org/show_bug.cgi?id=776810
2436 2017-01-04 16:11:08 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
2438 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
2439 pkgconfig: fix -uninstalled pc file
2440 pcfiledir was never defined so the paths were wrong.
2441 https://bugzilla.gnome.org/show_bug.cgi?id=776867
2443 2016-12-21 13:41:50 +0100 Patricia Muscalu <patricia@axis.com>
2445 * gst/rtsp-server/rtsp-stream.c:
2446 * tests/check/gst/rtspserver.c:
2447 rtsp-stream: Fixed TCP transport case
2448 Make sure that the appsink element is actually added to
2449 the bin before trying to link it with the elements in it.
2450 https://bugzilla.gnome.org/show_bug.cgi?id=776343
2452 2016-12-16 17:26:04 +0000 Tim-Philipp Müller <tim@centricular.com>
2458 Remove generated .spec file
2459 Likely extremely bitrotten, and we should not ship this anyway.
2461 2016-12-03 08:21:02 +0100 Edward Hervey <bilboed@bilboed.com>
2464 Automatic update of common submodule
2465 From f980fd9 to 39ac2f5
2467 2016-12-02 15:40:09 +0100 Edward Hervey <edward@centricular.com>
2469 * gst/rtsp-server/rtsp-media.c:
2470 media: Fix pt map caps
2471 Since decryption is handled within rtpbin, all outcoming stream
2472 caps will be application/x-rtp (i.e. regular rtp)
2473 Fixes RECORD with SRTP streams
2475 2016-12-02 15:38:04 +0100 Edward Hervey <edward@centricular.com>
2477 * gst/rtsp-server/rtsp-media-factory.c:
2478 media-factory: Create media objects with the proper transport mode
2479 The function called immediately afterwards (collect_streams()) will
2480 need it to work properly
2482 2016-12-02 14:36:50 +0200 Sebastian Dröge <sebastian@centricular.com>
2484 * gst/rtsp-server/rtsp-auth.c:
2485 rtsp-auth: Don't remove digest-auth nonces that already/still have a client connected
2487 2016-12-01 18:04:34 +0200 Sebastian Dröge <sebastian@centricular.com>
2489 * gst/rtsp-server/rtsp-media-factory.c:
2490 rtsp-media-factory: Don't create a pipeline for the media pipeline string
2491 We're going to put a pipeline into a pipeline otherwise, which is not
2494 2016-10-25 15:41:28 +0300 Kseniia Vasilchuk <vasilchukkseniia@gmail.com>
2496 * gst/rtsp-server/rtsp-media.c:
2497 media: Fix race condition around finish_unprepare() if called multiple time
2498 https://bugzilla.gnome.org/show_bug.cgi?id=755329
2500 2016-11-30 14:06:36 +1100 Jan Schmidt <jan@centricular.com>
2502 * gst/rtsp-sink/gstrtspclientsink.c:
2503 rtspclientsink: Don't leave stale pointer after unref
2504 Fix a warning on shutdown - don't keep a pointer to an
2505 alread-unreffed object.
2507 2016-11-26 11:24:50 +0000 Tim-Philipp Müller <tim@centricular.com>
2510 common: use https protocol for common submodule
2511 https://bugzilla.gnome.org/show_bug.cgi?id=775110
2513 2016-11-21 23:29:56 +1100 Matthew Waters <matthew@centricular.com>
2515 * gst/rtsp-server/rtsp-stream.c:
2516 stream: block the output of rtpbin instead of the source pipeline
2517 85c52e194bcb81928b96614be0ae47d59eccb1ce introduced a more correct
2518 detection of the srtp rollover counter to add to the SDP.
2519 Unfortunately, it was incomplete for live pipelines where the logic
2520 blocks the source bin before creating the SDP and thus would never have
2521 the necessary informaiton to create a correct SDP with srtp encryption.
2522 Move the pad blocks to rtpbin's output pads instead so that the
2523 necessary information can be created before we need the information for
2525 https://bugzilla.gnome.org/show_bug.cgi?id=770239
2527 2016-11-21 16:02:39 +0100 Dag Gullberg <dagg@axis.com>
2529 * gst/rtsp-server/rtsp-client.c:
2530 rtsp-client: add IDLE timeout, before session exists
2531 The RTSP server will not timeout an idle RTSP connection
2532 (note this is different from doing timeout on a RTSP
2534 At least for Apache this is a problem when running RTSP over
2535 HTTPS since it uses one of the threads (there is a rather
2536 limited number) that are available for handling requests.
2537 https://bugzilla.gnome.org/show_bug.cgi?id=771830
2539 2016-11-23 09:45:08 +0000 Tim-Philipp Müller <tim@centricular.com>
2544 2016-11-21 13:05:50 +0100 Göran Jönsson <goranjn@axis.com>
2546 * gst/rtsp-server/rtsp-stream.c:
2547 rtsp-stream: Set close-socket FALSE on UDP src:es
2548 With this RTSP server can use the sockets independent on the udpsrc
2550 When the udp src is finalized it will unref socket and when g_socket
2551 is finalized the socket will be closed.
2552 https://bugzilla.gnome.org/show_bug.cgi?id=765673
2554 2016-11-18 17:47:13 +0200 Sebastian Dröge <sebastian@centricular.com>
2556 * gst/rtsp-sink/gstrtspclientsink.c:
2557 rtspclientsink: Move to new helper function to parse authentication responses
2558 https://bugzilla.gnome.org/show_bug.cgi?id=774416
2560 2016-11-16 08:42:24 +0200 Sebastian Dröge <sebastian@centricular.com>
2562 * examples/Makefile.am:
2563 * examples/test-auth-digest.c:
2564 * gst/rtsp-server/rtsp-auth.c:
2565 * gst/rtsp-server/rtsp-auth.h:
2566 * win32/common/libgstrtspserver.def:
2567 rtsp-auth: Add support for Digest authentication
2568 https://bugzilla.gnome.org/show_bug.cgi?id=774416
2570 2016-11-17 09:41:53 -0800 Scott D Phillips <scott.d.phillips@intel.com>
2573 * gst/rtsp-server/meson.build:
2575 * tests/check/meson.build:
2577 * win32/common/libgstrtspserver.def:
2578 Enable building with MSVC
2579 https://bugzilla.gnome.org/show_bug.cgi?id=774640
2581 2016-11-18 20:23:14 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
2584 meson: gstreamer gst_check_dep does not exist on windows
2586 2016-11-17 09:43:37 -0800 Scott D Phillips <scott.d.phillips@intel.com>
2588 * gst/rtsp-server/rtsp-client.c:
2589 client: update do_send_message to match type GstRTSPClientSendFunc
2590 This type mismatch fails building with MSVC
2591 https://bugzilla.gnome.org/show_bug.cgi?id=774640
2593 2016-11-11 14:42:08 +0200 Sebastian Dröge <sebastian@centricular.com>
2595 * gst/rtsp-server/rtsp-sdp.c:
2596 rtsp-sdp: Fix indentation
2598 2016-11-10 05:16:00 +0000 Neha Arora <arora.neha@samsung.com>
2600 * gst/rtsp-server/rtsp-media.c:
2601 rtsp-media: Only signal "new-state" if the state has actually changed
2602 https://bugzilla.gnome.org/show_bug.cgi?id=774173
2604 2016-08-24 11:39:13 +0200 Branko Subasic <branko@axis.com>
2606 * gst/rtsp-server/rtsp-client.c:
2607 * gst/rtsp-server/rtsp-client.h:
2608 client: emit signal in the beginning of each rtsp request
2609 These signals let the application validate the requests, configure the
2610 media/stream in a certain way and also generate error status code in
2611 case of error or bad request.
2612 https://bugzilla.gnome.org/show_bug.cgi?id=758062
2614 2016-11-01 18:10:35 +0000 Tim-Philipp Müller <tim@centricular.com>
2617 meson: update version
2619 === release 1.11.0 ===
2621 2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com>
2626 === release 1.10.0 ===
2628 2016-11-01 18:06:46 +0200 Sebastian Dröge <sebastian@centricular.com>
2634 * gst-rtsp-server.doap:
2637 2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com>
2639 * tests/check/gst/rtspserver.c:
2640 * tests/check/gst/stream.c:
2641 tests: try to avoid using the same ports in different tests
2642 Causes problems with client multicast tests otherwise if
2643 tests are run in parallel.
2644 https://bugzilla.gnome.org/show_bug.cgi?id=773640
2646 2016-10-28 17:50:59 +0100 Tim-Philipp Müller <tim@centricular.com>
2648 * tests/check/gst/client.c:
2649 tests: client: use fail_unless_equals_foo() for better failure reporting
2651 2016-09-26 11:16:04 +0200 Göran Jönsson <goranjn@axis.com>
2653 * gst/rtsp-server/rtsp-client.c:
2654 rtsp-client: Session filter in unwatch session
2655 Call session filter with filter_session_media as paramer in
2656 client_unwatch_session if using drop_backlog = FALSE.
2657 In client_unwatch_session its allowed to grow the watchs backlog.
2658 If using drop_backlog = FALSE and the backlog is full it will cause
2659 a deadlock when setting session media state to NULL
2660 if the backlog is not allowed to grow.
2661 https://bugzilla.gnome.org/show_bug.cgi?id=771983
2663 2016-10-20 21:40:18 +0100 Tim-Philipp Müller <tim@centricular.com>
2666 meson: add fallbacks for gst modules
2669 2016-09-14 17:48:39 +0300 Nikita Bobkov <NikitaDBobkov@gmail.com>
2671 * gst/rtsp-server/rtsp-client.c:
2672 rtsp-client: Fix factory leaking in find_media() in error cases
2673 https://bugzilla.gnome.org/show_bug.cgi?id=771488
2675 2016-10-06 11:47:50 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2677 * gst/rtsp-server/rtsp-stream.c:
2678 stream: Fix randomly missing streams from SDP with dynamic elements
2679 When using dynamic elements, gst_rtsp_stream_join_bin() is called from
2680 "pad-added" signal. In that case priv->srcpad could already have its caps,
2681 and they'll be sent to priv->send_src[0] pad. That means that when it
2682 connects "notify::caps" signal, that pad could already have received its
2683 caps and the signal won't be emitted anymore.
2684 In that case priv->caps stay to NULL and when building the SDP that stream
2685 gets ignored. Leading to missing video or audio when playing in client side.
2686 https://bugzilla.gnome.org/show_bug.cgi?id=772478
2688 2016-09-30 11:42:08 +0100 Tim-Philipp Müller <tim@centricular.com>
2691 meson: update version
2693 === release 1.9.90 ===
2695 2016-09-30 13:04:12 +0300 Sebastian Dröge <sebastian@centricular.com>
2701 * gst-rtsp-server.doap:
2704 2016-09-17 13:17:19 +0100 Ian Jamison <ian.dev@arkver.com>
2706 * gst/rtsp-server/rtsp-media-factory.c:
2707 * gst/rtsp-server/rtsp-media.c:
2708 * gst/rtsp-server/rtsp-stream.c:
2709 rtsp-server: Hint that set_multicast_iface expects the name of the interface
2710 To prevent any possibly confusion with IPs or anything else.
2711 https://bugzilla.gnome.org/show_bug.cgi?id=771530
2713 2016-09-18 09:58:55 -0400 Sebastian Dröge <sebastian@centricular.com>
2715 * gst/rtsp-server/rtsp-media-factory.c:
2716 * gst/rtsp-server/rtsp-media.c:
2717 rtsp-media: Call g_free() instead of g_object_unref() on multicast-iface strings
2718 https://bugzilla.gnome.org/show_bug.cgi?id=763000#c5
2720 2016-09-14 11:31:15 +0200 Sebastian Dröge <sebastian@centricular.com>
2723 configure: Depend on gstreamer 1.9.2.1
2725 2016-09-10 20:52:31 +1000 Jan Schmidt <jan@centricular.com>
2729 Automatic update of common submodule
2730 From b18d820 to f980fd9
2732 2016-09-10 09:58:31 +1000 Jan Schmidt <jan@centricular.com>
2736 Automatic update of common submodule
2737 From 6f2d209 to b18d820
2739 2016-09-07 18:44:34 +0300 Sebastian Dröge <sebastian@centricular.com>
2741 * gst/rtsp-server/rtsp-stream.c:
2742 rtsp-stream: Remove unused _locked() variant of a function
2743 It was added during refactoring.
2745 2016-09-07 10:21:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2747 * gst/rtsp-server/rtsp-stream.c:
2748 stream: cosmetic cleanup
2749 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2751 2016-09-07 10:16:19 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2753 * gst/rtsp-server/rtsp-stream.c:
2754 stream: Compare IP addresses case insensitive in more places
2755 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2757 2016-09-07 10:12:18 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2760 * gst/rtsp-server/rtsp-stream.c:
2761 stream: Fix leaked joined_bin
2762 There is no need to keep a strong ref on it, and _leave_bin() was
2763 setting it to NULL before calling g_clear_object() so it was leaked.
2764 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2766 2016-09-06 19:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
2768 * gst/rtsp-server/rtsp-stream.c:
2769 rtsp-stream: Compare IP address strings case insensitive
2770 Otherwise IPv6 addresses might fail this comparision.
2772 2016-09-06 19:10:21 +0300 Sebastian Dröge <sebastian@centricular.com>
2774 * gst/rtsp-server/rtsp-stream.c:
2775 rtsp-stream: Bind multicast sockets to ANY as before
2776 https://bugzilla.gnome.org/show_bug.cgi?id=766612#c48
2778 2016-09-05 18:31:36 +0300 Kseniia <vasilchukkseniia@gmail.com>
2780 * gst/rtsp-server/rtsp-session.c:
2781 rtsp-session: Fix segfault when doing keep-alive after removing the session
2782 If keep-alive happens after removing the session but before finalizing the
2783 stream transport, we would segfault.
2784 https://bugzilla.gnome.org/show_bug.cgi?id=750544
2786 2016-09-05 18:04:50 +0300 Sebastian Dröge <sebastian@centricular.com>
2788 * gst/rtsp-server/rtsp-stream.c:
2789 rtsp-stream: Always create multicast UDP elements if the protocol flag is set
2790 Adding them later will cause deadlocks due to
2791 1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
2792 2) adding the multicast sink
2793 3) waiting for it to get data to preroll again
2794 3) never happens because the queues after the tee are full.
2796 2016-09-05 16:32:57 +0300 Sebastian Dröge <sebastian@centricular.com>
2798 * gst/rtsp-server/rtsp-stream.c:
2799 rtsp-stream: Fix up various multicast related issues
2801 2016-09-05 13:40:59 +0300 Sebastian Dröge <sebastian@centricular.com>
2803 * tests/check/gst/stream.c:
2804 tests: Fix compilation
2806 2016-07-28 15:33:05 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2808 * gst/rtsp-server/rtsp-client.c:
2809 * gst/rtsp-server/rtsp-stream.c:
2810 * tests/check/gst/stream.c:
2811 stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast
2812 This is basically reverting changes introduced in commit f62a9a7,
2813 because it was introducing various regressions:
2814 - It introduces a leak of udpsrc elements that got wrongly fixed by adding
2815 an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
2816 ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
2817 - If a mcast client connects, it creates a new socket in SETUP to try to respect
2818 the destination/port given by the client in the transport, and overrides the
2819 socket already set on the udpsink element. That means that if we already had a
2820 client connected, the source address on the udp packets it receives suddenly
2822 - If a 2nd mcast client connects, the destination/port in its transport is
2823 ignored but its transport wasn't updated.
2824 What this patch does:
2825 - Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
2826 - Always have a tee+queue when udp is enabled. This could be optimized
2827 again in a later patch, but is more complicated. If no unicast clients
2828 connects then those elements are useless, this could be also optimized
2830 - When mcast transport is added, it creates a new set of udpsrc/udpsink,
2831 seperated from those for unicast clients. Since we already support only
2832 one mcast address, we also create only one set of elements.
2833 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2835 2016-07-28 15:20:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2837 * gst/rtsp-server/rtsp-stream.c:
2838 stream: factor our plug_src function
2839 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2841 2016-07-21 21:46:16 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2843 * gst/rtsp-server/rtsp-stream.c:
2844 stream: factor out plug_sink function
2845 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2847 2016-07-20 23:05:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2849 * gst/rtsp-server/rtsp-stream.c:
2850 stream: small documentation clarification
2851 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2853 2016-07-20 15:35:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2855 * gst/rtsp-server/rtsp-stream.c:
2856 stream: rename addr_v4/6 to mcast_addr_v4/6 for clarity
2857 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2859 2016-07-14 11:10:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2861 * gst/rtsp-server/rtsp-stream.c:
2862 stream: Keep a ref on joined bin
2863 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2865 2016-07-20 15:11:32 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2867 * gst/rtsp-server/rtsp-stream.c:
2868 stream: code cleanup
2869 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2871 2016-07-20 23:18:23 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2873 * gst/rtsp-server/rtsp-stream.c:
2874 stream: small fix in error code path
2875 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2877 2016-07-20 20:09:57 -0400 Xavier Claessens <xavier.claessens@collabora.com>
2879 * gst/rtsp-server/rtsp-stream.c:
2880 Revert "rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc"
2881 This partly reverts commit cba045e1b19fad6e689e10206f57903e15f1229a,
2882 but keeps unit tests.
2883 https://bugzilla.gnome.org/show_bug.cgi?id=766612
2885 2016-09-01 12:33:00 +0300 Sebastian Dröge <sebastian@centricular.com>
2890 === release 1.9.2 ===
2892 2016-09-01 12:32:51 +0300 Sebastian Dröge <sebastian@centricular.com>
2898 * gst-rtsp-server.doap:
2901 2016-01-27 01:03:52 +0000 Tim-Philipp Müller <tim@centricular.com>
2904 * examples/meson.build:
2906 * gst/rtsp-server/meson.build:
2907 * gst/rtsp-sink/meson.build:
2909 * pkgconfig/meson.build:
2910 * tests/check/meson.build:
2911 * tests/meson.build:
2912 Add support for Meson as alternative/parallel build system
2913 https://github.com/mesonbuild/meson
2915 2016-08-26 21:56:13 +0200 Josep Torra <n770galaxy@gmail.com>
2918 * tests/check/Makefile.am:
2919 build: silence error about pthread for 'make check' in osx
2920 Fixes "clang: error: argument unused during compilation: '-pthread'"
2922 2015-09-25 15:04:00 +0000 Nikita Bobkov <NikitaDBobkov@gmail.com>
2924 * gst/rtsp-server/rtsp-client.c:
2925 rtsp-client: Fix leaking of media in error cases
2926 With additional fixes by Kseniya Vasilchuk <vasilchukkseniia@gmail.com>
2927 and myself to make the media refcounting a bit easier to follow.
2928 https://bugzilla.gnome.org/show_bug.cgi?id=755632
2930 2016-08-02 15:08:22 +0300 Sebastian Dröge <sebastian@centricular.com>
2932 * gst/rtsp-server/rtsp-client.c:
2933 rtsp-client: Fix leaking of session in error cases
2934 https://bugzilla.gnome.org/show_bug.cgi?id=755632
2936 2016-07-11 21:16:04 +0200 Stefan Sauer <ensonic@users.sf.net>
2939 Automatic update of common submodule
2940 From f363b32 to f49c55e
2942 2016-07-06 13:51:15 +0300 Sebastian Dröge <sebastian@centricular.com>
2947 === release 1.9.1 ===
2949 2016-07-06 13:28:12 +0300 Sebastian Dröge <sebastian@centricular.com>
2955 * gst-rtsp-server.doap:
2958 2016-06-24 02:02:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
2961 configure: Need to add -DGST_STATIC_COMPILATION when building only statically
2962 https://bugzilla.gnome.org/show_bug.cgi?id=767463
2964 2016-06-21 11:49:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
2967 Automatic update of common submodule
2968 From ac2f647 to f363b32
2970 2016-04-14 22:56:11 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
2972 * gst/rtsp-server/rtsp-sdp.c:
2973 * gst/rtsp-server/rtsp-sdp.h:
2974 * gst/rtsp-server/rtsp-stream.c:
2975 * gst/rtsp-server/rtsp-stream.h:
2976 sdp: add rollover counters for all sender SSRC
2977 We add different crypto sessions in MIKEY, one for each sender
2978 SSRC. Currently, all of them will have the same security policy, 0.
2979 The rollover counters are obtained from the srtpenc element using the
2981 https://bugzilla.gnome.org/show_bug.cgi?id=730539
2983 2016-06-07 20:44:42 +0100 Tim-Philipp Müller <tim@centricular.com>
2985 * gst/rtsp-server/rtsp-media-factory.h:
2986 * gst/rtsp-server/rtsp-server.h:
2987 docs: fix some typos
2989 2016-05-25 10:28:43 +0100 Tim-Philipp Müller <tim@centricular.com>
2991 * gst/rtsp-server/Makefile.am:
2992 g-i: pass compiler env to g-ir-scanner
2993 It's what introspection.mak does as well. Should
2994 fix spurious build failures on gnome-continuous
2995 (caused by g-ir-scanner getting compiler details
2996 via python which is broken in some environments
2997 so passing the compiler details bypasses that).
2999 2016-05-18 16:48:44 +0100 Ian <ian.arkver.dev@gmail.com>
3001 * gst/rtsp-server/rtsp-session.c:
3002 rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header
3003 This works with rtspsrc and live555, but fails with e.g. ffmpeg.
3004 https://bugzilla.gnome.org/show_bug.cgi?id=766619
3006 2016-03-07 14:48:38 +0100 Edward Hervey <bilboed@bilboed.com>
3008 * gst/rtsp-sink/gstrtspclientsink.c:
3009 rtspclientsink: Check return value of sscanf
3010 And just make sure we always have 0/0 if we have an error
3013 2016-04-25 08:55:25 -0400 Jake Foytik <jake.foytik@ipconfigure.com>
3015 * gst/rtsp-server/rtsp-stream.c:
3016 * tests/check/gst/rtspserver.c:
3017 * tests/check/gst/stream.c:
3018 rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
3019 - Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
3020 - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
3021 - Create unit test for shared media.
3022 https://bugzilla.gnome.org/show_bug.cgi?id=764744
3024 2016-04-11 10:55:23 +0300 Sebastian Dröge <sebastian@centricular.com>
3026 * gst/rtsp-server/rtsp-stream.c:
3027 rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
3028 For IPv6 addresses, binding to a multicast group does not work on Linux
3029 either. Always bind to ANY and then later join the multicast group.
3030 https://bugzilla.gnome.org/show_bug.cgi?id=764679
3032 2016-04-14 10:05:02 +0100 Julien Isorce <j.isorce@samsung.com>
3035 Automatic update of common submodule
3036 From 6f2d209 to ac2f647
3038 2016-04-06 10:09:46 +0200 Patricia Muscalu <patricia@axis.com>
3040 * gst/rtsp-server/rtsp-thread-pool.c:
3041 rtsp-thread-pool: explained why GSource is a part of ThreadImpl
3042 Clarified why it is necessary to add source information to
3043 GstRTSPThreadImpl. See the reported bug in GLib:
3044 https://bugzilla.gnome.org/show_bug.cgi?id=720186
3045 for more information.
3046 https://bugzilla.gnome.org/show_bug.cgi?id=761702
3048 2016-04-04 12:58:38 +0300 Sebastian Dröge <sebastian@centricular.com>
3050 * examples/Makefile.am:
3051 examples: Clean up CFLAGS/LDADD even more
3052 The internal .la should come first and is part of LDADD, as is
3055 2016-04-04 12:39:39 +0300 Sebastian Dröge <sebastian@centricular.com>
3057 * examples/Makefile.am:
3058 examples: Clean up CFLAGS/LDADD to link with the correct versions of all libraries
3060 2016-04-03 12:06:29 +0300 Sebastian Dröge <sebastian@centricular.com>
3062 * gst/rtsp-server/Makefile.am:
3063 rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS)
3065 2015-12-30 18:39:05 +0200 Sebastian Dröge <sebastian@centricular.com>
3067 * gst/rtsp-server/rtsp-client.c:
3068 * gst/rtsp-server/rtsp-media-factory.c:
3069 * gst/rtsp-server/rtsp-media-factory.h:
3070 * gst/rtsp-server/rtsp-media.c:
3071 * gst/rtsp-server/rtsp-media.h:
3072 * gst/rtsp-server/rtsp-sdp.c:
3073 * gst/rtsp-server/rtsp-stream.c:
3074 * gst/rtsp-server/rtsp-stream.h:
3075 rtsp-server: Implement clock signalling according to RFC7273
3076 For NTP and PTP clocks we signal the actual clock that is used and signal
3077 the direct media clock offset.
3078 For all other clocks we at least signal that it's the local sender clock.
3079 This allows receivers to know which clock was used to generate the media and
3080 its RTP timestamps. Receivers can then implement network synchronization,
3081 either absolute or at least relative by getting the sender clock rate directly
3082 via NTP/PTP instead of estimating it from RTP timestamps and packet receive
3084 https://bugzilla.gnome.org/show_bug.cgi?id=760005
3086 2016-03-02 19:42:58 +0200 Sebastian Dröge <sebastian@centricular.com>
3088 * gst/rtsp-sink/gstrtspclientsink.c:
3089 rtspclientsink: Add support for setting the multicast interface
3090 https://bugzilla.gnome.org/show_bug.cgi?id=763000
3092 2016-03-02 19:42:13 +0200 Sebastian Dröge <sebastian@centricular.com>
3094 * gst/rtsp-server/rtsp-media-factory.c:
3095 * gst/rtsp-server/rtsp-media-factory.h:
3096 * gst/rtsp-server/rtsp-media.c:
3097 * gst/rtsp-server/rtsp-media.h:
3098 * gst/rtsp-server/rtsp-stream.c:
3099 * gst/rtsp-server/rtsp-stream.h:
3100 rtsp-media: Add support for setting the multicast interface
3101 https://bugzilla.gnome.org/show_bug.cgi?id=763000
3103 2016-03-07 08:50:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
3105 * gst/rtsp-sink/gstrtspclientsink.c:
3106 rtspclientsink: use new gst_element_class_add_static_pad_template()
3107 https://bugzilla.gnome.org/show_bug.cgi?id=763196
3109 2016-03-24 13:33:43 +0200 Sebastian Dröge <sebastian@centricular.com>
3114 === release 1.8.0 ===
3116 2016-03-24 13:00:35 +0200 Sebastian Dröge <sebastian@centricular.com>
3122 * gst-rtsp-server.doap:
3125 2016-03-16 23:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
3127 * gst/rtsp-server/rtsp-stream.c:
3128 rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
3129 This would get us NO_PREROLL in the bin again and break seeking.
3130 Thanks to Carlos Rafael Giani for helping to debug this!
3131 https://bugzilla.gnome.org/show_bug.cgi?id=740509
3133 === release 1.7.91 ===
3135 2016-03-15 12:26:13 +0200 Sebastian Dröge <sebastian@centricular.com>
3141 * gst-rtsp-server.doap:
3144 2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
3146 * gst/rtsp-server/rtsp-stream.c:
3147 rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
3148 Without this, RECORD pipelines are broken because
3149 a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
3150 added later. Previously it was there earlier and due to NO_PREROLL caused the
3151 pipeline to preroll immediately
3152 b) the udpsrc for the pipeline is added later and never set to PLAYING state,
3153 as the corresponding code previously was only for PLAY pipelines.
3154 https://bugzilla.gnome.org/show_bug.cgi?id=763281
3156 2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
3158 * gst/rtsp-server/rtsp-stream.c:
3159 rtsp-stream: Fix typo in the docstring
3160 gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
3162 2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
3164 * gst/rtsp-server/rtsp-stream.c:
3165 rtsp-stream: Disable multicast loopback for all our sockets
3166 On Windows this is a receiver-side setting, on Linux a sender-side setting. As
3167 we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
3168 loopback setting on the socket... while udpsink does which unfortunately has
3169 no effect here on Windows but on Linux.
3170 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3172 2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
3174 * tests/check/gst/stream.c:
3175 stream tests: added new tests
3176 Test a case when the address pool only contains multicast addresses
3177 and the client is requesting unicast udp.
3178 Added tests for multicast ports allocation.
3179 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3181 2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
3183 * gst/rtsp-server/rtsp-stream.c:
3184 rtsp-stream: Only bind multicast sockets to ANY on Windows
3185 On Linux it is still needed to bind to the multicast address
3186 to filter out random other packets, while on Windows binding
3187 to multicast addresses just fails.
3189 2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
3191 * gst/rtsp-server/rtsp-stream.c:
3192 rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
3193 Otherwise we fail to allocate UDP ports if the pool only contains multicast
3194 addresses, which is something that used to work before. For unicast addresses
3195 if the pool contains none, we just allocate them as if there is no pool at
3197 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3199 2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
3201 * gst/rtsp-server/rtsp-client.c:
3202 * gst/rtsp-server/rtsp-stream.c:
3203 rtsp-server: Fix indentation
3205 2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
3207 * gst/rtsp-server/rtsp-stream.c:
3208 rtsp-stream: Don't bind the sockets to multicast addresses
3209 This works on Linux but fails completely on Windows. You're supposed
3210 to bind to ANY and then join the multicast group.
3211 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3213 === release 1.7.90 ===
3215 2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
3221 * gst-rtsp-server.doap:
3224 2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
3227 Automatic update of common submodule
3228 From b64f03f to 6f2d209
3230 2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
3232 * gst/rtsp-sink/gstrtspclientsink.c:
3233 * tests/check/gst/rtspclientsink.c:
3234 rtspsink: Fix some leaks in rtspclientsink and the unit test.
3235 https://bugzilla.gnome.org/show_bug.cgi?id=762525
3237 2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
3239 * tests/check/gst/media.c:
3240 * tests/check/gst/rtspclientsink.c:
3241 * tests/check/gst/rtspserver.c:
3242 * tests/check/gst/stream.c:
3243 tests: unit test fixes
3244 Removed port allocation test from the media suite.
3245 The port allocation failure is now in the stream suite.
3247 Make sure that the media is suspended after the DESCRIBE request
3248 before reconfiguring the UDP sinks.
3250 In the RECORD case we have to set async property to false
3251 for the appsink element in the test in order to make sure
3252 that the media pipeline doesn't hang in start_preroll().
3253 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3255 2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
3257 * gst/rtsp-server/rtsp-client.c:
3258 * gst/rtsp-server/rtsp-stream.c:
3259 * gst/rtsp-server/rtsp-stream.h:
3260 rtsp-stream: postpone UDP socket allocation until SETUP
3261 Postpone the allocation of the UDP sockets until we know
3262 what transport has been chosen by the client.
3263 Both unicast and multicast UDP sources are created in one
3265 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3267 2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
3269 * gst/rtsp-server/rtsp-stream.c:
3270 rtsp-stream: postpone the creation of the UDP sources
3271 Code refactoring: allocate the UDP ports after the sender and
3272 the reciver parts have been created.
3273 We postpone the creation of the UDP sources until the UDP
3274 ports have been allocated.
3275 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3277 2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
3279 * gst/rtsp-server/rtsp-stream.c:
3280 rtsp-stream: added function for setting UDP sources to PLAYING state
3281 Code refactoring: Introduced a function for setting UDP sources
3283 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3285 2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
3287 * gst/rtsp-server/rtsp-stream.c:
3288 rtsp-stream: added function for creating and configuring UDP sources
3289 Code refactoring: create and configure UDP sources in a separate function.
3290 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3292 2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
3294 * gst/rtsp-server/rtsp-stream.c:
3295 rtsp-stream: added function for RTP/RTCP socket configuration
3296 Code refactoring: configure RTP and RTCP sockets for UDP sinks
3297 in a separate function.
3298 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3300 2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
3302 * gst/rtsp-server/rtsp-stream.c:
3303 rtsp-stream: added function for creating and configuring UDP sinks
3304 Code refactoring: create and configure UDP sinks in a separate function.
3305 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3307 2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
3309 * gst/rtsp-server/rtsp-stream.c:
3310 rtsp-stream: added helper function for creating the sender/receiver parts
3311 Code refactoring: introduced helper function for creating
3312 the receiver and the sender parts of the streaming pipeline.
3313 https://bugzilla.gnome.org/show_bug.cgi?id=757488
3315 2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
3320 === release 1.7.2 ===
3322 2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
3328 * gst-rtsp-server.doap:
3331 2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
3333 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
3334 uninstalled.pc: add support for non libtool build systems
3335 Currently the .la path is provided which requires to use libtool as
3336 mentioned in the GStreamer manual section-helloworld-compilerun.html.
3337 It is fine as long as the application is built using libtool.
3338 So currently it is not possible to compile a GStreamer application
3339 within gst-uninstalled with CMake or other build system different
3341 This patch allows to do the following in gst-uninstalled env:
3342 gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
3343 gstreamer-rtsp-server-1.0)
3344 Previously it required to prepend libtool --mode=link
3345 https://bugzilla.gnome.org/show_bug.cgi?id=720778
3347 2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
3349 * gst/rtsp-sink/gstrtspclientsink.c:
3350 rtspclientsink: remove check for impossible condition
3351 Goto error label checks stream to see if it needs to be unreferenced before
3352 returning, but this goto jumps happens before the stream is ever set, so it
3353 will always be NULL in this error label.
3356 2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
3358 * gst/rtsp-sink/gstrtspclientsink.c:
3359 rtspclientsink: clean switch statements
3360 Coverity demands for fallthrough statements to be clearly commented,
3361 to distinguish from accidental fall throughs. And it also needs all
3362 cases to finish with a break, even if the break is never going to be
3363 executed like in the case of a continue jump.
3367 2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
3369 * tests/check/Makefile.am:
3370 tests: extend the AM_TESTS_ENVIRONMENT from check.mak
3371 To get the CK_DEFAULT_TIMEOUT defined for all tests
3372 Also removes a 120 seconds timeout that was set as default
3373 explicitly in this module
3374 https://bugzilla.gnome.org/show_bug.cgi?id=761472
3376 2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
3380 Automatic update of common submodule
3381 From 86e4663 to b64f03f
3383 2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
3385 * gst/rtsp-server/rtsp-media.c:
3386 rtsp-media: fix state_lock not locked again when preroll fails
3387 https://bugzilla.gnome.org/show_bug.cgi?id=761399
3389 2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
3392 configure: Move plugin specific flags below all the others
3393 They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
3394 -no-undefined. And -no-undefined is required on Windows to build DLLs.
3396 2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
3398 * gst/rtsp-sink/gstrtspclientsink.c:
3399 rtspclientsink: Simplify slightly using new -base API
3400 Use the new Mikey and SDP API in the base plugins libs
3401 to simplify some code.
3402 https://bugzilla.gnome.org/show_bug.cgi?id=758180
3404 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3409 * gst/rtsp-sink/Makefile.am:
3410 * gst/rtsp-sink/gstrtspclientsink.c:
3411 * gst/rtsp-sink/gstrtspclientsink.h:
3412 * gst/rtsp-sink/plugin.c:
3413 * tests/check/Makefile.am:
3414 * tests/check/gst/rtspclientsink.c:
3415 rtspsink: Add rtspclientsink element
3416 Add an rtspclientsink element that accepts streams for which
3417 there is a registered payloader and sends them to
3418 an RTSP server using RECORD.
3419 Sending is synchronised to the pipeline clock. Payload-types
3420 are automatically selected. The 'new-payloader' signal is fired
3421 for custom configuration of payloaders when they are created.
3422 Can now stream a movie like this:
3424 ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
3425 decodebin name=depay1 ! audioconvert ! autoaudiosink )"
3427 gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
3428 queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
3429 https://bugzilla.gnome.org/show_bug.cgi?id=758180
3431 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3433 * gst/rtsp-server/rtsp-stream.c:
3434 * gst/rtsp-server/rtsp-stream.h:
3435 rtsp-stream: Add functions for using rtsp-stream from the client
3436 Add a boolean to indicate that the rtsp-stream is running on the
3437 'client' side of an RTSP connection, for sending streams via
3438 RECORD. In that case, the roles of the client/server ports
3439 in transport setup are swapped.
3440 https://bugzilla.gnome.org/show_bug.cgi?id=758180
3442 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3444 * gst/rtsp-server/rtsp-sdp.c:
3445 * gst/rtsp-server/rtsp-sdp.h:
3446 rtsp-sdp: Add gst_rtsp_sdp_from_stream()
3447 A new function that adds info from a GstRTSPStream into an SDP message.
3448 https://bugzilla.gnome.org/show_bug.cgi?id=758180
3450 2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
3452 * gst/rtsp-server/rtsp-media.c:
3453 rtsp-media: Fix mutex beeing unlocked while they should be locked
3454 https://bugzilla.gnome.org/show_bug.cgi?id=761226
3456 2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
3458 * gst/rtsp-server/rtsp-media-factory.c:
3459 rtsp-media-factory: add missing break in "clock" property setter
3462 2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
3464 * gst/rtsp-server/rtsp-stream.c:
3465 rtsp-stream: fixed assert during update transport
3466 When RTSP server trying update transport during multicast, it throws an
3467 assert. The assert is thrown because it is trying to get the parent of
3468 an non-existing funnel element.
3469 https://bugzilla.gnome.org/show_bug.cgi?id=760150
3471 2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
3473 * gst/rtsp-server/rtsp-permissions.h:
3474 * gst/rtsp-server/rtsp-thread-pool.h:
3475 * gst/rtsp-server/rtsp-token.h:
3476 docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
3477 gtk-doc can handle static inline functions just fine these days,
3478 there's no need for this stuff any more.
3480 2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
3482 * gst/rtsp-server/rtsp-media.c:
3483 * gst/rtsp-server/rtsp-sdp.c:
3484 sdp: replace duplicated codes to call new base sdp apis
3485 https://bugzilla.gnome.org/show_bug.cgi?id=745880
3487 2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
3489 * examples/test-netclock.c:
3490 test-netclock: Use the new API to configure a clock directly
3492 2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
3494 * gst/rtsp-server/rtsp-media-factory.c:
3495 * gst/rtsp-server/rtsp-media-factory.h:
3496 * gst/rtsp-server/rtsp-media.c:
3497 * gst/rtsp-server/rtsp-media.h:
3498 rtsp-media: Add API to directly configure a clock on the media pipelines
3500 2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
3502 * gst/rtsp-server/rtsp-media.c:
3503 rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
3505 2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
3507 * gst/rtsp-server/rtsp-media-factory.c:
3508 rtsp-media-factory: Add FIXME for 2.0
3510 2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
3512 * gst/rtsp-server/rtsp-stream.c:
3513 rtsp-stream: Fix indentation
3515 2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3517 * gst/rtsp-server/rtsp-media.c:
3518 rtsp-media: Do not prepare media after media times out
3519 Deferred calls to start_prepare() can be deferred past the point until
3520 which wait_preroll() and by proxy gst_rtsp_media_get_status() is
3521 prepared to wait. Previously there was no lock and no check for this
3522 situation. This meant that a media could be prepared and unprepared
3523 simultaneously by two different threads. Now a lock is in place and a
3524 suitable check is done.
3525 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
3527 2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
3529 * gst/rtsp-server/rtsp-client.c:
3530 * gst/rtsp-server/rtsp-media-factory.c:
3531 * gst/rtsp-server/rtsp-media-factory.h:
3532 * gst/rtsp-server/rtsp-media.c:
3533 * gst/rtsp-server/rtsp-media.h:
3534 rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
3535 Without TEARDOWN it might be desireable to keep the media running and continue
3536 sending data to the client, even if the RTSP connection itself is
3538 Only do this for session medias that have only UDP transports. If there's at
3539 least on TCP transport, it will stop working and cause problems when the
3540 connection is disconnected.
3541 https://bugzilla.gnome.org/show_bug.cgi?id=758999
3543 2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
3548 === release 1.7.1 ===
3550 2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
3556 * gst-rtsp-server.doap:
3559 2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
3562 configure: Make -Bsymbolic check work with clang.
3563 Update the -Bsymbolic check with the version glib has. This version
3565 https://bugzilla.gnome.org/show_bug.cgi?id=759713
3567 2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
3569 * gst/rtsp-server/rtsp-session-pool.c:
3570 rtsp-session-pool: Avoid dollar sign ($) in session ids
3571 Live555 in VLC strips off dollar signs and then gets very confused,
3572 we don't loose too much entropy by just skipping it.
3574 2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
3576 * gst/rtsp-server/rtsp-address-pool.h:
3577 * gst/rtsp-server/rtsp-auth.h:
3578 * gst/rtsp-server/rtsp-client.h:
3579 * gst/rtsp-server/rtsp-media-factory-uri.h:
3580 * gst/rtsp-server/rtsp-media-factory.h:
3581 * gst/rtsp-server/rtsp-media.h:
3582 * gst/rtsp-server/rtsp-mount-points.h:
3583 * gst/rtsp-server/rtsp-permissions.h:
3584 * gst/rtsp-server/rtsp-server.h:
3585 * gst/rtsp-server/rtsp-session-media.h:
3586 * gst/rtsp-server/rtsp-session-pool.h:
3587 * gst/rtsp-server/rtsp-session.h:
3588 * gst/rtsp-server/rtsp-stream-transport.h:
3589 * gst/rtsp-server/rtsp-stream.h:
3590 * gst/rtsp-server/rtsp-thread-pool.h:
3591 * gst/rtsp-server/rtsp-token.h:
3592 rtsp-server: Add g_autoptr() support to all types
3593 https://bugzilla.gnome.org/show_bug.cgi?id=754464
3595 2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
3597 * gst/rtsp-server/rtsp-stream.c:
3598 rtsp-stream: fixed valgrind error
3599 Fixed the valgrind error in unit test. The UDP source created during
3600 gst_rtsp_stream_join_bin() was not released while destroying the rtp
3602 https://bugzilla.gnome.org/show_bug.cgi?id=759010
3604 2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
3608 Automatic update of common submodule
3609 From b319909 to 86e4663
3611 2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
3613 * gst/rtsp-server/rtsp-client.c:
3614 rtsp-client: suspend media during setup request
3615 SETUP request from clients needs to suspend the media to clear the
3616 prerolled buffers. Otherwise it will not affect the prerolled buffer
3617 and the prerolled buffers will be incorrect (for example block-size
3618 from setup request will not affect the prerolled buffer unless the
3619 media is suspended).
3620 https://bugzilla.gnome.org/show_bug.cgi?id=758268
3622 2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
3624 * gst/rtsp-server/rtsp-stream.c:
3625 rtsp-stream: create stream pipeline based on transport
3626 Based on the protocol, create the rtsp stream pipeline. If only TCP or
3627 only UDP is set as the transport protocol, it will not add the extra tee
3628 or queue element to the pipeline. Both these elements will be added, if
3629 it supports both TCP and UDP protocols. This improves the pipeline
3630 performance when one protocol is present.
3631 https://bugzilla.gnome.org/show_bug.cgi?id=758179
3633 2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
3635 * gst/rtsp-server/rtsp-stream.c:
3636 rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
3637 Adding them when not needed will start some logic inside rtpbin that might be
3638 problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
3639 would start up a rtpjitterbuffer and behave in weird ways.
3640 We still set up the UDP sources for RTP receiving for a sender media to be
3641 able to receive any packets sent by the client for NAT traversal. They will
3642 all go to a fakesink though.
3643 Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
3644 NO_PREROLL, which will cause deadlocks when seeking the media as it will never
3645 receive ASYNC_DONE after a seek.
3646 https://bugzilla.gnome.org/show_bug.cgi?id=758319
3648 2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
3650 * gst/rtsp-server/rtsp-stream.c:
3651 rtsp-stream: Disable multicast loopback for the multicast udp sources too
3652 On POSIX this setting is for sender sockets, on Windows for receiver sockets.
3653 Previously we were only setting this for sender sockets, which caused looped
3654 back packets to be received on Windows if a multicast transport was used.
3656 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3658 * examples/test-record-auth.c:
3659 * examples/test-record.c:
3660 examples: Actually use the provided port in the record examples
3662 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3664 * examples/test-record-auth.c:
3665 test-record-auth: Add the option to build in TLS support
3667 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3669 * examples/test-auth.c:
3670 test-auth: Use an 'anonymous' user for unauthenticated default
3671 There's a comment on one of the resources that 'user' and 'admin'
3672 shouldn't even be able to see it, but they can if the default
3673 token is 'admin2', since that gives them access anyway.
3675 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3677 * examples/.gitignore:
3678 * examples/Makefile.am:
3679 * examples/test-record-auth.c:
3680 Add test-record-auth example
3682 2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
3684 * gst/rtsp-server/rtsp-client.c:
3685 * tests/check/gst/client.c:
3686 rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
3688 2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
3690 * gst/rtsp-server/rtsp-server.c:
3691 rtsp-server: Change the logic so we don't pop a NULL context
3692 When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
3693 will sometimes fail. This call is made before any context is pushed
3694 resulting in an attempt to pop a NULL context.
3695 https://bugzilla.gnome.org/show_bug.cgi?id=757949
3697 2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
3699 * tests/check/gst/rtspserver.c:
3700 rtspserver: Add udp-mcast transport SETUP test
3701 Refactor utility functions in the test file so they can handle
3702 more than UDP and TCP as lower transport.
3703 https://bugzilla.gnome.org/show_bug.cgi?id=756969
3705 2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
3707 * gst/rtsp-server/rtsp-stream.c:
3708 rtsp-stream: Always unref return value of gst_object_get_parent()
3709 Fixes a leak of a GstBin in the udp-mcast case.
3710 https://bugzilla.gnome.org/show_bug.cgi?id=756968
3712 2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
3715 Automatic update of common submodule
3716 From b99800a to b319909
3718 2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
3721 Use new GST_ENABLE_EXTRA_CHECKS #define
3722 https://bugzilla.gnome.org/show_bug.cgi?id=756870
3724 2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
3727 Automatic update of common submodule
3728 From 6babecd to b99800a
3730 2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
3733 Update GLib dependency to 2.40.0
3735 2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
3737 * examples/test-mp4.c:
3738 * gst/rtsp-server/rtsp-stream.c:
3739 stream: listen to sender ssrc signals
3740 https://bugzilla.gnome.org/show_bug.cgi?id=746747
3742 2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
3745 common: update for new suppression
3746 Makes check-valgrind pass with glib 2.46
3748 2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
3750 * gst/rtsp-server/rtsp-media.c:
3751 rtsp-media: Take reference to media that will be prepared
3752 default_prepare() takes a transfer-none reference GstRTSPMedia object.
3753 Later on a g_idle_source_new() is created and a pointer to the media
3754 object is passed as user data. If the media is freed before the idle
3755 source is dispatched the media object pointer is invalid, but the idle
3756 source callback expects it to still be valid. To fix this a reference to
3757 the media object is taken when registering the source callback function
3758 and a corresponding release of the reference is done when the souce is
3760 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
3762 2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
3764 * examples/test-launch.c:
3765 * examples/test-mp4.c:
3766 * examples/test-ogg.c:
3767 * examples/test-record.c:
3768 * examples/test-uri.c:
3769 rtsp-server: Fix memory leaks when context parse fails
3770 When g_option_context_parse fails, context and error variables are not getting free'd
3771 which results in memory leaks. Free'ing the same.
3772 And replacing g_error_free with g_clear_error, which checks if the error being passed
3773 is not NULL and sets the variable to NULL on free'ing.
3774 https://bugzilla.gnome.org/show_bug.cgi?id=753863
3776 2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
3781 === release 1.6.0 ===
3783 2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
3789 * gst-rtsp-server.doap:
3792 === release 1.5.91 ===
3794 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
3800 * gst-rtsp-server.doap:
3803 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
3805 * docs/libs/gst-rtsp-server-sections.txt:
3806 * gst/rtsp-server/rtsp-stream.c:
3807 stream: fix docs for recently-added get/set_buffer_size API
3808 https://bugzilla.gnome.org/show_bug.cgi?id=749095
3810 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
3812 * gst/rtsp-server/rtsp-media.c:
3813 rtsp-media: Don't crash on encrypted RTX SDP
3814 In parse_keymgmt(), don't mutate the input string that's been passed
3815 as const, especially since we might need the original value again if
3816 the same key info applies to multiple streams (RTX, for example).
3817 https://bugzilla.gnome.org/show_bug.cgi?id=754753
3819 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
3821 * examples/test-mp4.c:
3822 test-mp4: Support filenames with spaces in them. Error out on too few arguments
3824 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
3826 * examples/test-record.c:
3827 test-record: Check parameter count and print out help
3828 If no launch pipeline was supplied, print out some help
3830 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
3832 * gst/rtsp-server/rtsp-media.c:
3833 * gst/rtsp-server/rtsp-stream.c:
3834 * gst/rtsp-server/rtsp-stream.h:
3835 rtsp-stream: Implement UDP buffer size setting.
3836 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
3838 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
3839 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
3841 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
3843 * gst/rtsp-server/rtsp-media.h:
3844 rtsp-media: Fix small typo causing gtk-doc to complain
3846 === release 1.5.90 ===
3848 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
3854 * gst-rtsp-server.doap:
3857 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
3859 * gst/rtsp-server/rtsp-media-factory.c:
3860 media-factory: get port number through gst_rtsp_url_get_port
3861 https://bugzilla.gnome.org/show_bug.cgi?id=753473
3863 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
3865 * tests/check/gst/media.c:
3866 media-test: Removing unnecessary assertion
3867 https://bugzilla.gnome.org/show_bug.cgi?id=753385
3869 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
3871 * gst/rtsp-server/rtsp-server.c:
3872 Document that source keeps a ref on server until it's destroyed
3873 https://bugzilla.gnome.org/show_bug.cgi?id=749227
3875 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
3877 * tests/check/gst/media.c:
3878 media-test: Test for multiple dynamic payload
3879 https://bugzilla.gnome.org/show_bug.cgi?id=753385
3881 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
3883 * gst/rtsp-server/rtsp-media.c:
3884 media: Only add fakesink once per pipeline
3885 The intention is to prevent going PLAYING state before pads are created.
3886 If there was mutilple dynamic payload, it would leak few fakesink and
3887 actually prevent from ever reaching playing state.
3888 https://bugzilla.gnome.org/show_bug.cgi?id=753385
3890 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
3892 * gst/rtsp-server/rtsp-media.c:
3893 Revert "rtsp-media: Only add 1 fakesink per pipeline"
3894 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
3896 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
3898 * gst/rtsp-server/rtsp-media.c:
3899 rtsp-media: Only add 1 fakesink per pipeline
3900 There should be only one fakesink per pipeline, not per dynpay. This
3901 would lead to element naming clash.
3903 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
3905 * gst/rtsp-server/rtsp-media.c:
3906 rtsp-media: assertion error due to wrong condition check
3907 In media to caps function, reserved_keys array is being used for variable i,
3908 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
3909 changed it to variable j
3910 https://bugzilla.gnome.org/show_bug.cgi?id=753009
3912 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
3914 * gst/rtsp-server/rtsp-media.c:
3915 rtsp-media: Strip keys from the fmtp that we use internally in our caps
3916 Skip keys from the fmtp, which we already use ourselves for the
3917 caps. Some software is adding random things like clock-rate into
3918 the fmtp, and we would otherwise here set a string-typed clock-rate
3919 in the caps... and thus fail to create valid RTP caps
3920 https://bugzilla.gnome.org/show_bug.cgi?id=753009
3922 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
3924 * gst/rtsp-server/rtsp-thread-pool.c:
3925 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
3926 https://bugzilla.gnome.org/show_bug.cgi?id=752640
3928 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
3931 Automatic update of common submodule
3932 From f74b2df to 9aed1d7
3934 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
3939 === release 1.5.2 ===
3941 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
3947 * gst-rtsp-server.doap:
3950 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
3952 * gst/rtsp-server/rtsp-client.c:
3953 * gst/rtsp-server/rtsp-client.h:
3954 * tests/check/gst/client.c:
3955 rtsp-client: allow application to decide what requirements are supported
3956 Add "check-requirements" signal and vfunc to allow application
3957 (and subclasses) to check the requirements.
3958 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
3959 https://bugzilla.gnome.org/show_bug.cgi?id=749417
3961 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
3964 Automatic update of common submodule
3965 From 6015d26 to f74b2df
3967 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
3969 * gst/rtsp-server/rtsp-media.c:
3970 rtsp-media: Always use real payloader when creating streams
3971 A bin that contains the real payloader might be used as payloader. In this
3972 case we have to get the real payloader for the various properties it provides.
3973 Example use cases for this are bins that payload some media and then have
3974 additional elements that add metadata or RTP extension headers to the stream.
3975 https://bugzilla.gnome.org/show_bug.cgi?id=750800
3977 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
3979 * examples/test-netclock-client.c:
3980 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
3982 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
3984 * examples/test-netclock-client.c:
3985 * examples/test-netclock.c:
3986 test-netclock: Use new ntp-time-source property on rtpbin
3987 Select the clock time to be used as NTP time source. This allows proper
3988 synchronization between receivers, independent of sharing base times, and just
3989 requires them to use the same clock.
3991 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
3993 * examples/test-netclock-client.c:
3994 * examples/test-netclock.c:
3995 test-netclock: Setting the same base time on sender and receiver is not necessary
3996 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
3998 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
4000 * gst/rtsp-server/rtsp-stream.c:
4001 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
4002 https://bugzilla.gnome.org/show_bug.cgi?id=750764
4004 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
4006 * docs/libs/gst-rtsp-server.types:
4007 docs: add missing types
4008 https://bugzilla.gnome.org/show_bug.cgi?id=750764
4010 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
4012 * docs/libs/gst-rtsp-server-sections.txt:
4013 docs: add missing apis
4014 https://bugzilla.gnome.org/show_bug.cgi?id=750764
4016 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
4018 * examples/test-netclock-client.c:
4019 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
4021 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
4023 * docs/libs/gst-rtsp-server-sections.txt:
4024 * gst/rtsp-server/rtsp-auth.c:
4025 * gst/rtsp-server/rtsp-auth.h:
4026 GstRTSPAuth: Add client certificate authentication support
4027 https://bugzilla.gnome.org/show_bug.cgi?id=750471
4029 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
4031 * examples/test-netclock-client.c:
4032 test-netclock-client: Use new GstClock API to wait for clock synchronization
4034 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
4036 * examples/test-netclock-client.c:
4037 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
4038 A mainloop is needed to get glimagesink to display something on OSX, and
4039 the source-setup signal just makes things a little bit easier.
4041 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
4044 Automatic update of common submodule
4045 From d9a3353 to 6015d26
4047 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
4050 Automatic update of common submodule
4051 From d37af32 to d9a3353
4053 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
4056 Automatic update of common submodule
4057 From 21ba2e5 to d37af32
4059 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
4062 Automatic update of common submodule
4063 From c408583 to 21ba2e5
4065 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
4067 * docs/libs/Makefile.am:
4068 docs: remove variables that we define in the snippet from common
4069 This is syncing our Makefile.am with upstream gtkdoc.
4071 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
4074 Automatic update of common submodule
4075 From 44a3517 to c408583
4077 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
4082 === release 1.5.1 ===
4084 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
4090 * gst-rtsp-server.doap:
4093 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
4095 * gst/rtsp-server/rtsp-client.c:
4096 rtsp-client: No flush during Teardown.
4097 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
4098 backlog is empty it can happen that just a part of a message will be
4099 sent and rest is in backlog queue. If then flush during teardown
4100 just a part of message will be sent.This can lead to client miss
4101 teardown response since it expect to get the last part of message.
4102 The flushing during teardown was introduced to fix a deadlock that now
4103 is fixed more generally in handle_request by temporary setting backlog
4105 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
4107 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
4109 * tests/check/Makefile.am:
4110 tests: Use AM_TESTS_ENVIRONMENT
4111 Needed by the new automake test runner and the
4112 current version of the common submodule.
4114 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
4116 * gst/rtsp-server/rtsp-media.h:
4117 * gst/rtsp-server/rtsp-stream.h:
4118 rtsp-server: Use single-include rtsp header to make sure we get all definitions
4120 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
4122 * gst/rtsp-server/rtsp-media.c:
4123 rtsp-media: Mark some more functions static
4125 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
4127 * gst/rtsp-server/rtsp-media.c:
4128 rtsp-media: Only unblock the media in suspend() when actually changing the state
4129 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
4131 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
4133 * examples/test-video-rtx.c:
4134 examples: Use AVPF profile for the RTX example
4136 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
4138 * gst/rtsp-server/rtsp-sdp.c:
4139 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
4141 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
4143 * gst/rtsp-server/rtsp-stream.c:
4144 rtsp-stream: get valid clock-rate from last-sample
4145 clock-rate in last-sample's caps is integer, not unsigned.
4146 To get this value properly, variable needs to be type-casted to int.
4147 https://bugzilla.gnome.org/show_bug.cgi?id=747614
4149 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
4153 autogen.sh: only run autopoint if gettext requested in configure.ac
4154 Not just because there happens to be a po directory.
4155 https://bugzilla.gnome.org/show_bug.cgi?id=748058
4157 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
4160 Revert "configure.ac: uncomment gettext version setup"
4161 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
4162 We don't need a gettext setup here and there's no po
4163 directory either, so no reason why autopoint would be
4164 run in the first place.
4165 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
4167 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
4169 * examples/test-multicast.c:
4170 * examples/test-multicast2.c:
4171 * examples/test-sdp.c:
4172 * examples/test-video-rtx.c:
4173 * examples/test-video.c:
4174 * tests/test-cleanup.c:
4175 * tests/test-reuse.c:
4176 Fix timeout function signatures across tests and examples
4178 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
4180 * tests/check/Makefile.am:
4181 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
4182 Make sure the test environment is set up.
4183 https://bugzilla.gnome.org//show_bug.cgi?id=747624
4185 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
4188 configure: bump automake requirement to 1.14 and autoconf to 2.69
4189 This is only required for builds from git, people can still
4190 build tarballs if they only have older autotools.
4191 https://bugzilla.gnome.org//show_bug.cgi?id=747624
4193 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
4196 configure.ac: uncomment gettext version setup
4197 Fixes autogen.sh. It would run autopoint, which would complain
4198 that it could not find the gettext version in configure.ac.
4199 https://bugzilla.gnome.org/show_bug.cgi?id=748058
4201 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
4203 * examples/test-video-rtx.c:
4204 test-video-rtx: set exact payload type to PCMA payloader
4205 Setting wrong payload type causes failure to do retransmission through audio stream
4206 https://bugzilla.gnome.org/show_bug.cgi?id=747839
4208 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
4210 * gst/rtsp-server/rtsp-media.c:
4211 * gst/rtsp-server/rtsp-stream.c:
4212 * gst/rtsp-server/rtsp-stream.h:
4213 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
4214 Because of duplicated g_signal_connect for request-aux-sender signal,
4215 wrong stream pointer is passed to the signal handler.
4216 Instead of passing each stream, pass stream array and get the relevant stream.
4217 https://bugzilla.gnome.org/show_bug.cgi?id=747839
4219 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
4223 Update autogen.sh to latest version from common
4224 Fixes build after aclocal_check etc. helpers have been removed.
4226 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
4229 Automatic update of common submodule
4230 From bc76a8b to c8fb372
4232 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
4234 * gst/rtsp-server/rtsp-stream.c:
4235 rtsp-stream: Limit the queues to 1 buffer
4236 We only need them to be able to pre-roll, queueing up more data here
4237 is only going to harm latency and memory usage.
4239 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
4241 * gst/rtsp-server/rtsp-stream.c:
4242 rtsp-stream: Update comment and ASCII art to the latest code
4243 We have a queue in front of the udpsink too to prevent the pipeline from
4246 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4248 * gst/rtsp-server/rtsp-stream.c:
4249 rtsp-media: Properly return first rtptime
4250 Instead we where returning first GstBuffer timestamp. This would result
4251 in clock skew and unwanted behaviour in RTSP playback.
4252 https://bugzilla.gnome.org/show_bug.cgi?id=746479
4254 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
4256 * gst/rtsp-server/rtsp-stream.c:
4257 rtsp-stream: Don't leave buffer mapped
4258 If the seq is NULL, the RTP buffer was left mapped. We should always
4261 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
4266 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
4268 * gst/rtsp-server/rtsp-media-factory.c:
4269 * tests/check/gst/client.c:
4270 Fix double semicolons
4272 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
4274 * gst/rtsp-server/rtsp-stream.c:
4275 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
4276 This gives more accurate values than asking the payloader. There might be
4277 queueing happening between the payloader and the sink.
4278 https://bugzilla.gnome.org/show_bug.cgi?id=745704
4280 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
4282 * gst/rtsp-server/rtsp-media.c:
4283 rtsp-media: Don't seek for PLAY if the position will not change
4284 https://bugzilla.gnome.org/show_bug.cgi?id=745704
4286 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
4288 * gst/rtsp-server/rtsp-media.c:
4289 rtsp-media: Don't include payload type in the caps for framesize
4290 When the sdp media attribute framesize are converted to caps
4291 the <payload> should not be included.
4292 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
4293 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
4295 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
4297 * gst/rtsp-server/rtsp-sdp.c:
4298 rtsp-sdp: add payload type to the sdp framesize attribute
4299 The sdp framesize attribute is desribed in RFC6064. It is specified
4300 for payloading of H263 and has the following form
4301 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
4302 should be added to the caps in a payloader and the <payload type> should
4303 be added by the rtsp-server.
4304 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
4306 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
4308 * examples/test-uri.c:
4309 examples: test-uri: fix tainted variable
4310 Insignificant but this keeps Coverity happy.
4313 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
4315 * examples/.gitignore:
4316 * examples/Makefile.am:
4317 * examples/test-netclock-client.c:
4318 * examples/test-netclock.c:
4319 examples: Add a simple example of network synch for live streams.
4320 An example server and client that works for synchronising live streams
4321 only - as it can't support pause/play.
4323 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
4325 * gst/rtsp-server/rtsp-media-factory.c:
4326 * gst/rtsp-server/rtsp-media-factory.h:
4327 rtsp-media-factory: Add functions to set/get the media gtype
4328 Allow specifying the GType of a GstRtspMedia subclass to create
4329 as a simpler way to get the factory to create a custom
4330 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
4332 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
4334 * gst/rtsp-server/rtsp-media.c:
4335 rtsp-media: fix double unlock in _get_buffer_size()
4336 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
4337 because of double g_mutex_unlock () usage.
4338 https://bugzilla.gnome.org/show_bug.cgi?id=745434
4340 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
4342 * gst/rtsp-server/rtsp-session-pool.c:
4343 * gst/rtsp-server/rtsp-session.c:
4344 * gst/rtsp-server/rtsp-session.h:
4345 rtsp-session: Use monotonic time for RTSP session timeout
4346 Changed RTSP session timeout handling to monotonic time
4347 and deprecating the API for current system time.
4348 This fixes timeouts when the system time changes.
4349 https://bugzilla.gnome.org/show_bug.cgi?id=743346
4351 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
4353 * gst/rtsp-server/rtsp-client.c:
4354 * gst/rtsp-server/rtsp-media.c:
4355 rtsp-client: Only error out in PLAY if seeking actually failed
4356 If the media was just not seekable, we continue from whatever position we are
4357 and let the client decide if that is what is wanted or not.
4358 Only if the actual seek failed, we can't really recover and should error out.
4360 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
4362 * gst/rtsp-server/rtsp-stream.c:
4363 rtsp-stream: Add necessary queues between tee and multiudpsink
4364 https://bugzilla.gnome.org/show_bug.cgi?id=744379
4366 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
4368 * gst/rtsp-server/rtsp-client.c:
4369 * gst/rtsp-server/rtsp-media.c:
4370 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
4371 Instead error out properly the same way as if the SEEKING query already
4374 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
4376 * gst/rtsp-server/rtsp-stream.h:
4377 rtsp-stream: minor code formatting fix
4379 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
4381 * gst/rtsp-server/rtsp-media.c:
4382 rtsp-media: fix logic for collect_streams
4383 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
4384 all streams it knows if it got any, and can check if the transport mode is OK.
4387 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
4389 * gst/rtsp-server/rtsp-media.c:
4390 rtsp-media: Don't set the transport mode based on what elements we find
4391 Just print a warning if the one that was set before disagrees with what
4392 elements we found. It must already be set to something before as this
4393 function is called after we received the SDP from ANNOUNCE in RECORD mode,
4394 and we would reject ANNOUNCE if the RECORD flag was not set.
4396 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
4398 * tests/check/gst/rtspserver.c:
4399 tests: rtspserver: rename shadowed variable
4400 We have two different 'sink' variables here,
4401 rename one of them for clarity.
4403 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
4405 * gst/rtsp-server/rtsp-client.c:
4406 rtsp-client: fix awkward if clause
4408 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
4410 * examples/test-uri.c:
4411 examples: test-uri: improve uri argument handling and accept file names
4412 Print an error if the argument passed is not a URI and can't
4413 be converted into one, or no arguments have been provided.
4415 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
4417 * examples/test-uri.c:
4418 examples: test-uri: don't remove mount point after 10 seconds
4419 It's very irritating when trying to test stuff repeatedly
4420 and serves no real purpose other than showing that it can
4423 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
4425 * examples/.gitignore:
4426 examples: add new test-record to .gitignore
4428 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
4430 * examples/test-record.c:
4431 * gst/rtsp-server/rtsp-client.c:
4432 * gst/rtsp-server/rtsp-media-factory.c:
4433 * gst/rtsp-server/rtsp-media-factory.h:
4434 * gst/rtsp-server/rtsp-media.c:
4435 * gst/rtsp-server/rtsp-media.h:
4436 * tests/check/gst/rtspserver.c:
4437 rtsp-media: Use flags to distinguish between PLAY and RECORD media
4439 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
4441 * examples/test-record.c:
4442 test-record: Set latency for playback-style example to 2s instead of 200ms
4444 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
4446 * tests/check/gst/rtspserver.c:
4447 tests: add some unit tests for ANNOUNCE and RECORD
4448 https://bugzilla.gnome.org/show_bug.cgi?id=743175
4450 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
4452 * gst/rtsp-server/rtsp-client.c:
4453 rtsp-client: fix a couple of leaks in handle_announce
4455 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
4457 * gst/rtsp-server/rtsp-media-factory.c:
4458 * gst/rtsp-server/rtsp-media-factory.h:
4459 * gst/rtsp-server/rtsp-media.c:
4460 * gst/rtsp-server/rtsp-media.h:
4461 rtsp-media: Expose latency setting for setting the rtpbin latency
4463 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
4465 * examples/test-record.c:
4466 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
4468 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
4470 * gst/rtsp-server/rtsp-stream.c:
4471 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
4473 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
4475 * examples/Makefile.am:
4476 * examples/test-record.c:
4477 * gst/rtsp-server/rtsp-client.c:
4478 * gst/rtsp-server/rtsp-client.h:
4479 * gst/rtsp-server/rtsp-media-factory.c:
4480 * gst/rtsp-server/rtsp-media-factory.h:
4481 * gst/rtsp-server/rtsp-media.c:
4482 * gst/rtsp-server/rtsp-media.h:
4483 * gst/rtsp-server/rtsp-session-media.c:
4484 * gst/rtsp-server/rtsp-stream.c:
4485 * gst/rtsp-server/rtsp-stream.h:
4486 Add initial support for RECORD
4487 We currently only support media that is RECORD or PLAY only, not both at once.
4488 https://bugzilla.gnome.org/show_bug.cgi?id=743175
4490 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
4492 * gst/rtsp-server/rtsp-stream.c:
4493 rtsp-stream: RTCP and RTP transport cache cookies seperated
4494 RTCP packets were not sent because the same tr_cache_cookie was used for
4495 both RTP and RTCP. So only one of the tr_cache lists were populated
4496 depending on which one was sent first. If the tr_cache list is not
4497 populated then no packets can be sent. Most often this happened to be
4498 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
4499 resulted in both the tr_cache_lists to be populated regardless of which
4501 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
4503 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
4505 * gst/rtsp-server/rtsp-stream.c:
4506 rtsp-stream: fix false compiler warning
4507 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
4509 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
4511 * gst/rtsp-server/rtsp-client.c:
4512 rtsp-client: log interleaved data received
4514 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
4516 * gst/rtsp-server/rtsp-client.c:
4517 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
4519 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
4521 * gst/rtsp-server/rtsp-client.c:
4522 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
4524 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
4526 * gst/rtsp-server/rtsp-client.c:
4527 rtsp-client: Use a random session ID in the SDP
4528 RFC4566 Section 5.2 says that it should make the username, session id,
4529 nettype, addrtype and unicast address tuple globally unique. Always using
4530 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
4531 Instead let's create a 64 bit random number, which at least brings us
4532 closer to the goal of global uniqueness.
4533 https://tools.ietf.org/html/rfc4566#section-5.2
4535 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
4537 * examples/test-launch.c:
4538 * examples/test-mp4.c:
4539 * examples/test-ogg.c:
4540 * examples/test-uri.c:
4541 examples: Don't call gst_init() and gst_get_option_group()
4542 The latter calls the former at the appropriate time.
4544 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
4546 * gst/rtsp-server/rtsp-client.c:
4547 rtsp-client: Drop trailing \0 of RTSP DATA messages
4548 We add a trailing \0 in GstRTSPConnection to make parsing of
4549 string message bodies easier (e.g. the SDP from DESCRIBE) but
4550 for actual data this means we have to drop it or otherwise
4551 create invalid data.
4553 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
4555 * gst/rtsp-server/rtsp-stream.c:
4556 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
4557 Fixes crash when two threads access handle_new_sample() at the same
4558 time, one for RTP, one for RTCP.
4559 Otherwise, when iterating over the transports cache, it might be modified by
4560 another thread at the same time if the transports cookie has changed.
4561 https://bugzilla.gnome.org/show_bug.cgi?id=742954
4563 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
4565 * gst/rtsp-server/rtsp-stream.c:
4566 rtsp-stream: Set format=TIME on our app sources for TCP
4568 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
4570 * gst/rtsp-server/rtsp-session-pool.c:
4571 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
4572 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
4573 RFC 2326 states that session IDs may consist of alphanumeric as well as
4574 the safe characters $-_.+ -- N.B. the percent character is not allowed.
4575 Previously the session ID was URI-escaped, this meant that any character
4576 which was not alphanumeric or any of the characters +-._~ would be
4577 percent encoded. While the RFC (surprisingly) mentions that linear white
4578 space in session IDs should be URI-escaped, it does not say anything
4579 about other characters. Moreover no white space is allowed in the
4580 session ID. Finally the percent character which is the result of
4581 URI-escaping is not allowed in a session ID.
4582 So there is no reason to do any URI-escaping, and now it is removed.
4583 https://bugzilla.gnome.org/show_bug.cgi?id=742869
4585 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
4588 Automatic update of common submodule
4589 From f2c6b95 to bc76a8b
4591 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
4594 Fix 'make check' from top-level directory
4596 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
4598 * examples/test-launch.c:
4599 * examples/test-mp4.c:
4600 * examples/test-ogg.c:
4601 * examples/test-uri.c:
4602 examples: Add command-line parsing and take a 'port' argument
4603 This allows users to run multiple servers on different ports for testing.
4604 Only done for examples that actually take arguments and hence are capable of
4605 outputting different streams for each instance on each port.
4606 https://bugzilla.gnome.org/show_bug.cgi?id=742115
4608 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
4610 * gst/rtsp-server/rtsp-client.c:
4611 * gst/rtsp-server/rtsp-client.h:
4612 rtsp-client: Add a send_message default signal handler
4613 This allows subclasses to easily hook into the response sending
4614 mechanism without doing everything from a signal, which seems
4615 awkward from subclasses.
4617 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
4620 Automatic update of common submodule
4621 From ef1ffdc to f2c6b95
4623 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4627 configure: add --disable-examples switch
4628 https://bugzilla.gnome.org/show_bug.cgi?id=741678
4630 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
4632 * examples/.gitignore:
4633 * examples/Makefile.am:
4634 * examples/test-video-rtx.c:
4635 examples: add a retransmisison example implementing RFC4588
4636 Currently only SSRC-multiplexed rtx streams are supported
4638 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
4640 * gst/rtsp-server/rtsp-stream.c:
4641 rtsp-stream: Fix some minor memory leaks
4643 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
4645 * gst/rtsp-server/rtsp-media.c:
4646 rtsp-media: Some minor cleanup
4648 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
4650 * gst/rtsp-server/rtsp-stream.c:
4651 rtsp-stream: Fix compiler warnings
4652 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
4653 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
4655 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
4656 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
4659 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
4661 * docs/libs/gst-rtsp-server-sections.txt:
4662 * gst/rtsp-server/rtsp-media-factory.c:
4663 * gst/rtsp-server/rtsp-media-factory.h:
4664 * gst/rtsp-server/rtsp-media.c:
4665 * gst/rtsp-server/rtsp-media.h:
4666 * gst/rtsp-server/rtsp-sdp.c:
4667 * gst/rtsp-server/rtsp-stream.c:
4668 * gst/rtsp-server/rtsp-stream.h:
4669 media: implement ssrc-multiplexed retransmission support
4670 based off RFC 4588 and the server-rtpaux example in -good
4672 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
4674 * gst/rtsp-server/rtsp-client.c:
4675 * gst/rtsp-server/rtsp-stream-transport.c:
4676 * gst/rtsp-server/rtsp-stream.c:
4677 rtsp: Ref transports in hash table.
4678 Also ref streams for transports.
4679 This solves a crash when reciving a rtcp after teardown but before
4681 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
4683 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
4686 Automatic update of common submodule
4687 From 7bb2bce to ef1ffdc
4689 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
4691 * gst/rtsp-server/rtsp-client.c:
4692 client: refactor cleanup of cached media
4694 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
4696 * tests/check/gst/client.c:
4698 The session leak is now fixed, lets remove those FIXME comments.
4700 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
4702 * tests/check/gst/rtspserver.c:
4703 tests: Test to setup two sessions on one connection
4704 https://bugzilla.gnome.org/show_bug.cgi?id=739112
4706 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
4708 * tests/check/gst/rtspserver.c:
4709 tests: Test setup with tcp transport
4710 https://bugzilla.gnome.org/show_bug.cgi?id=739112
4712 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
4714 * gst/rtsp-server/rtsp-client.c:
4715 client: Configure transport after creating session media
4716 The default implementation of configure_client_transport() in
4717 rtsp-client uses the session media when it chooses channels for
4718 interleaved traffic.
4719 https://bugzilla.gnome.org/show_bug.cgi?id=739112
4721 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
4723 * gst/rtsp-server/rtsp-client.c:
4724 * gst/rtsp-server/rtsp-session-media.c:
4725 client: Stop caching media in client when doing setup
4726 If the media has been managed by a session media, it should not be
4727 cached in the client any longer. The GstRTSPSessionMedia object is now
4728 responsible for unpreparing the GstRTSPMedia object using
4729 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
4731 https://bugzilla.gnome.org/show_bug.cgi?id=739112
4733 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4735 * gst/rtsp-server/rtsp-stream.c:
4736 rtsp-stream: unref srtp decoder when leaving bin
4737 https://bugzilla.gnome.org/show_bug.cgi?id=739481
4739 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4741 * gst/rtsp-server/rtsp-client.c:
4742 rtsp-client: mikey memory leaks
4743 https://bugzilla.gnome.org/show_bug.cgi?id=739383
4745 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
4748 Automatic update of common submodule
4749 From 84d06cd to 7bb2bce
4751 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
4754 Parallelise 'make check-valgrind'
4756 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
4759 Automatic update of common submodule
4760 From a8c8939 to 84d06cd
4762 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
4765 Automatic update of common submodule
4766 From 36388a1 to a8c8939
4768 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
4770 * gst/rtsp-server/rtsp-media.c:
4771 rtsp-media: deactivate media when shutting down from paused
4772 This was only done when going directly from playing.
4773 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
4775 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4777 * gst/rtsp-server/rtsp-client.c:
4778 * gst/rtsp-server/rtsp-context.h:
4779 rtsp-client: add stream transport to context
4780 We add the stream transport to the context so we can get the configured
4781 client stream transport in the setup request signal.
4782 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
4784 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4786 * gst/rtsp-server/rtsp-stream.c:
4787 stream: release lock even not all transports have been removed
4788 We don't want to keep the lock even we return FALSE because not all the
4789 transports have been removed. This could lead into a deadlock.
4790 https://bugzilla.gnome.org/show_bug.cgi?id=737797
4792 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
4794 * gst/rtsp-server/rtsp-sdp.c:
4795 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
4796 These were renamed in GstRTPBasePayload in 1.0
4798 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
4800 * gst/rtsp-server/rtsp-client.c:
4801 client: set session media to NULL without the lock
4802 We need to set session medias to NULL without the client lock otherwise
4803 we can end up in a deadlock if another thread is waiting for the lock
4804 and media unprepare is also waiting for that thread to end.
4805 https://bugzilla.gnome.org/show_bug.cgi?id=737690
4807 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
4809 * gst/rtsp-server/rtsp-media.c:
4810 rtsp-media: Set state to UNPREPARING in all cases
4812 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
4814 * gst/rtsp-server/rtsp-media.c:
4815 media: set state to unpreparing when unprepare is initiated
4816 https://bugzilla.gnome.org/show_bug.cgi?id=737675
4818 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
4820 * gst/rtsp-server/rtsp-client.c:
4821 rtsp-client: Remove backlog limit while processings requests
4822 If the backlog limit is kept two cases of deadlocks may be
4823 encountered when streaming over TCP. Without the backlog
4824 limit this deadlocks can not happen, at the expence of
4826 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
4828 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
4830 * gst/rtsp-server/rtsp-client.c:
4831 rtsp-client: do not free main context before rtsp watch
4832 https://bugzilla.gnome.org/show_bug.cgi?id=737110
4834 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
4836 * tests/check/gst/rtspserver.c:
4837 tests: Extend unit test timeout to accomodate for valgrind
4838 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
4840 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
4842 * gst/rtsp-server/rtsp-client.c:
4843 * gst/rtsp-server/rtsp-session.c:
4844 * gst/rtsp-server/rtsp-stream-transport.c:
4845 rtsp-*: Treat sending packets to clients as keepalive
4846 As long as gst-rtsp-server can successfully send RTP/RTCP data to
4847 clients then the client must be reading. This change makes the server
4848 timeout the connection if the client stops reading.
4849 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
4851 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
4853 * gst/rtsp-server/rtsp-client.c:
4854 rtsp-client: Allow backlog to grow while expiring session
4855 Allow the send backlog in the RTSP watch to grow to unlimited size while
4856 attempting to bring the media pipeline to NULL due to a session
4857 expiring. Without this change the appsink element cannot change state
4858 because it is blocked while rendering data in the new_sample callback.
4859 This callback will block until it has successfully put the data into the
4860 send backlog. There is a chance that the send backlog is full at this
4861 point which means that the callback may block for a long time, possibly
4862 forever. Therefore the media pipeline may also be prevented from
4863 changing state for a long time.
4864 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
4866 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
4868 * gst/rtsp-server/rtsp-client.c:
4869 rtsp-client: Make old compilers happy
4870 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
4871 Just in case that guint8 doesn't fit in a pointer. Just in case ...
4873 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
4875 * gst/rtsp-server/rtsp-client.c:
4876 client: raise the backlog limits before pausing
4877 We need to raise the backlog limits before pausing the pipeline or else
4878 the appsink might be blocking in the render method in wait_backlog() and
4879 we would deadlock waiting for paused.
4880 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
4882 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
4884 * gst/rtsp-server/rtsp-client.c:
4885 client: make define for the WATCH_BACKLOG
4886 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
4888 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
4890 * gst/rtsp-server/rtsp-client.c:
4891 client: simplify session transport handling
4892 link/unlink of the transport in a session was done to keep track of all
4893 TCP transports and to send RTP/RTCP data to the streams. We can simplify
4894 that by putting all the TCP transports in a hashtable indexed with the
4896 We also don't need to link/unlink the transports when we pause/resume
4897 the streams. The same effect is already achieved when we pause/play the
4898 media. Indeed, when we pause the media, the transport is removed from
4899 the media and the callbacks will not be called anymore.
4900 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
4902 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
4904 * gst/rtsp-server/rtsp-stream-transport.c:
4905 * gst/rtsp-server/rtsp-stream-transport.h:
4906 stream-transport: make method to handle received data
4907 Make a method to handle the data received on a channel. It sends the
4908 data to the stream of the transport on the RTP or RTCP pads based on
4911 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
4913 * examples/test-mp4.c:
4914 test: add example of dumping RTCP reports
4916 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
4918 * gst/rtsp-server/rtsp-media.c:
4919 * gst/rtsp-server/rtsp-stream.c:
4920 * gst/rtsp-server/rtsp-stream.h:
4921 rtsp-media: Make sure that sequence numbers are monotonic after pause
4922 The sequence number is not monotonic for RTP packets after pause. The
4923 reason is basepayloader generates a randon sequence number when the
4924 pipeline goes from ready to pause. With this fix generation of sequence
4925 number will be monotonic when going from pause to play request.
4926 https://bugzilla.gnome.org/show_bug.cgi?id=736017
4928 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
4930 * gst/rtsp-server/rtsp-client.c:
4931 rtsp-client: Protect saved clients watch with a mutex
4932 Fixes a crash when close() is called while merging clients
4933 in handle_tunnel(). In that case close() would destroy the
4934 watch while it is still being used in handle_tunnel().
4935 https://bugzilla.gnome.org/show_bug.cgi?id=735570
4937 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
4939 * gst/rtsp-server/rtsp-stream.c:
4940 rtsp-stream: Remove the multicast group udp sources when removing from the bin
4942 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
4944 * gst/rtsp-server/rtsp-media.c:
4945 * gst/rtsp-server/rtsp-stream.c:
4946 * gst/rtsp-server/rtsp-stream.h:
4947 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
4948 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
4949 seeking and will always continue counting the time. This leads to
4950 the NPT after a backwards seek to be something completely different
4951 to the actual seek position.
4952 https://bugzilla.gnome.org/show_bug.cgi?id=732644
4954 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
4956 * examples/test-appsrc.c:
4957 examples: fix another reference leak
4958 gst_rtsp_media_get_element() returns a new ref.
4960 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
4962 * examples/test-appsrc.c:
4963 examples: unref element after usage
4964 gst_bin_get_by_name_recurse_up() returns an element
4965 reference that must be unreffed after usage.
4966 https://bugzilla.gnome.org/show_bug.cgi?id=734546
4968 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
4970 * gst/rtsp-server/rtsp-media.c:
4971 signals: Fix copy-pasto in target-state signal offset
4973 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
4977 Makefile: Add usage of build-checks step
4978 Allows building checks without running them
4980 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
4982 * gst/rtsp-server/rtsp-stream.c:
4983 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
4984 When a UDP multicast transport is used it is expected that the server listens
4985 for RTP and RTCP packets on the multicast group with the corresponding port.
4986 Without this we will never get RTCP packets from clients in multicast mode.
4987 https://bugzilla.gnome.org/show_bug.cgi?id=732238
4989 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
4994 === release 1.4.0 ===
4996 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
5002 * gst-rtsp-server.doap:
5005 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
5007 * gst/rtsp-server/rtsp-media.h:
5008 media: correct misspelled words in description
5009 https://bugzilla.gnome.org/show_bug.cgi?id=733244
5011 === release 1.3.91 ===
5013 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
5019 * gst-rtsp-server.doap:
5022 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
5024 * docs/libs/gst-rtsp-server-sections.txt:
5027 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
5029 * gst/rtsp-server/rtsp-server.c:
5030 server: implement client REMOVE filter
5032 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
5034 * gst/rtsp-server/rtsp-client.c:
5035 * gst/rtsp-server/rtsp-client.h:
5036 client: expose _close() method
5037 Expose a previously internal close method to close the client
5040 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
5042 * gst/rtsp-server/rtsp-session-pool.c:
5043 session-pool: signal session-removed outside of the lock
5044 Release the lock before emiting the session-removed signal.
5046 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
5048 * gst/rtsp-server/rtsp-client.c:
5049 * gst/rtsp-server/rtsp-server.c:
5050 * gst/rtsp-server/rtsp-session-pool.c:
5051 * gst/rtsp-server/rtsp-session.c:
5052 * gst/rtsp-server/rtsp-stream.c:
5053 filter: Release lock in filter functions
5054 Release the object lock before calling the filter functions. We need to
5055 keep a cookie to detect when the list changed during the filter
5056 callback. We also keep a hashtable to make sure we only call the filter
5057 function once for each object in case of concurrent modification.
5058 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
5060 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
5062 * gst/rtsp-server/rtsp-client.c:
5063 client: check if watch is set in handle_teardown()
5064 The unit tests run without a watch
5066 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
5068 * tests/check/gst/client.c:
5069 client tests: send teardown to cleanup session
5071 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
5073 * tests/check/gst/rtspserver.c:
5074 server tests: send teardown to cleanup session
5076 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
5078 * gst/rtsp-server/rtsp-client.c:
5079 client: keep ref to client for the session removed handler
5080 This extra ref will be dropped when all client sessions have been
5081 removed. A session is removed when a client sends teardown, closes its
5082 endpoint of the TCP connection or the sessions expires.
5083 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
5085 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
5087 * gst/rtsp-server/rtsp-client.c:
5088 * gst/rtsp-server/rtsp-session.c:
5089 * tests/check/gst/client.c:
5090 client: manage media in session as a last step
5091 Once we manage a media in a session, we can't unmanage it anymore
5092 without destroying it. Therefore, first check everything before we
5093 manage the media, otherwise if something is wrong we have no way to
5095 If we created a new session and something went wrong, remove the session
5096 again. Fixes a leak in the unit test.
5098 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
5100 * examples/test-mp4.c:
5101 * examples/test-ogg.c:
5102 examples: print 'stream ready at url' for mp4 and ogg example
5104 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
5106 * gst/rtsp-server/rtsp-client.c:
5107 * gst/rtsp-server/rtsp-sdp.c:
5108 rtsp: fix for MIKEY api change
5110 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
5112 * gst/rtsp-server/rtsp-client.c:
5113 client: free watch context only once
5114 The watch context is freed when the source is destroyed. Avoids
5115 a CRITICAL when we try to unref the context twice.
5117 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
5119 * gst/rtsp-server/rtsp-client.c:
5122 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
5124 * gst/rtsp-server/rtsp-client.c:
5125 client: protect sessions with lock
5126 Protect the list of sessions with the lock.
5127 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
5129 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
5131 * gst/rtsp-server/rtsp-client.c:
5132 Client: keep a ref to the session
5133 Don't just keep a weak ref to the session objects but use a hard ref. We
5134 will be notified when a session is removed from the pool (expired) with
5135 the new session-removed signal.
5136 Don't automatically close the RTSP connection when all the sessions of
5137 a client are removed, a client can continue to operate and it can create
5138 a new session if it wants. If you want to remove the client from the
5139 server, you have to use gst_rtsp_server_client_filter() now.
5140 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
5141 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
5143 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
5145 * gst/rtsp-server/rtsp-session-pool.c:
5146 * gst/rtsp-server/rtsp-session-pool.h:
5147 session-pool: add session-removed signal
5148 Add a signal to be notified when a session is removed from the pool.
5150 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
5152 * gst/rtsp-server/Makefile.am:
5153 * gst/rtsp-server/rtsp-server.h:
5154 Make rtsp-server.h a single-include header, use it for G-I
5155 https://bugzilla.gnome.org/show_bug.cgi?id=732411
5157 === release 1.3.90 ===
5159 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
5165 * gst-rtsp-server.doap:
5168 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
5170 * gst/rtsp-server/rtsp-stream.c:
5171 stream: crypto can be NULL
5173 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
5175 * gst/rtsp-server/rtsp-client.c:
5176 * gst/rtsp-server/rtsp-media.c:
5177 * gst/rtsp-server/rtsp-mount-points.c:
5178 introspection: add missing allow-none annotations
5179 https://bugzilla.gnome.org/show_bug.cgi?id=730952
5181 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
5183 * gst/rtsp-server/rtsp-address-pool.c:
5184 * gst/rtsp-server/rtsp-media.c:
5185 * gst/rtsp-server/rtsp-session-media.c:
5186 * gst/rtsp-server/rtsp-session-pool.c:
5187 * gst/rtsp-server/rtsp-stream-transport.c:
5188 * gst/rtsp-server/rtsp-stream.c:
5189 * gst/rtsp-server/rtsp-token.c:
5190 introspection: add (nullable) annotations to return values
5191 https://bugzilla.gnome.org/show_bug.cgi?id=730952
5193 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
5195 * gst/rtsp-server/rtsp-client.c:
5196 * gst/rtsp-server/rtsp-stream.c:
5197 gi: improve annotations
5198 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
5200 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
5202 * gst/rtsp-server/rtsp-client.c:
5203 * gst/rtsp-server/rtsp-media-factory.c:
5204 * gst/rtsp-server/rtsp-media.c:
5205 * gst/rtsp-server/rtsp-server.c:
5206 signals: use generic marshal function
5207 Use the generic C marshal function.
5208 Use more explicit type instead of G_TYPE_POINTER
5210 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
5212 * gst/rtsp-server/rtsp-context.h:
5213 context: add type macro
5215 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
5217 * gst/rtsp-server/rtsp-client.c:
5218 * gst/rtsp-server/rtsp-sdp.c:
5219 * gst/rtsp-server/rtsp-sdp.h:
5220 sdp: hide key length defines
5221 They don't have a namespace.
5223 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
5228 === release 1.3.3 ===
5230 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
5236 * gst-rtsp-server.doap:
5239 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
5241 * gst/rtsp-server/rtsp-client.c:
5242 * gst/rtsp-server/rtsp-sdp.c:
5243 * gst/rtsp-server/rtsp-sdp.h:
5244 mikey: add different key length parameters
5245 Add encryption and authentication key length parameters to MIKEY. For
5246 the encoders, the key lengths are obtained from the cipher and auth
5247 algorithms set in the caps. For the decoders, they are obtained while
5248 parsing the key management from the client.
5249 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
5251 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
5253 * tests/check/gst/stream.c:
5254 stream tests: Make sure we get right multicast address from stream
5255 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
5257 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
5259 * gst/rtsp-server/rtsp-client.c:
5260 client: ref the context until rtsp watch is alive
5261 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
5263 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
5265 * gst/rtsp-server/rtsp-client.c:
5266 client: Destroy the rtsp watch after connection close
5268 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
5270 * gst/rtsp-server/rtsp-media.c:
5271 media: fix confusing comment
5273 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
5275 * gst/rtsp-server/rtsp-session.c:
5276 rtsp-session: Timeout in header.
5277 Adding the possbilty to always have timout in header.
5278 This is configurabe with setting "timeout-always-visible".
5279 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
5281 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
5286 === release 1.3.2 ===
5288 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
5295 * gst-rtsp-server.doap:
5298 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
5301 Automatic update of common submodule
5302 From 211fa5f to 1f5d3c3
5304 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
5306 * gst/rtsp-server/rtsp-client.c:
5307 client: store TCP ports in transport
5308 Store the TCP ports in the transport when we are doing RTSP over TCP.
5309 This way, we can easily get to the ports from the transport.
5310 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
5312 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
5314 * gst/rtsp-server/rtsp-stream.c:
5315 stream: add signals for new RTP/RTCP encoders
5316 New signals to allow the user to configure the dynamically created
5318 https://bugzilla.gnome.org/show_bug.cgi?id=730228
5320 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
5322 * gst/rtsp-server/rtsp-media.c:
5323 * gst/rtsp-server/rtsp-media.h:
5324 media: Make suspend()/unsuspend() virtual
5325 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
5327 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
5329 * gst/rtsp-server/rtsp-client.c:
5330 client: fix send-message signal marshaller
5331 Use generic marshalling for the send-message signal. It has
5332 two POINTER arguments, not just one.
5333 https://bugzilla.gnome.org/show_bug.cgi?id=729900
5335 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
5337 * tests/check/gst/media.c:
5338 tests: add and remove pads only once
5339 In this test we simulate a dynamic pad by watching the caps event.
5340 Because of renegotiation in the base payloader now, this caps is sent
5341 multiple times but we can only deal with 1 invocation, use a variable to
5342 only 'add and remove' the pad once.
5344 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
5346 * tests/check/gst/rtspserver.c:
5347 tests: add unit test for correct handling of Require headers
5348 https://bugzilla.gnome.org/show_bug.cgi?id=729426
5350 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
5352 * gst/rtsp-server/rtsp-client.c:
5353 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
5354 Servers must handle Require headers and must report a failure
5355 if they don't handle any of the Required options, see RFC 2326,
5356 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
5357 https://bugzilla.gnome.org/show_bug.cgi?id=729426
5359 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
5364 === release 1.3.1 ===
5366 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
5372 * gst-rtsp-server.doap:
5375 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
5378 Automatic update of common submodule
5379 From bcb1518 to 211fa5f
5381 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
5386 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
5388 * tests/check/gst/sessionmedia.c:
5389 tests: fix memory leak in sessionmedia unit test
5391 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
5393 * gst/rtsp-server/rtsp-client.c:
5394 client: emit a signal before sending a message
5395 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
5397 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
5399 * gst/rtsp-server/rtsp-client.c:
5400 client: pass context to send_message
5401 Pass the current context to send_message, we will need it later.
5403 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
5405 * gst/rtsp-server/rtsp-client.c:
5406 client: fix typo in comment
5408 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
5410 * gst/rtsp-server/rtsp-media.c:
5411 media: Do not stop thread twice if default_prepare() fails
5413 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
5415 * gst/rtsp-server/rtsp-client.c:
5416 client: set the watch to flushing before going to NULL
5417 First set the watch to flushing so that we unblock any current and
5418 future attempt to send data on the watch, Then set the pipeline to
5420 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
5422 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
5424 * gst/rtsp-server/rtsp-session-pool.c:
5425 * tests/check/gst/sessionpool.c:
5426 rtsp-session-pool: Fixes annotation
5427 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
5428 in the sessionpool test.
5429 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
5431 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
5433 * gst/rtsp-server/rtsp-media.c:
5434 * gst/rtsp-server/rtsp-media.h:
5435 media: make media_prepare virtual
5436 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
5438 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
5440 * gst/rtsp-server/rtsp-media.c:
5441 * tests/check/gst/media.c:
5442 media: stop the thread in more error cases
5444 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
5446 * gst/rtsp-server/rtsp-media.c:
5447 * tests/check/gst/media.c:
5448 media: allow NULL as the thread
5449 Use the default context whan passing a NULL thread.
5451 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
5453 * gst/rtsp-server/rtsp-client.c:
5454 rtsp-client: indent cleanup
5455 Coverity was moaning about unreachable code, and I think it was just
5456 confused by { being before the label. We'll see if it pops up again.
5459 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
5461 * gst/rtsp-server/rtsp-client.c:
5462 * gst/rtsp-server/rtsp-media.c:
5463 client: Add drop-backlog property
5464 When we have too many messages queued for a client (currently hardcoded
5465 to 100) we overflow and drop the messages. Add a drop-backlog property
5466 to control this behaviour. Setting this property to FALSE will retry
5467 to send the messages to the client by waiting for more room in the
5469 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
5471 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
5473 * gst/rtsp-server/rtsp-client.c:
5474 client: support for POST before GET when setting up a tunnel
5476 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
5478 * gst/rtsp-server/rtsp-client.c:
5479 client: remove watch of the second client after http tunnel setup
5480 The second client will be freed after the HTTP tunnel has been set up.
5481 Make sure it's RTSP watch is never dispatched again.
5482 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
5484 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
5486 * gst/rtsp-server/rtsp-media.c:
5487 * tests/check/gst/media.c:
5488 media: Make media_prepare() fail if port allocation fails
5489 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
5491 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
5493 * tests/check/gst/media.c:
5494 media test: cleanup the thread pool in tests
5496 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
5498 * gst/rtsp-server/rtsp-media.c:
5499 * tests/check/gst/media.c:
5500 rtsp-media: Unblock blocked streams in unprepare
5501 The streams will be blocked when a live media is prepared.
5502 The streams should be unblocked in gst_rtsp_media_unprepare.
5503 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
5505 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
5507 * gst/rtsp-server/rtsp-media.c:
5508 media: release the state lock when going to NULL
5509 Set our state to UNPREPARING and release the state-lock before
5510 setting the pipeline to the NULL state. This way, any pad-added
5511 callback will be able to take the state-lock and check that we are now
5512 unpreparing instead of deadlocking.
5513 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
5515 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
5517 * gst/rtsp-server/rtsp-media.c:
5518 media: protect status with lock
5519 Make sure we only update the status with the lock.
5521 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
5523 * gst/rtsp-server/rtsp-client.c:
5524 * gst/rtsp-server/rtsp-sdp.c:
5525 rtsp: update for MIKEY API changes
5527 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
5529 * gst/rtsp-server/rtsp-client.c:
5530 client: parse the mikey response from the client
5531 Parse the mikey response from the client and update the policy for
5534 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
5536 * gst/rtsp-server/rtsp-stream.c:
5537 * gst/rtsp-server/rtsp-stream.h:
5538 stream: add method to set crypto info
5539 Make a method to configure the crypto information of a stream.
5540 Set udpsrc in READY instead of PAUSED so that we can configure caps
5543 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
5545 * gst/rtsp-server/rtsp-client.c:
5546 client: cleanup error paths
5548 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
5550 * gst/rtsp-server/rtsp-media.c:
5553 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
5555 * examples/test-video.c:
5556 test: enable SRTP only on RTSPS
5557 We only want to enable SRTP when doing rtsp over TLS so that we can
5558 exchange the keys in a secure way.
5560 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
5562 * examples/test-video.c:
5563 test: print an error on failure
5565 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
5568 * examples/test-video.c:
5569 * gst/rtsp-server/rtsp-sdp.c:
5570 * gst/rtsp-server/rtsp-stream.c:
5571 * tests/check/Makefile.am:
5572 stream: add SRTP support
5573 Install srtp encoder and decoder elements in rtpbin
5576 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5578 * tests/check/Makefile.am:
5579 * tests/check/gst/sessionpool.c:
5580 tests: Add unit tests for sessionpool
5581 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
5583 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5585 * tests/check/gst/threadpool.c:
5586 tests: Improve code coverage of rtsp-threadpool tests
5587 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
5589 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5591 * tests/check/gst/sessionmedia.c:
5592 tests: Improve code coverage for rtsp-session-media
5593 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
5595 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5597 gobject-introspection: Add annotations to support language bindings
5598 In addition a few cosmetic changes:
5599 * Adjust the order of arguments
5600 * Fix typo: occured -> occurred
5601 * Fix indentation after Return:-clauses
5602 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
5604 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5606 * gst/rtsp-server/rtsp-stream.c:
5607 rtsp-stream: Don't mix IPv4 and IPv6 addresses
5608 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
5610 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
5612 * gst/rtsp-server/rtsp-stream.c:
5613 stream: take caps after the session manager
5614 Take the caps for the SDP after they leave the rtpbin so that we can
5615 also get the properties added by rtpbin elements.
5617 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
5619 * gst/rtsp-server/rtsp-stream.c:
5620 stream: release lock while pushing out packets
5621 Keep a cache of the transports and use this to iterate the transport
5622 while pushing packets. This allows us to release the lock early.
5623 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
5625 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
5627 * gst/rtsp-server/rtsp-client.c:
5628 * gst/rtsp-server/rtsp-client.h:
5629 rtsp-client: vmethod for modifying tunnel GET response
5630 Add a vmethod tunnel_http_response where the response to the HTTP GET
5631 for tunneled connections can be modified.
5632 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
5634 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
5636 * gst/rtsp-server/rtsp-sdp.c:
5637 sdp: make 1 media line per profile
5638 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
5639 line in the SDP for each profile. The client is then supposed to pick
5640 one of the profiles in the SETUP request. Because the m= lines have the
5641 same pt, the client also knows that only 1 option is possible.
5643 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
5645 * gst/rtsp-server/rtsp-media-factory.c:
5646 * gst/rtsp-server/rtsp-media-factory.h:
5647 * gst/rtsp-server/rtsp-media.c:
5648 factory: add profile property and pass to media and streams
5650 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
5652 * examples/test-multicast.c:
5653 * gst/rtsp-server/rtsp-sdp.c:
5654 sdp: pass multicast connection for multicast-only stream
5655 Pass the multicast address of the stream in the connection info in the
5656 SDP so that clients try a multicast connection first.
5657 Only allow multicast connections in the test-multicast example. Also
5658 increase the TTL a little.
5660 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5663 .gitignore: Ignore gcov intermediate files
5664 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
5666 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
5668 * gst/rtsp-server/rtsp-stream.c:
5669 stream: release some locks in error cases
5671 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
5673 docs: Enable and fix gtk-doc warnings
5674 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
5675 * addresspool/mediafactory: Add missing annotation colon
5676 * stream: Annotate return value
5677 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
5679 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
5682 Automatic update of common submodule
5683 From fe1672e to bcb1518
5685 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
5688 Automatic update of common submodule
5689 From 1a07da9 to fe1672e
5691 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
5693 * examples/Makefile.am:
5694 examples: use LDADD for libs instead of LDFLAGS
5696 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
5699 configure: make sure releases are in .doap file
5701 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
5703 * examples/test-cgroups.c:
5704 examples: test-cgroups: don't put code with side effects into g_assert()
5705 The g_assert() might get compiled out with the right
5706 compiler/preprocessor flags.
5708 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
5710 * examples/.gitignore:
5711 examples: add cgroup test binary to .gitignore
5713 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
5715 * examples/test-cgroups.c:
5716 examples: fix cgroup test build
5717 Fixes build failure caused by compiler warning:
5718 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
5720 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
5723 .gitignore: ignore temp files created in the course of 'make check'
5725 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
5727 * gst/rtsp-server/rtsp-media.c:
5728 rtsp-media: don't loose frames handling new PLAY request
5729 If client supplied a range check if the range specifies the start point.
5730 If not, then do an accurate seek to the current position. If a start
5731 point was specified do do a key unit seek to make sure the streaming
5732 starts with decodeable frames.
5733 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
5735 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
5737 * gst/rtsp-server/rtsp-media.c:
5738 Revert "media: only flush when setting a new start position"
5739 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
5740 We need to do the flush in all cases, demuxer block currently for
5743 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
5745 * gst/rtsp-server/rtsp-media.c:
5746 media: only flush when setting a new start position
5747 Only flush the pipeline when we change the start position with
5749 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
5751 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
5753 * gst/rtsp-server/rtsp-stream.c:
5754 stream: set ttl-mc before adding the socket
5755 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
5756 never be set on socket.
5757 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
5759 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
5761 * gst/rtsp-server/rtsp-media.c:
5762 media: stop thread if media is already prepared
5763 in gst_rtsp_media_prepare() the thread is not used if media is already
5764 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
5766 https://bugzilla.gnome.org/show_bug.cgi?id=724182
5768 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
5771 build: Ship gst-rtsp-server.doap file
5773 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
5775 * tests/check/gst/rtspserver.c:
5776 tests: Fix another compiler warning with gcc
5778 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
5780 * gst/rtsp-server/rtsp-client.c:
5781 * gst/rtsp-server/rtsp-mount-points.c:
5782 * gst/rtsp-server/rtsp-stream.c:
5783 * tests/check/gst/client.c:
5784 rtsp-server: Fix lots of compiler warnings with clang
5786 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
5789 * gst-rtsp-server.doap:
5790 * tests/Makefile.am:
5791 configure: Synchronise with the configure scripts of the other modules
5793 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
5796 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
5798 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
5800 * gst/rtsp-server/rtsp-media.c:
5801 * gst/rtsp-server/rtsp-stream.c:
5802 Revert "rtsp-server: support build against last stable release"
5803 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
5804 Let us require 1.2.3 now, which is going to be released in a few
5807 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
5809 * gst/rtsp-server/rtsp-session-media.c:
5810 * gst/rtsp-server/rtsp-stream-transport.c:
5811 session: improve RTP-Info
5812 Ignore streams that can't generate RTP-Info instead of failing.
5813 Don't return the empty string when all streams are unconfigured but
5814 return NULL so that we don't generate and empty RTP-Info header.
5815 Improve docs a little.
5817 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
5819 * gst/rtsp-server/rtsp-session-media.c:
5820 Don't free rtpinfo GString when it is NULL
5821 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
5823 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
5825 * gst/rtsp-server/rtsp-media.c:
5826 media: only set keyframe flag when modifying start
5827 Only set the keyframe flag when we modify the start position. The
5828 keyframe flag should probably be ignored when no change is requested but
5829 until we can claim this is all documented properly and all demuxer
5830 implement this, avoid setting the flag.
5831 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
5833 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
5835 * gst/rtsp-server/rtsp-thread-pool.c:
5836 thread-pool: Unref source after mainloop has quit to avoid races in GLib
5837 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
5839 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
5841 * gst/rtsp-server/rtsp-stream.c:
5842 stream: handle NULL seqnum and rtptime arguments
5844 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
5846 * gst/rtsp-server/rtsp-thread-pool.c:
5847 * tests/check/gst/threadpool.c:
5848 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
5849 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
5851 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
5853 * gst/rtsp-server/rtsp-stream.c:
5854 stream: add fallback for missing stats property
5855 Use a fallback when the payloader does not have a stats property
5856 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
5858 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
5861 Automatic update of common submodule
5862 From f7bc1c3 to 1a07da9
5864 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
5866 * gst/rtsp-server/rtsp-stream.c:
5867 stream: don't leak stats structure
5868 Don't leak the stats structure and deal with NULL stats.
5870 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
5872 * gst/rtsp-server/rtsp-stream.c:
5873 stream: Get rtpinfo properties atomically from payloader
5874 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
5876 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
5878 * gst/rtsp-server/rtsp-media.c:
5879 media: refactor state change functions and signals
5880 Make functions to set the target state and the pipeline state and emit
5881 the signals from those functions.
5883 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
5885 * gst/rtsp-server/rtsp-media.c:
5886 * gst/rtsp-server/rtsp-media.h:
5887 media: add signal to notify of pending state changes
5889 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
5891 * gst/rtsp-server/rtsp-media.c:
5892 * gst/rtsp-server/rtsp-stream.c:
5893 rtsp-server: support build against last stable release
5894 Until 1.2.3 is out with the new get_type function and we
5897 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
5899 * gst/rtsp-server/rtsp-stream.c:
5900 stream: fix compilation
5902 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
5904 * gst/rtsp-server/rtsp-media.c:
5905 * gst/rtsp-server/rtsp-media.h:
5906 * gst/rtsp-server/rtsp-stream.c:
5907 * gst/rtsp-server/rtsp-stream.h:
5908 stream: add property to configure profiles
5910 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
5912 * gst/rtsp-server/rtsp-client.c:
5913 client: let stream check supported transport
5914 Delegate the check if a transport is allowed to the stream.
5915 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
5917 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
5919 * gst/rtsp-server/rtsp-stream.c:
5920 * gst/rtsp-server/rtsp-stream.h:
5921 stream: add method to check supported transport
5922 Add a method to check if a transport is supported
5924 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
5927 configure.ac: Only check for gstreamer-check, not check
5928 We include check in gstreamer-check since quite some time now.
5930 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
5932 * gst/rtsp-server/rtsp-session-media.c:
5933 * gst/rtsp-server/rtsp-stream-transport.c:
5934 * gst/rtsp-server/rtsp-stream.c:
5935 * gst/rtsp-server/rtsp-stream.h:
5936 stream: return clock-rate from get_rtpinfo
5937 And use it to correct the rtptime to the requested start-time.
5938 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
5940 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
5942 * gst/rtsp-server/rtsp-session-media.c:
5943 * gst/rtsp-server/rtsp-stream-transport.c:
5944 * gst/rtsp-server/rtsp-stream-transport.h:
5945 session-media: calculate start-time
5947 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
5949 * gst/rtsp-server/rtsp-stream-transport.c:
5950 * gst/rtsp-server/rtsp-stream.c:
5951 * gst/rtsp-server/rtsp-stream.h:
5952 stream: also return the running-time
5953 Return the running-time in the rtpinfo as well.
5955 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
5957 * gst/rtsp-server/rtsp-client.c:
5958 * gst/rtsp-server/rtsp-session-media.c:
5959 * gst/rtsp-server/rtsp-session-media.h:
5960 * gst/rtsp-server/rtsp-stream-transport.c:
5961 * gst/rtsp-server/rtsp-stream-transport.h:
5962 session-media: let the session-media make the RTPInfo
5963 Add method to create the RTPInfo for a stream-transport.
5964 Add method to create the RTPInfo for all stream-transports in a
5966 Use the session-media RTPInfo code in client. This allows us to refactor
5967 another method to link the TCP callbacks.
5969 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
5971 mount-points: sort sequence before g_sequence_lookup
5972 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
5973 sort sequence if dirty, otherwise lookup will fail.
5974 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
5976 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
5979 configure: rename package from gst-rtsp to gst-rtsp-server
5980 To match git module name and avoid confusion with the
5981 rtsp lib in gst-plugins-base and rtsp plugin in -good.
5983 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
5986 configure: bump core/base/good requirement to 1.2.0
5987 Bump to released stable version and make implicit
5988 requirements explicit.
5990 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
5995 Fix broken gettext setup which is not used anyway
5997 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
6000 Automatic update of common submodule
6001 From dbedaa0 to d48bed3
6003 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
6005 * gst/rtsp-server/rtsp-client.c:
6006 * gst/rtsp-server/rtsp-media.c:
6007 * gst/rtsp-server/rtsp-media.h:
6008 media: add setup_sdp vmethod
6009 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
6010 gst_rtsp_media_setup_sdp.
6011 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
6013 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
6015 * gst/rtsp-server/rtsp-stream.c:
6016 rtsp-stream: Check return value of sscanf
6017 streamid is only valid if sscanf matched something.
6019 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
6021 * gst/rtsp-server/rtsp-client.c:
6022 rtsp-client: Fix iteration
6023 Wouldn't even enter the code block otherwise (i++ was used as the check
6024 and not the postfix).
6026 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
6028 * gst/rtsp-server/rtsp-client.c:
6029 * gst/rtsp-server/rtsp-client.h:
6030 client: add vmethod to configure media and streams
6031 Implement a vmethod that can be used to configure the media and the
6032 streams based on the current context. Handle the blocksize handling in
6033 the default handler.
6034 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
6036 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
6039 Make git ignore more unit test binaries
6041 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
6043 * gst/rtsp-server/rtsp-address-pool.h:
6044 * gst/rtsp-server/rtsp-auth.h:
6045 * gst/rtsp-server/rtsp-client.h:
6046 * gst/rtsp-server/rtsp-context.h:
6047 * gst/rtsp-server/rtsp-media-factory-uri.h:
6048 * gst/rtsp-server/rtsp-media-factory.h:
6049 * gst/rtsp-server/rtsp-media.h:
6050 * gst/rtsp-server/rtsp-mount-points.h:
6051 * gst/rtsp-server/rtsp-server.h:
6052 * gst/rtsp-server/rtsp-session-media.h:
6053 * gst/rtsp-server/rtsp-session-pool.h:
6054 * gst/rtsp-server/rtsp-session.h:
6055 * gst/rtsp-server/rtsp-stream-transport.h:
6056 * gst/rtsp-server/rtsp-stream.h:
6057 * gst/rtsp-server/rtsp-thread-pool.h:
6058 * gst/rtsp-server/rtsp-token.h:
6059 rtsp-server: add padding to many public structures
6060 Not mini objects though, since they are not subclassable
6061 anyway, nor kept on the stack or inlined in a structure.
6063 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
6065 media: add new create_rtpbin vmethod
6066 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
6067 https://bugzilla.gnome.org/show_bug.cgi?id=719734
6069 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
6071 * tests/check/gst/media.c:
6072 tests: fix memory leak, free test's thread pool
6073 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
6075 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
6077 * gst/rtsp-server/rtsp-stream-transport.c:
6078 stream-transport: free url in finalize
6080 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
6082 * gst/rtsp-server/rtsp-media.c:
6083 media: also do state change in suspended state
6085 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
6087 * gst/rtsp-server/rtsp-client.c:
6088 * gst/rtsp-server/rtsp-media.c:
6089 media: also handle prepare and range in suspended state
6090 When we are suspended, we are already prepared.
6091 We can get the range in the suspended state.
6093 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
6095 * tests/check/Makefile.am:
6096 * tests/check/gst/sessionmedia.c:
6097 check: add test for uri in setup
6098 Added unit tests for the new functionality in GstRTSPStreamTransport.
6099 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
6101 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
6103 * gst/rtsp-server/rtsp-client.c:
6104 client: store setup uri and use in PLAY response
6105 Store the uri used when doing the setup and use that in the PLAY
6107 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
6109 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
6111 * gst/rtsp-server/rtsp-stream-transport.c:
6112 * gst/rtsp-server/rtsp-stream-transport.h:
6113 stream-transport: add method to get/set url
6115 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
6117 * gst/rtsp-server/rtsp-client.c:
6118 client: suspend after SDP and unsuspend before PLAYING
6119 Based on patches by Ognyan Tonchev <ognyan@axis.com>
6120 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
6122 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
6124 * gst/rtsp-server/rtsp-media-factory.c:
6125 * gst/rtsp-server/rtsp-media-factory.h:
6126 * gst/rtsp-server/rtsp-media.c:
6127 * gst/rtsp-server/rtsp-media.h:
6128 * gst/rtsp-server/rtsp-session-media.c:
6129 * gst/rtsp-server/rtsp-session.c:
6130 * tests/check/gst/media.c:
6131 * tests/check/gst/mediafactory.c:
6132 media: add suspend modes
6133 Add support for different suspend modes. The stream is suspended right after
6134 producing the SDP and after PAUSE. Different suspend modes are available that
6135 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
6136 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
6137 state and RESET will bring the pipeline to the NULL state.
6138 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
6139 this means that the pipeline needs to be prerolled again.
6140 Base on patches by Ognyan Tonchev <ognyan@axis.com>
6141 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
6143 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
6145 * gst/rtsp-server/rtsp-media.c:
6146 media: start live streams in blocked state
6147 Start live streams in the blocked state and make them preroll using the
6148 messages. This ensure that no data is played by the sink until we explicitly
6149 unblock the stream right before going to PLAYING.
6150 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
6152 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
6154 * gst/rtsp-server/rtsp-media.c:
6155 media: refactor starting and waiting for preroll
6156 Based on patches from Ognyan Tonchev <ognyan@axis.com>
6157 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
6159 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
6161 * gst/rtsp-server/rtsp-stream.c:
6162 * gst/rtsp-server/rtsp-stream.h:
6163 stream: add API to block streams
6164 Add an API to block on the streams and make it post a message.
6165 Based on patch by Ognyan Tonchev <ognyan@axis.com>
6166 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
6168 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
6170 * docs/libs/Makefile.am:
6171 docs: Specify the override file
6172 Even if it's empty (for now) it avoids make distcheck complaining
6174 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
6176 * gst/rtsp-server/rtsp-media.c:
6177 media: move default implementations to where they are used
6179 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
6181 * gst/rtsp-server/rtsp-media.c:
6182 media: take the right lock in gst_rtsp_media_set_pipeline_state()
6183 We need to take the state_lock when calling this method.
6185 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
6187 * gst/rtsp-server/rtsp-media.c:
6188 media: handle add-added on non-bins too
6189 Handle dynamic payloaders that are not bins, as used in the unit-test.
6191 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
6193 * gst/rtsp-server/rtsp-media-factory.c:
6194 * gst/rtsp-server/rtsp-media-factory.h:
6195 * gst/rtsp-server/rtsp-media.c:
6196 rtsp-media/-factory: Fix request pad name comments
6197 These must be escaped for gtk-doc to parse the comments without warnings.
6199 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
6201 rtsp-media: remove transports if media is in error status
6202 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
6203 trying to change to GST_STATE_NULL and media is in error status, we
6204 remove all transports.
6205 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
6207 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
6209 * gst/rtsp-server/rtsp-media.c:
6210 rtsp-media: use element metadata to find payloader
6211 Use the element metadata to find the payloader instead of checking
6213 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
6215 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
6217 rtsp-stream: add getter for payload type
6218 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
6219 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
6220 element and create the stream with this one instead of the dynpay%d
6222 https://bugzilla.gnome.org/show_bug.cgi?id=712396
6224 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
6226 * gst/rtsp-server/rtsp-client.c:
6227 * gst/rtsp-server/rtsp-context.h:
6228 * gst/rtsp-server/rtsp-media.c:
6229 * gst/rtsp-server/rtsp-mount-points.c:
6230 * gst/rtsp-server/rtsp-server.c:
6231 * gst/rtsp-server/rtsp-token.c:
6232 rtsp-*: Refer to NULL as a constant in comments
6234 https://bugzilla.gnome.org/show_bug.cgi?id=714988
6236 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
6238 rtsp-*: Fix type name typos in comments
6239 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
6240 * rtsp-auth: Refer to part of constant name as text
6241 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
6242 * rtsp-session-media: Fix GstRTSPSessionMedia typo
6243 * rtsp-stream: Fix typo when refering to GstBin
6244 https://bugzilla.gnome.org/show_bug.cgi?id=714988
6246 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
6249 * docs/libs/gst-rtsp-server-docs.sgml:
6250 * docs/libs/gst-rtsp-server-sections.txt:
6251 docs: Improve documentation
6252 * Include annotation-glossary to quiet gtk-doc
6253 * Rename remaining ClientState -> Context
6254 * Rename object hierarchy file
6255 * Remove stale chapter references
6256 * Add missing function and object references
6257 * Include missing GstRTSPAddressPoolResult
6258 https://bugzilla.gnome.org/show_bug.cgi?id=714988
6260 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
6262 * gst/rtsp-server/rtsp-client.c:
6263 * gst/rtsp-server/rtsp-server.c:
6264 * gst/rtsp-server/rtsp-session-pool.c:
6265 * gst/rtsp-server/rtsp-session.c:
6266 * gst/rtsp-server/rtsp-stream.c:
6267 rtsp-server: sprinkle some allow-none annotations for g-i
6269 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
6271 * gst/rtsp-server/rtsp-stream.c:
6272 * gst/rtsp-server/rtsp-stream.h:
6273 stream: add method to filter transports
6274 Add a method to safely iterate and collect the stream transports
6275 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
6277 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
6279 * gst/rtsp-server/rtsp-client.c:
6280 * gst/rtsp-server/rtsp-server.c:
6281 * gst/rtsp-server/rtsp-session-pool.c:
6282 * gst/rtsp-server/rtsp-session.c:
6283 rtsp: allow NULL func in filters
6284 Passing a null function make the filters return a list of
6287 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
6289 * gst/rtsp-server/rtsp-address-pool.c:
6290 * tests/check/gst/addresspool.c:
6291 address-pool: fix address increment
6292 Use a guint instead of guint8 to increment the address. It's still not
6293 completely correct because a guint might not be able to hold the complete
6294 address range, but that's an enhacement for later.
6295 Add unit test to test improved behaviour.
6296 https://bugzilla.gnome.org/show_bug.cgi?id=708237
6298 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
6300 * gst/rtsp-server/rtsp-client.c:
6301 * tests/check/gst/client.c:
6302 client: allow absolute path in requests
6303 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
6305 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
6307 * gst/rtsp-server/rtsp-client.c:
6308 * gst/rtsp-server/rtsp-client.h:
6309 client: make make_path_from_uri a vmethod
6311 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
6313 * docs/libs/gst-rtsp-server-sections.txt:
6314 * gst/rtsp-server/rtsp-stream.c:
6315 * gst/rtsp-server/rtsp-stream.h:
6316 * tests/check/Makefile.am:
6317 * tests/check/gst/stream.c:
6318 stream: Add functions to get rtp and rtcp sockets
6319 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
6321 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
6323 * gst/rtsp-server/rtsp-context.c:
6324 * gst/rtsp-server/rtsp-context.h:
6325 context: defing a GType for the context
6326 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
6328 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6330 * gst/rtsp-server/Makefile.am:
6331 * gst/rtsp-server/rtsp-auth.c:
6332 * gst/rtsp-server/rtsp-context.c:
6333 * gst/rtsp-server/rtsp-media.c:
6334 * gst/rtsp-server/rtsp-mount-points.c:
6335 * gst/rtsp-server/rtsp-server.h:
6336 * gst/rtsp-server/rtsp-session-media.c:
6337 * gst/rtsp-server/rtsp-session.c:
6338 * gst/rtsp-server/rtsp-stream.c:
6339 Fixed several GIR warnings
6341 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
6343 * gst/rtsp-server/rtsp-auth.c:
6346 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6348 * tests/check/Makefile.am:
6349 * tests/check/gst/token.c:
6350 tests: Add unit tests for token
6351 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
6353 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6355 * gst/rtsp-server/rtsp-token.c:
6356 token: Validate args for gst_rtsp_token_is_allowed
6357 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
6359 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6361 * gst/rtsp-server/rtsp-token.c:
6362 token: Fix bug when creating empty token
6363 We always want to have a valid GstStructure in the token.
6364 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
6366 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
6368 * gst/rtsp-server/rtsp-thread-pool.c:
6369 thread-pool: avoid race in shutdown
6370 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
6371 don't actually stop the mainloop ever. Solve this race by adding an idle source
6372 to the mainloop that calls the _quit. This way we immediately exit the mainloop
6373 if quit was called before we started it.
6375 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6377 * tests/check/Makefile.am:
6378 * tests/check/gst/permissions.c:
6379 tests: Add unit tests for permissions
6380 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
6382 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6384 * tests/check/gst/mediafactory.c:
6385 tests: Test mediafactory permissions
6386 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
6388 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6390 * gst/rtsp-server/rtsp-permissions.c:
6391 permissions: Fix refcounting when adding/removing roles
6392 Previously a role that was removed was unreffed twice, and when
6393 replacing an existing role the replaced role was freed while still being
6394 referenced. Both bugs are now fixed.
6395 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
6397 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6399 * tests/check/gst/media.c:
6400 * tests/check/gst/mediafactory.c:
6401 * tests/check/gst/rtspserver.c:
6402 tests: Check gst_rtsp_url_parse return value
6403 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
6405 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
6408 Automatic update of common submodule
6409 From 865aa20 to dbedaa0
6411 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
6413 * gst/rtsp-server/rtsp-server.c:
6414 rtsp-server: Fix socket leak
6415 https://bugzilla.gnome.org/show_bug.cgi?id=710088
6417 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
6419 * gst/rtsp-server/rtsp-session-pool.c:
6420 rtsp-session-pool: Make sure session IDs are properly URI-escaped
6421 https://bugzilla.gnome.org/show_bug.cgi?id=643812
6423 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
6425 * examples/.gitignore:
6426 * examples/test-video.c:
6427 examples: fix compilation when WITH_AUTH is defined
6428 https://bugzilla.gnome.org/show_bug.cgi?id=710228
6430 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
6433 gitignore: Add new test binary
6435 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
6437 * tests/check/Makefile.am:
6438 * tests/check/gst/threadpool.c:
6439 thread-pool: Add unit test for the thread pools
6440 https://bugzilla.gnome.org/show_bug.cgi?id=710228
6442 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
6444 * gst/rtsp-server/rtsp-thread-pool.c:
6445 thread-pool: Fix thread leak when reusing threads
6446 https://bugzilla.gnome.org/show_bug.cgi?id=709730
6448 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
6450 * gst/rtsp-server/rtsp-server.c:
6451 * tests/check/gst/rtspserver.c:
6452 tests: fixed racy behavior in rtspserver tests
6453 https://bugzilla.gnome.org/show_bug.cgi?id=710078
6455 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
6457 * tests/check/gst/addresspool.c:
6458 tests: Improve address pool unit tests
6459 Add a range with mixed IPV4 and IPV6 addresses to pool.
6460 Get an IPV4 address from an IPV6-only pool.
6461 Get an IPV6 address from an IPV4-only pool.
6462 Reserve a IPV6 address from an IPV4-only pool.
6463 Check for unicast addresses in multicast-only pool.
6464 Check for unicast addresses in uni-/multicast-mixed pool.
6465 https://bugzilla.gnome.org/show_bug.cgi?id=710128
6467 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6469 * gst/rtsp-server/rtsp-client.c:
6470 client: append query string in PAUSE/PLAY/TEARDOWN as well
6472 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
6474 * gst/rtsp-server/rtsp-client.c:
6475 client: Add query to control path
6476 If the SETUP url contains a query it must be appended to the control
6477 path so that it matches any already created stream in the media. The
6478 query will also be appended to the session media path.
6480 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6482 * gst/rtsp-server/rtsp-media.c:
6483 rtsp-media: remove old line
6485 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
6487 * gst/rtsp-server/rtsp-stream.c:
6488 stream: Correct control comparison
6489 https://bugzilla.gnome.org/show_bug.cgi?id=709176
6491 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6493 * gst/rtsp-server/rtsp-media.c:
6494 media: Check dynamically if the pipeline supports seeking
6495 We should not depend on whether or not the pipeline state change
6496 returned NO_PREROLL or not. A media could dynamically change its
6497 element and switch from seekable to non seekable so it's best to test
6498 the seekable nature of the pipeline dynamically when we try to do a seek.
6500 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6502 * gst/rtsp-server/rtsp-media.c:
6503 media: Return FALSE if seeking is not supported
6505 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6507 * gst/rtsp-server/rtsp-media.c:
6508 rtsp-media: don't seek accurate by default
6509 Accurate seeking is perhaps a little overkill in the most common situation and
6510 causes some formats (mp3) over slow media to seek extremely slowly.
6512 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
6514 * tests/check/gst/rtspserver.c:
6515 tests: fix unit test
6516 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
6518 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
6520 * gst/rtsp-server/rtsp-client.c:
6521 client: Reply 400 if media cannot be constructed
6522 Reply 400 Bad Request instead of 503 Service Unavailable if media
6523 cannot be constructed in SETUP.
6524 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
6526 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
6528 * gst/rtsp-server/rtsp-client.c:
6529 client: Send setup reply once only
6530 If find_media() failed in handle_setup_request() two replies was sent.
6531 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
6533 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
6536 Automatic update of common submodule
6537 From 6b03ba7 to 865aa20
6539 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
6541 * gst/rtsp-server/rtsp-server.c:
6542 server: Emit client-connected signal earlier
6543 Emit client-connected before the client ref is given to a GSource,
6544 otherwise client-connected can be emitted after the client object has
6547 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
6549 * gst/rtsp-server/rtsp-address-pool.c:
6550 * gst/rtsp-server/rtsp-address-pool.h:
6551 * gst/rtsp-server/rtsp-stream.c:
6552 * tests/check/gst/addresspool.c:
6553 addresspool: return reason of failure
6554 Let gst_rtsp_address_pool_reserve_address() return the reason why
6555 the address could not be reserved.
6556 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
6558 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
6561 autogen.sh: Sync behaviour with other GStreamer modules
6562 Allows building from outside of tree amongst other things
6564 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
6567 Automatic update of common submodule
6568 From b613661 to 6b03ba7
6570 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
6573 Automatic update of common submodule
6574 From 74a6857 to b613661
6576 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
6579 Automatic update of common submodule
6580 From 01a7a46 to 74a6857
6582 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
6584 * gst/rtsp-server/rtsp-client.c:
6585 client: Do not read beyond end of path string
6586 If the setup was done without a control url, make sure we don't try to read the
6587 non-existing control string and crash.
6589 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6591 * gst/rtsp-server/rtsp-client.c:
6592 client: Fix RTPInfo header
6593 Refactor the method to make the content_base.
6594 Use the content-base and the control url to construct the RTPInfo
6597 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6599 * gst/rtsp-server/rtsp-client.c:
6600 client: map url to path only in describe
6601 Only map the request url to a path in the DESCRIBE method. The SDP then
6602 contains the base and control urls that should be used to SETUP/PAUSE/
6603 PLAY/TEARDOWN the media.
6605 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6607 * gst/rtsp-server/rtsp-client.c:
6608 Revert "client: map URL to path in requests"
6609 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
6610 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
6611 contains the base and control urls which are used in the SETUP, PLAY,
6612 PAUSE and TEARDOWN requests.
6614 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6616 * gst/rtsp-server/rtsp-client.c:
6617 client: map URL to path in requests
6619 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6621 * gst/rtsp-server/rtsp-client.c:
6622 * gst/rtsp-server/rtsp-mount-points.c:
6623 * gst/rtsp-server/rtsp-mount-points.h:
6624 mount-points: make vmethod to make path from uri
6625 Make a vmethod to transform an url into a path. The path is then used to lookup
6626 the factory. This makes it possible to also use other bits of the url, such as
6627 the query parameters, to locate the factory.
6629 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
6631 * gst/rtsp-server/rtsp-thread-pool.c:
6632 * gst/rtsp-server/rtsp-thread-pool.h:
6633 thread-pool: Add cleanup to wait for the threadpool to finish
6634 Also fix race condition if two threads are asking for the first
6635 thread from the thread pool at once. This would case two internal
6636 GThreadPools to be created.
6637 https://bugzilla.gnome.org/show_bug.cgi?id=707753
6639 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
6641 * gst/rtsp-server/rtsp-client.c:
6642 * tests/check/gst/client.c:
6643 client: free threadpool
6644 https://bugzilla.gnome.org/show_bug.cgi?id=707638
6646 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
6648 * tests/check/gst/mountpoints.c:
6649 mountpoints tests: unref matched factories
6650 https://bugzilla.gnome.org/show_bug.cgi?id=707638
6652 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
6654 * tests/check/gst/media.c:
6655 media tests: unref thread pool and caps
6656 https://bugzilla.gnome.org/show_bug.cgi?id=707638
6658 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
6660 * gst/rtsp-server/rtsp-auth.c:
6661 * gst/rtsp-server/rtsp-media-factory.c:
6662 * gst/rtsp-server/rtsp-media.c:
6663 auth, media, media-factory: unref permissions
6664 https://bugzilla.gnome.org/show_bug.cgi?id=707638
6666 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6668 * examples/Makefile.am:
6669 Makefile: add rule for appsrc example
6671 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6673 * examples/test-appsrc.c:
6674 tests: add appsrc example
6675 Add an example on how to use appsrc to feed the server pipeline with data.
6677 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
6679 * gst/rtsp-server/rtsp-client.c:
6680 rtsp-client: remove query part from content-base string
6681 Make sure that after the control url has been resolved, it's
6682 not a part of the query-string.
6683 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
6685 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6687 * gst/rtsp-server/rtsp-client.c:
6688 client: don't check url in response
6689 There is no url or method in the response to check
6691 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6693 * gst/rtsp-server/rtsp-client.c:
6694 * gst/rtsp-server/rtsp-client.h:
6695 Add handle-response signal for when we receive a GET_PARAMETER response
6697 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6699 * gst/rtsp-server/rtsp-server.c:
6700 Fix gst_rtsp_server_client_filter, using wrong variable type
6702 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
6704 * gst/rtsp-server/rtsp-media-factory-uri.c:
6705 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
6706 For AAC we need to check for framed=true instead of parsed=true.
6707 https://bugzilla.gnome.org/show_bug.cgi?id=701384
6709 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6711 * gst/rtsp-server/rtsp-stream.c:
6712 stream: optimize pipeline for protocols
6713 When TCP is not an allowed protocol for the stream, avoid creating the
6714 appsrc/appsink/queue and tee elements.
6716 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6718 * gst/rtsp-server/rtsp-media.c:
6719 media: set protocols on streams
6721 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6723 * gst/rtsp-server/rtsp-client.c:
6724 client: use protocols supported by stream
6726 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6728 * gst/rtsp-server/rtsp-media-factory.c:
6729 * gst/rtsp-server/rtsp-media.c:
6730 * gst/rtsp-server/rtsp-stream.c:
6731 media-factory: allow all protocols
6733 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6735 * gst/rtsp-server/rtsp-media.c:
6736 media: configure protocols in new streams
6738 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6740 * gst/rtsp-server/rtsp-stream.c:
6741 * gst/rtsp-server/rtsp-stream.h:
6742 stream: add protocols property
6744 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6746 * gst/rtsp-server/rtsp-media.c:
6747 rtsp-media: send state in "new-state" signal
6748 https://bugzilla.gnome.org/show_bug.cgi?id=705110
6750 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
6753 build: add subdir-objects to AM_INIT_AUTOMAKE
6754 Fixes warnings with automake 1.14
6755 https://bugzilla.gnome.org/show_bug.cgi?id=705350
6757 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6759 * docs/libs/gst-rtsp-server-sections.txt:
6760 * gst/rtsp-server/rtsp-client.c:
6761 * gst/rtsp-server/rtsp-server.c:
6762 * gst/rtsp-server/rtsp-server.h:
6763 server: add method to iterate clients of server
6765 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6767 * gst/rtsp-server/rtsp-media.c:
6768 * gst/rtsp-server/rtsp-media.h:
6769 Add vmethod for rtsp-media subclass to access rtpbin
6771 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6773 * gst/rtsp-server/rtsp-client.h:
6774 small documentation fix
6776 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6778 * gst/rtsp-server/rtsp-client.c:
6779 Do not take range header if range is invalid
6781 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6783 * docs/libs/gst-rtsp-server-sections.txt:
6784 * gst/rtsp-server/rtsp-media.c:
6785 media: add docs for new method
6787 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6789 * gst/rtsp-server/rtsp-media.c:
6790 * gst/rtsp-server/rtsp-media.h:
6791 Add API to rtsp-media set the pipeline's state
6793 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
6795 * gst/rtsp-server/rtsp-media.c:
6796 Update current position/duration when gst_rtsp_media_get_range_string is called
6798 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6800 * examples/test-cgroups.c:
6801 tests: add some more docs
6803 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6805 * examples/test-cgroups.c:
6806 * gst/rtsp-server/Makefile.am:
6807 * gst/rtsp-server/rtsp-auth.c:
6808 * gst/rtsp-server/rtsp-auth.h:
6809 * gst/rtsp-server/rtsp-client.c:
6810 * gst/rtsp-server/rtsp-client.h:
6811 * gst/rtsp-server/rtsp-context.c:
6812 * gst/rtsp-server/rtsp-context.h:
6813 * gst/rtsp-server/rtsp-params.c:
6814 * gst/rtsp-server/rtsp-params.h:
6815 * gst/rtsp-server/rtsp-server.c:
6816 * gst/rtsp-server/rtsp-thread-pool.c:
6817 * gst/rtsp-server/rtsp-thread-pool.h:
6818 * tests/check/gst/client.c:
6819 ClientState -> Context
6820 Rename the clientstate to context and put the code in a separate file.
6822 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6824 * examples/test-auth.c:
6825 * gst/rtsp-server/rtsp-auth.c:
6826 * gst/rtsp-server/rtsp-auth.h:
6827 auth: add support for default token
6828 The default token is used when the user is not authenticated and can be used to
6829 give minimal permissions.
6831 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6833 * examples/test-auth.c:
6834 * gst/rtsp-server/rtsp-auth.c:
6835 auth: use defines when possible
6837 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6839 * gst/rtsp-server/rtsp-address-pool.c:
6840 address-pool: improve docs
6842 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6844 * gst/rtsp-server/rtsp-permissions.c:
6845 permissions: add the role to the copy
6847 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
6849 * gst/rtsp-server/rtsp-permissions.c:
6850 permissions: Also copy the roles
6852 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
6854 * gst/rtsp-server/rtsp-permissions.c:
6855 permissions: Make it build
6857 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6859 * gst/rtsp-server/rtsp-address-pool.h:
6862 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6864 * docs/libs/gst-rtsp-server-sections.txt:
6865 * gst/rtsp-server/rtsp-auth.c:
6866 * gst/rtsp-server/rtsp-auth.h:
6867 * gst/rtsp-server/rtsp-media.c:
6868 * gst/rtsp-server/rtsp-session-media.c:
6869 * gst/rtsp-server/rtsp-stream-transport.c:
6870 * gst/rtsp-server/rtsp-stream-transport.h:
6871 * gst/rtsp-server/rtsp-stream.c:
6872 * tests/check/gst/client.c:
6875 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6877 * docs/libs/gst-rtsp-server-sections.txt:
6878 * gst/rtsp-server/rtsp-address-pool.c:
6879 * gst/rtsp-server/rtsp-address-pool.h:
6880 * tests/check/gst/addresspool.c:
6881 * tests/check/gst/rtspserver.c:
6882 address-pool: cleanups
6883 Remove redundant method, improve docs.
6885 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6887 * docs/libs/gst-rtsp-server-sections.txt:
6888 * gst/rtsp-server/rtsp-auth.h:
6889 * gst/rtsp-server/rtsp-permissions.c:
6890 * gst/rtsp-server/rtsp-permissions.h:
6891 * gst/rtsp-server/rtsp-token.c:
6894 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6896 * gst/rtsp-server/rtsp-permissions.c:
6897 permissions: implement _remove_role
6899 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6901 * gst/rtsp-server/rtsp-permissions.c:
6902 permissions: update docs
6904 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6906 * tests/check/gst/client.c:
6907 tests: simplify tests
6908 Client settings are now disabled by default so we don't need an auth
6909 module to disable them.
6911 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6913 * gst/rtsp-server/rtsp-auth.c:
6914 auth: add default authorizations
6915 When no auth module is specified, use our table of defaults to look up the
6916 default value of the check instead of always allowing everything. This was
6917 we can disallow client settings by default.
6919 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6922 README: update readme
6924 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6926 * gst/rtsp-server/rtsp-thread-pool.c:
6927 * gst/rtsp-server/rtsp-thread-pool.h:
6928 thread-pool: add more docs
6930 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6932 * gst/rtsp-server/rtsp-thread-pool.c:
6933 * gst/rtsp-server/rtsp-thread-pool.h:
6934 thread-pool: fix race in thread reuse
6935 If we try to reuse a thread right after we made it stop, we end up using a
6936 stopped thread. Catch this case and only reuse threads that are not stopping.
6938 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6940 * gst/rtsp-server/rtsp-server.c:
6941 server: add small debug
6943 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6945 * tests/check/gst/client.c:
6947 Add some permissions to media so we can use the auth and enable
6950 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6952 * gst/rtsp-server/rtsp-client.c:
6953 client: support pushed context in handle_request
6954 If we already have a pushed state, reuse it and add our own things. This makes
6955 it easier to write tests.
6957 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6959 * gst/rtsp-server/rtsp-auth.c:
6960 auth: don't auth on methods
6961 Don't authorize on methods anymore but on the resources that we
6962 try to access, this is more flexible.
6963 Move the authorization checks to where they are needed and let the
6964 check return the response on error.
6966 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6968 * gst/rtsp-server/rtsp-mount-points.c:
6969 mount-points: add some debug
6971 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6973 * tests/check/gst/client.c:
6974 tests: almost fix test
6976 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6978 * gst/rtsp-server/rtsp-auth.c:
6979 * gst/rtsp-server/rtsp-auth.h:
6980 * gst/rtsp-server/rtsp-client.c:
6981 * gst/rtsp-server/rtsp-client.h:
6982 * gst/rtsp-server/rtsp-server.c:
6983 * gst/rtsp-server/rtsp-server.h:
6984 auth: let the auth module check client_settings
6985 Let the auth module decide if client settings are allowed for the
6988 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6990 * gst/rtsp-server/rtsp-token.c:
6991 * gst/rtsp-server/rtsp-token.h:
6992 token: add method to check boolean permission
6994 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6996 * examples/test-auth.c:
6997 * examples/test-cgroups.c:
6998 * gst/rtsp-server/rtsp-token.c:
6999 * gst/rtsp-server/rtsp-token.h:
7000 token: simplify token constructor
7001 Use variable arguments to make easier API.
7003 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7005 * examples/test-auth.c:
7006 * examples/test-cgroups.c:
7007 * gst/rtsp-server/rtsp-media-factory.c:
7008 * gst/rtsp-server/rtsp-media-factory.h:
7009 media-factory: add convenience API for factory
7011 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7013 * examples/test-auth.c:
7014 * examples/test-cgroups.c:
7015 * gst/rtsp-server/rtsp-permissions.c:
7016 * gst/rtsp-server/rtsp-permissions.h:
7017 permissions: simplify API a little
7018 Avoid passing GstStructure in the add_role method, use varargs instead
7019 to construct the structure behind the scenes. We can then also use the
7020 structure name as the role and simplify some more logic.
7022 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7024 * gst/rtsp-server/rtsp-auth.c:
7027 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7029 * gst/rtsp-server/rtsp-auth.c:
7030 * gst/rtsp-server/rtsp-auth.h:
7031 * gst/rtsp-server/rtsp-client.c:
7032 auth: handle unauthorized response
7033 Move handling of the unauthorized response to the auth module, it can add
7034 the appropriate headers to request authorization for the required method
7035 much better than the client.
7037 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7039 * gst/rtsp-server/rtsp-client.c:
7040 * gst/rtsp-server/rtsp-client.h:
7041 client: allow for sending any message, not only requests
7042 Change the _send_request() method to _send_message() so that we
7043 can both send requests and replies.
7045 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7047 * docs/libs/gst-rtsp-server-sections.txt:
7048 * gst/rtsp-server/rtsp-server.h:
7051 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7053 * examples/test-video.c:
7054 * gst/rtsp-server/rtsp-auth.c:
7055 * gst/rtsp-server/rtsp-auth.h:
7056 * gst/rtsp-server/rtsp-server.c:
7057 * gst/rtsp-server/rtsp-server.h:
7058 auth: move TLS handling to auth module
7059 Remove the TLS settings on the server and move it to the auth module because
7060 that is where security related bits go.
7062 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7064 * gst/rtsp-server/rtsp-client.c:
7065 * gst/rtsp-server/rtsp-client.h:
7066 client: add state push/pop
7068 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7070 * gst/rtsp-server/rtsp-client.c:
7071 * gst/rtsp-server/rtsp-client.h:
7072 client: add connection to state
7074 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7076 * gst/rtsp-server/rtsp-mount-points.c:
7077 mount-points: fix debug
7079 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7081 * tests/check/gst/media.c:
7082 tests: fix media test
7084 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7086 * gst/rtsp-server/rtsp-thread-pool.c:
7087 thread-pool: we don't require a state
7089 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7091 * gst/rtsp-server/rtsp-server.c:
7092 server: let context ref the server
7093 So that we don't risk losing the server object early anc crash.
7095 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7097 * tests/check/gst/client.c:
7098 tests: fix client test
7100 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7103 * docs/libs/gst-rtsp-server-docs.sgml:
7104 * docs/libs/gst-rtsp-server-sections.txt:
7105 * gst/rtsp-server/rtsp-address-pool.c:
7106 * gst/rtsp-server/rtsp-auth.c:
7107 * gst/rtsp-server/rtsp-client.c:
7108 * gst/rtsp-server/rtsp-client.h:
7109 * gst/rtsp-server/rtsp-media-factory-uri.c:
7110 * gst/rtsp-server/rtsp-media-factory.c:
7111 * gst/rtsp-server/rtsp-media-factory.h:
7112 * gst/rtsp-server/rtsp-media.c:
7113 * gst/rtsp-server/rtsp-mount-points.c:
7114 * gst/rtsp-server/rtsp-params.c:
7115 * gst/rtsp-server/rtsp-permissions.c:
7116 * gst/rtsp-server/rtsp-sdp.c:
7117 * gst/rtsp-server/rtsp-server.c:
7118 * gst/rtsp-server/rtsp-server.h:
7119 * gst/rtsp-server/rtsp-session-media.c:
7120 * gst/rtsp-server/rtsp-session-pool.c:
7121 * gst/rtsp-server/rtsp-session.c:
7122 * gst/rtsp-server/rtsp-stream-transport.c:
7123 * gst/rtsp-server/rtsp-stream.c:
7124 * gst/rtsp-server/rtsp-thread-pool.c:
7125 * gst/rtsp-server/rtsp-token.c:
7128 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7130 * gst/rtsp-server/rtsp-session-pool.c:
7131 * gst/rtsp-server/rtsp-session-pool.h:
7132 session-pool: make vmethod to create a session
7133 Make a vmethod to create a sessions so that subclasses can create
7134 custom session objects
7136 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7138 * gst/rtsp-server/rtsp-auth.c:
7139 * gst/rtsp-server/rtsp-media-factory.h:
7140 * gst/rtsp-server/rtsp-media.h:
7141 * gst/rtsp-server/rtsp-mount-points.h:
7142 * gst/rtsp-server/rtsp-session-pool.h:
7143 * gst/rtsp-server/rtsp-stream.h:
7146 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7148 * docs/libs/gst-rtsp-server-docs.sgml:
7149 * docs/libs/gst-rtsp-server-sections.txt:
7150 * gst/rtsp-server/rtsp-address-pool.c:
7151 * gst/rtsp-server/rtsp-address-pool.h:
7152 * gst/rtsp-server/rtsp-auth.c:
7153 * gst/rtsp-server/rtsp-client.h:
7154 * gst/rtsp-server/rtsp-media-factory.h:
7155 * gst/rtsp-server/rtsp-media.c:
7156 * gst/rtsp-server/rtsp-media.h:
7157 * gst/rtsp-server/rtsp-permissions.c:
7158 * gst/rtsp-server/rtsp-permissions.h:
7159 * gst/rtsp-server/rtsp-server.h:
7160 * gst/rtsp-server/rtsp-session-media.c:
7161 * gst/rtsp-server/rtsp-session-media.h:
7162 * gst/rtsp-server/rtsp-session-pool.h:
7163 * gst/rtsp-server/rtsp-session.h:
7164 * gst/rtsp-server/rtsp-stream-transport.h:
7165 * gst/rtsp-server/rtsp-stream.c:
7166 * gst/rtsp-server/rtsp-thread-pool.h:
7169 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7172 * examples/Makefile.am:
7173 configure: compile cgroup example conditionally
7174 Only compile the cgroup example when we have libcgroup
7176 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7179 * examples/Makefile.am:
7180 * examples/test-cgroups.c:
7181 examples: add cgroups example
7183 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7185 * tests/check/gst/rtspserver.c:
7186 tests: fix compilation
7188 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7190 * gst/rtsp-server/rtsp-thread-pool.c:
7191 thread-pool: fix vmethod invocation
7193 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7195 * gst/rtsp-server/rtsp-thread-pool.c:
7196 * gst/rtsp-server/rtsp-thread-pool.h:
7197 thread-pool: store thread type in thread
7199 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7201 * gst/rtsp-server/rtsp-client.c:
7202 client: pass thread from pool to media _prepare
7203 Get a thread from the configured threadpool and pass it to the prepare method of
7206 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7208 * gst/rtsp-server/rtsp-media.c:
7209 * gst/rtsp-server/rtsp-media.h:
7210 media: Accept a thread in _prepare
7211 Remove out own threadpool handling and use the provided thread and
7212 maincontext for the bus messages and the state changes.
7214 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7216 * gst/rtsp-server/rtsp-server.c:
7217 server: configure client thread pool
7219 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7221 * gst/rtsp-server/rtsp-client.c:
7222 * gst/rtsp-server/rtsp-client.h:
7223 client: add method to configure thread pool
7225 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7227 * gst/rtsp-server/rtsp-client.h:
7228 * gst/rtsp-server/rtsp-server.c:
7229 * gst/rtsp-server/rtsp-server.h:
7230 server: use thread pool
7231 Use the thread pool instead of doing our own thing.
7233 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7235 * gst/rtsp-server/Makefile.am:
7236 * gst/rtsp-server/rtsp-thread-pool.c:
7237 * gst/rtsp-server/rtsp-thread-pool.h:
7238 thread-pool: add object to manage threads
7239 Add an object to manage the client and media threads.
7241 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7243 * gst/rtsp-server/rtsp-auth.c:
7244 auth: debug authorization check
7246 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7248 * gst/rtsp-server/rtsp-media.c:
7249 media: start media pipeline in context
7250 Start the media pipeline in the provided context (or our default one
7251 when NULL). This makes sure that we run the bus thread in this context and that
7252 all media threads are children of this context.
7254 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7256 * gst/rtsp-server/rtsp-media-factory.c:
7257 factory: pass permissions to media by default
7259 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7261 * examples/test-auth.c:
7262 test: add permissions to auth test
7263 Ass some permissions to the media factory in the test.
7265 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7267 * gst/rtsp-server/rtsp-auth.c:
7268 * gst/rtsp-server/rtsp-auth.h:
7269 * gst/rtsp-server/rtsp-client.c:
7270 auth: simplify auth checks
7271 Remove client from methods, it's now in the state
7272 Perform the check specified by the string, use the information from the
7273 thread local context.
7275 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7277 * gst/rtsp-server/rtsp-client.c:
7278 * gst/rtsp-server/rtsp-client.h:
7279 client: add state to current thread
7280 Add the client to the ClientState object.
7281 Place the ClientState on the current thread.
7283 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7285 * gst/rtsp-server/rtsp-media-factory.c:
7286 * gst/rtsp-server/rtsp-media-factory.h:
7287 * gst/rtsp-server/rtsp-media.c:
7288 * gst/rtsp-server/rtsp-media.h:
7289 media: make it possible to set permissions
7290 Make it possible to set permissions on media and media factory objects
7292 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7294 * gst/rtsp-server/Makefile.am:
7295 * gst/rtsp-server/rtsp-permissions.c:
7296 * gst/rtsp-server/rtsp-permissions.h:
7297 permissions: add permissions object
7298 Add a mini object to store permissions based on a role.
7300 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7302 * examples/test-auth.c:
7303 * gst/rtsp-server/rtsp-auth.c:
7304 * gst/rtsp-server/rtsp-auth.h:
7305 * gst/rtsp-server/rtsp-client.c:
7306 auth: add auth checks
7307 Add an enum with auth checks and implement the checks in the auth object.
7308 Perform the checks from the client.
7310 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7312 * examples/test-auth.c:
7313 * gst/rtsp-server/rtsp-auth.c:
7314 * gst/rtsp-server/rtsp-auth.h:
7315 * gst/rtsp-server/rtsp-client.h:
7316 auth: use the token after authentication
7317 After we authenticated a user, keep the Token around in the state.
7319 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7321 * gst/rtsp-server/rtsp-client.c:
7322 * gst/rtsp-server/rtsp-media.c:
7323 * gst/rtsp-server/rtsp-media.h:
7324 * tests/check/gst/media.c:
7325 media: add optional context for bus messages
7326 Add an optional mainloop to _prepare that will handle the bus messages instead
7327 of always using the shared mainloop.
7329 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7331 * gst/rtsp-server/Makefile.am:
7332 * gst/rtsp-server/rtsp-token.c:
7333 * gst/rtsp-server/rtsp-token.h:
7334 token: add authorization token
7335 Add a simply miniobject that contains the authorizations. The object contains a
7336 GstStructure that hold all authorization fields. When a user is authenticated,
7337 the auth module will create a Token for the user. The token is then used to
7338 check what operations the user is allowed to do and various other configuration
7341 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7343 * examples/test-auth.c:
7344 * gst/rtsp-server/rtsp-auth.c:
7345 * gst/rtsp-server/rtsp-auth.h:
7346 * gst/rtsp-server/rtsp-client.c:
7347 * gst/rtsp-server/rtsp-client.h:
7348 * gst/rtsp-server/rtsp-media-factory.c:
7349 * gst/rtsp-server/rtsp-media-factory.h:
7350 * gst/rtsp-server/rtsp-media.c:
7351 * gst/rtsp-server/rtsp-media.h:
7352 auth: remove auth from media and factory
7353 Remove the auth object from media and factory. We want to have the RTSPClient
7354 authenticate and authorize resources, there is no need to place another auth
7355 manager on the media/factory.
7357 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7359 * examples/test-auth.c:
7360 * gst/rtsp-server/rtsp-auth.c:
7361 * gst/rtsp-server/rtsp-auth.h:
7362 * gst/rtsp-server/rtsp-client.h:
7363 auth: add support for multiple basic auth tokens
7364 Make it possible to add multiple basic authorisation tokens to one authorization
7365 object. Associate with each token an authorization group that will define what
7366 capabilities are allowed.
7368 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7370 * gst/rtsp-server/rtsp-client.c:
7371 client: error out on non-aggregate control
7372 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
7374 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7376 * gst/rtsp-server/rtsp-client.c:
7377 client: rework setup request a little
7378 Cache the media in DESCRIBE based on the longest matching path with the uri
7379 that we can find in the mount points.
7380 Rework the setup request a little to get the media from the session or from
7381 the longest matching path, this way we can derive the control string as
7382 everything after the path instead of hardcoding it.
7383 Find the stream based on the control string and only open a session when all
7386 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7388 * gst/rtsp-server/rtsp-media.c:
7389 * gst/rtsp-server/rtsp-media.h:
7390 media: add method to find a stream by control url
7392 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7394 * gst/rtsp-server/rtsp-stream.c:
7395 * gst/rtsp-server/rtsp-stream.h:
7396 stream: add method to check control url of stream
7398 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7400 * gst/rtsp-server/rtsp-client.c:
7401 * gst/rtsp-server/rtsp-session-media.c:
7402 * gst/rtsp-server/rtsp-session-media.h:
7403 * gst/rtsp-server/rtsp-session.c:
7404 * gst/rtsp-server/rtsp-session.h:
7405 session: use path matching for session media
7406 Use a path string instead of a uri to lookup session media in the sessions. Also
7407 use path matching to find the largest possible path that matches.
7409 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7411 * gst/rtsp-server/rtsp-client.c:
7412 * gst/rtsp-server/rtsp-mount-points.c:
7413 * gst/rtsp-server/rtsp-mount-points.h:
7414 * tests/check/gst/mountpoints.c:
7415 mount-points: remove useless vmethod
7416 Making lookups in the mount points should not be done with a URL, if there is a
7417 mapping to be done from URL to mount points, we'll need to do it somewhere
7420 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7422 * gst/rtsp-server/rtsp-mount-points.c:
7423 * gst/rtsp-server/rtsp-mount-points.h:
7424 * tests/check/gst/mountpoints.c:
7425 mount-points: improve mount point searching
7426 Use a GSequence to keep track of the mount points.
7427 Match a URL to the longest matching registered mount point. This should be the
7428 URL to perform aggreagate control and the remainder is the stream specific
7430 Add some unit tests for this.
7432 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
7434 * gst/rtsp-server/Makefile.am:
7435 rtsp-server: Allow building of static library
7437 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7439 * tests/check/gst/mediafactory.c:
7440 tests: fix compilation
7442 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7444 * gst/rtsp-server/rtsp-sdp.c:
7445 sdp: get control string from stream
7446 Use the control string as configured in the stream.
7448 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7450 * gst/rtsp-server/rtsp-stream.c:
7451 * gst/rtsp-server/rtsp-stream.h:
7452 stream: add methods and property to set control string
7454 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7456 * gst/rtsp-server/rtsp-client.c:
7458 Rename variables for clarity
7459 Keep media in state when we can
7461 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7463 * gst/rtsp-server/rtsp-client.c:
7464 * gst/rtsp-server/rtsp-stream.c:
7465 * gst/rtsp-server/rtsp-stream.h:
7466 stream: add more support for IPv6
7467 Rename _get_address to _get_multicast_address in GstRTSPStream to
7468 make it clear that this function only deals with multicast.
7469 Make it possible to have both an IPv4 and IPv6 multicast address on
7470 a stream. Give the client an IPv4 or IPv6 address depending on the
7471 address it used to connect to the server.
7472 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
7474 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7476 * gst/rtsp-server/rtsp-client.c:
7479 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7481 * gst/rtsp-server/rtsp-stream.c:
7482 stream: handle failed port allocation
7483 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
7484 can't allocate any family at all. Also keep track of what port families we
7486 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
7488 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7490 * gst/rtsp-server/rtsp-stream.c:
7491 stream: improve docs
7493 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7495 * gst/rtsp-server/rtsp-stream-transport.c:
7496 stream-transport: remove old if 0 block
7498 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
7500 * tests/check/gst/client.c:
7502 gst_rtsp_client_get_uri() has been removed
7503 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
7505 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7507 * gst/rtsp-server/rtsp-client.c:
7508 * gst/rtsp-server/rtsp-client.h:
7509 client: add method to filter managed sessions
7510 Add a method to filter the sessions managed by this client connection.
7511 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
7513 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7515 * gst/rtsp-server/rtsp-client.c:
7516 * gst/rtsp-server/rtsp-client.h:
7517 client: remove _get_uri() method
7518 Remove the get_uri() method on the client. A client has no uri, the uri
7519 property is an internal property to manage the last cached media for
7522 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7524 * gst/rtsp-server/rtsp-media-factory.h:
7525 media-factory: fix typo
7527 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
7529 * gst/rtsp-server/rtsp-media.c:
7530 rtsp-media: Do not leak the query in default_query_stop
7531 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
7533 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7535 * gst/rtsp-server/rtsp-media.c:
7536 media: don't unlock when conversion fails
7537 Don't unlock the state lock when conversion fails because it was not locked.
7539 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
7541 * gst/rtsp-server/rtsp-media.c:
7542 * gst/rtsp-server/rtsp-media.h:
7543 Add query_position and query_stop vmethods to rtsp-media
7545 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
7547 * gst/rtsp-server/rtsp-media.c:
7548 Fix typo in property install for rtsp-media's time-provider
7550 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7552 * gst/rtsp-server/rtsp-client.c:
7553 * gst/rtsp-server/rtsp-client.h:
7554 client: clean some variables
7555 Clean some variables and add some guards to _send_request()
7557 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
7559 * gst/rtsp-server/rtsp-client.c:
7560 * gst/rtsp-server/rtsp-client.h:
7561 Add gst_rtsp_client_send_request API
7562 This makes it possible to send arbitrary messages to a client, such as
7563 SET_PARAMETER or GET_PARAMETER
7565 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7567 * gst/rtsp-server/rtsp-media.c:
7568 * gst/rtsp-server/rtsp-media.h:
7569 media: add _get_element() method
7570 Add method to get the element used when creating the media.
7571 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
7573 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7575 * gst/rtsp-server/rtsp-media.c:
7578 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
7580 * gst/rtsp-server/rtsp-stream.c:
7581 * gst/rtsp-server/rtsp-stream.h:
7582 stream: allow access to the rtp session
7583 https://bugzilla.gnome.org/show_bug.cgi?id=703004
7585 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
7587 * gst/rtsp-server/rtsp-stream.c:
7588 * gst/rtsp-server/rtsp-stream.h:
7589 dscp qos support in gst-rtsp-stream
7590 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
7592 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7594 * tests/check/gst/rtspserver.c:
7596 Actually do what the comment says. Also keep the old code around, not sure what
7597 should happen when you get a 454 from a TEARDOWN, does it close the connection?
7598 it currently doesn't.
7600 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7602 * gst/rtsp-server/rtsp-client.c:
7603 client: also watch newly created session
7604 When we newly created a session, start watching it immediately instead of
7605 on the next request.
7607 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
7609 * tests/check/gst/client.c:
7610 tests: add unit test for new-session
7611 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
7613 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7615 * gst/rtsp-server/rtsp-client.c:
7616 client: emit new-session when new session is created
7617 Only emit new-session when we created a new session for a client, not when a
7618 client picked up a previous session.
7619 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
7621 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
7623 * gst/rtsp-server/rtsp-client.c:
7624 client: handle asterisk as path in requests
7625 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
7627 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7629 * gst/rtsp-server/rtsp-media.c:
7630 media: handle segment query format mismatch
7631 It's possible that the segment query returns with a different format than what
7632 we asked for, handle this case also.
7634 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
7636 * gst/rtsp-server/rtsp-media.c:
7637 media: use segment stop in collect_media_stats
7638 Use segment stop instead of duration as range end point.
7639 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
7641 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
7643 * gst/rtsp-server/rtsp-media.c:
7644 * tests/check/gst/media.c:
7645 rtsp-media: Do not leak the element in take_pipeline
7646 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
7648 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
7650 * gst/rtsp-server/rtsp-client.c:
7651 * gst/rtsp-server/rtsp-client.h:
7652 rtsp-client: Make configure_client_transport virtual
7653 This patch makes configure_client_transport virtual. The functionality is
7654 needed to handle some weird clients sending multicast transport settings as url
7656 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
7658 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
7660 * gst/rtsp-server/rtsp-client.c:
7661 * gst/rtsp-server/rtsp-client.h:
7662 rtsp-client: Make param_set and param_get virtual
7663 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
7665 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
7667 * gst/rtsp-server/rtsp-client.c:
7668 * gst/rtsp-server/rtsp-media.c:
7669 * gst/rtsp-server/rtsp-media.h:
7670 media: convert_range replaces get_range_times
7671 get_range_times worked for handling UTC ranges for seeks, but we also
7672 need to convert back from NPT to the requested unit in
7673 get_range_string. convert_range is now used for both.
7674 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
7676 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7678 * gst/rtsp-server/rtsp-client.c:
7679 * gst/rtsp-server/rtsp-sdp.c:
7680 * gst/rtsp-server/rtsp-sdp.h:
7681 sdp: cleanup sdp info
7682 We don't need to pass the proto, we can more easily check a boolean.
7683 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
7685 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
7687 * gst/rtsp-server/rtsp-sdp.c:
7688 use 0.0.0.0 or :: for c= line instead of server address
7690 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
7692 * gst/rtsp-server/rtsp-client.c:
7693 use local address, not remote, in SDP
7694 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
7696 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7699 Automatic update of common submodule
7700 From 098c0d7 to 01a7a46
7702 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
7704 * gst/rtsp-server/rtsp-media.c:
7705 * gst/rtsp-server/rtsp-media.h:
7706 media: possibility to override range time conversion
7707 Make it possible to override the conversion from GstRTSPTimeRange to
7708 GstClockTimes, that is done before seeking on the media
7709 pipeline. Overriding can be useful for UTC ranges, where the default
7710 conversion gives nanoseconds since 1900.
7711 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
7713 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
7715 * gst/rtsp-server/rtsp-server.c:
7716 * gst/rtsp-server/rtsp-server.h:
7717 rtsp-server: Expose the use_client_settings API
7718 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
7720 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
7722 * gst/rtsp-server/rtsp-client.c:
7723 * gst/rtsp-server/rtsp-stream.c:
7724 * gst/rtsp-server/rtsp-stream.h:
7725 rtspstream: handle both ipv4 and ipv6 clients
7726 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
7728 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7730 * gst/rtsp-server/rtsp-sdp.c:
7731 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
7732 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
7733 We already have a way to place extra attributes in the SDP by using a string
7734 property with prefix x- or a- in the caps.
7736 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7738 * gst/rtsp-server/rtsp-sdp.c:
7739 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
7740 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
7741 We already have a way to place extra attributes in the SDP, just make a string
7742 property in the payloader with a- or x- prefix.
7744 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7746 * gst/rtsp-server/rtsp-sdp.c:
7747 rtsp: place a- and x- properties as attributes
7748 application/x-rtp has properties with a- and x- prefixes that should be
7749 placed as attributes in the SDP for the media instead of being added to the
7752 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7754 * examples/Makefile.am:
7755 * examples/test-video.c:
7756 example: add TLS example
7758 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7760 * gst/rtsp-server/rtsp-server.c:
7761 * gst/rtsp-server/rtsp-server.h:
7762 server: add support for TLS
7763 Add methods to set and get a TLS certificate.
7764 Add vmethod to configure a new connection. By default, configure the TLS
7765 certificate in a new connection if needed.
7767 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7769 * gst/rtsp-server/rtsp-server.c:
7770 * gst/rtsp-server/rtsp-server.h:
7771 server: remove accept_client vmethod
7772 This vmethod is not very useful so remove it.
7774 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7776 * gst/rtsp-server/rtsp-server.c:
7777 server: don't crash on NULL GError
7779 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
7781 * gst/rtsp-server/rtsp-session-pool.c:
7782 rtsp-session-pool: corrected session timeout detection
7783 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
7785 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7787 * gst/rtsp-server/rtsp-client.c:
7788 client: improve debug
7790 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7792 * gst/rtsp-server/rtsp-client.c:
7793 * gst/rtsp-server/rtsp-client.h:
7794 * gst/rtsp-server/rtsp-server.c:
7795 server: refactor connection setup
7796 Let the server accept the socket connection and construct a GstRTSPConnection
7797 from it. Remove the code from the client and let the client only deal with
7798 a fully configure GstRTSPConnection object.
7799 We will need this later when the server will configure the connection for
7802 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7804 * gst/rtsp-server/rtsp-stream.c:
7805 stream: keep the transport object alive
7806 Keep the transport object alive while we have it as qdata on the
7809 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
7811 * gst/rtsp-server/rtsp-client.c:
7812 * gst/rtsp-server/rtsp-server.c:
7813 rtsp-server: Do not crash on nmapping of server
7814 * generate error when gst_rtsp_connection_accept fails
7815 * do not stop accepting incoming connections because
7816 accepting a client fails
7817 https://bugzilla.gnome.org/show_bug.cgi?id=701072
7819 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
7821 * gst/rtsp-server/rtsp-client.c:
7822 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
7823 https://bugzilla.gnome.org/show_bug.cgi?id=700953
7825 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
7827 * gst/rtsp-server/rtsp-sdp.c:
7828 rtsp-sdp: Parse framerate caps field and set SDP attribute
7829 The SDP attribute and its format is described in RFC4566.
7830 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
7832 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
7834 * gst/rtsp-server/rtsp-sdp.c:
7835 rtsp-sdp: Parse width/height from caps and set SDP attribute
7836 The SDP attribute and its format is described in RFC6064.
7837 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
7839 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
7841 * gst/rtsp-server/rtsp-sdp.c:
7842 * tests/check/gst/client.c:
7843 rtsp-sdp: add bandwidth line
7844 https://bugzilla.gnome.org/show_bug.cgi?id=699220
7846 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7849 Automatic update of common submodule
7850 From 5edcd85 to 098c0d7
7852 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
7854 * tests/check/gst/media.c:
7855 tests: add dynamic payloader prepare/unprepare check
7857 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7859 * gst/rtsp-server/rtsp-media.c:
7860 media: release lock when removing fakesink
7862 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7864 * gst/rtsp-server/rtsp-stream.c:
7865 stream: set elements to NULL before removing
7866 When removing a stream, set the elements to NULL first. This avoids
7867 element-is-not-in-NULL-state errors when we dispose the elements.
7869 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
7872 Automatic update of common submodule
7873 From 3cb3d3c to 5edcd85
7875 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7877 * gst/rtsp-server/rtsp-media.c:
7878 * gst/rtsp-server/rtsp-media.h:
7879 media: listen to pad-removed signals
7880 Listen to the pad-removed signal and remove the stream associated with the
7882 Add signal to be notified of the removed pad.
7883 Remove the fakesink in unprepare()
7884 Fix signatures of the signal methods
7886 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7888 * examples/test-sdp.c:
7889 tests: add example of reusable pipelines
7891 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
7893 * gst/rtsp-server/rtsp-stream.c:
7894 * gst/rtsp-server/rtsp-stream.h:
7895 stream: add method to get the srcpad
7897 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
7899 * tests/check/gst/media.c:
7900 check: add media prepare/unprepare test
7901 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
7903 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
7905 * gst/rtsp-server/rtsp-media.c:
7906 media: disconnect from signal handlers in unprepare()
7907 We connected to the pad-added and no-more-pads signals in prepare() so
7908 we need to disconnect from them in unprepare().
7909 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
7911 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
7913 * gst/rtsp-server/rtsp-media.c:
7914 media: don't free streams array
7915 Don't free the streams array in the unprepare() method, they were not
7917 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
7919 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
7921 * gst/rtsp-server/rtsp-media.c:
7922 media: don't unref the pipeline in unprepare
7923 Unprepare() should undo what prepare() does. Because the pipeline is
7924 not created in prepare(), we should not unref it in unprepare()
7926 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
7928 * gst/rtsp-server/rtsp-stream.c:
7929 stream: clear session and caps for reuse
7930 Set the session and caps to NULL after unref otherwise we might unref
7932 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
7934 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
7936 * gst/rtsp-server/rtsp-client.c:
7937 client: send out teardown signal before tearing down
7938 The advantage is that in the signal handler you get direct access to
7939 information about what streams are about to get torn down (in the
7940 GstRTSPClientState).
7941 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
7943 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
7945 * gst/rtsp-server/rtsp-client.c:
7946 * gst/rtsp-server/rtsp-client.h:
7947 client: expose connection
7948 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
7950 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
7953 Automatic update of common submodule
7954 From aed87ae to 3cb3d3c
7956 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7958 * gst/rtsp-server/rtsp-media.c:
7959 * gst/rtsp-server/rtsp-media.h:
7960 * gst/rtsp-server/rtsp-session-media.c:
7961 * gst/rtsp-server/rtsp-session-media.h:
7962 media: add method to get the base_time of the pipeline
7963 Together with a shared clock, this base-time could eventually be sent to
7964 the client so that it can reconstruct the exact running-time of the clock
7967 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7969 * gst/rtsp-server/Makefile.am:
7970 * gst/rtsp-server/rtsp-media.c:
7971 * gst/rtsp-server/rtsp-media.h:
7972 * gst/rtsp-server/rtsp-sdp.c:
7973 media: add GstNetTimeProvider support
7974 Add a property to let the media provide a GstNetTimeProvider for its clock.
7975 Make methods to get the clock and nettimeprovider
7976 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
7977 provider and also the current time of the clock. This should make it possible
7978 for (GStreamer) clients to slave their clock to the server clock.
7980 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
7983 Automatic update of common submodule
7984 From 04c7a1e to aed87ae
7986 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7988 * gst/rtsp-server/rtsp-media.c:
7989 media: wait for buffering to complete
7990 Wait for buffering to complete before changing the state to the target state.
7992 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7994 * gst/rtsp-server/rtsp-media.c:
7995 media: small cleanup
7997 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
7999 * tests/check/gst/rtspserver.c:
8000 tests: remove extra unref in test_setup_non_existing_stream
8001 The unref is not needed anymore, teardown runs without it.
8002 https://bugzilla.gnome.org/show_bug.cgi?id=696542
8004 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
8006 * tests/check/gst/rtspserver.c:
8007 tests: GSocketService cleanup in test_bind_already_in_use
8008 Use g_socket_service_stop so the rtspserver test stops listening for
8009 incoming connections in test_bind_already_in_use.
8010 https://bugzilla.gnome.org/show_bug.cgi?id=696541
8012 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
8014 * gst/rtsp-server/rtsp-media-factory.c:
8015 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
8016 Instead use a GWeakRef which is safe to use
8017 This is a known GLib bug, see:
8018 https://bugzilla.gnome.org/show_bug.cgi?id=667145
8020 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
8022 * gst/rtsp-server/rtsp-client.c:
8023 * gst/rtsp-server/rtsp-media.c:
8024 * gst/rtsp-server/rtsp-media.h:
8025 * gst/rtsp-server/rtsp-sdp.c:
8026 * tests/check/gst/media.c:
8027 * tests/check/gst/rtspserver.c:
8028 rtsp-media/client: Reply to PLAY request with same type of Range
8029 Remember the type of Range from the PLAY request and use the same type for
8032 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
8034 * gst/rtsp-server/rtsp-client.c:
8035 * gst/rtsp-server/rtsp-client.h:
8036 * tests/check/gst/client.c:
8037 rtsp-client: expose uri
8039 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
8041 * tests/check/gst/mediafactory.c:
8042 tests: Hold ref while creating second media
8043 To test if the media aren't shared, make sure we keep the first one while creating a second
8044 otherwise the same memory address may be reused.
8046 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
8049 configure: remove out-of-date comment
8051 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
8054 .gitignore: ignore more build files
8056 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
8058 * tests/check/Makefile.am:
8059 tests: use right _LIBS variable for gst-plugins-base libs
8061 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8063 * tests/check/Makefile.am:
8064 check: add librtp to libs
8066 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
8068 * tests/check/gst/rtspserver.c:
8069 tests: Add test to check selecting a port the server will send from
8071 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
8073 * tests/check/gst/rtspserver.c:
8074 tests: Make sure packets are actually received
8076 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
8078 * gst/rtsp-server/rtsp-stream.c:
8079 stream: Select unicast address from pool if appropriate
8081 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
8083 * gst/rtsp-server/rtsp-stream.c:
8084 stream: Properties are always there in Gst 1.0
8086 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
8088 * tests/check/gst/addresspool.c:
8089 tests: Add tests for unicast addresses in pool
8091 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
8093 * gst/rtsp-server/rtsp-address-pool.c:
8094 * tests/check/gst/addresspool.c:
8095 address-pool: Verify that multicast addresses are used for multicast and vice-versa
8097 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
8099 * docs/libs/gst-rtsp-server-sections.txt:
8100 * gst/rtsp-server/rtsp-address-pool.c:
8101 * gst/rtsp-server/rtsp-address-pool.h:
8102 * gst/rtsp-server/rtsp-stream.c:
8103 * tests/check/gst/addresspool.c:
8104 address-pool: Add unicast addresses
8106 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
8109 * gst/rtsp-server/rtsp-server.c:
8110 * tests/check/gst/rtspserver.c:
8111 rtsp-server: Limit the number of threads per server instance
8112 If we exceed the maximum, just round robin the clients over the existing
8115 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
8117 * gst/rtsp-server/rtsp-server.c:
8118 rtsp-server: No need to store the GMainContext in the client context
8120 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
8122 * tests/check/gst/rtspserver.c:
8123 tests: Add test for client disconnection
8125 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
8127 * tests/check/gst/rtspserver.c:
8128 tests: Test client and session timeouts with multiple threads
8130 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
8132 * gst/rtsp-server/rtsp-address-pool.c:
8133 * gst/rtsp-server/rtsp-auth.c:
8134 * gst/rtsp-server/rtsp-client.c:
8135 * gst/rtsp-server/rtsp-media-factory-uri.c:
8136 * gst/rtsp-server/rtsp-media-factory.c:
8137 * gst/rtsp-server/rtsp-media.c:
8138 * gst/rtsp-server/rtsp-mount-points.c:
8139 * gst/rtsp-server/rtsp-server.c:
8140 * gst/rtsp-server/rtsp-session-media.c:
8141 * gst/rtsp-server/rtsp-session-pool.c:
8142 * gst/rtsp-server/rtsp-session.c:
8143 Document locking and its order
8145 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
8147 * tests/check/gst/rtspserver.c:
8148 tests: Test that slow DESCRIBE don't block other clients
8150 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
8152 * tests/check/gst/client.c:
8153 tests: Add tests for client-requested multicast address
8155 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
8157 * docs/libs/gst-rtsp-server-sections.txt:
8158 docs: Put the various functions in the right sections
8160 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
8162 * docs/libs/gst-rtsp-server-docs.sgml:
8163 * docs/libs/gst-rtsp-server-sections.txt:
8164 * gst/rtsp-server/rtsp-address-pool.c:
8165 * gst/rtsp-server/rtsp-address-pool.h:
8166 docs: Generate docs for GstRTSPAddressPool
8168 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
8170 * gst/rtsp-server/rtsp-client.c:
8171 * gst/rtsp-server/rtsp-stream.c:
8172 * gst/rtsp-server/rtsp-stream.h:
8173 client: Check client provided addresses against the address pool
8175 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
8177 * gst/rtsp-server/rtsp-address-pool.c:
8178 * gst/rtsp-server/rtsp-address-pool.h:
8179 * tests/check/gst/addresspool.c:
8180 address-pool: Add API to request a specific address from the pool
8181 Also add relevant unit tests.
8183 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
8185 * tests/check/gst/mediafactory.c:
8186 tests: Check the passing around of a RTSPAddressPool
8187 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
8188 way down to the stream.
8190 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
8192 * tests/check/gst/addresspool.c:
8193 tests: Add more tests for the address pool
8195 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
8197 * gst/rtsp-server/rtsp-address-pool.c:
8198 address-pool: Fix off by one error
8199 When splitting a port range, the port after a skip is not part of range.
8201 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
8204 Automatic update of common submodule
8205 From 2de221c to 04c7a1e
8207 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
8210 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
8211 AM_CONFIG_HEADER was removed in automake 1.13
8212 https://bugzilla.gnome.org/show_bug.cgi?id=693368
8214 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
8217 Automatic update of common submodule
8218 From a942293 to 2de221c
8220 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8222 * gst/rtsp-server/rtsp-client.c:
8223 client: make sure the watch exists while sending data
8224 Protect the send_func with a lock. This allows us to wait for sending
8225 to complete before changing the send_func and user_data. We add an
8226 extra ref to the watch to make sure that it remains valid during
8228 When closing the connection, set the send_func to NULL
8229 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
8231 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8233 * tests/check/Makefile.am:
8234 tests: use GST_*_1_0 environment variables everywhere
8235 The _1_0 suffixed environment variables override the
8236 non-suffixed ones, so if we're in an environment that
8237 sets the _1_0 suffixed ones, such as jhbuild, we need
8238 to set those to make sure ours actually always get
8241 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8244 Automatic update of common submodule
8245 From acb04d9 to a942293
8247 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8249 * gst/rtsp-server/rtsp-client.c:
8250 rtsp-client: set the client backlog
8251 Set the client backlog to a reasonable default
8253 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
8255 * gst/rtsp-server/rtsp-media.c:
8256 rtsp-media: Make the element a constructor parameter
8257 https://bugzilla.gnome.org/show_bug.cgi?id=689594
8259 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
8261 * docs/libs/Makefile.am:
8262 docs: Link with gcov library when gcov is enabled
8263 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
8265 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8267 * gst/rtsp-server/rtsp-media.c:
8268 media: match prepare with unprepare
8269 Really unprepare when there were an equal amount of prepare calls.
8271 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8273 * gst/rtsp-server/rtsp-media.c:
8274 media: media has to be unprepared in finalize
8275 Because unprepare takes away the last ref on the media.
8277 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8279 * gst/rtsp-server/rtsp-client.c:
8280 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
8281 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
8282 We can't use the refcount to trigger unprepare because it is the unprepare call
8283 that removes the last refcount after all messages are consumed. What we should
8284 probably do is make a prepared refcount and only unprepare when the refcount
8287 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8289 * gst/rtsp-server/rtsp-media.c:
8290 media: let the source unref the last media ref
8291 the last ref to the media is held by the source so we don't need to add more ref
8292 and unrefs, we simply destroy the media when the source is gone.
8294 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8296 * gst/rtsp-server/rtsp-media.c:
8297 media: improve debug
8299 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8301 * gst/rtsp-server/rtsp-media.c:
8303 Make sure we are in the right state when collecting the position and duration.
8304 Only make ourselves PREPARED when we were previously PREPARING.
8306 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8308 * gst/rtsp-server/rtsp-media.c:
8309 media: use g_object_ref/unref for GObjects
8311 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
8313 * gst/rtsp-server/rtsp-client.c:
8314 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
8315 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
8316 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
8317 isn't being used anymore.
8319 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
8321 * gst/rtsp-server/rtsp-media.c:
8322 Fix compiler warning
8324 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
8326 * gst/rtsp-server/rtsp-media-factory-uri.c:
8327 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
8329 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8331 * gst/rtsp-server/rtsp-session-media.h:
8334 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8336 * gst/rtsp-server/rtsp-media.c:
8337 * tests/check/gst/media.c:
8338 media: avoid element leak
8340 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8342 * gst/rtsp-server/rtsp-media.c:
8343 media: require an element in media constructor
8345 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8347 * gst/rtsp-server/rtsp-client.c:
8348 Revert "client: TEARDOWN brings that state to Init again"
8349 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
8350 The object is already disposed, there is no point in setting the state.
8352 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8354 * gst/rtsp-server/rtsp-client.c:
8355 client: TEARDOWN brings that state to Init again
8357 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8359 * docs/libs/gst-rtsp-server-sections.txt:
8360 * examples/test-auth.c:
8361 * gst/rtsp-server/rtsp-auth.c:
8362 * gst/rtsp-server/rtsp-auth.h:
8363 * gst/rtsp-server/rtsp-client.c:
8364 * gst/rtsp-server/rtsp-client.h:
8365 * gst/rtsp-server/rtsp-media-factory-uri.c:
8366 * gst/rtsp-server/rtsp-media-factory-uri.h:
8367 * gst/rtsp-server/rtsp-media-factory.c:
8368 * gst/rtsp-server/rtsp-media-factory.h:
8369 * gst/rtsp-server/rtsp-media.c:
8370 * gst/rtsp-server/rtsp-media.h:
8371 * gst/rtsp-server/rtsp-mount-points.c:
8372 * gst/rtsp-server/rtsp-mount-points.h:
8373 * gst/rtsp-server/rtsp-sdp.c:
8374 * gst/rtsp-server/rtsp-server.c:
8375 * gst/rtsp-server/rtsp-server.h:
8376 * gst/rtsp-server/rtsp-session-media.c:
8377 * gst/rtsp-server/rtsp-session-media.h:
8378 * gst/rtsp-server/rtsp-session-pool.c:
8379 * gst/rtsp-server/rtsp-session-pool.h:
8380 * gst/rtsp-server/rtsp-session.c:
8381 * gst/rtsp-server/rtsp-session.h:
8382 * gst/rtsp-server/rtsp-stream-transport.c:
8383 * gst/rtsp-server/rtsp-stream-transport.h:
8384 * gst/rtsp-server/rtsp-stream.c:
8385 * gst/rtsp-server/rtsp-stream.h:
8386 * tests/check/gst/media.c:
8387 rtsp: make object details private
8388 Make all object details private
8389 Add methods to access private bits
8391 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8393 * tests/check/Makefile.am:
8394 * tests/check/gst/media.c:
8395 tests: add media tests
8397 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8399 * gst/rtsp-server/rtsp-media.c:
8400 media: check if prepared for some methods
8401 Check that the media object is prepared before doing seek and getting the
8402 current position etc.
8403 Add some g_return checks.
8405 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8407 * tests/check/Makefile.am:
8408 * tests/check/gst/mediafactory.c:
8409 tests: add mediafactory test
8411 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8413 * gst/rtsp-server/rtsp-stream.c:
8414 stream: improve debug
8416 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8418 * gst/rtsp-server/rtsp-media.c:
8419 * gst/rtsp-server/rtsp-media.h:
8420 media: unref pipeline in finalize to avoid leaking it
8422 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8424 * gst/rtsp-server/rtsp-media-factory-uri.c:
8425 * gst/rtsp-server/rtsp-media.c:
8426 rtsp: use gst_object_unref on GstObjects
8428 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8430 * gst/rtsp-server/rtsp-media-factory.c:
8431 media-factory: require an url
8433 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8435 * examples/test-uri.c:
8436 examples: fix include
8438 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8440 * gst/rtsp-server/rtsp-server.h:
8441 server: remove unused include
8443 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8445 * tests/check/Makefile.am:
8446 * tests/check/gst/mountpoints.c:
8447 tests: add test for mountpoints
8449 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8451 * gst/rtsp-server/rtsp-client.c:
8452 client: fix factory leak
8453 Keep the factory in the state object only for authorization checks and make
8454 sure we unref it on failure. Also don't keep invalid objects in the state
8457 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8459 * gst/rtsp-server/rtsp-mount-points.c:
8460 mounts: add g_return_if guards
8462 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8464 * tests/check/gst/client.c:
8465 tests: add more tests
8467 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8469 * gst/rtsp-server/rtsp-client.c:
8470 client: improve debug
8472 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8474 * gst/rtsp-server/rtsp-client.c:
8475 client: improve debug and fix leaks
8476 Cleanup the uri and session when there is a bad request.
8478 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8483 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8485 * tests/check/gst/client.c:
8486 test: add test for session in options request
8488 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8490 * gst/rtsp-server/rtsp-client.c:
8491 client: use 454 when session can't be found
8492 We should use 454 when a session can't be found because there was no session
8493 pool configured in the server. This is not a server configuration problem
8494 because the server on which the request is done might not be the same one that
8495 will keep the sessions for us and so it does not need to support sessions.
8497 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8499 * gst/rtsp-server/rtsp-client.c:
8500 client: only free connection when there is one
8501 It's possible that the client doesn't have a connection when we try to free it.
8503 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8505 * tests/check/Makefile.am:
8506 * tests/check/gst/client.c:
8507 tests: add unit test for the client object
8509 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8511 * gst/rtsp-server/rtsp-client.c:
8512 client: small cleanup
8514 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8516 * gst/rtsp-server/rtsp-client.h:
8517 client: remove unused include
8519 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8521 * gst/rtsp-server/rtsp-client.c:
8522 client: fix compilation
8524 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8526 * gst/rtsp-server/rtsp-client.c:
8527 client: call destroy without the lock
8529 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8531 * gst/rtsp-server/rtsp-client.c:
8532 * gst/rtsp-server/rtsp-client.h:
8533 client: make the client usable without a socket
8534 Make a method to let the client handle a message and a callback when the client
8535 wants us to send a response message back. This makes it possible to also use the
8536 client object without the sockets, which should make it easier to test.
8538 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8540 * gst/rtsp-server/rtsp-client.c:
8541 * gst/rtsp-server/rtsp-client.h:
8542 client: small cleanup
8544 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8546 * docs/libs/gst-rtsp-server-sections.txt:
8547 * gst/rtsp-server/rtsp-client.c:
8548 * gst/rtsp-server/rtsp-client.h:
8549 * gst/rtsp-server/rtsp-server.c:
8550 client: remove reference to server
8551 We don't need to keep a ref to the server
8553 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8555 * gst/rtsp-server/rtsp-client.c:
8556 * gst/rtsp-server/rtsp-client.h:
8558 Also add some g_return_if()
8560 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8562 * gst/rtsp-server/rtsp-client.c:
8563 client: log more errors
8565 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8567 * gst/rtsp-server/rtsp-client.c:
8568 client: fix compilation
8570 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8572 * gst/rtsp-server/rtsp-client.c:
8573 * gst/rtsp-server/rtsp-client.h:
8574 client: add generic close-after-send support
8575 Add a property to send_response() to close the connection after the response has
8576 been sent to the client.
8578 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8581 * docs/libs/gst-rtsp-server-docs.sgml:
8582 * docs/libs/gst-rtsp-server-sections.txt:
8583 * docs/libs/gst-rtsp-server.types:
8584 * examples/test-auth.c:
8585 * examples/test-launch.c:
8586 * examples/test-mp4.c:
8587 * examples/test-multicast.c:
8588 * examples/test-multicast2.c:
8589 * examples/test-ogg.c:
8590 * examples/test-readme.c:
8591 * examples/test-sdp.c:
8592 * examples/test-uri.c:
8593 * examples/test-video.c:
8594 * gst/rtsp-server/Makefile.am:
8595 * gst/rtsp-server/rtsp-auth.h:
8596 * gst/rtsp-server/rtsp-client.c:
8597 * gst/rtsp-server/rtsp-client.h:
8598 * gst/rtsp-server/rtsp-media-mapping.c:
8599 * gst/rtsp-server/rtsp-media-mapping.h:
8600 * gst/rtsp-server/rtsp-mount-points.c:
8601 * gst/rtsp-server/rtsp-mount-points.h:
8602 * gst/rtsp-server/rtsp-server.c:
8603 * gst/rtsp-server/rtsp-server.h:
8604 * gst/rtsp-server/rtsp-session-media.c:
8605 * gst/rtsp-server/rtsp-session-pool.c:
8606 * gst/rtsp-server/rtsp-session-pool.h:
8607 * tests/check/gst/rtspserver.c:
8608 MediaMapping -> MountPoints
8609 Describes better what the object manages.
8611 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8614 configure: bump required version of -base
8616 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8618 * gst/rtsp-server/rtsp-media.c:
8621 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8623 * gst/rtsp-server/rtsp-media.c:
8624 * gst/rtsp-server/rtsp-media.h:
8625 media: support more Range formats
8626 Use the new -base methods to convert the Range string into a seek start and stop
8629 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8631 * examples/test-launch.c:
8632 examples: fix whitespace
8634 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8636 * examples/test-auth.c:
8637 test-auth: add example of how to remove sessions
8638 Add an example of the session filter api.
8640 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8642 * examples/test-uri.c:
8643 test-uri: remove mapping example
8645 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8647 * examples/test-uri.c:
8648 test-uri: fix callback signature
8650 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8652 * gst/rtsp-server/rtsp-media-factory.c:
8653 factory: keep ref to factory while media active
8654 While the media from a factory is alive, keep a ref to the factory.
8655 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
8657 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8659 * gst/rtsp-server/rtsp-media-factory-uri.c:
8660 factory-uri: add some debug
8662 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8664 * gst/rtsp-server/rtsp-stream.c:
8665 stream: set udp sources to PLAYING
8666 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
8667 so that it doesn't cause our pipeline to produce ASYNC-DONE.
8669 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8671 * gst/rtsp-server/rtsp-media-factory-uri.c:
8672 factory-uri: take ref to factory
8673 Take a ref to the factory that we place in our list.
8675 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8677 * tests/Makefile.am:
8678 * tests/test-reuse.c:
8679 test: add test for server reuse
8680 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
8682 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
8684 * gst/rtsp-server/rtsp-server.c:
8685 server: start and stop multiple times
8686 Stop listening on the RTSP port when the GSource is removed, so clients
8687 can't connect and the server can be started again.
8688 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
8690 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8692 * gst/rtsp-server/rtsp-server.c:
8693 server: fix small leak
8695 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8697 * gst/rtsp-server/rtsp-media.c:
8698 media: unref source in finish_unprepare
8699 The source is created in prepare, unref it in finish_unprepare.
8700 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
8702 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
8704 * gst/rtsp-server/rtsp-client.c:
8705 * gst/rtsp-server/rtsp-media.c:
8706 rtsp-media: remove bus watch before finalizing
8707 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
8708 * An extra media ref is added for the bus watch. This extra ref is unreffed by
8709 the GDestroyNotify function.
8710 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
8711 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
8712 gst_rtsp_media_unprepare before unreffing the media.
8713 This way, the bus watch will be removed before the media is finalized.
8714 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
8716 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
8718 * gst/rtsp-server/rtsp-client.c:
8719 * gst/rtsp-server/rtsp-client.h:
8720 client: wait until the TEARDOWN response is sent to close the connection
8721 Responses can be sent async so we need to wait until the TEARDOWN response has
8722 been written before we close the connection to the client. This avoids the risk
8723 of writing/polling closed sockets.
8724 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
8726 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
8728 * gst/rtsp-server/rtsp-stream.c:
8729 rtsp-stream: plug socket leak
8730 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
8732 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
8735 Automatic update of common submodule
8736 From 6bb6951 to a72faea
8738 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
8740 * gst/rtsp-server/rtsp-media-factory-uri.c:
8741 rtsp-server: don't use deprecated API
8743 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
8745 * gst/rtsp-server/rtsp-client.c:
8746 rtsp-client: fix unused-but-set-variable compiler warning
8747 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
8749 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8752 * docs/libs/gst-rtsp-server-sections.txt:
8753 * gst/rtsp-server/rtsp-client.c:
8756 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8758 * examples/Makefile.am:
8759 * examples/test-multicast2.c:
8760 examples: add another multicast example
8761 Add an example for how to configure separate multicast ranges for each media
8764 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8766 * examples/test-multicast.c:
8769 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8771 * gst/rtsp-server/rtsp-client.c:
8772 * gst/rtsp-server/rtsp-media.c:
8773 * gst/rtsp-server/rtsp-session-media.c:
8774 * gst/rtsp-server/rtsp-session-media.h:
8775 * gst/rtsp-server/rtsp-stream-transport.c:
8776 * gst/rtsp-server/rtsp-stream-transport.h:
8777 stream: use the address managed by the stream
8778 Use the address managed by the stream for multicast. This allows us to have 1
8779 multicast address for each stream.
8780 Because the address is now managed by the stream we don't have to pass it around
8782 Set the address pool on the streams.
8784 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8786 * gst/rtsp-server/rtsp-client.c:
8787 * gst/rtsp-server/rtsp-media.c:
8788 * gst/rtsp-server/rtsp-stream.c:
8791 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8793 * gst/rtsp-server/rtsp-media.c:
8794 * gst/rtsp-server/rtsp-media.h:
8795 media: add signal for new streams
8796 This allows applications to listen for new streams and configure properties on
8797 them, like the address pool.
8799 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8801 * gst/rtsp-server/rtsp-media.c:
8802 media: configure address pool in new streams
8804 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8806 * gst/rtsp-server/rtsp-stream.c:
8807 * gst/rtsp-server/rtsp-stream.h:
8808 stream: add methods to deal with address pool
8809 Add methods to get and set the address pool for the stream
8810 Add method to allocate and get the multicast addresses for this stream.
8812 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8814 * docs/libs/gst-rtsp-server-sections.txt:
8815 * gst/rtsp-server/rtsp-media.c:
8816 * gst/rtsp-server/rtsp-media.h:
8817 media: remove MTU property
8818 It is a stream property
8820 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8822 * gst/rtsp-server/rtsp-client.c:
8823 client: set blocksize only on stream
8824 Set the blocksize only on the current stream.
8826 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8828 * gst/rtsp-server/rtsp-stream.c:
8829 stream: share src and sink sockets
8830 the allocated socket is in the used-socket property, not socket.
8832 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8834 * gst/rtsp-server/rtsp-address-pool.c:
8835 * gst/rtsp-server/rtsp-address-pool.h:
8836 * gst/rtsp-server/rtsp-client.c:
8837 * gst/rtsp-server/rtsp-session-media.c:
8838 * gst/rtsp-server/rtsp-session-media.h:
8839 * gst/rtsp-server/rtsp-stream-transport.c:
8840 * gst/rtsp-server/rtsp-stream-transport.h:
8841 * tests/check/gst/addresspool.c:
8842 rtsp: make address-pool return an address object
8843 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
8844 store more info in the structure and allows us to more easily return the address
8845 to the right pool when no longer needed.
8846 Pass the address to the StreamTransport so that we can return it to the pool
8847 when the stream transport is freed or changed.
8849 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8851 * examples/Makefile.am:
8852 * examples/test-multicast.c:
8853 examples: add multicast example
8854 Show how to set up the multicast address pool so that media can be
8855 server with multicast.
8857 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8859 * gst/rtsp-server/rtsp-client.c:
8860 * gst/rtsp-server/rtsp-media-factory.c:
8861 * gst/rtsp-server/rtsp-media-factory.h:
8862 * gst/rtsp-server/rtsp-media.c:
8863 * gst/rtsp-server/rtsp-media.h:
8864 rtsp: use AddressPool
8865 Remove the multicast_group property.
8866 Use the configured addresspool to allocate multicast addresses.
8868 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8870 * gst/rtsp-server/rtsp-address-pool.c:
8871 * gst/rtsp-server/rtsp-address-pool.h:
8872 address-pool: add clear method
8874 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8876 * gst/rtsp-server/rtsp-address-pool.c:
8877 address-pool: small cleanups
8879 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8881 * tests/check/Makefile.am:
8882 * tests/check/gst/addresspool.c:
8883 tests: add addresspool unit test
8885 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8887 * gst/rtsp-server/Makefile.am:
8888 * gst/rtsp-server/rtsp-address-pool.c:
8889 * gst/rtsp-server/rtsp-address-pool.h:
8890 address-pool: add object to manage multicast addresses
8891 Make an object that can manage a rage of multicast addresses and ports.
8893 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8895 * gst/rtsp-server/rtsp-server.c:
8896 server: set default max-threads property
8898 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8900 * gst/rtsp-server/rtsp-media.c:
8901 media: wait for concurrent _prepare
8902 If a prepare is busy, wait for the result.
8904 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8906 * gst/rtsp-server/rtsp-media.c:
8907 media: add lock around message handler
8908 We don't want to dispatch messages while we are still processing the result of
8911 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8913 * gst/rtsp-server/rtsp-media.c:
8914 * gst/rtsp-server/rtsp-media.h:
8915 media: add lock to protect state changes
8917 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8919 * gst/rtsp-server/rtsp-stream.c:
8920 * gst/rtsp-server/rtsp-stream.h:
8923 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8925 * gst/rtsp-server/rtsp-stream-transport.c:
8926 * gst/rtsp-server/rtsp-stream-transport.h:
8927 * gst/rtsp-server/rtsp-stream.c:
8928 stream-transport: add keep-alive method
8930 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8932 * gst/rtsp-server/rtsp-stream-transport.c:
8933 * gst/rtsp-server/rtsp-stream-transport.h:
8934 * gst/rtsp-server/rtsp-stream.c:
8935 stream-transport: add method to handle RTP/RTCP
8936 Call new methods instead of poking into the structures directly.
8938 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8940 * gst/rtsp-server/rtsp-session-media.c:
8941 * gst/rtsp-server/rtsp-session-media.h:
8942 session-media: add locking
8944 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8946 * gst/rtsp-server/rtsp-session.c:
8947 * gst/rtsp-server/rtsp-session.h:
8948 session: add locking
8950 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8952 * gst/rtsp-server/rtsp-server.c:
8953 server: free old socket
8955 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8957 * gst/rtsp-server/rtsp-media-mapping.c:
8958 * gst/rtsp-server/rtsp-media-mapping.h:
8959 mapping: add locking
8961 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8963 * gst/rtsp-server/rtsp-media-factory.c:
8964 media-factory: add locking
8966 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8968 * gst/rtsp-server/rtsp-auth.c:
8969 * gst/rtsp-server/rtsp-auth.h:
8972 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8974 * gst/rtsp-server/rtsp-server.c:
8975 * gst/rtsp-server/rtsp-server.h:
8976 server: add max-thread property
8978 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8980 * gst/rtsp-server/rtsp-server.c:
8981 * gst/rtsp-server/rtsp-server.h:
8982 server: use a threadpool for the mainloops
8984 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8986 * gst/rtsp-server/rtsp-client.c:
8987 * gst/rtsp-server/rtsp-client.h:
8988 client: rename method
8989 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
8990 don't really create the client from the socket, we use the socket for the
8993 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8995 * gst/rtsp-server/rtsp-client.c:
8996 * gst/rtsp-server/rtsp-client.h:
8997 * gst/rtsp-server/rtsp-server.c:
8998 server: rework maincontext handling in clients
8999 Make a separate method to attach a client to a MainContext.
9000 Let the server decide in what GMainContext the client will operate and give this
9001 context to the client in attach. Then the server can later decide to use a
9002 separate thread for each client or just use the mainthread.
9004 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9006 * gst/rtsp-server/rtsp-client.c:
9007 * gst/rtsp-server/rtsp-session.c:
9008 * gst/rtsp-server/rtsp-session.h:
9009 session: move session header code in session object
9011 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
9015 * examples/test-auth.c:
9016 * examples/test-launch.c:
9017 * examples/test-mp4.c:
9018 * examples/test-ogg.c:
9019 * examples/test-readme.c:
9020 * examples/test-sdp.c:
9021 * examples/test-uri.c:
9022 * examples/test-video.c:
9023 * gst/rtsp-server/rtsp-auth.c:
9024 * gst/rtsp-server/rtsp-auth.h:
9025 * gst/rtsp-server/rtsp-client.c:
9026 * gst/rtsp-server/rtsp-client.h:
9027 * gst/rtsp-server/rtsp-media-factory-uri.c:
9028 * gst/rtsp-server/rtsp-media-factory-uri.h:
9029 * gst/rtsp-server/rtsp-media-factory.c:
9030 * gst/rtsp-server/rtsp-media-factory.h:
9031 * gst/rtsp-server/rtsp-media-mapping.c:
9032 * gst/rtsp-server/rtsp-media-mapping.h:
9033 * gst/rtsp-server/rtsp-media.c:
9034 * gst/rtsp-server/rtsp-media.h:
9035 * gst/rtsp-server/rtsp-params.c:
9036 * gst/rtsp-server/rtsp-params.h:
9037 * gst/rtsp-server/rtsp-sdp.c:
9038 * gst/rtsp-server/rtsp-sdp.h:
9039 * gst/rtsp-server/rtsp-server.c:
9040 * gst/rtsp-server/rtsp-server.h:
9041 * gst/rtsp-server/rtsp-session-media.c:
9042 * gst/rtsp-server/rtsp-session-media.h:
9043 * gst/rtsp-server/rtsp-session-pool.c:
9044 * gst/rtsp-server/rtsp-session-pool.h:
9045 * gst/rtsp-server/rtsp-session.c:
9046 * gst/rtsp-server/rtsp-session.h:
9047 * gst/rtsp-server/rtsp-stream-transport.c:
9048 * gst/rtsp-server/rtsp-stream-transport.h:
9049 * gst/rtsp-server/rtsp-stream.c:
9050 * gst/rtsp-server/rtsp-stream.h:
9051 * tests/check/gst/rtspserver.c:
9052 * tests/test-cleanup.c:
9055 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
9057 * gst/rtsp-server/rtsp-media.c:
9058 * gst/rtsp-server/rtsp-session-media.c:
9059 * gst/rtsp-server/rtsp-session.c:
9060 rtsp-server: added annotations to indicate type of ownership transfer of return values
9061 https://bugzilla.gnome.org/show_bug.cgi?id=680777
9063 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
9066 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
9068 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
9071 * bindings/Makefile.am:
9072 * bindings/vala/Makefile.am:
9073 * bindings/vala/gst-rtsp-server-0.10.deps:
9074 * bindings/vala/gst-rtsp-server-0.10.vapi:
9075 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
9076 * bindings/vala/packages/gst-rtsp-server-0.10.files:
9077 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
9078 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9079 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
9081 bindings: remove vala bindings
9082 They'll be reunited with the other GStreamer bindings
9083 https://bugzilla.gnome.org/show_bug.cgi?id=680777
9085 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9087 * gst/rtsp-server/rtsp-client.c:
9088 * gst/rtsp-server/rtsp-session-media.c:
9089 * gst/rtsp-server/rtsp-session-media.h:
9090 * gst/rtsp-server/rtsp-stream-transport.c:
9091 * gst/rtsp-server/rtsp-stream-transport.h:
9092 rtsp: only create transport when needed
9093 Only create the StreamTransport when configured.
9095 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9097 * gst/rtsp-server/rtsp-client.c:
9098 client: small cleanup
9100 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9102 * gst/rtsp-server/rtsp-client.c:
9103 * gst/rtsp-server/rtsp-client.h:
9104 * gst/rtsp-server/rtsp-stream-transport.c:
9105 * gst/rtsp-server/rtsp-stream-transport.h:
9106 rtsp: refactor configuration of transport
9107 Move the configuration of the transport to a place where it makes
9110 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9112 * gst/rtsp-server/rtsp-client.c:
9113 client: refactor transport parsing
9115 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9117 * gst/rtsp-server/rtsp-client.c:
9118 client: refuse to change the MTU on shared media
9119 If we change the MTU of chared media, it changes for all clients.
9120 We don't want to set the MTU to something large for clients that
9123 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9125 * examples/test-mp4.c:
9126 * gst/rtsp-server/rtsp-media.c:
9127 small fixes to docs and debug
9129 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9131 * gst/rtsp-server/rtsp-stream.c:
9132 stream: transports must already have been removed
9134 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9136 * gst/rtsp-server/rtsp-media.c:
9137 * gst/rtsp-server/rtsp-stream.c:
9138 * gst/rtsp-server/rtsp-stream.h:
9139 stream: improve join and leave of the pipeline
9141 Do the cleanup properly
9144 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9146 * gst/rtsp-server/rtsp-media.c:
9147 media: move unprepare below default implementation
9148 Makes it easier to find the default implementation
9150 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9152 * gst/rtsp-server/rtsp-media.c:
9153 media: signal unprepared when we actually finish
9155 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9157 * gst/rtsp-server/rtsp-media.c:
9158 media: no need to unlock, unprepare does that when needed
9160 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9162 * docs/libs/gst-rtsp-server-sections.txt:
9163 * gst/rtsp-server/rtsp-media-factory.h:
9164 * gst/rtsp-server/rtsp-media-mapping.c:
9165 * gst/rtsp-server/rtsp-media.h:
9166 * gst/rtsp-server/rtsp-params.c:
9167 * gst/rtsp-server/rtsp-server.c:
9168 * gst/rtsp-server/rtsp-session-pool.h:
9169 * gst/rtsp-server/rtsp-session.c:
9170 * gst/rtsp-server/rtsp-session.h:
9171 * gst/rtsp-server/rtsp-stream-transport.h:
9172 * gst/rtsp-server/rtsp-stream.h:
9175 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9177 * gst/rtsp-server/rtsp-client.c:
9178 * gst/rtsp-server/rtsp-media-mapping.h:
9179 * gst/rtsp-server/rtsp-media.c:
9180 * gst/rtsp-server/rtsp-media.h:
9181 * gst/rtsp-server/rtsp-server.h:
9182 * gst/rtsp-server/rtsp-stream.c:
9183 * gst/rtsp-server/rtsp-stream.h:
9184 rtsp: fix MTU setting
9185 Fix setting of the MTU. There is no need for a vmethod.
9187 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9192 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9195 configure: bump version number after refactoring
9197 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9199 * gst/rtsp-server/Makefile.am:
9200 * gst/rtsp-server/rtsp-client.c:
9201 * gst/rtsp-server/rtsp-client.h:
9202 * gst/rtsp-server/rtsp-media-factory-uri.c:
9203 * gst/rtsp-server/rtsp-media-factory.c:
9204 * gst/rtsp-server/rtsp-media-factory.h:
9205 * gst/rtsp-server/rtsp-media.c:
9206 * gst/rtsp-server/rtsp-media.h:
9207 * gst/rtsp-server/rtsp-sdp.c:
9208 * gst/rtsp-server/rtsp-session-media.c:
9209 * gst/rtsp-server/rtsp-session-media.h:
9210 * gst/rtsp-server/rtsp-session.c:
9211 * gst/rtsp-server/rtsp-session.h:
9212 * gst/rtsp-server/rtsp-stream-transport.c:
9213 * gst/rtsp-server/rtsp-stream-transport.h:
9214 * gst/rtsp-server/rtsp-stream.c:
9215 * gst/rtsp-server/rtsp-stream.h:
9216 rtsp: massive refactoring
9217 Make GObjects from the remaining simple structures.
9218 Remove GstRTSPSessionStream, it's not needed.
9219 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
9220 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
9221 a GstRTSPStream should be transported to a client.
9222 Rename GstRTSPMediaFactory::get_element -> create_element because that
9223 more accurately describes what it does.
9224 Make nice methods instead of poking in the structures.
9225 Move some methods inside the relevant object source code.
9226 Use GPtrArray to store objects instead of plain arrays, it is more
9227 natural and allows us to more easily clean up.
9228 Move the allocation of udp ports to the Stream object. The Stream object
9229 contains the elements needed to stream the media to a client.
9230 Improve the prepare and unprepare methods. Unprepare should now undo
9231 everything prepare did. Improve also async unprepare when doing EOS on
9232 shutdown. Make sure we always unprepare correctly.
9234 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
9236 * gst/rtsp-server/rtsp-client.c:
9237 rtsp-client: Unref server address clients connected to
9238 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
9240 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
9242 * gst/rtsp-server/rtsp-server.c:
9243 rtsp-server: don't ref server socket if it is NULL
9244 Fixes test_bind_already_in_use unit test again after commit 6a497440.
9245 https://bugzilla.gnome.org/show_bug.cgi?id=686644
9247 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
9249 * tests/check/Makefile.am:
9250 tests: Add libgio link dependency
9251 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
9253 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9255 * gst/rtsp-server/rtsp-media-mapping.c:
9256 * gst/rtsp-server/rtsp-media-mapping.h:
9257 rtsp-media-mapping: rename find_media vfunc to find_factory
9258 The virtual method and class method should have the same name
9259 so it is correctly represented in GIR file
9260 https://bugzilla.gnome.org/show_bug.cgi?id=680777
9262 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
9264 * gst/rtsp-server/rtsp-auth.c:
9265 * gst/rtsp-server/rtsp-client.c:
9266 * gst/rtsp-server/rtsp-media-factory-uri.c:
9267 * gst/rtsp-server/rtsp-media-factory.c:
9268 * gst/rtsp-server/rtsp-media-mapping.c:
9269 * gst/rtsp-server/rtsp-media.c:
9270 * gst/rtsp-server/rtsp-server.c:
9271 * gst/rtsp-server/rtsp-session-pool.c:
9272 * gst/rtsp-server/rtsp-session.c:
9273 rtsp-server: fixed comments and GIR annotations
9274 https://bugzilla.gnome.org/show_bug.cgi?id=680777
9276 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
9278 * gst/rtsp-server/rtsp-media-mapping.c:
9279 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
9281 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
9283 * gst/rtsp-server/rtsp-server.c:
9284 rtsp-server: allow binding on port 0 (binds on a random port)
9286 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
9288 * gst/rtsp-server/rtsp-server.c:
9289 * gst/rtsp-server/rtsp-server.h:
9290 rtsp-server: add bound-port property
9291 bound-port can be used to retrieve the port number when the server is bound on
9292 port 0, which binds on a random port.
9294 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
9296 * gst/rtsp-server/rtsp-media-factory.c:
9297 * gst/rtsp-server/rtsp-media-factory.h:
9298 rtsp-media-factory: make ::get_element overridable by GI bindings
9299 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
9300 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
9301 as the invoker for ::get_element(), making it overridable by GI generated
9304 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
9306 * gst/rtsp-server/rtsp-media-factory-uri.c:
9307 rtsp-media-factory-uri: don't autoplug parsers in a loop
9308 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
9311 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
9313 * gst/rtsp-server/Makefile.am:
9314 Explicitly link against gio. Fix link error on mac.
9316 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
9318 * gst/rtsp-server/rtsp-session.c:
9319 session: add ttl to the transport header in SETUP
9320 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
9322 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
9324 * gst/rtsp-server/rtsp-client.c:
9325 * gst/rtsp-server/rtsp-client.h:
9326 * gst/rtsp-server/rtsp-media.c:
9327 client: Use client transport settings for multicast if allowed.
9328 This patch makes it possible for the client to send transport settings for
9329 multicast (destination && ttl). Client settings must be explicitly allowed or
9330 the server will use its own settings.
9331 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
9333 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
9336 Automatic update of common submodule
9337 From 6c0b52c to 6bb6951
9339 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
9341 * gst/rtsp-server/rtsp-client.c:
9342 rtsp-client: do not destroy the rtsp watch
9343 Don't destroy the client watch while dispatching. The rtsp watch is
9344 automatically destroyed after the rtsp watch function closed() has
9346 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
9348 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
9351 Automatic update of common submodule
9352 From 4f962f7 to 6c0b52c
9354 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
9356 * gst/rtsp-server/rtsp-media.c:
9357 media: fix check for seekability
9359 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9361 * gst/rtsp-server/rtsp-client.c:
9362 client: use more GIO
9363 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
9365 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9367 * gst/rtsp-server/rtsp-server.c:
9368 server: remove obsolete includes
9370 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
9372 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
9373 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
9374 be available in "on_new_ssrc". The transports are added in
9375 gst_rtsp_media_set_state when going to PLAYING state. However,
9376 "on_new_ssrc" might be called before this happens.
9377 https://bugzilla.gnome.org/show_bug.cgi?id=683304
9379 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
9381 * gst/rtsp-server/rtsp-client.c:
9382 * gst/rtsp-server/rtsp-client.h:
9383 rtsp-client: add signals for rtsp requests (fixes #683287)
9385 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
9387 * gst/rtsp-server/rtsp-client.c:
9388 * gst/rtsp-server/rtsp-client.h:
9389 add new-session signal to rtsp-client (fixes #683058)
9391 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
9394 Automatic update of common submodule
9395 From 668acee to 4f962f7
9397 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
9399 * gst/rtsp-server/rtsp-server.c:
9400 * tests/check/gst/rtspserver.c:
9401 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
9402 Do not assume that *error is set in g_socket_address_enumerator_next.
9403 Added test_bind_already_in_use unit-test.
9404 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
9406 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
9409 Automatic update of common submodule
9410 From 94ccf4c to 668acee
9412 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
9414 * gst/rtsp-server/rtsp-client.c:
9415 * gst/rtsp-server/rtsp-client.h:
9416 rtsp-client: make create_sdp virtual method
9417 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
9419 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9422 Automatic update of common submodule
9423 From 98e386f to 94ccf4c
9425 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9427 * gst/rtsp-server/rtsp-client.c:
9430 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
9432 * gst/rtsp-server/rtsp-client.c:
9433 * gst/rtsp-server/rtsp-client.h:
9434 * gst/rtsp-server/rtsp-server.c:
9435 * gst/rtsp-server/rtsp-server.h:
9436 rtsp-server: use an existing socket to establish HTTP tunnel
9437 Make it possible to transfer a socket from an HTTP server to be used as
9438 an RTSP over HTTP tunnel.
9440 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
9442 * gst/rtsp-server/rtsp-client.c:
9443 * gst/rtsp-server/rtsp-media.c:
9444 * gst/rtsp-server/rtsp-media.h:
9445 rtsp: Handle the blocksize parameter
9446 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
9448 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
9450 * tests/check/Makefile.am:
9451 * tests/check/gst/rtspserver.c:
9452 Have unit test get header from source dir, not installed dir
9453 This makes compilation of unit tests work in a build directory other
9454 than the source directory.
9455 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
9457 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
9459 * gst/rtsp-server/rtsp-media.c:
9460 rtsp-media: update for gst_element_make_from_uri() changes
9462 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
9465 * tests/Makefile.am:
9466 * tests/check/Makefile.am:
9467 * tests/check/gst/rtspserver.c:
9469 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
9471 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
9473 * gst/rtsp-server/rtsp-media.c:
9474 rtsp-media: don't collect media stats when going to NULL
9475 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
9477 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9479 * gst/rtsp-server/rtsp-client.c:
9480 client: don't leak transports
9482 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
9484 * gst/rtsp-server/rtsp-client.c:
9485 rtsp-client: free transport on no_stream in SETUP handler
9487 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
9489 * gst/rtsp-server/rtsp-client.c:
9490 rtsp-client: changed session media iteration
9491 In client_unlink_session: now don't iterate in session->medias
9492 list where items are removed by gst_rtsp_session_release_media.
9493 Instead, repeatedly remove the first item.
9495 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
9497 * gst/rtsp-server/rtsp-client.c:
9498 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
9499 GstRTSPSessionMedia is not a GObject type. When the
9500 GstRTSPSession is freed, it will free the media.
9502 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
9504 * gst/rtsp-server/rtsp-media-factory.c:
9505 factory: plug pad leak in collect_streams
9506 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
9507 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
9508 will take one reference, and the other reference will otherwise
9511 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
9514 configure: suppress some warnings when debug is disabled
9515 Warnings about unused variables should be suppressed if core has the
9516 debug system disabled.
9517 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
9519 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9521 * docs/libs/Makefile.am:
9522 docs: fix build in uninstalled setup
9523 Include gst-plugins-base libs properly.
9525 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
9527 * docs/libs/gst-rtsp-server.types:
9528 docs: include headers defining rtsp-server object types
9529 Fixes compiler warnings during docs build.
9530 https://bugzilla.gnome.org/show_bug.cgi?id=676824
9532 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
9535 configure: Add warning flags for compiler when configuring
9536 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
9538 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
9541 Automatic update of common submodule
9542 From 03a0e57 to 98e386f
9544 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
9547 Automatic update of common submodule
9548 From 1fab359 to 03a0e57
9550 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
9552 * gst/rtsp-server/rtsp-client.c:
9553 client: fix GSocketAddress leak in gst_rtsp_client_accept
9554 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
9556 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
9559 Automatic update of common submodule
9560 From f1b5a96 to 1fab359
9562 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9565 Automatic update of common submodule
9566 From 92b7266 to f1b5a96
9568 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9571 Automatic update of common submodule
9572 From ec1c4a8 to 92b7266
9574 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9577 Automatic update of common submodule
9578 From 3429ba6 to ec1c4a8
9580 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
9582 * gst/rtsp-server/rtsp-auth.c:
9583 * gst/rtsp-server/rtsp-client.c:
9584 * gst/rtsp-server/rtsp-media-factory-uri.c:
9585 * gst/rtsp-server/rtsp-server.c:
9586 rtsp: fix compiler warnings
9587 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
9589 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9592 Automatic update of common submodule
9593 From dc70203 to 3429ba6
9595 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9597 * gst/rtsp-server/rtsp-client.c:
9598 * gst/rtsp-server/rtsp-media-factory.c:
9599 * gst/rtsp-server/rtsp-media-factory.h:
9600 * gst/rtsp-server/rtsp-media.c:
9601 * gst/rtsp-server/rtsp-media.h:
9602 * gst/rtsp-server/rtsp-server.c:
9603 * gst/rtsp-server/rtsp-server.h:
9604 * gst/rtsp-server/rtsp-session-pool.c:
9605 * gst/rtsp-server/rtsp-session-pool.h:
9606 rtsp-server: port to new thread API
9608 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9611 Automatic update of common submodule
9612 From 6db25be to dc70203
9614 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9616 * gst/rtsp-server/rtsp-auth.c:
9617 * gst/rtsp-server/rtsp-auth.h:
9618 * gst/rtsp-server/rtsp-client.c:
9619 rtsp-server: Fix compilation and compiler warnings
9621 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9625 * gst/rtsp-server/Makefile.am:
9626 configure: Modernize autotools setup a bit
9627 Also we now only create tar.bz2 and tar.xz tarballs.
9629 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9632 Automatic update of common submodule
9633 From 464fe15 to 6db25be
9635 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9638 Automatic update of common submodule
9639 From 7fda524 to 464fe15
9641 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9644 * docs/libs/Makefile.am:
9645 * docs/version.entities.in:
9647 * gst/rtsp-server/Makefile.am:
9648 * pkgconfig/Makefile.am:
9649 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
9650 * pkgconfig/gstreamer-rtsp-server.pc.in:
9651 * tests/Makefile.am:
9652 rtsp-server: Update versioning
9654 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9656 Merge remote-tracking branch 'origin/0.10'
9658 gst/rtsp-server/rtsp-session-pool.c
9660 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9662 * gst/rtsp-server/rtsp-session-pool.c:
9663 rtsp-server: Don't use deprecated GLib API
9665 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9667 Replace master with 0.11
9669 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9671 Merge branch 'master' into 0.11
9673 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9675 Merge branch 'master' into 0.11
9677 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
9680 A couple minor typo fixes
9682 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9684 * gst/rtsp-server/rtsp-media.c:
9685 media: fix state of the appqueue
9687 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9689 * gst/rtsp-server/rtsp-media-factory-uri.c:
9690 factory: use videoconvert
9692 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9694 * gst/rtsp-server/rtsp-media-factory-uri.c:
9695 factory: change to new style caps
9697 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9699 * gst/rtsp-server/rtsp-client.c:
9700 * gst/rtsp-server/rtsp-client.h:
9701 * gst/rtsp-server/rtsp-media-factory-uri.c:
9702 * gst/rtsp-server/rtsp-media.c:
9703 * gst/rtsp-server/rtsp-server.c:
9704 * gst/rtsp-server/rtsp-server.h:
9705 * gst/rtsp-server/rtsp-session-pool.c:
9706 rtsp-server: port to GIO
9709 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9712 configure: fix build
9714 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9717 docs: fix for gst_rtsp_server_set_port() -> _set_service()
9718 https://bugzilla.gnome.org/show_bug.cgi?id=666548
9720 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9723 * examples/Makefile.am:
9724 First rule of gst-rtsp-server club: don't talk about gst-phonon
9726 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9729 * pkgconfig/Makefile.am:
9730 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
9731 * pkgconfig/gstreamer-rtsp-server.pc.in:
9732 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
9733 For consistency with all other modules.
9735 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9737 * gst/rtsp-server/rtsp-client.c:
9738 rtsp-client: update for new map API
9740 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9743 * bindings/Makefile.am:
9744 * bindings/python/Makefile.am:
9745 * bindings/python/arg-types.py:
9746 * bindings/python/codegen/Makefile.am:
9747 * bindings/python/codegen/__init__.py:
9748 * bindings/python/codegen/argtypes.py:
9749 * bindings/python/codegen/code-coverage.py:
9750 * bindings/python/codegen/codegen.py:
9751 * bindings/python/codegen/definitions.py:
9752 * bindings/python/codegen/defsparser.py:
9753 * bindings/python/codegen/docextract.py:
9754 * bindings/python/codegen/docgen.py:
9755 * bindings/python/codegen/fileprefix.override:
9756 * bindings/python/codegen/fileprefixmodule.c:
9757 * bindings/python/codegen/h2def.py:
9758 * bindings/python/codegen/mergedefs.py:
9759 * bindings/python/codegen/mkskel.py:
9760 * bindings/python/codegen/override.py:
9761 * bindings/python/codegen/reversewrapper.py:
9762 * bindings/python/codegen/scmexpr.py:
9763 * bindings/python/rtspserver-types.defs:
9764 * bindings/python/rtspserver.defs:
9765 * bindings/python/rtspserver.override:
9766 * bindings/python/rtspservermodule.c:
9767 * bindings/python/test.py:
9769 python: remove pygst-based python bindings
9770 pygi is the future, apparently.
9772 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
9775 Automatic update of common submodule
9776 From c463bc0 to 7fda524
9778 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9781 Automatic update of common submodule
9782 From 2a59016 to c463bc0
9784 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9787 Automatic update of common submodule
9788 From 0807187 to 2a59016
9790 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9793 Automatic update of common submodule
9794 From 11f0cd5 to 0807187
9796 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9798 * examples/test-auth.c:
9799 example: update for new caps
9801 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9803 * examples/test-video.c:
9804 * gst/rtsp-server/rtsp-client.c:
9805 * gst/rtsp-server/rtsp-media-factory-uri.c:
9806 * gst/rtsp-server/rtsp-media.c:
9807 * gst/rtsp-server/rtsp-media.h:
9808 * gst/rtsp-server/rtsp-session.c:
9809 * gst/rtsp-server/rtsp-session.h:
9810 rtsp-server: port some more to 0.11
9812 Remove bufferlist stuff
9814 Add queue before appsink now that preroll-queue-len is gone.
9815 Update for request pad changes.
9817 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9819 Merge branch 'master' into 0.11
9821 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
9823 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9824 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
9825 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
9827 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
9829 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
9830 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
9831 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
9833 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9835 Merge branch 'master' into 0.11
9837 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9839 * gst/rtsp-server/rtsp-media.c:
9840 * gst/rtsp-server/rtsp-media.h:
9841 media: add a seekable boolean
9842 Maintain the seekable state with a new variable instead of reusing the
9845 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
9847 * gst/rtsp-server/rtsp-media.c:
9848 Disallow seek in live media
9850 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9852 Merge branch 'master' into 0.11
9854 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
9856 * gst/rtsp-server/rtsp-server.c:
9857 #ifdef statements for windows socket creation were missing
9859 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
9862 Automatic update of common submodule
9863 From a39eb83 to 11f0cd5
9865 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
9868 Automatic update of common submodule
9869 From 605cd9a to a39eb83
9871 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9873 Merge branch 'master' into 0.11
9875 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9877 * gst/rtsp-server/rtsp-client.c:
9878 client: use method to access property
9880 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9882 * gst/rtsp-server/rtsp-media-factory.c:
9883 * gst/rtsp-server/rtsp-media-factory.h:
9884 media-factory: add protocols property
9885 Add a property to configure the allowed protocols in the media created from the
9888 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9890 * gst/rtsp-server/rtsp-media-factory.c:
9891 * gst/rtsp-server/rtsp-media-factory.h:
9892 media-factory: add media-configure signal
9893 Add signal to allow the application to configure the media after it was created
9896 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9898 * gst/rtsp-server/rtsp-client.c:
9899 client: use method to access property
9901 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9903 * gst/rtsp-server/rtsp-media-factory.c:
9904 * gst/rtsp-server/rtsp-media-factory.h:
9905 media-factory: add protocols property
9906 Add a property to configure the allowed protocols in the media created from the
9909 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9911 * gst/rtsp-server/rtsp-media-factory.c:
9912 * gst/rtsp-server/rtsp-media-factory.h:
9913 media-factory: add media-configure signal
9914 Add signal to allow the application to configure the media after it was created
9917 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9919 Merge branch 'master' into 0.11
9921 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9923 * gst/rtsp-server/rtsp-client.c:
9924 client: use media multicast group
9926 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9928 * gst/rtsp-server/rtsp-media-factory.h:
9929 * gst/rtsp-server/rtsp-server.h:
9930 * gst/rtsp-server/rtsp-session-pool.h:
9931 * gst/rtsp-server/rtsp-session.h:
9934 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
9936 * gst/rtsp-server/rtsp-client.c:
9937 * gst/rtsp-server/rtsp-sdp.h:
9938 sdp: copy and free the server ip address
9939 Copy and free the server ip address to make memory management easier later.
9941 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9943 * gst/rtsp-server/rtsp-media-factory.c:
9944 media-factory: configure multicast in media
9946 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9948 * gst/rtsp-server/rtsp-media.c:
9949 * gst/rtsp-server/rtsp-media.h:
9950 media: add property for multicast group
9951 Add a property to configure the multicast group in the media.
9952 Based on patches from Marc Leeman and Robert Krakora.
9954 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9956 * gst/rtsp-server/rtsp-media-factory.c:
9957 * gst/rtsp-server/rtsp-media-factory.h:
9958 media-factory: add property for multicast group
9959 Add a property to configure the multicast group in the media factory.
9960 Based on patches from Marc Leeman and Robert Krakora.
9962 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9964 * gst/rtsp-server/rtsp-client.c:
9965 client: do configuration of transport in one place
9966 Move the configuration of the transport destination address to where we also
9967 configure the other bits.
9969 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9971 * gst/rtsp-server/rtsp-client.c:
9972 client: use media multicast group
9974 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9976 * gst/rtsp-server/rtsp-media-factory.h:
9977 * gst/rtsp-server/rtsp-server.h:
9978 * gst/rtsp-server/rtsp-session-pool.h:
9979 * gst/rtsp-server/rtsp-session.h:
9982 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
9984 * gst/rtsp-server/rtsp-client.c:
9985 * gst/rtsp-server/rtsp-sdp.h:
9986 sdp: copy and free the server ip address
9987 Copy and free the server ip address to make memory management easier later.
9989 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9991 * gst/rtsp-server/rtsp-media-factory.c:
9992 media-factory: configure multicast in media
9994 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9996 * gst/rtsp-server/rtsp-media.c:
9997 * gst/rtsp-server/rtsp-media.h:
9998 media: add property for multicast group
9999 Add a property to configure the multicast group in the media.
10000 Based on patches from Marc Leeman and Robert Krakora.
10002 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10004 * gst/rtsp-server/rtsp-media-factory.c:
10005 * gst/rtsp-server/rtsp-media-factory.h:
10006 media-factory: add property for multicast group
10007 Add a property to configure the multicast group in the media factory.
10008 Based on patches from Marc Leeman and Robert Krakora.
10010 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10012 * gst/rtsp-server/rtsp-client.c:
10013 client: do configuration of transport in one place
10014 Move the configuration of the transport destination address to where we also
10015 configure the other bits.
10017 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10019 Merge branch 'master' into 0.11
10021 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
10023 * gst/rtsp-server/rtsp-client.c:
10024 client: destroy pipeline on client disconnect with no prior TEARDOWN.
10025 The problem occurs when the client abruptly closes the connection without
10026 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
10027 server is where the pipeline gets torn down. Since this handler is not called,
10028 the pipeline remains and is up and running. Subsequent clients get their own
10029 pipelines and if the do not issue TEARDOWNs then those pipelines will also
10030 remain up and running. This is a resource leak.
10032 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10034 Merge branch 'master' into 0.11
10036 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
10038 * gst/rtsp-server/rtsp-media-factory.c:
10039 * gst/rtsp-server/rtsp-media-factory.h:
10040 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
10041 For example, it can be used to retrieve source elements like appsrc, in a more
10042 convenient way than subclassing get_element.
10044 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10046 Merge branch 'master' into 0.11
10048 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
10050 * gst/rtsp-server/rtsp-server.c:
10051 rtsp-server: hold on to reference while using object
10053 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10055 * gst/rtsp-server/rtsp-media.c:
10058 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10061 configure: use unstable api
10063 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
10065 * gst/rtsp-server/rtsp-client.c:
10066 client: fix reference counting
10068 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
10070 * gst/rtsp-server/rtsp-client.c:
10071 * gst/rtsp-server/rtsp-media.c:
10072 fix compiler warnings about unused variables
10074 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
10076 * examples/test-launch.c:
10077 * examples/test-readme.c:
10078 * examples/test-uri.c:
10079 * examples/test-video.c:
10080 examples: tell rtsp uri when ready
10082 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
10085 Automatic update of common submodule
10086 From 69b981f to 605cd9a
10088 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10090 * gst/rtsp-server/rtsp-client.c:
10091 client: update for buffer API change
10093 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
10095 * gst/rtsp-server/Makefile.am:
10096 Makefile.am: 0.10 => @GST_MAJORMINOR@
10098 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
10100 * gst/rtsp-server/rtsp-media-factory-uri.c:
10101 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
10103 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
10105 * gst/rtsp-server/.gitignore:
10106 .gitignore: 0.10 => 0.11
10108 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
10110 * gst/rtsp-server/Makefile.am:
10111 Makefile.am: 0.10 => @GST_MAJORMINOR@
10113 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10115 Merge branch 'master' into 0.11
10117 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
10120 Automatic update of common submodule
10121 From 9e5bbd5 to 69b981f
10123 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
10126 Automatic update of common submodule
10127 From fd35073 to 9e5bbd5
10129 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
10132 Automatic update of common submodule
10133 From 46dfcea to fd35073
10135 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10137 * gst/rtsp-server/rtsp-media-factory-uri.c:
10138 * gst/rtsp-server/rtsp-media.c:
10139 media: port to new caps API
10141 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10143 Merge branch 'master' into 0.11
10145 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
10147 * bindings/vala/gst-rtsp-server-0.10.vapi:
10148 Updated Vala bindings.
10149 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
10151 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
10153 * gst/rtsp-server/rtsp-server.c:
10154 * gst/rtsp-server/rtsp-server.h:
10155 Add a signal for newly connected clients.
10156 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
10158 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
10160 * bindings/python/rtspserver.override:
10161 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
10163 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10165 * gst/rtsp-server/Makefile.am:
10166 * gst/rtsp-server/rtsp-client.c:
10167 * gst/rtsp-server/rtsp-funnel.c:
10168 * gst/rtsp-server/rtsp-funnel.h:
10169 * gst/rtsp-server/rtsp-media.c:
10170 rtsp-server: port to 0.11
10172 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10177 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
10179 Merge branch 'master' into 0.11
10184 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10187 Automatic update of common submodule
10188 From c3cafe1 to 46dfcea
10190 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
10192 * bindings/python/Makefile.am:
10193 * bindings/python/rtspserver.defs:
10194 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
10196 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
10198 * bindings/python/arg-types.py:
10199 python bindings: add GstRTSPUrlParam
10200 Needed to implement MediaFactory virtual proxies
10202 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
10204 * bindings/python/arg-types.py:
10205 python bindings: fix returning GstRTSPUrl types
10207 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
10209 * bindings/python/arg-types.py:
10210 python bindings: add arg type for GstRTSPUrl
10212 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
10214 * bindings/python/rtspserver.defs:
10215 python bindings: fix the definition of MediaFactory.collect_stream
10217 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
10220 Automatic update of common submodule
10221 From 1ccbe09 to c3cafe1
10223 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10226 Automatic update of common submodule
10227 From 193b717 to 1ccbe09
10229 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
10232 Automatic update of common submodule
10233 From b77e2bf to 193b717
10235 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10238 build: Include lcov.mak to allow test coverage report generation
10240 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10243 Automatic update of common submodule
10244 From d8814b6 to b77e2bf
10246 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10249 Automatic update of common submodule
10250 From 6aaa286 to d8814b6
10252 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
10255 Automatic update of common submodule
10256 From 6aec6b9 to 6aaa286
10258 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
10261 autogen: wingo signed comment
10263 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
10265 * gst/rtsp-server/rtsp-session-pool.c:
10266 session: use full charset for RTSP session ID
10267 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
10268 session ID more difficult.
10269 https://bugzilla.gnome.org/show_bug.cgi?id=643812
10271 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10273 * gst/rtsp-server/Makefile.am:
10274 rtsp-server: Don't install the funnel header
10276 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
10279 Automatic update of common submodule
10280 From 1de7f6a to 6aec6b9
10282 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10285 configure: require core/base 0.10.31
10286 Needed at least for gst_plugin_feature_rank_compare_func().
10288 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
10291 Automatic update of common submodule
10292 From f94d739 to 1de7f6a
10294 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10296 * gst/rtsp-server/rtsp-media.c:
10297 media: remove more unused code
10299 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10301 * gst/rtsp-server/rtsp-media.c:
10302 * gst/rtsp-server/rtsp-media.h:
10303 media: remove duplicate filtering
10304 Remove the duplicate filtering code now that we have a released -good version.
10305 Give a warning instead.
10307 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10309 * gst/rtsp-server/rtsp-media-factory.c:
10310 * gst/rtsp-server/rtsp-media.c:
10311 media: fix default buffer size
10313 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10315 * gst/rtsp-server/rtsp-media-factory.c:
10316 * gst/rtsp-server/rtsp-media-factory.h:
10317 media-factory: add property to configure the buffer-size
10318 Add a property to configure the kernel UDP buffer size.
10320 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10322 * gst/rtsp-server/rtsp-media.c:
10323 * gst/rtsp-server/rtsp-media.h:
10324 media: add property to configure kernel buffer sizes
10325 Add a property to configure the kernel UDP buffer size.
10327 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10330 configure: set PYGOBJECT_REQ before using it
10331 https://bugzilla.gnome.org/show_bug.cgi?id=640641
10333 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10335 * docs/Makefile.am:
10336 docs: recursive into sub-directories on 'make upload'
10338 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10340 * docs/libs/gst-rtsp-server-docs.sgml:
10341 * docs/version.entities.in:
10342 docs: mention full version these docs are for, not just major-minor
10344 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10347 back to development
10349 === release 0.10.8 ===
10351 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10356 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10358 * gst/rtsp-server/rtsp-server.c:
10359 rtsp-server: clarify docs a little
10361 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10363 * gst/rtsp-server/rtsp-media.c:
10364 media: init debug category before starting thread
10366 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10368 * gst/rtsp-server/rtsp-auth.c:
10369 auth: add realm to make it more spec compliant
10371 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10373 * gst/rtsp-server/rtsp-server.c:
10374 * gst/rtsp-server/rtsp-server.h:
10375 server: add locking
10377 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10379 * examples/test-video.c:
10380 example: improve example docs a little
10382 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10384 * gst/rtsp-server/rtsp-server.c:
10385 server: ensure the watch has a ref to the server
10387 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10389 * gst/rtsp-server/rtsp-server.c:
10390 server: simpify channel function
10392 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10394 * gst/rtsp-server/rtsp-server.c:
10395 * gst/rtsp-server/rtsp-server.h:
10396 server: simplify management of channel and source
10397 We don't need to keep around the channel and source objects. Let the mainloop
10398 and the source manage the source and channel respectively.
10400 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10406 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10408 * tests/.gitignore:
10409 * tests/Makefile.am:
10410 * tests/test-cleanup.c:
10411 tests: add tests directory and cleanup test
10413 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10415 * gst/rtsp-server/rtsp-media-factory-uri.c:
10416 * gst/rtsp-server/rtsp-media-factory.c:
10417 * gst/rtsp-server/rtsp-media-mapping.c:
10418 * gst/rtsp-server/rtsp-media.c:
10419 * gst/rtsp-server/rtsp-session-pool.c:
10420 * gst/rtsp-server/rtsp-session.c:
10421 server: improve debugging in various objects
10423 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10425 * gst/rtsp-server/rtsp-server.c:
10426 server: chain up to the parent finalize
10428 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
10430 * bindings/python/rtspserver-types.defs:
10431 * bindings/python/rtspserver.defs:
10432 * bindings/python/rtspserver.override:
10433 * bindings/python/test.py:
10434 gst-rtsp-server: update python bindings
10436 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10438 * gst/rtsp-server/rtsp-client.c:
10439 client: use the response from the clientstate
10440 Create the response object only once and store in the client state.
10441 Make all methods use the state response,
10443 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10445 * gst/rtsp-server/rtsp-server.c:
10446 server: use signal to keep track of clients
10447 Keep track of all the clients that the server creates and remove them when they
10448 fire the 'closed' signal.
10450 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10452 * gst/rtsp-server/rtsp-client.c:
10453 * gst/rtsp-server/rtsp-client.h:
10454 client: emit signal when closing
10456 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10458 * examples/.gitignore:
10459 * examples/Makefile.am:
10460 * examples/test-auth.c:
10461 * examples/test-video.c:
10462 * gst/rtsp-server/rtsp-auth.c:
10463 * gst/rtsp-server/rtsp-auth.h:
10464 * gst/rtsp-server/rtsp-client.c:
10465 * gst/rtsp-server/rtsp-media-factory.c:
10466 * gst/rtsp-server/rtsp-media.c:
10467 * gst/rtsp-server/rtsp-media.h:
10468 * gst/rtsp-server/rtsp-session-pool.h:
10469 * gst/rtsp-server/rtsp-session.h:
10470 media: enable per factory authorisations
10471 Allow for adding a GstRTSPAuth on the factory and media level and check
10472 permissions when accessing the factory.
10473 Add hints to the auth methods for future more fine grained authorisation.
10474 Add example application for per factory authentication.
10476 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10478 * gst/rtsp-server/rtsp-auth.c:
10479 * gst/rtsp-server/rtsp-auth.h:
10480 * gst/rtsp-server/rtsp-client.c:
10481 * gst/rtsp-server/rtsp-client.h:
10482 * gst/rtsp-server/rtsp-params.c:
10483 * gst/rtsp-server/rtsp-params.h:
10484 rtsp-server: Pass ClientState structure arround
10485 Pass the collected information for the ongoing request in a GstRTSPClientState
10486 structure that we can then pass around to simplify the method arguments. This
10487 will also be handy when we implement logging functionality.
10489 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10491 * gst/rtsp-server/rtsp-media-factory.c:
10492 * gst/rtsp-server/rtsp-media-factory.h:
10493 media-factory: add methods to configure authorisation
10495 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10497 * gst/rtsp-server/rtsp-client.c:
10498 client: unref auth in finalize
10500 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10502 * gst/rtsp-server/rtsp-server.c:
10503 server: unref auth in finalize
10505 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10507 * docs/libs/gst-rtsp-server-docs.sgml:
10508 * docs/libs/gst-rtsp-server-sections.txt:
10509 * docs/libs/gst-rtsp-server.types:
10510 docs: add more docs
10512 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10514 * gst/rtsp-server/rtsp-server.c:
10515 * gst/rtsp-server/rtsp-server.h:
10516 server: separate create and accept
10517 Create separate create and accept methods so that subclasses can create custom
10519 Configure the server in the client object and prepare for keeping track of
10522 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10524 * gst/rtsp-server/rtsp-client.c:
10525 * gst/rtsp-server/rtsp-client.h:
10526 client: add support for setting the server.
10527 Add support for keeping a ref to the server that started this client
10530 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10532 * gst/rtsp-server/rtsp-auth.c:
10533 auth: fix memleak and add some docs
10534 Fix a memleak of the basic auth token.
10535 Add docs for the helper function
10537 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10539 * gst/rtsp-server/rtsp-auth.c:
10540 * gst/rtsp-server/rtsp-auth.h:
10541 * gst/rtsp-server/rtsp-client.c:
10542 client: delegate setup of auth to the manager
10543 Delegate the configuration of the authentication tokens to the manager object
10546 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10548 * examples/test-video.c:
10549 * gst/rtsp-server/Makefile.am:
10550 * gst/rtsp-server/rtsp-auth.c:
10551 * gst/rtsp-server/rtsp-auth.h:
10552 * gst/rtsp-server/rtsp-client.c:
10553 * gst/rtsp-server/rtsp-client.h:
10554 * gst/rtsp-server/rtsp-server.c:
10555 * gst/rtsp-server/rtsp-server.h:
10556 auth: add authentication object
10557 Add an object that can check the authorization of requests.
10558 Implement basic authentication.
10559 Add example authentication to test-video
10561 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10563 * gst/rtsp-server/rtsp-server.c:
10564 * gst/rtsp-server/rtsp-server.h:
10565 server: move includes back
10566 the includes are needed for sockaddr_in.
10568 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10570 * gst/rtsp-server/rtsp-client.c:
10571 * gst/rtsp-server/rtsp-client.h:
10572 * gst/rtsp-server/rtsp-server.c:
10573 * gst/rtsp-server/rtsp-server.h:
10574 rtsp: move network includes where they are needed
10576 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
10578 * gst/rtsp-server/rtsp-media.h:
10579 rtsp-media.h: Minor corrections in comments.
10582 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
10585 Automatic update of common submodule
10586 From e572c87 to f94d739
10588 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
10592 * docs/libs/.gitignore:
10593 * examples/.gitignore:
10594 * gst/rtsp-server/.gitignore:
10597 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
10599 * docs/libs/Makefile.am:
10600 docs: We don't build ps/pdf for API reference docs
10602 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10605 Automatic update of common submodule
10606 From ccbaa85 to e572c87
10608 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10611 Automatic update of common submodule
10612 From 46445ad to ccbaa85
10614 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10616 * gst/rtsp-server/Makefile.am:
10617 * gst/rtsp-server/rtsp-funnel.c:
10618 * gst/rtsp-server/rtsp-funnel.h:
10619 * gst/rtsp-server/rtsp-media.c:
10620 funnel: rename fsfunnel to rtspfunnel
10621 Rename the funnel to avoid conflicts with the farsight one.
10623 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10625 * gst/rtsp-server/Makefile.am:
10626 * gst/rtsp-server/fs-funnel.c:
10627 * gst/rtsp-server/fs-funnel.h:
10628 * gst/rtsp-server/rtsp-media.c:
10629 rtsp-media: add and use fsfunnel
10630 Add a copy of fsfunnel to the build because input-selector removed the (broken)
10631 select-all property that we need.
10633 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10635 * gst/rtsp-server/Makefile.am:
10636 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
10637 Use PKG_CONFIG_PATH specified at configure time (if any) as well
10638 for the g-ir-compiler, rather than just assuming the env var has
10641 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10648 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
10650 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10653 * gst/rtsp-server/Makefile.am:
10654 gobject-introspection: fix g-i build for uninstalled setup
10655 Requires gst-plugins-base git (> 0.10.31.2).
10657 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10659 * examples/test-uri.c:
10660 examples: add some more options and comments
10662 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10664 * gst/rtsp-server/rtsp-media-factory-uri.c:
10665 factory-uri: use right property type
10667 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10669 * gst/rtsp-server/rtsp-media-factory-uri.c:
10670 factory-uri: attempt to configure buffer-lists
10671 Attempt to configure buffer lists in the payloader for improved performance.
10673 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10675 * gst/rtsp-server/rtsp-media.c:
10676 media: attempt to configure bigger UDP buffers
10677 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
10678 send buffers with high bitrate streams.
10680 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
10682 * gst/rtsp-server/rtsp-client.c:
10683 client: use the socket length from getsockname
10684 Use the length returned by getsockname to perform the getnameinfo call because
10685 the size can depend on the socket type and platform.
10688 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10690 * docs/libs/gst-rtsp-server-docs.sgml:
10691 * docs/libs/gst-rtsp-server-sections.txt:
10692 docs: add uri factory to the docs
10694 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10696 * gst/rtsp-server/rtsp-client.c:
10697 * gst/rtsp-server/rtsp-media.h:
10700 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10702 * gst/rtsp-server/rtsp-client.c:
10703 * gst/rtsp-server/rtsp-media.c:
10704 * gst/rtsp-server/rtsp-media.h:
10705 * gst/rtsp-server/rtsp-session.c:
10706 * gst/rtsp-server/rtsp-session.h:
10707 rtsp-server: add support for buffer lists
10708 Add support for sending bufferlists received from appsink.
10711 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10713 * gst/rtsp-server/rtsp-client.c:
10714 * gst/rtsp-server/rtsp-media.c:
10715 * gst/rtsp-server/rtsp-media.h:
10716 * gst/rtsp-server/rtsp-sdp.c:
10717 media: make method to retrieve the play range
10718 Make a method to retrieve the playback range so that we can conditionally create
10719 a different range for the SDP and the PLAY requests.
10721 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10723 * gst/rtsp-server/rtsp-media.c:
10724 * gst/rtsp-server/rtsp-media.h:
10725 media: add signal to notify of state changes
10727 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10729 * gst/rtsp-server/rtsp-client.h:
10730 client: cleanup headers
10732 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10734 * gst/rtsp-server/rtsp-client.c:
10737 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10739 * gst/rtsp-server/rtsp-media-factory-uri.c:
10740 * gst/rtsp-server/rtsp-media-factory-uri.h:
10741 factory-uri: add support for gstpay
10742 Add an option to prefer gstpay over decoder + raw payloader.
10744 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10746 * gst/rtsp-server/rtsp-media-factory-uri.c:
10747 * gst/rtsp-server/rtsp-media-factory-uri.h:
10748 factory-uri: rework the autoplugger.
10749 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
10752 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10754 * gst/rtsp-server/rtsp-media-factory-uri.c:
10755 factory-uri: use better factory filter
10756 Make better payloader filter based on autoplug rank and RTP use case.
10758 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
10761 Automatic update of common submodule
10762 From 169462a to 46445ad
10764 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10766 * gst/rtsp-server/rtsp-server.c:
10767 server: set SO_REUSEADDR before bind
10768 Set the SO_REUSEADDR _before_ bind() to make it actually work.
10770 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10772 * gst/rtsp-server/rtsp-media.c:
10773 * gst/rtsp-server/rtsp-media.h:
10774 media: emit prepared signal when prepared
10775 Make a 'prepared' signal and emit it when we successfully prepared the element.
10776 This signal can be used to configure the media object after it has been prepared
10779 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
10782 Automatic update of common submodule
10783 From 011bcc8 to 169462a
10785 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
10787 python an optional dependency
10788 * configure.ac: Move up valgrind and g-i checks. Make the python
10789 dependency optional, as it was before.
10791 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10793 Merge branch 'master' into 0.11
10798 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10800 * gst/rtsp-server/rtsp-media.c:
10801 media: update range when active clients changed
10802 When we changed the number of active clients, update the current range
10803 information because we want the second client connecting to a shared resource
10804 continue from where the stream currently.
10806 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10808 * gst/rtsp-server/rtsp-media-factory-uri.c:
10809 * gst/rtsp-server/rtsp-media-factory-uri.h:
10810 factory-uri: add colorspace and fix pt
10811 Rework the way we pass data to the autoplugger.
10812 When we have raw caps, plug a converter element to make pluggin to raw
10813 payloaders more successful.
10814 Make sure all dynamically plugged payloaders have a unique payload types.
10816 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10818 * examples/Makefile.am:
10819 * examples/test-uri.c:
10820 example: add example of the uri factory
10822 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10824 * gst/rtsp-server/Makefile.am:
10825 * gst/rtsp-server/rtsp-media-factory-uri.c:
10826 * gst/rtsp-server/rtsp-media-factory-uri.h:
10827 * gst/rtsp-server/rtsp-server.h:
10828 factory-uri: add a factory to stream any URI
10829 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
10832 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10834 * gst/rtsp-server/rtsp-media.c:
10835 * gst/rtsp-server/rtsp-media.h:
10836 media: ignore spurious ASYNC_DONE messages
10837 When we are dynamically adding pads, the addition of the udpsrc elements will
10838 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
10839 the real ASYNC_DONE when everything is prerolled.
10841 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10843 * gst/rtsp-server/rtsp-media-factory.c:
10844 * gst/rtsp-server/rtsp-media-factory.h:
10845 media-factory: make lock macro
10847 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
10849 * gst/rtsp-server/rtsp-client.c:
10850 rtsp-server: Remove unused variable and dead assignment
10852 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
10854 * examples/test-launch.c:
10855 * examples/test-mp4.c:
10856 * examples/test-ogg.c:
10857 * examples/test-readme.c:
10858 * examples/test-sdp.c:
10859 * examples/test-video.c:
10860 examples: Run gst-indent
10862 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
10864 * gst/rtsp-server/rtsp-client.c:
10865 * gst/rtsp-server/rtsp-media-factory.c:
10866 * gst/rtsp-server/rtsp-media-mapping.c:
10867 * gst/rtsp-server/rtsp-media.c:
10868 * gst/rtsp-server/rtsp-params.c:
10869 * gst/rtsp-server/rtsp-sdp.c:
10870 * gst/rtsp-server/rtsp-server.c:
10871 * gst/rtsp-server/rtsp-session-pool.c:
10872 * gst/rtsp-server/rtsp-session.c:
10873 rtsp-server: Run gst-indent
10874 Since it wasn't using the upstream common previously, there was no
10875 indentation check before commiting.
10877 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
10879 * gst/rtsp-server/rtsp-media-mapping.h:
10880 * gst/rtsp-server/rtsp-media.c:
10881 * gst/rtsp-server/rtsp-media.h:
10882 * gst/rtsp-server/rtsp-sdp.c:
10883 * gst/rtsp-server/rtsp-session-pool.h:
10884 * gst/rtsp-server/rtsp-session.c:
10885 * gst/rtsp-server/rtsp-session.h:
10886 rtsp-server: Some more doc fixups
10888 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
10891 Makefile: Add cruft-cleaning support
10893 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
10897 * docs/Makefile.am:
10898 * docs/libs/Makefile.am:
10899 * docs/libs/gst-rtsp-server-docs.sgml:
10900 * docs/libs/gst-rtsp-server-sections.txt:
10901 * docs/libs/gst-rtsp-server.types:
10902 * docs/version.entities.in:
10903 docs: Add gtk-doc build system
10905 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
10907 * gst/rtsp-server/Makefile.am:
10908 Makefile.am: Use standard GIR make behaviour
10910 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
10914 autogen/configure: Bring more in sync to standard gst module behaviour
10916 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10918 * gst/rtsp-server/rtsp-media.c:
10919 media: warn and fail when gstrtpbin is not found
10921 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10924 configure: open 0.11 branch
10926 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
10930 Add common submodule
10932 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
10934 * common/ChangeLog:
10935 * common/Makefile.am:
10936 * common/c-to-xml.py:
10937 * common/check.mak:
10938 * common/coverage/coverage-report-entry.pl:
10939 * common/coverage/coverage-report.pl:
10940 * common/coverage/coverage-report.xsl:
10941 * common/coverage/lcov.mak:
10942 * common/gettext.patch:
10943 * common/glib-gen.mak:
10944 * common/gst-autogen.sh:
10945 * common/gst-xmlinspect.py:
10947 * common/gstdoc-scangobj:
10948 * common/gtk-doc-plugins.mak:
10949 * common/gtk-doc.mak:
10950 * common/m4/.gitignore:
10951 * common/m4/Makefile.am:
10952 * common/m4/README:
10953 * common/m4/as-ac-expand.m4:
10954 * common/m4/as-auto-alt.m4:
10955 * common/m4/as-compiler-flag.m4:
10956 * common/m4/as-compiler.m4:
10957 * common/m4/as-docbook.m4:
10958 * common/m4/as-libtool-tags.m4:
10959 * common/m4/as-libtool.m4:
10960 * common/m4/as-python.m4:
10961 * common/m4/as-scrub-include.m4:
10962 * common/m4/as-version.m4:
10963 * common/m4/ax_create_stdint_h.m4:
10964 * common/m4/check.m4:
10965 * common/m4/glib-gettext.m4:
10966 * common/m4/gst-arch.m4:
10967 * common/m4/gst-args.m4:
10968 * common/m4/gst-check.m4:
10969 * common/m4/gst-debuginfo.m4:
10970 * common/m4/gst-default.m4:
10971 * common/m4/gst-doc.m4:
10972 * common/m4/gst-error.m4:
10973 * common/m4/gst-feature.m4:
10974 * common/m4/gst-function.m4:
10975 * common/m4/gst-gettext.m4:
10976 * common/m4/gst-glib2.m4:
10977 * common/m4/gst-libxml2.m4:
10978 * common/m4/gst-plugindir.m4:
10979 * common/m4/gst-valgrind.m4:
10980 * common/m4/gtk-doc.m4:
10981 * common/m4/introspection.m4:
10982 * common/m4/pkg.m4:
10983 * common/mangle-tmpl.py:
10984 * common/plugins.xsl:
10986 * common/release.mak:
10987 * common/scangobj-merge.py:
10988 * common/upload.mak:
10989 common: Remove static version
10991 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
10993 * common/m4/introspection.m4:
10994 Update introspection.m4 to match usage
10996 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11000 Remove old stuff from the README
11002 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11005 back to development
11007 === release 0.10.7 ===
11009 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11014 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11016 * examples/test-ogg.c:
11017 test-ogg: remove parsers
11018 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
11019 buffers with timestamps. Using the parsers also seems to break things.
11021 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11023 * bindings/vala/gst-rtsp-server-0.10.vapi:
11024 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11025 Updated Vala bindings
11027 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11029 * common/m4/introspection.m4:
11031 * gst/rtsp-server/Makefile.am:
11032 Added initial gobject-introspection support
11034 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11036 * gst/rtsp-server/rtsp-media-factory.c:
11037 media-factory: don't use host for shared hash key
11038 When we generate the key to share made between connections, don't include the
11039 host used to connect so that we can share media even if between clients that
11040 connected with localhost and ones with the ip address.
11042 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11044 * bindings/vala/Makefile.am:
11045 build: fix distcheck
11047 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11049 * bindings/vala/gst-rtsp-server-0.10.vapi:
11050 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11051 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11052 Update Vala bindings
11054 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
11056 * bindings/vala/Makefile.am:
11058 Fix configure checks and installation location for Vala bindings
11061 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11064 back to development
11066 === release 0.10.6 ===
11068 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11071 configure: release 0.10.6
11073 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11075 * gst/rtsp-server/rtsp-media.c:
11076 media: help the compiler a little
11078 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11080 * gst/rtsp-server/rtsp-media.c:
11081 * gst/rtsp-server/rtsp-media.h:
11082 * gst/rtsp-server/rtsp-session.c:
11083 media: cleanup media transport before freeing
11084 Cleanup the media transport data before freeing. In particular, remove the qdata
11085 from the rtpsource object.
11087 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11089 * gst/rtsp-server/rtsp-media-factory.c:
11090 * gst/rtsp-server/rtsp-media-factory.h:
11091 * gst/rtsp-server/rtsp-media.c:
11092 * gst/rtsp-server/rtsp-media.h:
11093 media-factory: add eos-shutdown property
11094 Add an eos-shutdown property that will send an EOS to the pipeline before
11095 shutting it down. This allows for nice cleanup in case of a muxer.
11098 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11100 * gst/rtsp-server/rtsp-media.c:
11101 * gst/rtsp-server/rtsp-media.h:
11102 media: use multiudpsink send-duplicates when we can
11103 If we have a new enough multiudpsink with the send-duplicates property, use this
11104 instead of doing our own filtering. Our custom filtering code should eventually
11105 be removed when we can depend on a released -good.
11107 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11109 * gst/rtsp-server/rtsp-media.c:
11110 media: don't leak destinations
11111 Refactor and cleanup the destinations array when the stream is destroyed.
11113 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11115 * gst/rtsp-server/rtsp-media.c:
11116 * gst/rtsp-server/rtsp-media.h:
11117 media: don't add udp addresses multiple times
11118 Keep track of the udp addresses we added to udpsink and never add the same udp
11119 destination twice. This avoids duplicate packets when using multicast.
11121 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11123 * gst/rtsp-server/rtsp-server.c:
11124 server: disable use of SO_LINGER
11125 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
11126 server close()s the connection.
11128 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11130 * gst/rtsp-server/rtsp-server.c:
11131 server: use 5 second linger period in SO_LINGER
11132 Wait 5 seconds before clearing the send buffers and reseting the connection with
11133 the client when we do a close. This should be enough time to get the message to
11137 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
11139 * gst/rtsp-server/rtsp-server.c:
11140 server: use SO_LINGER
11141 SO_LINGER on the socket will make sure that any pending data on the socket is
11142 flushed ASAP and that the socket connection is reset. This makes sure that the
11143 socket can be reused immediately.
11146 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11149 README: add blurb about shared media factories
11151 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
11153 * gst/rtsp-server/rtsp-media.c:
11154 Add stdlib.h for atoi()
11156 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11158 * bindings/python/Makefile.am:
11159 * bindings/vala/Makefile.am:
11160 build: distcheck fixes
11161 Fix 'make distcheck', somewhat (it still fails because it tries to
11162 install files into /usr/share/vala/vapi/ irrespective of the
11163 configured prefix).
11165 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11168 configure: bump core/base requirements to released version
11169 Makes things less confusing for people.
11171 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11174 configure: fail if GStreamer core/base requirements are not met
11176 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11178 * gst/rtsp-server/rtsp-client.c:
11179 client: improve client cleanups
11180 Make sure the session does not timeout when using TCP. We need to do this
11181 because quicktime player does not send RTCP for some reason in tunneled
11183 Refactor some cleanup code.
11186 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11188 * gst/rtsp-server/rtsp-session.c:
11189 * gst/rtsp-server/rtsp-session.h:
11190 session: add support for prevent session timeouts
11191 Add an atomix counter to prevent session timeouts when we are, for example,
11192 streaming over TCP.
11194 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11196 * gst/rtsp-server/rtsp-client.c:
11197 client: fix unlink on session timeouts
11198 When our session times out, make sure we unlink all streams in this
11200 Remove the tunnelid when closing the connection.
11202 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11204 * gst/rtsp-server/rtsp-session.c:
11205 session: small cleanups
11207 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11209 * gst/rtsp-server/rtsp-client.c:
11210 client: handle lost_tunnel callbacks
11211 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
11212 hashtable so that we can reuse it for when the client reopens the POST
11214 Close the connection after a TEARDOWN.
11215 Make sure or watchid is cleared when the watch is removed.
11218 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11220 * gst/rtsp-server/rtsp-client.c:
11221 * gst/rtsp-server/rtsp-media.c:
11222 * gst/rtsp-server/rtsp-sdp.c:
11223 rtsp-server: add more support for multicast
11225 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11228 * gst/rtsp-server/rtsp-media.c:
11229 * gst/rtsp-server/rtsp-media.h:
11230 media: allow configuration of allowed lower transport
11232 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11234 * gst/rtsp-server/rtsp-client.h:
11235 * gst/rtsp-server/rtsp-media.c:
11236 * gst/rtsp-server/rtsp-media.h:
11237 * gst/rtsp-server/rtsp-sdp.c:
11238 * gst/rtsp-server/rtsp-sdp.h:
11239 * gst/rtsp-server/rtsp-server.c:
11240 rtsp: keep track of server ip and ipv6
11241 Keep track of how the client connected to the server and setup the udp ports
11242 with the same protocol.
11243 Copy the server ip address in the SDP so that clients can send RTCP back to
11246 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11248 * gst/rtsp-server/rtsp-session.c:
11251 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11253 * gst/rtsp-server/rtsp-client.c:
11254 client: use right size for malloc
11256 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11258 * gst/rtsp-server/rtsp-server.c:
11259 server: comment ipv6 server listening address
11261 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11263 * gst/rtsp-server/rtsp-media.c:
11264 media: allow for ipv6 sockets
11266 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11268 * gst/rtsp-server/rtsp-server.c:
11269 * gst/rtsp-server/rtsp-server.h:
11270 server: rework server part
11271 Allow setting a bind address, make sure we can deal with ipv6.
11272 Remove the port property and change with the service property.
11274 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11276 * gst/rtsp-server/rtsp-media.h:
11277 media: update comments a little
11279 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11281 * gst/rtsp-server/rtsp-client.c:
11282 client: make content-base better
11283 Use the URI formatting functions to make a content-base. Also make sure that
11284 there is a trailing / at the end.
11286 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11288 * gst/rtsp-server/rtsp-client.c:
11289 client: guard against invalid paths
11291 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11293 * examples/test-video.c:
11294 test: catch server bind errors
11296 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
11298 * gst/rtsp-server/rtsp-media.c:
11299 rtspmedia: emit "unprepared" if _prepare fails.
11300 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
11301 media object is removed from its factory's cache.
11303 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11305 * gst/rtsp-server/rtsp-media.c:
11306 media: collect media position when seek completes
11308 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
11310 * gst/rtsp-server/rtsp-client.c:
11311 client: call unlink_streams in client finalize
11314 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11316 * gst/rtsp-server/rtsp-media.c:
11317 media: limit the time to wait to something huge
11318 Avoid waiting forever but limit the timeout to 20 seconds.
11320 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11322 * gst/rtsp-server/rtsp-sdp.c:
11323 sdp: reindent and check for prepared status
11325 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11327 * gst/rtsp-server/rtsp-media.c:
11328 * gst/rtsp-server/rtsp-media.h:
11329 * gst/rtsp-server/rtsp-session.c:
11330 media: avoid doing _get_state() for state changes
11331 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
11332 until the media is prerolled or in error. This avoids doing a blocking call of
11333 gst_element_get_state() that can cause lockups when there is an error.
11336 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11338 * gst/rtsp-server/rtsp-media.c:
11341 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11343 * gst/rtsp-server/rtsp-media-factory.c:
11344 media-factory: better error handling
11345 Improve the error handling a bit.
11347 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11349 * gst/rtsp-server/rtsp-client.c:
11350 client: rework transport parsing
11351 Rework the transport parsing code so that we can ignore transports we don't
11352 support instead of just picking the first one we can parse.
11353 Configure a (for now hardcoded) destination for multicast transports.
11355 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11357 * gst/rtsp-server/rtsp-media.c:
11358 media: set multicast sink parameters
11359 Disable loop and automatic multicast join on the udpsink elements.
11360 Add some more debug info.
11361 Reset some state variables in the right place.
11362 Use the right port numbers for multicast.
11364 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11366 * gst/rtsp-server/rtsp-session.c:
11367 session: handle transport setup correctly
11368 Handle UDP, MCAST and TCP transport negotiation more correctly.
11369 Store the server session SSRC in the transport.
11371 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11373 * gst/rtsp-server/rtsp-client.c:
11374 rtsp-client: implement error_full
11375 Implement error_full to avoid some segfaults when the rtspconnection calls it.
11378 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11381 * gst/rtsp-server/rtsp-client.c:
11382 * gst/rtsp-server/rtsp-server.c:
11383 docs: update docs and comments
11385 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
11387 * gst/rtsp-server/rtsp-sdp.c:
11388 sdp: make server work better when behind a proxy
11390 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11392 * gst/rtsp-server/rtsp-client.c:
11393 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
11395 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11397 * gst/rtsp-server/rtsp-client.c:
11398 * gst/rtsp-server/rtsp-media-factory.c:
11399 * gst/rtsp-server/rtsp-media-mapping.c:
11400 * gst/rtsp-server/rtsp-media.c:
11401 * gst/rtsp-server/rtsp-server.c:
11402 * gst/rtsp-server/rtsp-session-pool.c:
11403 * gst/rtsp-server/rtsp-session.c:
11404 Use GStreamer's debugging subsystem
11406 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
11408 * gst/rtsp-server/rtsp-media-factory.c:
11409 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
11411 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11414 back to development
11416 === release 0.10.5 ===
11418 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11423 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11426 configure: bump required versions
11428 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
11430 * gst/rtsp-server/rtsp-client.c:
11431 client: call weak-unref on client->sessions from finalize
11434 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11436 * gst/rtsp-server/rtsp-media.c:
11437 media: Fixed crasher where caps got unref'ed too often
11439 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11442 * pkgconfig/.gitignore:
11443 * pkgconfig/Makefile.am:
11444 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
11445 Added pkg-config file to use gst-rtsp-server uninstalled
11447 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11449 * gst/rtsp-server/rtsp-media.c:
11450 media: add some docs
11452 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
11454 * gst/rtsp-server/rtsp-client.c:
11455 rtsp: Use gst_rtsp_watch_send_message().
11456 Use gst_rtsp_watch_send_message() since the old API which used
11457 gst_rtsp_watch_queue_message() has been deprecated.
11459 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11462 back to development
11464 === release 0.10.4 ===
11466 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11471 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11473 * gst/rtsp-server/rtsp-client.c:
11474 * gst/rtsp-server/rtsp-session.c:
11475 * gst/rtsp-server/rtsp-session.h:
11476 rtsp: allocate channels in TCP mode
11477 When the client does not provide us with channels in TCP mode, allocate channels
11480 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11482 * gst/rtsp-server/rtsp-client.c:
11483 client: don't crash when tunnelid is missing
11484 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
11485 don't crash but return an error response to the client.
11488 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11490 * bindings/vala/gst-rtsp-server-0.10.vapi:
11491 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11492 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11493 bindings: update vala bindings with new method
11495 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11497 * gst/rtsp-server/rtsp-session-pool.c:
11498 * gst/rtsp-server/rtsp-session-pool.h:
11499 sessionpool: add function to filter sessions
11500 Add generic function to retrieve/remove sessions.
11502 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11505 configure: bump core/base requirements to release
11507 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11509 * gst/rtsp-server/rtsp-media.c:
11510 media: fix indentation
11512 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11514 * gst/rtsp-server/rtsp-media.c:
11515 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
11517 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11519 * gst/rtsp-server/rtsp-media.c:
11520 set state and remove elements of media in for loop
11522 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
11524 * bindings/vala/gst-rtsp-server-0.10.vapi:
11525 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11526 Added gst_rtsp_media_remove_elements function to Vala bindings
11528 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
11530 * gst/rtsp-server/rtsp-media.c:
11531 * gst/rtsp-server/rtsp-media.h:
11532 Added gst_rtsp_media_remove_elements function
11534 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
11536 * gst/rtsp-server/rtsp-media.c:
11537 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
11539 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11541 * bindings/vala/gst-rtsp-server-0.10.vapi:
11542 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11543 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11544 Updated Vala bindings
11546 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11548 * gst/rtsp-server/rtsp-media.c:
11549 * gst/rtsp-server/rtsp-media.h:
11550 Added vmethod unprepare to GstRTSPMedia
11551 The default implementation sets the state of the pipeline to GST_STATE_NULL
11553 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11555 * gst/rtsp-server/rtsp-media-factory.c:
11556 * gst/rtsp-server/rtsp-media-factory.h:
11557 Made collect_streams function public
11559 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11561 * gst/rtsp-server/rtsp-media-factory.c:
11562 * gst/rtsp-server/rtsp-media-factory.h:
11563 * gst/rtsp-server/rtsp-media.c:
11564 Added vmethod create_pipeline to GstRTSPMediaFactory
11565 The pipeline is created in this method and the GstRTSPMedia's element is added to it
11567 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11569 * gst/rtsp-server/rtsp-client.c:
11570 client: use g_source_destroy()
11571 We need to use g_source_destroy() because we might have added the source to a
11572 different main context than the default one.
11574 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11576 * gst/rtsp-server/Makefile.am:
11577 * gst/rtsp-server/rtsp-client.c:
11578 * gst/rtsp-server/rtsp-params.c:
11579 * gst/rtsp-server/rtsp-params.h:
11580 rtsp: prepare for handling GET/SET_PARAMETER
11581 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
11583 Fix return codes of handlers.
11585 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11587 * gst/rtsp-server/rtsp-media.c:
11588 media: don't leak session pads
11590 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11592 * gst/rtsp-server/rtsp-media.c:
11593 media: clean up the messages a bit
11595 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11597 * gst/rtsp-server/rtsp-sdp.c:
11598 sdp: warn and skip streams without media
11600 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11602 * bindings/vala/gst-rtsp-server-0.10.vapi:
11603 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11604 vala: Fixed typo in header file of RTSPMediaStream
11606 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11608 * gst/rtsp-server/rtsp-media.c:
11610 Fix a debug message
11611 Make dumping RTCP stats configurable
11613 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11615 * gst/rtsp-server/rtsp-media.c:
11616 media: be less verbose and leak less
11618 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11620 * gst/rtsp-server/rtsp-media.c:
11621 media: don't leak the destination address
11623 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11625 * gst/rtsp-server/rtsp-client.c:
11626 * gst/rtsp-server/rtsp-media.c:
11627 * gst/rtsp-server/rtsp-media.h:
11628 * gst/rtsp-server/rtsp-session.c:
11629 * gst/rtsp-server/rtsp-session.h:
11630 rtsp: use RTCP to keep the session alive
11631 Use the RTCP rtcp-from stats field to find the associated session and use this
11632 to keep the session alive.
11634 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11636 * gst/rtsp-server/rtsp-session.c:
11637 session: add 5sec to the real session timeout
11638 Allow the session to live 5sec longer before really timing out. This should give
11639 clients some extra time to keep the session active.
11641 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11643 * gst/rtsp-server/rtsp-client.c:
11644 client: replay OK to GET/SET_PARAMETER
11645 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
11646 so that we return OK for those requests.
11648 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11650 * gst/rtsp-server/rtsp-media.c:
11651 * gst/rtsp-server/rtsp-media.h:
11652 media: keep track of active transports
11653 Keep track of which transport is active to avoid closing the connection too
11655 Remove the destination transport also when going to NULL.
11656 Print some stats about the SDES and other RTCP messages we receive from the
11659 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11661 * examples/.gitignore:
11662 * examples/Makefile.am:
11663 * examples/test-sdp.c:
11664 example: add SDP relay example
11666 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11668 * gst/rtsp-server/rtsp-media.c:
11669 media: also count active TCP connections
11671 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11673 * gst/rtsp-server/rtsp-media-factory.c:
11674 * gst/rtsp-server/rtsp-media.c:
11675 * gst/rtsp-server/rtsp-media.h:
11676 rtsp: add support for dynamic elements
11677 Add support for dynamic elements.
11678 Don't set live pipelines back to paused.
11680 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11682 * gst/rtsp-server/rtsp-sdp.c:
11683 sdp: don't add encoding name when absent in caps
11685 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11687 * gst/rtsp-server/rtsp-client.c:
11688 client: warn when we can't do RTP-Info
11690 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11692 * gst/rtsp-server/rtsp-media-factory.c:
11693 factory: factor out the stream construction
11695 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11697 * gst/rtsp-server/rtsp-client.c:
11698 client: only add RTP-Info when we have the info
11699 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
11702 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11705 back to development
11707 === release 0.10.3 ===
11709 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11713 - Fixes a bug where it put the wrong verion in pkgconfig
11714 - Link RTP and RTCP sources
11716 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11718 * gst/rtsp-server/rtsp-media.c:
11719 * gst/rtsp-server/rtsp-media.h:
11720 media: link the RTP udpsrc to the session manager
11721 Link the RTP udpsrc and the appsrc to the session manager so that they don't
11722 shut down when the client sends a packet to open firewalls.
11724 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11726 * pkgconfig/gst-rtsp-server.pc.in:
11727 Don't use hard-coded version number in pkg-config file
11729 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11732 back to development
11734 === release 0.10.2 ===
11736 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11741 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11744 * common/m4/.gitignore:
11745 * examples/.gitignore:
11746 * pkgconfig/.gitignore:
11747 add some .gitignore files
11749 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11751 * gst/rtsp-server/rtsp-media.c:
11752 media: seek to key frames
11754 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11756 * gst/rtsp-server/rtsp-media.c:
11757 media: emit the unprepared signal by id
11758 Emit the unprepared signal by id instead of name and set the media as
11761 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11763 * gst/rtsp-server/rtsp-media.c:
11764 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
11766 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11768 * gst/rtsp-server/rtsp-server.c:
11769 Added finalize function to GstRTPSPServer to unref session pool and media mapping
11771 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
11773 * bindings/vala/gst-rtsp-server-0.10.vapi:
11774 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
11775 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
11776 Updated vala bindings
11778 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11780 * gst/rtsp-server/Makefile.am:
11781 * gst/rtsp-server/rtsp-client.c:
11782 * gst/rtsp-server/rtsp-media.c:
11783 server: use appsink and appsrc with the API
11784 Use the appsink/appsrc API instead of the signals for higher
11787 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11789 * examples/test-ogg.c:
11790 tests: set the payload type correctly
11792 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11794 * gst/rtsp-server/rtsp-media-factory.c:
11795 factory: connect to the unprepare signal
11796 Connect to the unprepare signal for non-reusable media so that we can remove
11797 them from the cache.
11799 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11801 * gst/rtsp-server/rtsp-media.c:
11802 * gst/rtsp-server/rtsp-media.h:
11803 media: add signal to notify of unprepare
11805 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11807 * gst/rtsp-server/rtsp-media.c:
11808 * gst/rtsp-server/rtsp-media.h:
11809 media: more work on making the media shared
11810 Add a reusable flag to medias, indicating that they can be reused after a state
11814 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11816 * examples/test-readme.c:
11817 examples: mark the example as shared for testing
11819 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11821 * gst/rtsp-server/rtsp-media.c:
11822 * gst/rtsp-server/rtsp-media.h:
11823 client: support shared media
11824 Always perform the state actions even if the target state of the pipeline is
11825 already correct, we still want to add/remove the transports when we are dealing
11827 Keep a counter of the number of active transports for a media so that we can use
11828 this to perform a state change when needed.
11829 Perform a state change of the pipeline only when the first transport was added
11830 or when there are no active transports.
11832 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
11834 * gst/rtsp-server/rtsp-client.c:
11835 client: fix refcounting crasher
11836 Don't need to remove the weak refs in the finalize methods, they are already
11837 removed in the dispose.
11838 Don't register the callback with a DestroyNofity.
11840 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11842 * gst/rtsp-server/rtsp-client.c:
11843 Fix rtsp client refcount management in TCP mode.
11844 Don't unref a client ref we never had. Fixes an unref
11845 of an already-free client object after a client
11846 teardown request for me.
11848 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
11850 * gst/rtsp-server/rtsp-session.c:
11851 docs: fix typo in API docs
11853 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11855 * gst/rtsp-server/rtsp-media.c:
11856 More seeking fixes.
11857 Keep the udp sources in playing even if we go to paused. unlock the sources when
11859 Add some more debug info.
11860 Only seek when we need to.
11861 Keep track of the position when we go to paused.
11863 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11865 * gst/rtsp-server/rtsp-client.c:
11866 * gst/rtsp-server/rtsp-media.c:
11867 * gst/rtsp-server/rtsp-media.h:
11868 Add beginnings of seeking.
11869 Parse the Range header and perform a seek on the pipeline for the requested
11870 position. It's disabled currently until I figure out what's going wrong.
11872 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11874 * gst/rtsp-server/rtsp-client.c:
11875 allow pause requests for now.
11878 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11880 * gst/rtsp-server/rtsp-client.c:
11881 Remove weak ref on the session in teardown
11882 We need to remove our weakref from the session when we do a teardown because
11883 else we close the TCP connection prematurely.
11885 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11887 * gst/rtsp-server/rtsp-client.c:
11888 * gst/rtsp-server/rtsp-client.h:
11889 * gst/rtsp-server/rtsp-session-pool.c:
11890 Do some more session cleanup
11891 Make session timeout kill the TCP connection that currently watches the
11893 Remove the client timeout property.
11895 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11897 * gst/rtsp-server/rtsp-client.c:
11898 * gst/rtsp-server/rtsp-client.h:
11899 * gst/rtsp-server/rtsp-media.c:
11900 * gst/rtsp-server/rtsp-media.h:
11901 * gst/rtsp-server/rtsp-server.c:
11902 * gst/rtsp-server/rtsp-session.c:
11903 * gst/rtsp-server/rtsp-session.h:
11905 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
11908 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11910 * examples/Makefile.am:
11911 * examples/test-launch.c:
11912 Add example server that takes launch lines
11913 Add an example server that streams any -launch line.
11915 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11917 * examples/test-readme.c:
11918 * gst/rtsp-server/rtsp-client.c:
11919 * gst/rtsp-server/rtsp-media.c:
11920 * gst/rtsp-server/rtsp-media.h:
11921 Add support for live streams
11922 Add support for live streams and ranges
11923 Start on handling TCP data transfer.
11925 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11927 * gst/rtsp-server/rtsp-media.c:
11928 Free the pipeline before other things
11931 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11933 * gst/rtsp-server/rtsp-client.c:
11934 Only free the pending tunnel if there is one
11937 2009-03-04 12:44:01 +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-media.c:
11942 rtsp-server: Add support for tunneling
11943 Add support for tunneling over HTTP.
11944 Use new connection methods to retrieve the url.
11945 Dispatch messages based on the message type instead of blindly
11946 assuming it's always a request.
11947 Keep track of the watch id so that we can remove it later.
11948 Set the media pipeline to NULL before unreffing the pipeline.
11950 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11952 * gst/rtsp-server/rtsp-client.c:
11953 * gst/rtsp-server/rtsp-client.h:
11954 Fix for channel -> watch rename in gstreamer
11955 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
11957 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11959 * gst/rtsp-server/rtsp-client.c:
11960 * gst/rtsp-server/rtsp-client.h:
11962 Use the async RTSP channels instead of spawning a new thread for each client.
11963 If a sessionid is specified in a request, fail if we don't have the session.
11965 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11967 * gst/rtsp-server/rtsp-media.c:
11968 Add better debug info
11969 Add some better debug info.
11971 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11973 * examples/test-video.c:
11975 Add support for session timeouts in the example.
11977 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11979 * gst/rtsp-server/rtsp-session-pool.c:
11980 * gst/rtsp-server/rtsp-session-pool.h:
11981 Pass GTimeVal around for performance reasons
11982 Get the current time only once and pass it around so that sessions don't have to
11983 get the current time anymore.
11984 Add experimental support for a GSource that dispatches when the session needs to
11987 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11989 * gst/rtsp-server/rtsp-session.c:
11990 * gst/rtsp-server/rtsp-session.h:
11991 Add better support for session timeouts
11992 Add a method to request the number of milliseconds when a session will timeout.
11994 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
11996 * gst/rtsp-server/rtsp-media.c:
11997 * gst/rtsp-server/rtsp-media.h:
11998 Add suport for RTP manager monitoring
11999 Add the first stage in monitoring the rtp manager.
12000 Make sure we don't update the state to something we don't want.
12002 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12004 * gst/rtsp-server/rtsp-client.c:
12005 Add support for session keepalive
12006 Get and update the session timeout for all requests. get the session as early as
12009 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12011 * gst/rtsp-server/rtsp-media-factory.h:
12012 * gst/rtsp-server/rtsp-media.c:
12013 * gst/rtsp-server/rtsp-media.h:
12014 Handle media bus messages
12015 Handle media bus messages in a custom mainloop and dispatch them to the
12016 RTSPMedia objects. Let the default implementation handle some common messages.
12018 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12020 * gst/rtsp-server/rtsp-client.c:
12021 * gst/rtsp-server/rtsp-session-pool.c:
12022 * gst/rtsp-server/rtsp-session.c:
12023 Some more session timeout handling
12024 Move the session header setting code to a central place so that we always add
12025 the timeout parameter too.
12026 Handle timeouts by running the session cleanup code.
12027 Stop media before cleaning up.
12029 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12031 * gst/rtsp-server/rtsp-client.c:
12032 * gst/rtsp-server/rtsp-client.h:
12033 Add timeout property
12034 Add a timeout property ot the client and make the other properties into GObject
12037 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12039 * gst/rtsp-server/rtsp-session-pool.c:
12040 Use getters and setters in property code
12041 Use the getters and setters for the timeout property instead of locking
12044 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12046 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
12048 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12050 * gst/rtsp-server/rtsp-session-pool.c:
12051 * gst/rtsp-server/rtsp-session-pool.h:
12052 * gst/rtsp-server/rtsp-session.c:
12053 * gst/rtsp-server/rtsp-session.h:
12054 Add more timeout stuff
12055 Add method to check if a session is expired.
12056 Add method to perform cleanup on a session pool.
12058 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12060 * gst/rtsp-server/rtsp-client.c:
12061 * gst/rtsp-server/rtsp-session-pool.c:
12062 * gst/rtsp-server/rtsp-session-pool.h:
12063 * gst/rtsp-server/rtsp-session.c:
12064 * gst/rtsp-server/rtsp-session.h:
12065 Add beginnings of session timeouts and limits
12066 Add the timeout value to the Session header for unusual timeout values.
12067 Allow us to configure a limit to the amount of active sessions in a pool. Set a
12068 limit on the amount of retry we do after a sessionid collision.
12069 Add properties to the sessionid and the timeout of a session. Keep track of
12070 creation time and last access time for sessions.
12072 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12074 * gst/rtsp-server/rtsp-client.c:
12075 * gst/rtsp-server/rtsp-media.c:
12076 * gst/rtsp-server/rtsp-media.h:
12077 * gst/rtsp-server/rtsp-sdp.c:
12078 * gst/rtsp-server/rtsp-session-pool.c:
12079 * gst/rtsp-server/rtsp-session.c:
12080 * gst/rtsp-server/rtsp-session.h:
12081 Cleanup of sessions and more
12082 Fix the refcounting of media and sessions in the client. Properly clean up the
12083 session data when the client performs a teardown.
12084 Add Server header to responses.
12085 Allow for multiple uri setups in one session.
12086 Add Range header to the PLAY response and add the range attribute to the SDP
12088 Fix the session pool remove method, it used the wrong key in the hashtable. Also
12089 give the ownership of the sessionid to the session object.
12091 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12093 * gst/rtsp-server/rtsp-server.c:
12094 * gst/rtsp-server/rtsp-server.h:
12096 Rename the 'server_port' variable to simply 'port'.
12098 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12101 * gst/rtsp-server/rtsp-client.c:
12102 * gst/rtsp-server/rtsp-media.c:
12103 * gst/rtsp-server/rtsp-media.h:
12104 * gst/rtsp-server/rtsp-session.c:
12105 * gst/rtsp-server/rtsp-session.h:
12106 Rework the way we handle transports for streams
12107 Make the media accept an array of transports for the streams that we have
12108 configured for the play/pause requests.
12109 Implement server states for a client and its media.
12110 Require 0.10.22.1 (git HEAD) of gstreamer.
12112 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12114 * gst/rtsp-server/rtsp-client.c:
12115 * gst/rtsp-server/rtsp-media-factory.c:
12116 Drop const from functions dealing with urls
12117 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
12118 have the right const in them.
12120 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12122 * gst/rtsp-server/rtsp-client.c:
12123 * gst/rtsp-server/rtsp-media.c:
12124 * gst/rtsp-server/rtsp-sdp.c:
12128 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12130 * gst/rtsp-server/rtsp-client.c:
12131 * gst/rtsp-server/rtsp-media-factory.c:
12132 * gst/rtsp-server/rtsp-media.c:
12133 * gst/rtsp-server/rtsp-media.h:
12135 Don't keep a reference to the GstRTSPMedia in the stream.
12136 Free more things when freeing the GstRTSPMedia.
12138 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12141 * gst/rtsp-server/rtsp-media-factory.c:
12142 * gst/rtsp-server/rtsp-media-factory.h:
12143 * gst/rtsp-server/rtsp-media.c:
12144 * gst/rtsp-server/rtsp-media.h:
12145 * gst/rtsp-server/rtsp-server.c:
12146 * gst/rtsp-server/rtsp-server.h:
12147 More docs and small cleanups
12148 Add some more docs and update the README
12149 Cleanup some method names.
12150 Remove an unneeded idx field in the GstRTSPMediaStream
12152 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12155 * examples/Makefile.am:
12156 * examples/test-readme.c:
12157 Add a README and more example code
12158 Add a README file that contains a small introduction on how to use the server
12159 along with the example code explained in the readme.
12161 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12163 * gst/rtsp-server/rtsp-media.c:
12164 * gst/rtsp-server/rtsp-server.c:
12165 Fix some leaks and change default port
12166 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
12167 we finished the initial preroll. If we keep them locked, setting the pipeline to
12168 NULL will not stop and clean up the sources correctly.
12169 Change the default RTSP port to 8554 aka the official alternative RTSP port.
12171 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12173 * gst/rtsp-server/rtsp-session.c:
12174 * gst/rtsp-server/rtsp-session.h:
12175 Cleanups to the session object
12176 Remove some unneeded variables in the session state of a stream such as the
12177 owner media and the server transport.
12178 Get the configuration of a media stream in a session based on the media_stream
12179 in the original object instead of our cached index.
12180 Free more data in the finalize method.
12182 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12184 * gst/rtsp-server/rtsp-client.c:
12185 * gst/rtsp-server/rtsp-client.h:
12186 Cleanups and reuse media from DESCRIBE
12187 Handle thread create errors.
12188 Rename some internal methods to better match what they actually do.
12189 Handle misconfiguration of session_pool and media_mapping gracefully.
12190 Cache the DESCRIBE media and uri in the client connection and reuse them when
12191 we receive a SETUP request in the same connection for the same uri.
12192 Cleanup the client connection object.
12194 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12196 * gst/rtsp-server/rtsp-media-factory.c:
12197 * gst/rtsp-server/rtsp-media-factory.h:
12198 * gst/rtsp-server/rtsp-media.c:
12199 * gst/rtsp-server/rtsp-media.h:
12200 Add shared properties to media and factory
12201 Add the shared property to media.
12202 Implement some simple caching in the factory depending on if the media is shared
12205 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12207 * gst/rtsp-server/rtsp-client.c:
12208 Add a little comment
12209 Add some comment about the content-base header.
12211 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12213 * examples/Makefile.am:
12214 * examples/test-mp4.c:
12215 * examples/test-ogg.c:
12216 * examples/test-video.c:
12217 * gst/rtsp-server/Makefile.am:
12218 * gst/rtsp-server/rtsp-client.c:
12219 * gst/rtsp-server/rtsp-client.h:
12220 * gst/rtsp-server/rtsp-media-factory.c:
12221 * gst/rtsp-server/rtsp-media-factory.h:
12222 * gst/rtsp-server/rtsp-media.c:
12223 * gst/rtsp-server/rtsp-media.h:
12224 * gst/rtsp-server/rtsp-sdp.c:
12225 * gst/rtsp-server/rtsp-sdp.h:
12226 * gst/rtsp-server/rtsp-server.c:
12227 * gst/rtsp-server/rtsp-server.h:
12228 * gst/rtsp-server/rtsp-session.c:
12229 * gst/rtsp-server/rtsp-session.h:
12230 Reorganize things, prepare for media sharing
12231 Added various other test server examples
12232 Move the SDP message generation to a separate helper.
12233 Refactor common code for finding the session.
12234 Add content-base for realplayer compatibility
12235 Clean up request uris before processing for better vlc compatibility.
12236 Move prerolling and pipeline construction to the RTSPMedia object.
12237 Use multiudpsink for future pipeline reuse.
12239 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12242 Back to development
12245 === release 0.10.1 ===
12247 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12250 Make 0.10.1 release
12253 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12255 * bindings/vala/Makefile.am:
12257 Add more directories and files to the dist.
12259 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
12261 * bindings/python/Makefile.am:
12262 * bindings/python/rtspserver.override:
12263 Fixed compile error of python bindings
12265 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
12267 * bindings/vala/gst-rtsp-server-0.10.vapi:
12268 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
12269 Marked values as nullable accordingly
12271 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
12273 * bindings/vala/gst-rtsp-server-0.10.vapi:
12274 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
12275 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
12276 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
12277 Updated Vala bindings
12279 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12281 * gst/rtsp-server/rtsp-client.c:
12282 * gst/rtsp-server/rtsp-media-mapping.c:
12283 * gst/rtsp-server/rtsp-media-mapping.h:
12284 * gst/rtsp-server/rtsp-media.h:
12285 * gst/rtsp-server/rtsp-session-pool.h:
12286 Cleanups and doc updates
12287 Add some more documentation and do some minor cleanups here and there.
12289 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12291 * gst/rtsp-server/rtsp-client.c:
12292 * gst/rtsp-server/rtsp-media-factory.c:
12293 * gst/rtsp-server/rtsp-media-factory.h:
12294 * gst/rtsp-server/rtsp-media.c:
12295 * gst/rtsp-server/rtsp-media.h:
12296 * gst/rtsp-server/rtsp-session.c:
12297 * gst/rtsp-server/rtsp-session.h:
12299 Rename GstRTSPMediaBin to GstRTSPMedia
12300 Parse the request url into a GstRTSPUri object and pass this object to the
12301 various handlers and methods that require the uri.
12303 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12307 Add some more docs and remove some old code from the example.
12309 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12311 * gst/rtsp-server/rtsp-client.c:
12312 Handle state change failures better
12313 Handle state change failures better when changing the state of the pipeline to
12316 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12318 * gst/rtsp-server/rtsp-media-factory.c:
12319 * gst/rtsp-server/rtsp-media-factory.h:
12320 Make element creation more extendible
12321 Add get_element vmethod to the default MediaFactory so that subclasses can just
12322 override that method and still use the default logic for making a MediaBin from
12325 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12328 * gst/rtsp-server/Makefile.am:
12329 * gst/rtsp-server/rtsp-client.c:
12330 * gst/rtsp-server/rtsp-client.h:
12331 * gst/rtsp-server/rtsp-media-factory.c:
12332 * gst/rtsp-server/rtsp-media-factory.h:
12333 * gst/rtsp-server/rtsp-media-mapping.c:
12334 * gst/rtsp-server/rtsp-media-mapping.h:
12335 * gst/rtsp-server/rtsp-media.c:
12336 * gst/rtsp-server/rtsp-media.h:
12337 * gst/rtsp-server/rtsp-server.c:
12338 * gst/rtsp-server/rtsp-server.h:
12339 * gst/rtsp-server/rtsp-session.c:
12340 * gst/rtsp-server/rtsp-session.h:
12341 Make the server handle arbitrary pipelines
12342 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
12343 The GstMediaBin object has a handle to a bin with elements and to a list of
12344 GstMediaStream objects that this bin produces.
12345 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
12346 with methods to register and remove those mappings.
12347 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
12348 used by the server instance.
12349 Modify the example application so that it shows how to create custom pipelines
12350 attached to a specific mount point.
12351 Various misc cleanps.
12353 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12355 * gst/rtsp-server/rtsp-server.c:
12356 * gst/rtsp-server/rtsp-server.h:
12357 Allow setting a custom media factory for a server
12359 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12361 * gst/rtsp-server/rtsp-client.c:
12362 * gst/rtsp-server/rtsp-client.h:
12363 Allow setting a custom media factory for a client.
12365 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12367 * gst/rtsp-server/Makefile.am:
12368 Add Makefile entry for the media factory
12370 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12372 * gst/rtsp-server/rtsp-media-factory.c:
12373 * gst/rtsp-server/rtsp-media-factory.h:
12374 Add media factory to map urls to media pipeline objects.
12376 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12378 * gst/rtsp-server/rtsp-media.c:
12379 * gst/rtsp-server/rtsp-media.h:
12380 Add comments. Remove unused field
12382 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12384 * gst/rtsp-server/rtsp-session-pool.c:
12385 * gst/rtsp-server/rtsp-session-pool.h:
12386 Allow custom session pools to override the session id allocation algorithms Add some comments.
12388 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12390 * gst/rtsp-server/rtsp-session.h:
12393 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12395 * gst/rtsp-server/rtsp-client.c:
12396 * gst/rtsp-server/rtsp-client.h:
12397 Move the connection code in one place Add some comments
12399 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12401 * gst/rtsp-server/rtsp-server.c:
12402 * gst/rtsp-server/rtsp-server.h:
12403 Make vmethod to create and accept new clients. Add some docs.
12405 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12407 * gst/rtsp-server/rtsp-server.c:
12408 * gst/rtsp-server/rtsp-server.h:
12409 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
12411 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12413 * gst/rtsp-server/rtsp-client.c:
12414 * gst/rtsp-server/rtsp-client.h:
12415 Name the parameters more appropriately.
12417 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12419 * gst/rtsp-server/rtsp-session-pool.c:
12420 Do some more cleanup of the session pool.
12422 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12424 * gst/rtsp-server/Makefile.am:
12425 * gst/rtsp-server/rtsp-client.c:
12426 Check if return value of gst_rtsp_session_get_media is not NULL
12428 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12430 * gst/rtsp-server/Makefile.am:
12431 Install rtsp-session and rtsp-session-pool headers
12433 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12438 * bindings/python/Makefile.am:
12439 * bindings/python/arg-types.py:
12440 * bindings/python/codegen/Makefile.am:
12441 * bindings/python/codegen/__init__.py:
12442 * bindings/python/codegen/argtypes.py:
12443 * bindings/python/codegen/code-coverage.py:
12444 * bindings/python/codegen/codegen.py:
12445 * bindings/python/codegen/definitions.py:
12446 * bindings/python/codegen/defsparser.py:
12447 * bindings/python/codegen/docextract.py:
12448 * bindings/python/codegen/docgen.py:
12449 * bindings/python/codegen/fileprefix.override:
12450 * bindings/python/codegen/fileprefixmodule.c:
12451 * bindings/python/codegen/h2def.py:
12452 * bindings/python/codegen/mergedefs.py:
12453 * bindings/python/codegen/mkskel.py:
12454 * bindings/python/codegen/override.py:
12455 * bindings/python/codegen/reversewrapper.py:
12456 * bindings/python/codegen/scmexpr.py:
12457 * bindings/python/rtspserver-types.defs:
12458 * bindings/python/rtspserver.defs:
12459 * bindings/python/rtspserver.override:
12460 * bindings/python/rtspservermodule.c:
12462 Add python bindings.
12464 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12466 * bindings/Makefile.am:
12468 Don't go into python dir when requirements for python bindings are missing
12470 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12472 * bindings/Makefile.am:
12473 * bindings/vala/Makefile.am:
12475 Install Vala bindings if vala is available
12477 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
12479 * bindings/vala/gst-rtsp-server-0.10.deps:
12480 * bindings/vala/gst-rtsp-server-0.10.vapi:
12481 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
12482 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
12483 * bindings/vala/packages/gst-rtsp-server-0.10.files:
12484 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
12485 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
12486 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
12487 Regenerated Vala bindings
12489 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
12491 * bindings/vala/gst-rtsp-server.vapi:
12492 * bindings/vala/packages/gst-rtsp-server.metadata:
12493 Fixed typo in included headers for vala bindings
12495 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12499 * pkgconfig/Makefile.am:
12500 * pkgconfig/gst-rtsp-server.pc.in:
12501 Added pkgconfig file
12503 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
12505 * bindings/vala/gst-rtsp-server.vapi:
12506 * bindings/vala/packages/gst-rtsp-server.excludes:
12507 * bindings/vala/packages/gst-rtsp-server.gi:
12508 * bindings/vala/packages/gst-rtsp-server.metadata:
12509 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
12511 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
12513 * bindings/vala/gst-rtsp-server.vapi:
12514 * bindings/vala/packages/gst-rtsp-server.deps:
12515 * bindings/vala/packages/gst-rtsp-server.files:
12516 * bindings/vala/packages/gst-rtsp-server.gi:
12517 * bindings/vala/packages/gst-rtsp-server.metadata:
12518 * bindings/vala/packages/gst-rtsp-server.namespace:
12519 Added Vala bindings
12521 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
12523 * gst/rtsp-server/rtsp-session.c:
12524 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
12526 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
12528 * examples/Makefile.am:
12529 * gst/rtsp-server/Makefile.am:
12530 Put GStreamer version in library name
12532 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12534 * examples/Makefile.am:
12535 * gst/rtsp-server/Makefile.am:
12536 Fix some issues to pass distcheck
12538 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12540 * gst/rtsp-server/rtsp-server.c:
12541 Added port property to GstRTSPServer class.
12543 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12548 * examples/Makefile.am:
12551 * gst/rtsp-server/Makefile.am:
12552 * gst/rtsp-server/rtsp-client.c:
12553 * gst/rtsp-server/rtsp-client.h:
12554 * gst/rtsp-server/rtsp-media.c:
12555 * gst/rtsp-server/rtsp-media.h:
12556 * gst/rtsp-server/rtsp-server.c:
12557 * gst/rtsp-server/rtsp-server.h:
12558 * gst/rtsp-server/rtsp-session-pool.c:
12559 * gst/rtsp-server/rtsp-session-pool.h:
12560 * gst/rtsp-server/rtsp-session.c:
12561 * gst/rtsp-server/rtsp-session.h:
12563 Split in library and example program
12565 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
12567 * src/rtsp-client.h:
12568 Removed obsolete variable
12570 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
12572 * src/rtsp-client.c:
12573 * src/rtsp-client.h:
12574 Removed pipeline variable GstRTSPClient, because it's only used in one function
12576 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
12578 * src/rtsp-media.c:
12579 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
12581 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
12583 * src/rtsp-session.c:
12584 Initialize some more vars.
12586 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
12588 * src/rtsp-session.c:
12589 Initialize variable to avoid compiler warning.
12591 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
12594 Add a reasonable generic .gitignore